Merge commit 'AU_LINUX_ANDROID_ICS.04.00.04.00.126' into msm-3.4
AU_LINUX_ANDROID_ICS.04.00.04.00.126 from msm-3.0.
First parent is from google/android-3.4.
* commit 'AU_LINUX_ANDROID_ICS.04.00.04.00.126': (8712 commits)
PRNG: Device tree entry for qrng device.
vidc:1080p: Set video core timeout value for Thumbnail mode
msm: sps: improve the debugging support in SPS driver
board-8064 msm: Overlap secure and non secure video firmware heaps.
msm: clock: Add handoff ops for 7x30 and copper XO clocks
msm_fb: display: Wait for external vsync before DTV IOMMU unmap
msm: Fix ciruclar dependency in debug UART settings
msm: gdsc: Add GDSC regulator driver for msm-copper
defconfig: Enable Mobicore Driver.
mobicore: Add mobicore driver.
mobicore: rename variable to lower case.
mobicore: rename folder.
mobicore: add makefiles
mobicore: initial import of kernel driver
ASoC: msm: Add SLIMBUS_2_RX CPU DAI
board-8064-gpio: Update FUNC for EPM SPI CS
msm_fb: display: Remove chicken bit config during video playback
mmc: msm_sdcc: enable the sanitize capability
msm-fb: display: lm2 writeback support on mpq platfroms
msm_fb: display: Disable LVDS phy & pll during panel off
...
Signed-off-by: Steve Muckle <smuckle@codeaurora.org>
diff --git a/sound/compress_offload/core.c b/sound/compress_offload/core.c
new file mode 100644
index 0000000..987594a
--- /dev/null
+++ b/sound/compress_offload/core.c
@@ -0,0 +1,658 @@
+/*
+ * core.c - compress offload core
+ *
+ * Copyright (C) 2011 Intel Corporation
+ * Authors: Vinod Koul <vinod.koul@linux.intel.com>
+ * Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+#include <linux/file.h>
+#include <linux/fs.h>
+#include <linux/list.h>
+#include <linux/miscdevice.h>
+#include <linux/mm.h>
+#include <linux/mutex.h>
+#include <linux/poll.h>
+#include <linux/slab.h>
+#include <linux/sched.h>
+#include <linux/uio.h>
+#include <linux/uaccess.h>
+#include <sound/snd_compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+
+/* TODO:
+ * - Integrate with alsa, compressed devices should register as alsa devices
+ * as /dev/snd_compr_xxx
+ * - Integrate with ASoC:
+ * Opening compressed path should also start the codec dai
+ * TBD how the cpu dai will be viewed and started.
+ * ASoC should always be optional part
+ * (we should be able to use this framework in non asoc systems
+ * - Multiple node representation
+ * driver should be able to register multiple nodes
+ * - Version numbering for API
+ */
+
+static DEFINE_MUTEX(device_mutex);
+static LIST_HEAD(device_list);
+static LIST_HEAD(misc_list);
+
+/*
+ * currently we are using misc device for registration and exposing ioctls
+ * this is temporary and will be moved to snd
+ * the device should be registered as /dev/snd_compr.....
+ */
+
+struct snd_compr_misc {
+ struct miscdevice misc;
+ struct list_head list;
+ struct snd_compr *compr;
+};
+
+struct snd_ioctl_data {
+ struct snd_compr_misc *misc;
+ unsigned long caps;
+ unsigned int minor;
+ struct snd_compr_stream stream;
+};
+
+static struct snd_compr_misc *snd_compr_get_device(unsigned int minor)
+{
+ struct snd_compr_misc *misc;
+
+ list_for_each_entry(misc, &misc_list, list) {
+ if (minor == misc->misc.minor)
+ return misc;
+ }
+ return NULL;
+}
+
+static int snd_compr_open(struct inode *inode, struct file *f)
+{
+ unsigned int minor = iminor(inode);
+ struct snd_compr_misc *misc = snd_compr_get_device(minor);
+ struct snd_ioctl_data *data;
+ struct snd_compr_runtime *runtime;
+ unsigned int direction;
+ int ret;
+
+ mutex_lock(&device_mutex);
+ if (f->f_flags & O_WRONLY)
+ direction = SNDRV_PCM_STREAM_PLAYBACK;
+ else {
+ ret = -ENXIO;
+ goto out;
+ }
+ /* curently only encoded playback is supported, above needs to be
+ * removed once we have recording support */
+
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto out;
+ }
+ data->misc = misc;
+ data->minor = minor;
+ data->stream.ops = misc->compr->ops;
+ data->stream.direction = direction;
+ data->stream.private_data = misc->compr->private_data;
+ data->stream.device = misc->compr;
+ runtime = kzalloc(sizeof(*runtime), GFP_KERNEL);
+ if (!runtime) {
+ ret = -ENOMEM;
+ kfree(data);
+ goto out;
+ }
+ runtime->state = SNDRV_PCM_STATE_OPEN;
+ init_waitqueue_head(&runtime->sleep);
+ data->stream.runtime = runtime;
+ f->private_data = (void *)data;
+ ret = misc->compr->ops->open(&data->stream);
+ if (ret) {
+ kfree(runtime);
+ kfree(data);
+ goto out;
+ }
+out:
+ mutex_unlock(&device_mutex);
+ return ret;
+}
+
+static int snd_compr_free(struct inode *inode, struct file *f)
+{
+ struct snd_ioctl_data *data = f->private_data;
+ mutex_lock(&device_mutex);
+ data->stream.ops->free(&data->stream);
+ kfree(data->stream.runtime->buffer);
+ kfree(data->stream.runtime);
+ kfree(data);
+ mutex_unlock(&device_mutex);
+ return 0;
+}
+
+static void snd_compr_update_tstamp(struct snd_compr_stream *stream,
+ struct snd_compr_tstamp *tstamp)
+{
+ stream->ops->pointer(stream, tstamp);
+ stream->runtime->hw_pointer = tstamp->copied_bytes;
+}
+
+static size_t snd_compr_calc_avail(struct snd_compr_stream *stream,
+ struct snd_compr_avail *avail)
+{
+ size_t avail_calc;
+
+ snd_compr_update_tstamp(stream, &avail->tstamp);
+ avail_calc = stream->runtime->app_pointer - stream->runtime->hw_pointer;
+ if (avail_calc < 0)
+ avail_calc = stream->runtime->buffer_size + avail_calc;
+ avail->avail = avail_calc;
+ return avail_calc;
+}
+
+static size_t snd_compr_get_avail(struct snd_compr_stream *stream)
+{
+ struct snd_compr_avail avail;
+
+ return snd_compr_calc_avail(stream, &avail);
+}
+
+static int
+snd_compr_ioctl_avail(struct snd_compr_stream *stream, unsigned long arg)
+{
+ struct snd_compr_avail ioctl_avail;
+
+ snd_compr_calc_avail(stream, &ioctl_avail);
+
+ if (copy_to_user((unsigned long __user *)arg, &ioctl_avail, sizeof(ioctl_avail)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_compr_write_data(struct snd_compr_stream *stream,
+ const char __user *buf, size_t count)
+{
+ void *dstn;
+ size_t copy;
+
+ dstn = stream->runtime->buffer + stream->runtime->app_pointer;
+ if (count < stream->runtime->buffer_size - stream->runtime->app_pointer) {
+ if (copy_from_user(dstn, buf, count))
+ return -EFAULT;
+ stream->runtime->app_pointer += count;
+ } else {
+ copy = stream->runtime->buffer_size - stream->runtime->app_pointer;
+ if (copy_from_user(dstn, buf, copy))
+ return -EFAULT;
+ if (copy_from_user(stream->runtime->buffer, buf + copy, count - copy))
+ return -EFAULT;
+ stream->runtime->app_pointer = count - copy;
+ }
+ /* if DSP cares, let it know data has been written */
+ if (stream->ops->ack)
+ stream->ops->ack(stream);
+ return count;
+}
+
+static ssize_t snd_compr_write(struct file *f, const char __user *buf,
+ size_t count, loff_t *offset)
+{
+ struct snd_ioctl_data *data = f->private_data;
+ struct snd_compr_stream *stream;
+ size_t avail;
+ int retval;
+
+ BUG_ON(!data);
+ stream = &data->stream;
+ mutex_lock(&stream->device->lock);
+ /* write is allowed when stream is running or has been steup */
+ if (stream->runtime->state != SNDRV_PCM_STATE_SETUP &&
+ stream->runtime->state != SNDRV_PCM_STATE_RUNNING) {
+ mutex_unlock(&stream->device->lock);
+ return -EPERM;
+ }
+
+ avail = snd_compr_get_avail(stream);
+ /* calculate how much we can write to buffer */
+ if (avail > count)
+ avail = count;
+
+ if (stream->ops->copy)
+ retval = stream->ops->copy(stream, buf, avail);
+ else
+ retval = snd_compr_write_data(stream, buf, avail);
+
+ /* while initiating the stream, write should be called before START
+ * call, so in setup move state */
+ if (stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+ stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+
+ mutex_unlock(&stream->device->lock);
+ return retval;
+}
+
+
+static ssize_t snd_compr_read(struct file *f, char __user *buf,
+ size_t count, loff_t *offset)
+{
+ return -ENXIO;
+}
+
+static int snd_compr_mmap(struct file *f, struct vm_area_struct *vma)
+{
+ return -ENXIO;
+}
+
+unsigned int snd_compr_poll(struct file *f, poll_table *wait)
+{
+ struct snd_ioctl_data *data = f->private_data;
+ struct snd_compr_stream *stream;
+ int retval = 0;
+
+ BUG_ON(!data);
+ stream = &data->stream;
+
+ mutex_lock(&stream->device->lock);
+ if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) {
+ retval = -ENXIO;
+ goto out;
+ }
+ poll_wait(f, &stream->runtime->sleep, wait);
+
+ /* this would change after read is implemented, we would need to
+ * check for direction here */
+ if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
+ retval = POLLOUT | POLLWRNORM;
+out:
+ mutex_unlock(&stream->device->lock);
+ return retval;
+}
+
+void snd_compr_fragment_elapsed(struct snd_compr_stream *stream)
+{
+ size_t avail;
+
+ if (stream->direction != SNDRV_PCM_STREAM_PLAYBACK)
+ return;
+ avail = snd_compr_get_avail(stream);
+ if (avail >= stream->runtime->fragment_size)
+ wake_up(&stream->runtime->sleep);
+}
+EXPORT_SYMBOL_GPL(snd_compr_fragment_elapsed);
+
+void snd_compr_frame_elapsed(struct snd_compr_stream *stream)
+{
+ size_t avail;
+
+ if (stream->direction != SNDRV_PCM_STREAM_CAPTURE)
+ return;
+ avail = snd_compr_get_avail(stream);
+ if (avail)
+ wake_up(&stream->runtime->sleep);
+}
+EXPORT_SYMBOL_GPL(snd_compr_frame_elapsed);
+
+static int snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg)
+{
+ int retval;
+ struct snd_compr_caps caps;
+
+ if (!stream->ops->get_caps)
+ return -ENXIO;
+
+ retval = stream->ops->get_caps(stream, &caps);
+ if (retval)
+ goto out;
+ if (copy_to_user((void __user *)arg, &caps, sizeof(caps)))
+ retval = -EFAULT;
+out:
+ return retval;
+}
+
+static int snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg)
+{
+ int retval;
+ struct snd_compr_codec_caps *caps;
+
+ if (!stream->ops->get_codec_caps)
+ return -ENXIO;
+
+ caps = kmalloc(sizeof(*caps), GFP_KERNEL);
+ if (!caps)
+ return -ENOMEM;
+
+ retval = stream->ops->get_codec_caps(stream, caps);
+ if (retval)
+ goto out;
+ if (copy_to_user((void __user *)arg, caps, sizeof(*caps)))
+ retval = -EFAULT;
+
+out:
+ kfree(caps);
+ return retval;
+}
+
+/* revisit this with snd_pcm_preallocate_xxx */
+static int snd_compr_allocate_buffer(struct snd_compr_stream *stream,
+ struct snd_compr_params *params)
+{
+ unsigned int buffer_size;
+ void *buffer;
+
+ buffer_size = params->buffer.fragment_size * params->buffer.fragments;
+ if (stream->ops->copy) {
+ buffer = NULL;
+ /* if copy is defined the driver will be required to copy
+ * the data from core
+ */
+ } else {
+ buffer = kmalloc(buffer_size, GFP_KERNEL);
+ if (!buffer)
+ return -ENOMEM;
+ }
+ stream->runtime->fragment_size = params->buffer.fragment_size;
+ stream->runtime->fragments = params->buffer.fragments;
+ stream->runtime->buffer = buffer;
+ stream->runtime->buffer_size = buffer_size;
+ return 0;
+}
+
+static int snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
+{
+ struct snd_compr_params *params;
+ int retval;
+
+ if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) {
+ /*
+ * we should allow parameter change only when stream has been
+ * opened not in other cases
+ */
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params)
+ return -ENOMEM;
+ if (copy_from_user(params, (void __user *)arg, sizeof(*params)))
+ return -EFAULT;
+ retval = snd_compr_allocate_buffer(stream, params);
+ if (retval) {
+ kfree(params);
+ return -ENOMEM;
+ }
+ retval = stream->ops->set_params(stream, params);
+ if (retval)
+ goto out;
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ } else
+ return -EPERM;
+out:
+ kfree(params);
+ return retval;
+}
+
+static int snd_compr_get_params(struct snd_compr_stream *stream, unsigned long arg)
+{
+ struct snd_compr_params *params;
+ int retval;
+
+ if (!stream->ops->get_params)
+ return -ENXIO;
+
+ params = kmalloc(sizeof(*params), GFP_KERNEL);
+ if (!params)
+ return -ENOMEM;
+ retval = stream->ops->get_params(stream, params);
+ if (retval)
+ goto out;
+ if (copy_to_user((char __user *)arg, params, sizeof(*params)))
+ retval = -EFAULT;
+
+out:
+ kfree(params);
+ return retval;
+}
+
+static int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg)
+{
+ struct snd_compr_tstamp tstamp;
+
+ snd_compr_update_tstamp(stream, &tstamp);
+ if (copy_to_user((struct snd_compr_tstamp __user *)arg, &tstamp, sizeof(tstamp)))
+ return -EFAULT;
+ return 0;
+}
+
+static int snd_compr_pause(struct snd_compr_stream *stream)
+{
+ int retval;
+
+ if (stream->runtime->state == SNDRV_PCM_STATE_PAUSED)
+ return 0;
+ retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH);
+ if (!retval) {
+ stream->runtime->state = SNDRV_PCM_STATE_PAUSED;
+ wake_up(&stream->runtime->sleep);
+ }
+ return retval;
+}
+
+static int snd_compr_resume(struct snd_compr_stream *stream)
+{
+ int retval;
+
+ if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED)
+ return -EPERM;
+ retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE);
+ if (!retval)
+ stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
+ return retval;
+}
+
+static int snd_compr_start(struct snd_compr_stream *stream)
+{
+ int retval;
+
+ if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+ return -EPERM;
+ retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
+ if (!retval)
+ stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
+ return retval;
+}
+
+static int snd_compr_stop(struct snd_compr_stream *stream)
+{
+ int retval;
+
+ if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+ return -EPERM;
+ retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
+ if (!retval) {
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ wake_up(&stream->runtime->sleep);
+ }
+ return retval;
+}
+
+static int snd_compr_drain(struct snd_compr_stream *stream)
+{
+ int retval;
+
+ if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED ||
+ stream->runtime->state != SNDRV_PCM_STATE_PAUSED)
+ return -EPERM;
+ retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
+ if (!retval) {
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
+ wake_up(&stream->runtime->sleep);
+ }
+ return retval;
+}
+
+static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg)
+{
+ struct snd_ioctl_data *data = f->private_data;
+ struct snd_compr_stream *stream;
+ int retval = -ENOTTY;
+
+ BUG_ON(!data);
+ stream = &data->stream;
+ mutex_lock(&stream->device->lock);
+ switch (_IOC_NR(cmd)) {
+ case _IOC_NR(SNDRV_COMPRESS_GET_CAPS):
+ retval = snd_compr_get_caps(stream, arg);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_GET_CODEC_CAPS):
+ retval = snd_compr_get_codec_caps(stream, arg);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS):
+ retval = snd_compr_set_params(stream, arg);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS):
+ retval = snd_compr_get_params(stream, arg);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_TSTAMP):
+ retval = snd_compr_tstamp(stream, arg);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_AVAIL):
+ retval = snd_compr_ioctl_avail(stream, arg);
+ case _IOC_NR(SNDRV_COMPRESS_PAUSE):
+ retval = snd_compr_pause(stream);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_RESUME):
+ retval = snd_compr_resume(stream);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_START):
+ retval = snd_compr_start(stream);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_STOP):
+ retval = snd_compr_stop(stream);
+ break;
+ case _IOC_NR(SNDRV_COMPRESS_DRAIN):
+ cmd = SND_COMPR_TRIGGER_DRAIN;
+ retval = snd_compr_drain(stream);
+ break;
+ }
+ mutex_unlock(&stream->device->lock);
+ return retval;
+}
+
+static const struct file_operations snd_comp_file = {
+ .owner = THIS_MODULE,
+ .open = snd_compr_open,
+ .release = snd_compr_free,
+ .read = snd_compr_read,
+ .write = snd_compr_write,
+ .unlocked_ioctl = snd_compr_ioctl,
+ .mmap = snd_compr_mmap,
+ .poll = snd_compr_poll,
+};
+
+static int snd_compress_add_device(struct snd_compr *device)
+{
+ int ret;
+
+ struct snd_compr_misc *misc = kzalloc(sizeof(*misc), GFP_KERNEL);
+
+ misc->misc.name = device->name;
+ misc->misc.fops = &snd_comp_file;
+ misc->misc.minor = MISC_DYNAMIC_MINOR;
+ misc->compr = device;
+ ret = misc_register(&misc->misc);
+ if (ret) {
+ pr_err("couldn't register misc device\n");
+ kfree(misc);
+ } else {
+ pr_debug("Got minor %d\n", misc->misc.minor);
+ list_add_tail(&misc->list, &misc_list);
+ }
+ return ret;
+}
+
+static int snd_compress_remove_device(struct snd_compr *device)
+{
+ struct snd_compr_misc *misc, *__misc;
+
+ list_for_each_entry_safe(misc, __misc, &misc_list, list) {
+ if (device == misc->compr) {
+ misc_deregister(&misc->misc);
+ list_del(&device->list);
+ kfree(misc);
+ }
+ }
+ return 0;
+}
+/**
+ * snd_compress_register - register compressed device
+ *
+ * @device: compressed device to register
+ */
+int snd_compress_register(struct snd_compr *device)
+{
+ int retval;
+
+ if (device->name == NULL || device->dev == NULL || device->ops == NULL)
+ return -EINVAL;
+ BUG_ON(!device->ops->open);
+ BUG_ON(!device->ops->free);
+ BUG_ON(!device->ops->set_params);
+ BUG_ON(!device->ops->get_params);
+ BUG_ON(!device->ops->trigger);
+ BUG_ON(!device->ops->pointer);
+ BUG_ON(!device->ops->get_caps);
+ BUG_ON(!device->ops->get_codec_caps);
+
+ INIT_LIST_HEAD(&device->list);
+ /* todo register the compressed streams */
+ /* todo integrate with asoc */
+
+ /* register a compressed card TBD if this needs change */
+
+ pr_debug("Registering compressed device %s\n", device->name);
+ mutex_lock(&device_mutex);
+ /* register a msic device for now */
+ retval = snd_compress_add_device(device);
+ if (!retval)
+ list_add_tail(&device->list, &device_list);
+ mutex_unlock(&device_mutex);
+ return retval;
+}
+EXPORT_SYMBOL_GPL(snd_compress_register);
+
+int snd_compress_deregister(struct snd_compr *device)
+{
+ pr_debug("Removing compressed device %s\n", device->name);
+ mutex_lock(&device_mutex);
+ snd_compress_remove_device(device);
+ list_del(&device->list);
+ mutex_unlock(&device_mutex);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_compress_deregister);
+
+static int __init snd_compress_init(void)
+{
+ return 0;
+}
+
+static void __exit snd_compress_exit(void)
+{
+}
+
+module_init(snd_compress_init);
+module_exit(snd_compress_exit);
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index b413ed0..87dcaee 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -149,6 +149,7 @@
config SND_DYNAMIC_MINORS
bool "Dynamic device file minor numbers"
+ default y if SND_OMAP_SOC_ABE_DSP
help
If you say Y here, the minor numbers of ALSA device files in
/dev/snd/ are allocated dynamically. This allows you to have
diff --git a/sound/core/init.c b/sound/core/init.c
index d8ec849..f300bd3 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -22,13 +22,12 @@
#include <linux/init.h>
#include <linux/sched.h>
#include <linux/module.h>
-#include <linux/device.h>
#include <linux/file.h>
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/ctype.h>
#include <linux/pm.h>
-
+#include <linux/device.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/info.h>
@@ -481,104 +480,74 @@
EXPORT_SYMBOL(snd_card_free);
-/* retrieve the last word of shortname or longname */
-static const char *retrieve_id_from_card_name(const char *name)
+static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid)
{
- const char *spos = name;
-
- while (*name) {
- if (isspace(*name) && isalnum(name[1]))
- spos = name + 1;
- name++;
- }
- return spos;
-}
-
-/* return true if the given id string doesn't conflict any other card ids */
-static bool card_id_ok(struct snd_card *card, const char *id)
-{
- int i;
- if (!snd_info_check_reserved_words(id))
- return false;
- for (i = 0; i < snd_ecards_limit; i++) {
- if (snd_cards[i] && snd_cards[i] != card &&
- !strcmp(snd_cards[i]->id, id))
- return false;
- }
- return true;
-}
-
-/* copy to card->id only with valid letters from nid */
-static void copy_valid_id_string(struct snd_card *card, const char *src,
- const char *nid)
-{
- char *id = card->id;
-
- while (*nid && !isalnum(*nid))
- nid++;
- if (isdigit(*nid))
- *id++ = isalpha(*src) ? *src : 'D';
- while (*nid && (size_t)(id - card->id) < sizeof(card->id) - 1) {
- if (isalnum(*nid))
- *id++ = *nid;
- nid++;
- }
- *id = 0;
-}
-
-/* Set card->id from the given string
- * If the string conflicts with other ids, add a suffix to make it unique.
- */
-static void snd_card_set_id_no_lock(struct snd_card *card, const char *src,
- const char *nid)
-{
- int len, loops;
- bool with_suffix;
- bool is_default = false;
+ int i, len, idx_flag = 0, loops = SNDRV_CARDS;
+ const char *spos, *src;
char *id;
- copy_valid_id_string(card, src, nid);
- id = card->id;
-
- again:
- /* use "Default" for obviously invalid strings
- * ("card" conflicts with proc directories)
- */
- if (!*id || !strncmp(id, "card", 4)) {
- strcpy(id, "Default");
- is_default = true;
+ if (nid == NULL) {
+ id = card->shortname;
+ spos = src = id;
+ while (*id != '\0') {
+ if (*id == ' ')
+ spos = id + 1;
+ id++;
+ }
+ } else {
+ spos = src = nid;
}
+ id = card->id;
+ while (*spos != '\0' && !isalnum(*spos))
+ spos++;
+ if (isdigit(*spos))
+ *id++ = isalpha(src[0]) ? src[0] : 'D';
+ while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) {
+ if (isalnum(*spos))
+ *id++ = *spos;
+ spos++;
+ }
+ *id = '\0';
- with_suffix = false;
- for (loops = 0; loops < SNDRV_CARDS; loops++) {
- if (card_id_ok(card, id))
- return; /* OK */
+ id = card->id;
+
+ if (*id == '\0')
+ strcpy(id, "Default");
+ while (1) {
+ if (loops-- == 0) {
+ snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
+ strcpy(card->id, card->proc_root->name);
+ return;
+ }
+ if (!snd_info_check_reserved_words(id))
+ goto __change;
+ for (i = 0; i < snd_ecards_limit; i++) {
+ if (snd_cards[i] && !strcmp(snd_cards[i]->id, id))
+ goto __change;
+ }
+ break;
+
+ __change:
len = strlen(id);
- if (!with_suffix) {
- /* add the "_X" suffix */
- char *spos = id + len;
- if (len > sizeof(card->id) - 3)
- spos = id + sizeof(card->id) - 3;
- strcpy(spos, "_1");
- with_suffix = true;
- } else {
- /* modify the existing suffix */
- if (id[len - 1] != '9')
- id[len - 1]++;
+ if (idx_flag) {
+ if (id[len-1] != '9')
+ id[len-1]++;
else
- id[len - 1] = 'A';
+ id[len-1] = 'A';
+ } else if ((size_t)len <= sizeof(card->id) - 3) {
+ strcat(id, "_1");
+ idx_flag++;
+ } else {
+ spos = id + len - 2;
+ if ((size_t)len <= sizeof(card->id) - 2)
+ spos++;
+ *(char *)spos++ = '_';
+ *(char *)spos++ = '1';
+ *(char *)spos++ = '\0';
+ idx_flag++;
}
}
- /* fallback to the default id */
- if (!is_default) {
- *id = 0;
- goto again;
- }
- /* last resort... */
- snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
- if (card->proc_root->name)
- strcpy(card->id, card->proc_root->name);
}
/**
@@ -595,7 +564,7 @@
if (card->id[0] != '\0')
return;
mutex_lock(&snd_card_mutex);
- snd_card_set_id_no_lock(card, nid, nid);
+ snd_card_set_id_no_lock(card, nid);
mutex_unlock(&snd_card_mutex);
}
EXPORT_SYMBOL(snd_card_set_id);
@@ -627,12 +596,22 @@
memcpy(buf1, buf, copy);
buf1[copy] = '\0';
mutex_lock(&snd_card_mutex);
- if (!card_id_ok(NULL, buf1)) {
+ if (!snd_info_check_reserved_words(buf1)) {
+ __exist:
mutex_unlock(&snd_card_mutex);
return -EEXIST;
}
+ for (idx = 0; idx < snd_ecards_limit; idx++) {
+ if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) {
+ if (card == snd_cards[idx])
+ goto __ok;
+ else
+ goto __exist;
+ }
+ }
strcpy(card->id, buf1);
snd_info_card_id_change(card);
+__ok:
mutex_unlock(&snd_card_mutex);
return count;
@@ -686,18 +665,7 @@
mutex_unlock(&snd_card_mutex);
return 0;
}
- if (*card->id) {
- /* make a unique id name from the given string */
- char tmpid[sizeof(card->id)];
- memcpy(tmpid, card->id, sizeof(card->id));
- snd_card_set_id_no_lock(card, tmpid, tmpid);
- } else {
- /* create an id from either shortname or longname */
- const char *src;
- src = *card->shortname ? card->shortname : card->longname;
- snd_card_set_id_no_lock(card, src,
- retrieve_id_from_card_name(src));
- }
+ snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id);
snd_cards[card->number] = card;
mutex_unlock(&snd_card_mutex);
init_info_for_card(card);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 471e1e3..9e2e085 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -25,13 +25,16 @@
#include <sound/jack.h>
#include <sound/core.h>
-static int jack_switch_types[SND_JACK_SWITCH_TYPES] = {
+static int jack_switch_types[] = {
SW_HEADPHONE_INSERT,
SW_MICROPHONE_INSERT,
SW_LINEOUT_INSERT,
SW_JACK_PHYSICAL_INSERT,
SW_VIDEOOUT_INSERT,
SW_LINEIN_INSERT,
+ SW_HPHL_OVERCURRENT,
+ SW_HPHR_OVERCURRENT,
+ SW_UNSUPPORT_INSERT,
};
static int snd_jack_dev_free(struct snd_device *device)
@@ -128,7 +131,7 @@
jack->type = type;
- for (i = 0; i < SND_JACK_SWITCH_TYPES; i++)
+ for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++)
if (type & (1 << i))
input_set_capability(jack->input_dev, EV_SW,
jack_switch_types[i]);
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 1a3070b..09bf06e 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -24,7 +24,6 @@
#include <linux/module.h>
#include <linux/time.h>
#include <linux/mutex.h>
-#include <linux/device.h>
#include <sound/core.h>
#include <sound/minors.h>
#include <sound/pcm.h>
@@ -42,6 +41,7 @@
static int snd_pcm_free(struct snd_pcm *pcm);
static int snd_pcm_dev_free(struct snd_device *device);
static int snd_pcm_dev_register(struct snd_device *device);
+static int snd_pcm_dev_register_soc_be(struct snd_device *device);
static int snd_pcm_dev_disconnect(struct snd_device *device);
static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device)
@@ -651,7 +651,7 @@
pstr->stream = stream;
pstr->pcm = pcm;
pstr->substream_count = substream_count;
- if (substream_count > 0 && !pcm->internal) {
+ if (substream_count > 0) {
err = snd_pcm_stream_proc_init(pstr);
if (err < 0) {
snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
@@ -675,18 +675,15 @@
pstr->substream = substream;
else
prev->next = substream;
-
- if (!pcm->internal) {
- err = snd_pcm_substream_proc_init(substream);
- if (err < 0) {
- snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
- if (prev == NULL)
- pstr->substream = NULL;
- else
- prev->next = NULL;
- kfree(substream);
- return err;
- }
+ err = snd_pcm_substream_proc_init(substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
+ if (prev == NULL)
+ pstr->substream = NULL;
+ else
+ prev->next = NULL;
+ kfree(substream);
+ return err;
}
substream->group = &substream->self_group;
spin_lock_init(&substream->self_group.lock);
@@ -700,9 +697,25 @@
EXPORT_SYMBOL(snd_pcm_new_stream);
-static int _snd_pcm_new(struct snd_card *card, const char *id, int device,
- int playback_count, int capture_count, bool internal,
- struct snd_pcm **rpcm)
+/**
+ * snd_pcm_new - create a new PCM instance
+ * @card: the card instance
+ * @id: the id string
+ * @device: the device index (zero based)
+ * @playback_count: the number of substreams for playback
+ * @capture_count: the number of substreams for capture
+ * @rpcm: the pointer to store the new pcm instance
+ *
+ * Creates a new PCM instance.
+ *
+ * The pcm operators have to be set afterwards to the new instance
+ * via snd_pcm_set_ops().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_new(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count,
+ struct snd_pcm ** rpcm)
{
struct snd_pcm *pcm;
int err;
@@ -723,7 +736,7 @@
}
pcm->card = card;
pcm->device = device;
- pcm->internal = internal;
+
if (id)
strlcpy(pcm->id, id, sizeof(pcm->id));
if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) {
@@ -745,32 +758,49 @@
return 0;
}
-/**
- * snd_pcm_new - create a new PCM instance
- * @card: the card instance
- * @id: the id string
- * @device: the device index (zero based)
- * @playback_count: the number of substreams for playback
- * @capture_count: the number of substreams for capture
- * @rpcm: the pointer to store the new pcm instance
- *
- * Creates a new PCM instance.
- *
- * The pcm operators have to be set afterwards to the new instance
- * via snd_pcm_set_ops().
- *
- * Returns zero if successful, or a negative error code on failure.
- */
-int snd_pcm_new(struct snd_card *card, const char *id, int device,
- int playback_count, int capture_count, struct snd_pcm **rpcm)
-{
- return _snd_pcm_new(card, id, device, playback_count, capture_count,
- false, rpcm);
-}
EXPORT_SYMBOL(snd_pcm_new);
+static int snd_pcm_new_stream_soc_be(struct snd_pcm *pcm, int stream,
+ int substream_count)
+{
+ int idx;
+ struct snd_pcm_str *pstr = &pcm->streams[stream];
+ struct snd_pcm_substream *substream, *prev;
+
+ pstr->stream = stream;
+ pstr->pcm = pcm;
+ pstr->substream_count = substream_count;
+
+ prev = NULL;
+ for (idx = 0, prev = NULL; idx < substream_count; idx++) {
+ substream = kzalloc(sizeof(*substream), GFP_KERNEL);
+ if (substream == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate BE PCM substream\n");
+ return -ENOMEM;
+ }
+ substream->pcm = pcm;
+ substream->pstr = pstr;
+ substream->number = idx;
+ substream->stream = stream;
+ sprintf(substream->name, "subdevice #%i", idx);
+ substream->buffer_bytes_max = UINT_MAX;
+ if (prev == NULL)
+ pstr->substream = substream;
+ else
+ prev->next = substream;
+
+ substream->group = &substream->self_group;
+ spin_lock_init(&substream->self_group.lock);
+ INIT_LIST_HEAD(&substream->self_group.substreams);
+ list_add_tail(&substream->link_list, &substream->self_group.substreams);
+ atomic_set(&substream->mmap_count, 0);
+ prev = substream;
+ }
+ return 0;
+}
+
/**
- * snd_pcm_new_internal - create a new internal PCM instance
+ * snd_pcm_new_soc_be - create a new PCM instance for ASoC BE DAI link
* @card: the card instance
* @id: the id string
* @device: the device index (zero based - shared with normal PCMs)
@@ -778,25 +808,62 @@
* @capture_count: the number of substreams for capture
* @rpcm: the pointer to store the new pcm instance
*
- * Creates a new internal PCM instance with no userspace device or procfs
- * entries. This is used by ASoC Back End PCMs in order to create a PCM that
- * will only be used internally by kernel drivers. i.e. it cannot be opened
- * by userspace. It provides existing ASoC components drivers with a substream
- * and access to any private data.
+ * Creates a new PCM instance with no userspace device or procfs entries.
+ * This is used by ASoC Back End PCMs in order to create a PCM that will only
+ * be used internally by kernel drivers. i.e. it cannot be opened by userspace.
+ * It also provides existing ASoC components drivers with a substream and
+ * access to any private data.
*
* The pcm operators have to be set afterwards to the new instance
* via snd_pcm_set_ops().
*
* Returns zero if successful, or a negative error code on failure.
*/
-int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
+int snd_pcm_new_soc_be(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
- struct snd_pcm **rpcm)
+ struct snd_pcm ** rpcm)
{
- return _snd_pcm_new(card, id, device, playback_count, capture_count,
- true, rpcm);
+ struct snd_pcm *pcm;
+ int err;
+ static struct snd_device_ops ops = {
+ .dev_free = snd_pcm_dev_free,
+ .dev_register = snd_pcm_dev_register_soc_be,
+ .dev_disconnect = snd_pcm_dev_disconnect,
+ };
+
+ if (snd_BUG_ON(!card))
+ return -ENXIO;
+ if (rpcm)
+ *rpcm = NULL;
+ pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
+ if (pcm == NULL) {
+ snd_printk(KERN_ERR "Cannot allocate virtual PCM\n");
+ return -ENOMEM;
+ }
+ pcm->card = card;
+ pcm->device = device;
+
+ if (id)
+ strlcpy(pcm->id, id, sizeof(pcm->id));
+ if ((err = snd_pcm_new_stream_soc_be(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) {
+ snd_pcm_free(pcm);
+ return err;
+ }
+ if ((err = snd_pcm_new_stream_soc_be(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) {
+ snd_pcm_free(pcm);
+ return err;
+ }
+
+ if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) {
+ snd_pcm_free(pcm);
+ return err;
+ }
+ if (rpcm)
+ *rpcm = pcm;
+ return 0;
}
-EXPORT_SYMBOL(snd_pcm_new_internal);
+
+EXPORT_SYMBOL(snd_pcm_new_soc_be);
static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
{
@@ -1034,7 +1101,7 @@
}
for (cidx = 0; cidx < 2; cidx++) {
int devtype = -1;
- if (pcm->streams[cidx].substream == NULL || pcm->internal)
+ if (pcm->streams[cidx].substream == NULL)
continue;
switch (cidx) {
case SNDRV_PCM_STREAM_PLAYBACK:
@@ -1075,6 +1142,29 @@
return 0;
}
+static int snd_pcm_dev_register_soc_be(struct snd_device *device)
+{
+ int err;
+ struct snd_pcm_notify *notify;
+ struct snd_pcm *pcm;
+
+ if (snd_BUG_ON(!device || !device->device_data))
+ return -ENXIO;
+ pcm = device->device_data;
+ mutex_lock(®ister_mutex);
+ err = snd_pcm_add(pcm);
+ if (err) {
+ mutex_unlock(®ister_mutex);
+ return err;
+ }
+
+ list_for_each_entry(notify, &snd_pcm_notify_list, list)
+ notify->n_register(pcm);
+
+ mutex_unlock(®ister_mutex);
+ return 0;
+}
+
static int snd_pcm_dev_disconnect(struct snd_device *device)
{
struct snd_pcm *pcm = device->device_data;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4d18941..b5d5a75 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -6,8 +6,7 @@
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * the Free Software Foundation; only version 2 of the License.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
@@ -1030,7 +1029,7 @@
* Returns non-zero if the value is changed, zero if not changed.
*/
int snd_interval_list(struct snd_interval *i, unsigned int count,
- const unsigned int *list, unsigned int mask)
+ unsigned int *list, unsigned int mask)
{
unsigned int k;
struct snd_interval list_range;
@@ -1992,6 +1991,9 @@
struct snd_pcm_runtime *runtime;
if (PCM_RUNTIME_CHECK(substream))
return -ENXIO;
+ /* TODO: consider and -EINVAL here */
+ if (substream->hw_no_buffer)
+ snd_printd("%s: warning this PCM is host less\n", __func__);
runtime = substream->runtime;
if (snd_BUG_ON(!substream->ops->copy && !runtime->dma_area))
return -EINVAL;
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 9c9eff9..1dbd5ad 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -148,7 +148,9 @@
.le = -1, .signd = -1,
},
[SNDRV_PCM_FORMAT_SPECIAL] = {
- .le = -1, .signd = -1,
+ /* set the width and phys same as S16_LE */
+ .width = 16, .phys = 16, .le = -1, .signd = -1,
+ .silence = {},
},
[SNDRV_PCM_FORMAT_S24_3LE] = {
.width = 24, .phys = 24, .le = 1, .signd = 1,
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 3fe99e6..5034393 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -5,8 +5,7 @@
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * the Free Software Foundation; only version 2 of the License.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
@@ -29,6 +28,7 @@
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/control.h>
+#include <sound/compress_offload.h>
#include <sound/info.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -843,6 +843,7 @@
if (runtime->status->state != SNDRV_PCM_STATE_PREPARED)
return -EBADFD;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+ !substream->hw_no_buffer &&
!snd_pcm_playback_data(substream))
return -EPIPE;
runtime->trigger_master = substream;
@@ -1520,6 +1521,19 @@
return result;
}
+static int snd_compressed_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void __user *arg)
+{
+ struct snd_pcm_runtime *runtime;
+ int err = 0;
+
+ if (PCM_RUNTIME_CHECK(substream))
+ return -ENXIO;
+ runtime = substream->runtime;
+ pr_debug("%s called with cmd = %d\n", __func__, cmd);
+ err = substream->ops->ioctl(substream, cmd, arg);
+ return err;
+}
/*
* drop ioctl
*
@@ -2041,6 +2055,12 @@
goto error;
}
+ if (substream->ops == NULL) {
+ snd_printd("cannot open back end PCMs directly\n");
+ err = -ENODEV;
+ goto error;
+ }
+
if ((err = substream->ops->open(substream)) < 0)
goto error;
@@ -2567,6 +2587,12 @@
snd_pcm_stream_unlock_irq(substream);
return res;
}
+ case SNDRV_COMPRESS_GET_CAPS:
+ case SNDRV_COMPRESS_GET_CODEC_CAPS:
+ case SNDRV_COMPRESS_SET_PARAMS:
+ case SNDRV_COMPRESS_GET_PARAMS:
+ case SNDRV_COMPRESS_TSTAMP:
+ return snd_compressed_ioctl(substream, cmd, arg);
}
snd_printd("unknown ioctl = 0x%x\n", cmd);
return -ENOTTY;
@@ -2739,7 +2765,7 @@
pcm_file = file->private_data;
- if (((cmd >> 8) & 0xff) != 'A')
+ if ((((cmd >> 8) & 0xff) != 'A') && (((cmd >> 8) & 0xff) != 'C'))
return -ENOTTY;
return snd_pcm_playback_ioctl1(file, pcm_file->substream, cmd,
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 91c9855..7aa695f 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -41,6 +41,7 @@
source "sound/soc/omap/Kconfig"
source "sound/soc/kirkwood/Kconfig"
source "sound/soc/mid-x86/Kconfig"
+source "sound/soc/msm/Kconfig"
source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/samsung/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 2feaf37..a1b1523 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -15,6 +15,7 @@
obj-$(CONFIG_SND_SOC) += imx/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += mid-x86/
+obj-$(CONFIG_SND_SOC) += msm/
obj-$(CONFIG_SND_SOC) += mxs/
obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 59d8efa..80204f5 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -40,7 +40,6 @@
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
select SND_SOC_MAX9850 if I2C
- select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
select SND_SOC_RT5631 if I2C
@@ -63,7 +62,6 @@
select SND_SOC_WL1273 if MFD_WL1273_CORE
select SND_SOC_WM1250_EV1 if I2C
select SND_SOC_WM2000 if I2C
- select SND_SOC_WM2200 if I2C
select SND_SOC_WM5100 if I2C
select SND_SOC_WM8350 if MFD_WM8350
select SND_SOC_WM8400 if MFD_WM8400
@@ -107,6 +105,7 @@
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
select SND_SOC_WM9712 if SND_SOC_AC97_BUS
select SND_SOC_WM9713 if SND_SOC_AC97_BUS
+ select SND_SOC_TIMPANI if MARIMBA_CORE
help
Normally ASoC codec drivers are only built if a machine driver which
uses them is also built since they are only usable with a machine
@@ -284,6 +283,15 @@
config SND_SOC_UDA1380
tristate
+config SND_SOC_WCD9304
+ tristate
+
+config SND_SOC_WCD9310
+ tristate
+
+config SND_SOC_CS8427
+ tristate
+
config SND_SOC_WL1273
tristate
@@ -429,11 +437,11 @@
config SND_SOC_LM4857
tristate
-config SND_SOC_MAX9768
- tristate
-
config SND_SOC_MAX9877
tristate
config SND_SOC_TPA6130A2
tristate
+
+config SND_SOC_MSM_STUB
+ tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6662eb0..965d6a1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -25,7 +25,6 @@
snd-soc-jz4740-codec-objs := jz4740.o
snd-soc-l3-objs := l3.o
snd-soc-lm4857-objs := lm4857.o
-snd-soc-max9768-objs := max9768.o
snd-soc-max98088-objs := max98088.o
snd-soc-max98095-objs := max98095.o
snd-soc-max9850-objs := max9850.o
@@ -49,10 +48,12 @@
snd-soc-twl6040-objs := twl6040.o
snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
+snd-soc-wcd9304-objs := wcd9304.o wcd9304-tables.o
+snd-soc-wcd9310-objs := wcd9310.o wcd9310-tables.o
+snd-soc-cs8427-objs := cs8427.o
snd-soc-wl1273-objs := wl1273.o
snd-soc-wm1250-ev1-objs := wm1250-ev1.o
snd-soc-wm2000-objs := wm2000.o
-snd-soc-wm2200-objs := wm2200.o
snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8400-objs := wm8400.o
@@ -98,6 +99,8 @@
snd-soc-wm9713-objs := wm9713.o
snd-soc-wm-hubs-objs := wm_hubs.o
+snd-soc-timpani-objs := timpani.o
+snd-soc-msm-stub-objs := msm_stub.o
# Amp
snd-soc-max9877-objs := max9877.o
snd-soc-tpa6130a2-objs := tpa6130a2.o
@@ -131,7 +134,6 @@
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o
obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
-obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
@@ -153,10 +155,12 @@
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WCD9304) += snd-soc-wcd9304.o
+obj-$(CONFIG_SND_SOC_WCD9310) += snd-soc-wcd9310.o
+obj-$(CONFIG_SND_SOC_CS8427) += snd-soc-cs8427.o
obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o
obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o
-obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o
obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
@@ -201,6 +205,7 @@
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o
+obj-$(CONFIG_SND_SOC_MSM_STUB) += snd-soc-msm-stub.o
# Amp
obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o
diff --git a/sound/soc/codecs/cs8427.c b/sound/soc/codecs/cs8427.c
new file mode 100644
index 0000000..e406a32
--- /dev/null
+++ b/sound/soc/codecs/cs8427.c
@@ -0,0 +1,904 @@
+/*
+ * Routines for control of the CS8427 via i2c bus
+ * IEC958 (S/PDIF) receiver & transmitter by Cirrus Logic
+ * Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ * Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/bitrev.h>
+#include <linux/bitops.h>
+#include <linux/module.h>
+//#include <linux/export.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <asm/unaligned.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/cs8427.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#define CS8427_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_96000)
+
+#define CS8427_FORMATS (SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FORMAT_S16_LE |\
+ SNDRV_PCM_FORMAT_S20_3LE)
+
+struct cs8427_stream {
+ struct snd_pcm_substream *substream;
+ char hw_status[CHANNEL_STATUS_SIZE]; /* hardware status */
+ char def_status[CHANNEL_STATUS_SIZE]; /* default status */
+ char pcm_status[CHANNEL_STATUS_SIZE]; /* PCM private status */
+ char hw_udata[32];
+ struct snd_kcontrol *pcm_ctl;
+};
+
+struct cs8427 {
+ struct i2c_client *client;
+ struct i2c_msg xfer_msg[2];
+ unsigned char regmap[0x14]; /* map of first 1 + 13 registers */
+ unsigned int reset_timeout;
+ struct cs8427_stream playback;
+};
+
+static int cs8427_i2c_write_device(struct cs8427 *cs8427_i2c,
+ u16 reg, u8 *value, u32 bytes)
+{
+ struct i2c_msg *msg;
+ int ret = 0;
+ u8 reg_addr = 0;
+ u8 data[bytes + 1];
+
+ if (cs8427_i2c->client == NULL) {
+ pr_err("%s: failed to get device info\n", __func__);
+ return -ENODEV;
+ }
+ reg_addr = (u8)reg;
+ msg = &cs8427_i2c->xfer_msg[0];
+ msg->addr = cs8427_i2c->client->addr;
+ msg->len = bytes + 1;
+ msg->flags = 0;
+ data[0] = reg_addr;
+ data[1] = *value;
+ msg->buf = data;
+ ret = i2c_transfer(cs8427_i2c->client->adapter,
+ cs8427_i2c->xfer_msg, 1);
+ /* Try again if the write fails
+ * checking with ebusy and number of bytes executed
+ * for write ret value should be 1
+ */
+ if ((ret != 1) || (ret == -EBUSY)) {
+ ret = i2c_transfer(
+ cs8427_i2c->client->adapter,
+ cs8427_i2c->xfer_msg, 1);
+ if ((ret != 1) || (ret < 0)) {
+ dev_err(&cs8427_i2c->client->dev,
+ "failed to write the"
+ " device reg %d\n", reg);
+ return ret;
+ }
+ }
+ return 0;
+}
+
+static int cs8427_i2c_write(struct cs8427 *chip, unsigned short reg,
+ int bytes, void *src)
+{
+ return cs8427_i2c_write_device(chip, reg, src, bytes);
+}
+static int cs8427_i2c_read_device(struct cs8427 *cs8427_i2c,
+ unsigned short reg,
+ int bytes, unsigned char *dest)
+{
+ struct i2c_msg *msg;
+ int ret = 0;
+ u8 reg_addr = 0;
+ u8 i = 0;
+
+ if (cs8427_i2c->client == NULL) {
+ pr_err("%s: failed to get device info\n", __func__);
+ return -ENODEV;
+ }
+ for (i = 0; i < bytes; i++) {
+ reg_addr = (u8)reg++;
+ msg = &cs8427_i2c->xfer_msg[0];
+ msg->addr = cs8427_i2c->client->addr;
+ msg->len = 1;
+ msg->flags = 0;
+ msg->buf = ®_addr;
+
+ msg = &cs8427_i2c->xfer_msg[1];
+ msg->addr = cs8427_i2c->client->addr;
+ msg->len = 1;
+ msg->flags = I2C_M_RD;
+ msg->buf = dest++;
+ ret = i2c_transfer(cs8427_i2c->client->adapter,
+ cs8427_i2c->xfer_msg, 2);
+
+ /* Try again if read fails first time
+ checking with ebusy and number of bytes executed
+ for read ret value should be 2*/
+ if ((ret != 2) || (ret == -EBUSY)) {
+ ret = i2c_transfer(
+ cs8427_i2c->client->adapter,
+ cs8427_i2c->xfer_msg, 2);
+ if ((ret != 2) || (ret < 0)) {
+ dev_err(&cs8427_i2c->client->dev,
+ "failed to read cs8427"
+ " register %d\n", reg);
+ return ret;
+ }
+ }
+ }
+ return 0;
+}
+
+static int cs8427_i2c_read(struct cs8427 *chip,
+ unsigned short reg,
+ int bytes, void *dest)
+{
+ return cs8427_i2c_read_device(chip, reg,
+ bytes, dest);
+}
+
+static int cs8427_i2c_sendbytes(struct cs8427 *chip,
+ char *reg_addr, char *data,
+ int bytes)
+{
+ u32 ret = 0;
+ u8 i = 0;
+
+ if (!chip) {
+ pr_err("%s, invalid device info\n", __func__);
+ return -ENODEV;
+ }
+ if (!data) {
+ dev_err(&chip->client->dev, "%s:"
+ "invalid data pointer\n", __func__);
+ return -EINVAL;
+ }
+ for (i = 0; i < bytes; i++) {
+ ret = cs8427_i2c_write_device(chip, (*reg_addr + i),
+ &data[i], 1);
+ if (ret < 0) {
+ dev_err(&chip->client->dev,
+ "%s: failed to send the data to"
+ " cs8427 chip\n", __func__);
+ break;
+ }
+ }
+ return i;
+}
+
+/*
+ * Reset the chip using run bit, also lock PLL using ILRCK and
+ * put back AES3INPUT. This workaround is described in latest
+ * CS8427 datasheet, otherwise TXDSERIAL will not work.
+ */
+static void snd_cs8427_reset(struct cs8427 *chip)
+{
+ unsigned long end_time;
+ int data, aes3input = 0;
+ unsigned char val = 0;
+
+ if (snd_BUG_ON(!chip))
+ return;
+ if ((chip->regmap[CS8427_REG_CLOCKSOURCE] & CS8427_RXDAES3INPUT) ==
+ CS8427_RXDAES3INPUT) /* AES3 bit is set */
+ aes3input = 1;
+ chip->regmap[CS8427_REG_CLOCKSOURCE] &= ~(CS8427_RUN | CS8427_RXDMASK);
+ cs8427_i2c_write(chip, CS8427_REG_CLOCKSOURCE,
+ 1, &chip->regmap[CS8427_REG_CLOCKSOURCE]);
+ udelay(200);
+ chip->regmap[CS8427_REG_CLOCKSOURCE] |= CS8427_RUN | CS8427_RXDILRCK;
+ cs8427_i2c_write(chip, CS8427_REG_CLOCKSOURCE,
+ 1, &chip->regmap[CS8427_REG_CLOCKSOURCE]);
+ udelay(200);
+ end_time = jiffies + chip->reset_timeout;
+ while (time_after_eq(end_time, jiffies)) {
+ data = cs8427_i2c_read(chip, CS8427_REG_RECVERRORS,
+ 1, &val);
+ if (!(val & CS8427_UNLOCK))
+ break;
+ schedule_timeout_uninterruptible(1);
+ }
+ chip->regmap[CS8427_REG_CLOCKSOURCE] &= ~CS8427_RXDMASK;
+ if (aes3input)
+ chip->regmap[CS8427_REG_CLOCKSOURCE] |= CS8427_RXDAES3INPUT;
+ cs8427_i2c_write(chip, CS8427_REG_CLOCKSOURCE,
+ 1, &chip->regmap[CS8427_REG_CLOCKSOURCE]);
+}
+
+static int snd_cs8427_in_status_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 255;
+ return 0;
+}
+
+static int snd_cs8427_in_status_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cs8427 *chip = kcontrol->private_data;
+ unsigned char val = 0;
+ int err = 0;
+
+ err = cs8427_i2c_read(chip, kcontrol->private_value, 1, &val);
+ if (err < 0)
+ return err;
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+static int snd_cs8427_qsubcode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+ uinfo->count = 10;
+ return 0;
+}
+
+static int snd_cs8427_qsubcode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cs8427 *chip = kcontrol->private_data;
+ unsigned char reg = CS8427_REG_QSUBCODE;
+ int err;
+ unsigned char val[20];
+
+ if (!chip) {
+ pr_err("%s: invalid device info\n", __func__);
+ return -ENODEV;
+ }
+
+ err = cs8427_i2c_write(chip, reg, 1, &val[0]);
+ if (err != 1) {
+ dev_err(&chip->client->dev, "unable to send register"
+ " 0x%x byte to CS8427\n", reg);
+ return err < 0 ? err : -EIO;
+ }
+ err = cs8427_i2c_read(chip, *ucontrol->value.bytes.data, 10, &val);
+ if (err != 10) {
+ dev_err(&chip->client->dev, "unable to read"
+ " Q-subcode bytes from CS8427\n");
+ return err < 0 ? err : -EIO;
+ }
+ return 0;
+}
+
+static int snd_cs8427_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cs8427_select_corudata(struct cs8427 *cs8427_i2c, int udata)
+{
+ struct cs8427 *chip = cs8427_i2c;
+ int err;
+
+ udata = udata ? CS8427_BSEL : 0;
+ if (udata != (chip->regmap[CS8427_REG_CSDATABUF] & udata)) {
+ chip->regmap[CS8427_REG_CSDATABUF] &= ~CS8427_BSEL;
+ chip->regmap[CS8427_REG_CSDATABUF] |= udata;
+ err = cs8427_i2c_write(cs8427_i2c, CS8427_REG_CSDATABUF,
+ 1, &chip->regmap[CS8427_REG_CSDATABUF]);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_cs8427_send_corudata(struct cs8427 *obj,
+ int udata,
+ unsigned char *ndata,
+ int count)
+{
+ struct cs8427 *chip = obj;
+ char *hw_data = udata ?
+ chip->playback.hw_udata : chip->playback.hw_status;
+ char data[32];
+ int err, idx;
+ unsigned char addr = 0;
+ int ret = 0;
+
+ if (!memcmp(hw_data, ndata, count))
+ return 0;
+ err = snd_cs8427_select_corudata(chip, udata);
+ if (err < 0)
+ return err;
+ memcpy(hw_data, ndata, count);
+ if (udata) {
+ memset(data, 0, sizeof(data));
+ if (memcmp(hw_data, data, count) == 0) {
+ chip->regmap[CS8427_REG_UDATABUF] &= ~CS8427_UBMMASK;
+ chip->regmap[CS8427_REG_UDATABUF] |= CS8427_UBMZEROS |
+ CS8427_EFTUI;
+ err = cs8427_i2c_write(chip, CS8427_REG_UDATABUF,
+ 1, &chip->regmap[CS8427_REG_UDATABUF]);
+ return err < 0 ? err : 0;
+ }
+ }
+ idx = 0;
+ memcpy(data, ndata, CHANNEL_STATUS_SIZE);
+ /* address from where the bufferhas to write*/
+ addr = 0x20;
+ ret = cs8427_i2c_sendbytes(chip, &addr, data, count);
+ if (ret != count)
+ return -EIO;
+ return 1;
+}
+
+static int snd_cs8427_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cs8427 *chip = kcontrol->private_data;
+ if (!chip) {
+ pr_err("%s: invalid device info\n", __func__);
+ return -ENODEV;
+ }
+
+ memcpy(ucontrol->value.iec958.status,
+ chip->playback.def_status, CHANNEL_STATUS_SIZE);
+ return 0;
+}
+
+static int snd_cs8427_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct cs8427 *chip = kcontrol->private_data;
+ unsigned char *status;
+ int err, change;
+
+ if (!chip) {
+ pr_err("%s: invalid device info\n", __func__);
+ return -ENODEV;
+ }
+ status = kcontrol->private_value ?
+ chip->playback.pcm_status : chip->playback.def_status;
+
+ change = memcmp(ucontrol->value.iec958.status, status,
+ CHANNEL_STATUS_SIZE) != 0;
+
+ if (!change) {
+ memcpy(status, ucontrol->value.iec958.status,
+ CHANNEL_STATUS_SIZE);
+ err = snd_cs8427_send_corudata(chip, 0, status,
+ CHANNEL_STATUS_SIZE);
+ if (err < 0)
+ change = err;
+ }
+ return change;
+}
+
+static int snd_cs8427_spdif_mask_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_cs8427_spdif_mask_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ memset(ucontrol->value.iec958.status, 0xff, CHANNEL_STATUS_SIZE);
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_cs8427_iec958_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_cs8427_in_status_info,
+ .name = "IEC958 CS8427 Input Status",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .get = snd_cs8427_in_status_get,
+ .private_value = 15,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_cs8427_in_status_info,
+ .name = "IEC958 CS8427 Error Status",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .get = snd_cs8427_in_status_get,
+ .private_value = 16,
+ },
+ {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK),
+ .info = snd_cs8427_spdif_mask_info,
+ .get = snd_cs8427_spdif_mask_get,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK,
+ DEFAULT),
+ .info = snd_cs8427_spdif_info,
+ .get = snd_cs8427_spdif_get,
+ .put = snd_cs8427_spdif_put,
+ .private_value = 0
+ },
+ {
+ .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE),
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM),
+ .info = snd_cs8427_spdif_info,
+ .get = snd_cs8427_spdif_get,
+ .put = snd_cs8427_spdif_put,
+ .private_value = 1
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .info = snd_cs8427_qsubcode_info,
+ .name = "IEC958 Q-subcode Capture Default",
+ .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+ .get = snd_cs8427_qsubcode_get
+ }
+};
+
+static int cs8427_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cs8427 *chip = dev_get_drvdata(codec->dev);
+ int ret = 0;
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return -ENODEV;
+ }
+ chip->regmap[CS8427_REG_SERIALINPUT] &= CS8427_BITWIDTH_MASK;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ chip->regmap[CS8427_REG_SERIALINPUT] |= CS8427_SIRES16;
+ ret = cs8427_i2c_write(chip, CS8427_REG_SERIALINPUT, 1,
+ &chip->regmap[CS8427_REG_SERIALINPUT]);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ chip->regmap[CS8427_REG_SERIALINPUT] |= CS8427_SIRES20;
+ ret = cs8427_i2c_write(chip, CS8427_REG_SERIALINPUT, 1,
+ &chip->regmap[CS8427_REG_SERIALINPUT]);
+
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ chip->regmap[CS8427_REG_SERIALINPUT] |= CS8427_SIRES24;
+ ret = cs8427_i2c_write(chip, CS8427_REG_SERIALINPUT, 1,
+ &chip->regmap[CS8427_REG_SERIALINPUT]);
+ break;
+ default:
+ pr_err("invalid format\n");
+ break;
+ }
+ dev_dbg(&chip->client->dev,
+ "%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+ return ret;
+}
+
+static int snd_cs8427_iec958_register_kcontrol(struct cs8427 *cs8427,
+ struct snd_card *card)
+{
+ struct cs8427 *chip = cs8427;
+ struct snd_kcontrol *kctl;
+ unsigned int idx;
+ int err;
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_cs8427_iec958_controls); idx++) {
+ kctl = snd_ctl_new1(&snd_cs8427_iec958_controls[idx], chip);
+ if (kctl == NULL)
+ return -ENOMEM;
+ err = snd_ctl_add(card, kctl);
+ if (err < 0) {
+ dev_err(&chip->client->dev,
+ "failed to add the kcontrol\n");
+ return err;
+ }
+ }
+ return err;
+}
+
+static int cs8427_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct cs8427 *chip = dev_get_drvdata(dai->codec->dev);
+
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return -ENODEV;
+ }
+ /*
+ * we need to make the pll lock for the I2S tranfers
+ * reset the cs8427 chip for this.
+ */
+ snd_cs8427_reset(chip);
+ dev_dbg(&chip->client->dev,
+ "%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+
+ return 0;
+}
+
+static void cs8427_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct cs8427 *chip = dev_get_drvdata(dai->codec->dev);
+
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return;
+ }
+ dev_dbg(&chip->client->dev,
+ "%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+}
+
+static int cs8427_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct cs8427 *chip = dev_get_drvdata(dai->codec->dev);
+
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return -ENODEV;
+ }
+ dev_dbg(&chip->client->dev, "%s\n", __func__);
+ return 0;
+}
+
+static struct snd_soc_dai_ops cs8427_dai_ops = {
+ .startup = cs8427_startup,
+ .shutdown = cs8427_shutdown,
+ .hw_params = cs8427_hw_params,
+ .set_fmt = cs8427_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver cs8427_dai[] = {
+ {
+ .name = "spdif_rx",
+ .id = 1,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .rates = CS8427_RATES,
+ .formats = CS8427_FORMATS,
+ .rate_max = 192000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &cs8427_dai_ops,
+ },
+};
+
+
+static unsigned int cs8427_soc_i2c_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct cs8427 *chip = dev_get_drvdata(codec->dev);
+
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return -ENODEV;
+ }
+ dev_dbg(&chip->client->dev, "cs8427 soc i2c read\n");
+ return 0;
+}
+
+static int cs8427_soc_i2c_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ struct cs8427 *chip = dev_get_drvdata(codec->dev);
+
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return -ENODEV;
+ }
+ dev_dbg(&chip->client->dev, "cs8427 soc i2c write\n");
+ return 0;
+}
+
+static int cs8427_soc_probe(struct snd_soc_codec *codec)
+{
+ int ret = 0;
+ struct cs8427 *chip;
+ codec->control_data = dev_get_drvdata(codec->dev);
+ chip = codec->control_data;
+
+ if (chip == NULL) {
+ pr_err("invalid device private data\n");
+ return -ENODEV;
+ }
+ snd_cs8427_iec958_register_kcontrol(chip, codec->card->snd_card);
+ dev_set_drvdata(codec->dev, chip);
+ return ret;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cs8427 = {
+ .read = cs8427_soc_i2c_read,
+ .write = cs8427_soc_i2c_write,
+ .probe = cs8427_soc_probe,
+};
+
+int poweron_cs8427(struct cs8427 *chip)
+{
+ struct cs8427_platform_data *pdata = chip->client->dev.platform_data;
+ int ret = 0;
+
+ /*enable the 100KHz level shifter*/
+ if (pdata->enable) {
+ ret = pdata->enable(1);
+ if (ret < 0) {
+ dev_err(&chip->client->dev,
+ "failed to enable the level shifter\n");
+ return ret;
+ }
+ }
+
+ ret = gpio_request(pdata->reset_gpio, "cs8427 reset");
+ if (ret < 0) {
+ dev_err(&chip->client->dev,
+ "failed to request the gpio %d\n",
+ pdata->reset_gpio);
+ return ret;
+ }
+ /*bring the chip out of reset*/
+ gpio_direction_output(pdata->reset_gpio, 1);
+ msleep(20);
+ gpio_direction_output(pdata->reset_gpio, 0);
+ msleep(20);
+ gpio_direction_output(pdata->reset_gpio, 1);
+ msleep(20);
+ return ret;
+}
+
+static __devinit int cs8427_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ static unsigned char initvals1[] = {
+ CS8427_REG_CONTROL1 | CS8427_REG_AUTOINC,
+ /* CS8427_REG_CONTROL1: RMCK to OMCK, valid PCM audio, disable mutes,
+ * TCBL=output
+ */
+ CS8427_SWCLK | CS8427_TCBLDIR,
+ /* CS8427_REG_CONTROL2: hold last valid audio sample, RMCK=256*Fs,
+ * normal stereo operation
+ */
+ 0x08,
+ /* CS8427_REG_DATAFLOW:
+ * AES3 Transmitter data source => Serial Audio input port
+ * Serial audio output port data source => reserved
+ */
+ CS8427_TXDSERIAL,
+ /* CS8427_REG_CLOCKSOURCE: Run off, CMCK=256*Fs,
+ * output time base = OMCK, input time base = recovered input clock,
+ * recovered input clock source is ILRCK changed to AES3INPUT
+ * (workaround, see snd_cs8427_reset)
+ */
+ CS8427_RXDILRCK | CS8427_OUTC,
+ /* CS8427_REG_SERIALINPUT: Serial audio input port data format = I2S,
+ * 24-bit, 64*Fsi
+ */
+ CS8427_SIDEL | CS8427_SILRPOL | CS8427_SORES16,
+ /* CS8427_REG_SERIALOUTPUT: Serial audio output port data format
+ * = I2S, 24-bit, 64*Fsi
+ */
+ CS8427_SODEL | CS8427_SOLRPOL | CS8427_SIRES16,
+ };
+ static unsigned char initvals2[] = {
+ CS8427_REG_RECVERRMASK | CS8427_REG_AUTOINC,
+ /* CS8427_REG_RECVERRMASK: unmask the input PLL clock, V, confidence,
+ * biphase, parity status bits
+ * CS8427_UNLOCK | CS8427_V | CS8427_CONF | CS8427_BIP | CS8427_PAR,
+ */
+ 0xff, /* set everything */
+ /* CS8427_REG_CSDATABUF:
+ * Registers 32-55 window to CS buffer
+ * Inhibit D->E transfers from overwriting first 5 bytes of CS data.
+ * Inhibit D->E transfers (all) of CS data.
+ * Allow E->F transfer of CS data.
+ * One byte mode; both A/B channels get same written CB data.
+ * A channel info is output to chip's EMPH* pin.
+ */
+ CS8427_CBMR | CS8427_DETCI,
+ /* CS8427_REG_UDATABUF:
+ * Use internal buffer to transmit User (U) data.
+ * Chip's U pin is an output.
+ * Transmit all O's for user data.
+ * Inhibit D->E transfers.
+ * Inhibit E->F transfers.
+ */
+ CS8427_UD | CS8427_EFTUI | CS8427_DETUI,
+ };
+ int err;
+ unsigned char buf[CHANNEL_STATUS_SIZE];
+ unsigned char val = 0;
+ char addr = 0;
+ unsigned int reset_timeout = 100;
+ int ret = 0;
+ struct cs8427 *chip;
+
+ if (!client) {
+ pr_err("%s: invalid device info\n", __func__);
+ return -EINVAL;
+ }
+
+ chip = kzalloc(sizeof(struct cs8427), GFP_KERNEL);
+ if (chip == NULL) {
+ dev_err(&client->dev,
+ "%s: error, allocation failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ chip->client = client;
+
+ dev_set_drvdata(&chip->client->dev, chip);
+
+ ret = poweron_cs8427(chip);
+
+ if (ret) {
+ dev_err(&chip->client->dev,
+ "failed to bring chip out of reset\n");
+ return -ENODEV;
+ }
+
+ err = cs8427_i2c_read(chip, CS8427_REG_ID_AND_VER, 1, &val);
+ if (err < 0) {
+ /* give second chance */
+ dev_err(&chip->client->dev,
+ "failed to read cs8427 trying once again\n");
+ err = cs8427_i2c_read(chip, CS8427_REG_ID_AND_VER,
+ 1, &val);
+ if (err < 0) {
+ dev_err(&chip->client->dev,
+ "failed to read version number\n");
+ return -ENODEV;
+ }
+ dev_dbg(&chip->client->dev,
+ "version number read = %x\n", val);
+ }
+ if (val != CS8427_VER8427A) {
+ dev_err(&chip->client->dev,
+ "unable to find CS8427 signature "
+ "(expected 0x%x, read 0x%x),\n",
+ CS8427_VER8427A, val);
+ dev_err(&chip->client->dev,
+ " initialization is not completed\n");
+ return -EFAULT;
+ }
+ val = 0;
+ /* turn off run bit while making changes to configuration */
+ err = cs8427_i2c_write(chip, CS8427_REG_CLOCKSOURCE, 1, &val);
+ if (err < 0)
+ goto __fail;
+ /* send initial values */
+ memcpy(chip->regmap + (initvals1[0] & 0x7f), initvals1 + 1, 6);
+ addr = 1;
+ err = cs8427_i2c_sendbytes(chip, &addr, &initvals1[1], 6);
+ if (err != 6) {
+ err = err < 0 ? err : -EIO;
+ goto __fail;
+ }
+ /* Turn off CS8427 interrupt stuff that is not used in hardware */
+ memset(buf, 0, 7);
+ /* from address 9 to 15 */
+ addr = 9;
+ err = cs8427_i2c_sendbytes(chip, &addr, buf, 7);
+ if (err != 7)
+ goto __fail;
+ /* send transfer initialization sequence */
+ addr = 0x11;
+ memcpy(chip->regmap + (initvals2[0] & 0x7f), initvals2 + 1, 3);
+ err = cs8427_i2c_sendbytes(chip, &addr, &initvals2[1], 3);
+ if (err != 3) {
+ err = err < 0 ? err : -EIO;
+ goto __fail;
+ }
+ /* write default channel status bytes */
+ put_unaligned_le32(SNDRV_PCM_DEFAULT_CON_SPDIF, buf);
+ memset(buf + 4, 0, CHANNEL_STATUS_SIZE - 4);
+ if (snd_cs8427_send_corudata(chip, 0, buf, CHANNEL_STATUS_SIZE) < 0)
+ goto __fail;
+ memcpy(chip->playback.def_status, buf, CHANNEL_STATUS_SIZE);
+ memcpy(chip->playback.pcm_status, buf, CHANNEL_STATUS_SIZE);
+
+ /* turn on run bit and rock'n'roll */
+ if (reset_timeout < 1)
+ reset_timeout = 1;
+ chip->reset_timeout = reset_timeout;
+ snd_cs8427_reset(chip);
+
+ ret = snd_soc_register_codec(&chip->client->dev, &soc_codec_dev_cs8427,
+ cs8427_dai, ARRAY_SIZE(cs8427_dai));
+
+ return 0;
+
+__fail:
+ kfree(chip);
+ return err < 0 ? err : -EIO;
+}
+
+static int __devexit cs8427_remove(struct i2c_client *client)
+{
+ struct cs8427 *chip;
+ struct cs8427_platform_data *pdata;
+ chip = dev_get_drvdata(&client->dev);
+ if (!chip) {
+ pr_err("invalid device info\n");
+ return -ENODEV;
+ }
+ pdata = chip->client->dev.platform_data;
+ gpio_free(pdata->reset_gpio);
+ if (pdata->enable)
+ pdata->enable(0);
+ kfree(chip);
+ return 0;
+}
+
+static struct i2c_device_id cs8427_id_table[] = {
+ {"cs8427", CS8427_ADDR0},
+ {"cs8427", CS8427_ADDR2},
+ {"cs8427", CS8427_ADDR3},
+ {"cs8427", CS8427_ADDR4},
+ {"cs8427", CS8427_ADDR5},
+ {"cs8427", CS8427_ADDR6},
+ {"cs8427", CS8427_ADDR7},
+ {}
+};
+MODULE_DEVICE_TABLE(i2c, cs8427_id_table);
+
+static struct i2c_driver cs8427_i2c_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "cs8427-spdif",
+ },
+ .id_table = cs8427_id_table,
+ .probe = cs8427_i2c_probe,
+ .remove = __devexit_p(cs8427_remove),
+};
+
+static int __init cs8427_module_init(void)
+{
+ int ret = 0;
+ ret = i2c_add_driver(&cs8427_i2c_driver);
+ if (ret != 0)
+ pr_err("failed to add the I2C driver\n");
+ return ret;
+}
+
+static void __exit cs8427_module_exit(void)
+{
+ pr_info("module exit\n");
+}
+
+module_init(cs8427_module_init)
+module_exit(cs8427_module_exit)
+
+MODULE_DESCRIPTION("CS8427 interface driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/msm_stub.c b/sound/soc/codecs/msm_stub.c
new file mode 100644
index 0000000..0a3157f
--- /dev/null
+++ b/sound/soc/codecs/msm_stub.c
@@ -0,0 +1,80 @@
+/* Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+/* A dummy driver useful only to advertise hardware parameters */
+static struct snd_soc_dai_driver msm_stub_dais[] = {
+ {
+ .name = "msm-stub-rx",
+ .playback = { /* Support maximum range */
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+ {
+ .name = "msm-stub-tx",
+ .capture = { /* Support maximum range */
+ .stream_name = "Record",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+};
+
+static struct snd_soc_codec_driver soc_msm_stub = {};
+
+static int __devinit msm_stub_dev_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev,
+ &soc_msm_stub, msm_stub_dais, ARRAY_SIZE(msm_stub_dais));
+}
+
+static int __devexit msm_stub_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_stub_driver = {
+ .driver = {
+ .name = "msm-stub-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_stub_dev_probe,
+ .remove = __devexit_p(msm_stub_dev_remove),
+};
+
+static int __init msm_stub_init(void)
+{
+ return platform_driver_register(&msm_stub_driver);
+}
+module_init(msm_stub_init);
+
+static void __exit msm_stub_exit(void)
+{
+ platform_driver_unregister(&msm_stub_driver);
+}
+module_exit(msm_stub_exit);
+
+MODULE_DESCRIPTION("Generic MSM CODEC driver");
+MODULE_VERSION("1.0");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/timpani.c b/sound/soc/codecs/timpani.c
new file mode 100644
index 0000000..786b2d6
--- /dev/null
+++ b/sound/soc/codecs/timpani.c
@@ -0,0 +1,482 @@
+/* Copyright (c) 2010, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/err.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/marimba.h>
+#include <linux/mfd/timpani-audio.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+/* Debug purpose */
+#include <linux/gpio.h>
+#include <linux/clk.h>
+#include <mach/mpp.h>
+/* End of debug purpose */
+
+#define ADIE_CODEC_MAX 2
+
+struct adie_codec_register {
+ u8 reg;
+ u8 mask;
+ u8 val;
+};
+
+static struct adie_codec_register dmic_on[] = {
+ {0x80, 0x05, 0x05},
+ {0x80, 0x05, 0x00},
+ {0x83, 0x0C, 0x00},
+ {0x8A, 0xF0, 0x30},
+ {0x86, 0xFF, 0xAC},
+ {0x87, 0xFF, 0xAC},
+ {0x8A, 0xF0, 0xF0},
+ {0x82, 0x1F, 0x1E},
+ {0x83, 0x0C, 0x0C},
+ {0x92, 0x3F, 0x21},
+ {0x94, 0x3F, 0x24},
+ {0xA3, 0x39, 0x01},
+ {0xA8, 0x0F, 0x00},
+ {0xAB, 0x3F, 0x00},
+ {0x86, 0xFF, 0x00},
+ {0x87, 0xFF, 0x00},
+ {0x8A, 0xF0, 0xC0},
+};
+
+static struct adie_codec_register dmic_off[] = {
+ {0x8A, 0xF0, 0xF0},
+ {0x83, 0x0C, 0x00},
+ {0x92, 0xFF, 0x00},
+ {0x94, 0xFF, 0x1B},
+};
+
+static struct adie_codec_register spk_on[] = {
+ {0x80, 0x02, 0x02},
+ {0x80, 0x02, 0x00},
+ {0x83, 0x03, 0x00},
+ {0x8A, 0x0F, 0x03},
+ {0xA3, 0x02, 0x02},
+ {0x84, 0xFF, 0x00},
+ {0x85, 0xFF, 0x00},
+ {0x8A, 0x0F, 0x0C},
+ {0x81, 0xFF, 0x0E},
+ {0x83, 0x03, 0x03},
+ {0x24, 0x6F, 0x6C},
+ {0xB7, 0x01, 0x01},
+ {0x31, 0x01, 0x01},
+ {0x32, 0xF8, 0x08},
+ {0x32, 0xF8, 0x48},
+ {0x32, 0xF8, 0xF8},
+ {0xE0, 0xFE, 0xAC},
+ {0xE1, 0xFE, 0xAC},
+ {0x3A, 0x24, 0x24},
+ {0xE0, 0xFE, 0x3C},
+ {0xE1, 0xFE, 0x3C},
+ {0xE0, 0xFE, 0x1C},
+ {0xE1, 0xFE, 0x1C},
+ {0xE0, 0xFE, 0x10},
+ {0xE1, 0xFE, 0x10},
+};
+
+static struct adie_codec_register spk_off[] = {
+ {0x8A, 0x0F, 0x0F},
+ {0xE0, 0xFE, 0x1C},
+ {0xE1, 0xFE, 0x1C},
+ {0xE0, 0xFE, 0x3C},
+ {0xE1, 0xFE, 0x3C},
+ {0xE0, 0xFC, 0xAC},
+ {0xE1, 0xFC, 0xAC},
+ {0x32, 0xF8, 0x00},
+ {0x31, 0x05, 0x00},
+ {0x3A, 0x24, 0x00},
+};
+
+static struct adie_codec_register spk_mute[] = {
+ {0x84, 0xFF, 0xAC},
+ {0x85, 0xFF, 0xAC},
+ {0x8A, 0x0F, 0x0C},
+};
+
+static struct adie_codec_register spk_unmute[] = {
+ {0x84, 0xFF, 0x00},
+ {0x85, 0xFF, 0x00},
+ {0x8A, 0x0F, 0x0C},
+};
+
+struct adie_codec_path {
+ int rate; /* sample rate of path */
+ u32 reg_owner;
+};
+
+struct timpani_drv_data { /* member undecided */
+ struct snd_soc_codec codec;
+ struct adie_codec_path path[ADIE_CODEC_MAX];
+ u32 ref_cnt;
+ struct marimba_codec_platform_data *codec_pdata;
+};
+
+static struct snd_soc_codec *timpani_codec;
+
+enum /* regaccess blk id */
+{
+ RA_BLOCK_RX1 = 0,
+ RA_BLOCK_RX2,
+ RA_BLOCK_TX1,
+ RA_BLOCK_TX2,
+ RA_BLOCK_LB,
+ RA_BLOCK_SHARED_RX_LB,
+ RA_BLOCK_SHARED_TX,
+ RA_BLOCK_TXFE1,
+ RA_BLOCK_TXFE2,
+ RA_BLOCK_PA_COMMON,
+ RA_BLOCK_PA_EAR,
+ RA_BLOCK_PA_HPH,
+ RA_BLOCK_PA_LINE,
+ RA_BLOCK_PA_AUX,
+ RA_BLOCK_ADC,
+ RA_BLOCK_DMIC,
+ RA_BLOCK_TX_I2S,
+ RA_BLOCK_DRV,
+ RA_BLOCK_TEST,
+ RA_BLOCK_RESERVED,
+ RA_BLOCK_NUM,
+};
+
+enum /* regaccess onwer ID */
+{
+ RA_OWNER_NONE = 0,
+ RA_OWNER_PATH_RX1,
+ RA_OWNER_PATH_RX2,
+ RA_OWNER_PATH_TX1,
+ RA_OWNER_PATH_TX2,
+ RA_OWNER_PATH_LB,
+ RA_OWNER_DRV,
+ RA_OWNER_NUM,
+};
+
+struct reg_acc_blk_cfg {
+ u8 valid_owners[RA_OWNER_NUM];
+};
+
+struct timpani_regaccess {
+ u8 reg_addr;
+ u8 blk_mask[RA_BLOCK_NUM];
+ u8 reg_mask;
+ u8 reg_default;
+};
+
+static unsigned int timpani_codec_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ struct marimba *pdrv = codec->control_data;
+ int rc;
+ u8 val;
+
+ rc = marimba_read(pdrv, reg, &val, 1);
+ if (IS_ERR_VALUE(rc)) {
+ pr_err("%s: fail to write reg %x\n", __func__, reg);
+ return 0;
+ }
+ return val;
+}
+
+static int timpani_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ struct marimba *pdrv = codec->control_data;
+ int rc;
+
+ rc = marimba_write_bit_mask(pdrv, reg, (u8 *)&value, 1, 0xFF);
+ if (IS_ERR_VALUE(rc)) {
+ pr_err("%s: fail to write reg %x\n", __func__, reg);
+ return -EIO;
+ }
+ pr_debug("%s: write reg %x val %x\n", __func__, reg, value);
+ return 0;
+}
+
+static void timpani_codec_bring_up(struct snd_soc_codec *codec)
+{
+ struct timpani_drv_data *timpani = snd_soc_codec_get_drvdata(codec);
+ int rc;
+
+ if (timpani->codec_pdata &&
+ timpani->codec_pdata->marimba_codec_power) {
+ if (timpani->ref_cnt)
+ return;
+ /* Codec power up sequence */
+ rc = timpani->codec_pdata->marimba_codec_power(1);
+ if (rc)
+ pr_err("%s: could not power up timpani "
+ "codec\n", __func__);
+ else {
+ timpani_codec_write(codec, 0xFF, 0x08);
+ timpani_codec_write(codec, 0xFF, 0x0A);
+ timpani_codec_write(codec, 0xFF, 0x0E);
+ timpani_codec_write(codec, 0xFF, 0x07);
+ timpani_codec_write(codec, 0xFF, 0x17);
+ timpani_codec_write(codec, TIMPANI_A_MREF, 0x22);
+ msleep(15);
+ timpani->ref_cnt++;
+ }
+ }
+}
+
+static void timpani_codec_bring_down(struct snd_soc_codec *codec)
+{
+ struct timpani_drv_data *timpani = snd_soc_codec_get_drvdata(codec);
+ int rc;
+
+ if (timpani->codec_pdata &&
+ timpani->codec_pdata->marimba_codec_power) {
+ timpani->ref_cnt--;
+ if (timpani->ref_cnt >= 1)
+ return;
+ timpani_codec_write(codec, TIMPANI_A_MREF, TIMPANI_MREF_POR);
+ timpani_codec_write(codec, 0xFF, 0x07);
+ timpani_codec_write(codec, 0xFF, 0x06);
+ timpani_codec_write(codec, 0xFF, 0x0E);
+ timpani_codec_write(codec, 0xFF, 0x08);
+ rc = timpani->codec_pdata->marimba_codec_power(0);
+ if (rc)
+ pr_err("%s: could not power down timpani "
+ "codec\n", __func__);
+ }
+}
+
+static void timpani_dmic_config(struct snd_soc_codec *codec, int on)
+{
+ struct adie_codec_register *regs;
+ int regs_sz, i;
+
+ if (on) {
+ regs = dmic_on;
+ regs_sz = ARRAY_SIZE(dmic_on);
+ } else {
+ regs = dmic_off;
+ regs_sz = ARRAY_SIZE(dmic_off);
+ }
+
+ for (i = 0; i < regs_sz; i++)
+ timpani_codec_write(codec, regs[i].reg,
+ (regs[i].mask & regs[i].val));
+}
+
+static void timpani_spk_config(struct snd_soc_codec *codec, int on)
+{
+ struct adie_codec_register *regs;
+ int regs_sz, i;
+
+ if (on) {
+ regs = spk_on;
+ regs_sz = ARRAY_SIZE(spk_on);
+ } else {
+ regs = spk_off;
+ regs_sz = ARRAY_SIZE(spk_off);
+ }
+
+ for (i = 0; i < regs_sz; i++)
+ timpani_codec_write(codec, regs[i].reg,
+ (regs[i].mask & regs[i].val));
+}
+
+static int timpani_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ pr_info("%s()\n", __func__);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_info("%s: playback\n", __func__);
+ timpani_codec_bring_up(codec);
+ timpani_spk_config(codec, 1);
+ } else {
+ pr_info("%s: Capture\n", __func__);
+ timpani_codec_bring_up(codec);
+ timpani_dmic_config(codec, 1);
+ }
+ return 0;
+}
+
+static void timpani_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ pr_info("%s()\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ timpani_codec_bring_down(codec);
+ timpani_spk_config(codec, 0);
+ } else {
+ timpani_codec_bring_down(codec);
+ timpani_dmic_config(codec, 0);
+ }
+ return;
+}
+
+int digital_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adie_codec_register *regs;
+ int regs_sz, i;
+
+ if (mute) {
+ regs = spk_mute;
+ regs_sz = ARRAY_SIZE(spk_mute);
+ } else {
+ regs = spk_unmute;
+ regs_sz = ARRAY_SIZE(spk_unmute);
+ }
+
+ for (i = 0; i < regs_sz; i++) {
+ timpani_codec_write(codec, regs[i].reg,
+ (regs[i].mask & regs[i].val));
+ msleep(10);
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops timpani_dai_ops = {
+ .startup = timpani_startup,
+ .shutdown = timpani_shutdown,
+};
+
+struct snd_soc_dai timpani_codec_dai[] = {
+ {
+ .name = "TIMPANI Rx",
+ .playback = {
+ .stream_name = "Handset Playback",
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_max = 96000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &timpani_dai_ops,
+ },
+ {
+ .name = "TIMPANI Tx",
+ .capture = {
+ .stream_name = "Handset Capture",
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_max = 96000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &timpani_dai_ops,
+ }
+};
+
+static int timpani_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (!timpani_codec) {
+ dev_err(&pdev->dev, "core driver not yet probed\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = timpani_codec;
+ codec = timpani_codec;
+
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ dev_err(codec->dev, "failed to create pcms\n");
+ return ret;
+}
+
+/* power down chip */
+static int timpani_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_timpani = {
+ .probe = timpani_soc_probe,
+ .remove = timpani_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_timpani);
+
+static int timpani_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_codec *codec;
+ struct timpani_drv_data *priv;
+
+ pr_info("%s()\n", __func__);
+ priv = kzalloc(sizeof(struct timpani_drv_data), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
+ codec = &priv->codec;
+ snd_soc_codec_set_drvdata(codec, priv);
+ priv->codec_pdata = pdev->dev.platform_data;
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->name = "TIMPANI";
+ codec->owner = THIS_MODULE;
+ codec->read = timpani_codec_read;
+ codec->write = timpani_codec_write;
+ codec->dai = timpani_codec_dai;
+ codec->num_dai = ARRAY_SIZE(timpani_codec_dai);
+ codec->control_data = platform_get_drvdata(pdev);
+ timpani_codec = codec;
+
+ snd_soc_register_dais(timpani_codec_dai, ARRAY_SIZE(timpani_codec_dai));
+ snd_soc_register_codec(codec);
+
+ return 0;
+}
+
+static struct platform_driver timpani_codec_driver = {
+ .probe = timpani_codec_probe,
+ .driver = {
+ .name = "timpani_codec",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init timpani_codec_init(void)
+{
+ return platform_driver_register(&timpani_codec_driver);
+}
+
+static void __exit timpani_codec_exit(void)
+{
+ platform_driver_unregister(&timpani_codec_driver);
+}
+
+module_init(timpani_codec_init);
+module_exit(timpani_codec_exit);
+
+MODULE_DESCRIPTION("Timpani codec driver");
+MODULE_VERSION("1.0");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/timpani.h b/sound/soc/codecs/timpani.h
new file mode 100644
index 0000000..bd14eea
--- /dev/null
+++ b/sound/soc/codecs/timpani.h
@@ -0,0 +1,15 @@
+/* Copyright (c) 2010, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+#define NUM_I2S 2
+extern struct snd_soc_dai timpani_codec_dai[NUM_I2S];
+extern struct snd_soc_codec_device soc_codec_dev_timpani;
diff --git a/sound/soc/codecs/wcd9304-tables.c b/sound/soc/codecs/wcd9304-tables.c
new file mode 100644
index 0000000..252cb0e
--- /dev/null
+++ b/sound/soc/codecs/wcd9304-tables.c
@@ -0,0 +1,722 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/mfd/wcd9xxx/wcd9xxx_registers.h>
+#include <linux/mfd/wcd9xxx/wcd9304_registers.h>
+#include "wcd9304.h"
+
+const u8 sitar_reg_defaults[SITAR_CACHE_SIZE] = {
+ [WCD9XXX_A_CHIP_CTL] = WCD9XXX_A_CHIP_CTL__POR,
+ [WCD9XXX_A_CHIP_STATUS] = WCD9XXX_A_CHIP_STATUS__POR,
+ [WCD9XXX_A_CHIP_ID_BYTE_0] = WCD9XXX_A_CHIP_ID_BYTE_0__POR,
+ [WCD9XXX_A_CHIP_ID_BYTE_1] = WCD9XXX_A_CHIP_ID_BYTE_1__POR,
+ [WCD9XXX_A_CHIP_ID_BYTE_2] = WCD9XXX_A_CHIP_ID_BYTE_2__POR,
+ [WCD9XXX_A_CHIP_ID_BYTE_3] = WCD9XXX_A_CHIP_ID_BYTE_3__POR,
+ [WCD9XXX_A_CHIP_VERSION] = WCD9XXX_A_CHIP_VERSION__POR,
+ [WCD9XXX_A_SB_VERSION] = WCD9XXX_A_SB_VERSION__POR,
+ [WCD9XXX_A_SLAVE_ID_1] = WCD9XXX_A_SLAVE_ID_1__POR,
+ [WCD9XXX_A_SLAVE_ID_2] = WCD9XXX_A_SLAVE_ID_2__POR,
+ [WCD9XXX_A_SLAVE_ID_3] = WCD9XXX_A_SLAVE_ID_3__POR,
+ [SITAR_A_PIN_CTL_OE0] = SITAR_A_PIN_CTL_OE0__POR,
+ [SITAR_A_PIN_CTL_OE1] = SITAR_A_PIN_CTL_OE1__POR,
+ [SITAR_A_PIN_CTL_DATA0] = SITAR_A_PIN_CTL_DATA0__POR,
+ [SITAR_A_PIN_CTL_DATA1] = SITAR_A_PIN_CTL_DATA1__POR,
+ [SITAR_A_HDRIVE_GENERIC] = SITAR_A_HDRIVE_GENERIC__POR,
+ [SITAR_A_HDRIVE_OVERRIDE] = SITAR_A_HDRIVE_OVERRIDE__POR,
+ [SITAR_A_ANA_CSR_WAIT_STATE] = SITAR_A_ANA_CSR_WAIT_STATE__POR,
+ [SITAR_A_PROCESS_MONITOR_CTL0] = SITAR_A_PROCESS_MONITOR_CTL0__POR,
+ [SITAR_A_PROCESS_MONITOR_CTL1] = SITAR_A_PROCESS_MONITOR_CTL1__POR,
+ [SITAR_A_PROCESS_MONITOR_CTL2] = SITAR_A_PROCESS_MONITOR_CTL2__POR,
+ [SITAR_A_PROCESS_MONITOR_CTL3] = SITAR_A_PROCESS_MONITOR_CTL3__POR,
+ [SITAR_A_QFUSE_CTL] = SITAR_A_QFUSE_CTL__POR,
+ [SITAR_A_QFUSE_STATUS] = SITAR_A_QFUSE_STATUS__POR,
+ [SITAR_A_QFUSE_DATA_OUT0] = SITAR_A_QFUSE_DATA_OUT0__POR,
+ [SITAR_A_QFUSE_DATA_OUT1] = SITAR_A_QFUSE_DATA_OUT1__POR,
+ [SITAR_A_QFUSE_DATA_OUT2] = SITAR_A_QFUSE_DATA_OUT2__POR,
+ [SITAR_A_QFUSE_DATA_OUT3] = SITAR_A_QFUSE_DATA_OUT3__POR,
+ [SITAR_A_CDC_CTL] = SITAR_A_CDC_CTL__POR,
+ [SITAR_A_LEAKAGE_CTL] = SITAR_A_LEAKAGE_CTL__POR,
+ [SITAR_A_INTR_MODE] = SITAR_A_INTR_MODE__POR,
+ [SITAR_A_INTR_MASK0] = SITAR_A_INTR_MASK0__POR,
+ [SITAR_A_INTR_MASK1] = SITAR_A_INTR_MASK1__POR,
+ [SITAR_A_INTR_MASK2] = SITAR_A_INTR_MASK2__POR,
+ [SITAR_A_INTR_STATUS0] = SITAR_A_INTR_STATUS0__POR,
+ [SITAR_A_INTR_STATUS1] = SITAR_A_INTR_STATUS1__POR,
+ [SITAR_A_INTR_STATUS2] = SITAR_A_INTR_STATUS2__POR,
+ [SITAR_A_INTR_CLEAR0] = SITAR_A_INTR_CLEAR0__POR,
+ [SITAR_A_INTR_CLEAR1] = SITAR_A_INTR_CLEAR1__POR,
+ [SITAR_A_INTR_CLEAR2] = SITAR_A_INTR_CLEAR2__POR,
+ [SITAR_A_INTR_LEVEL0] = SITAR_A_INTR_LEVEL0__POR,
+ [SITAR_A_INTR_LEVEL1] = SITAR_A_INTR_LEVEL1__POR,
+ [SITAR_A_INTR_LEVEL2] = SITAR_A_INTR_LEVEL2__POR,
+ [SITAR_A_INTR_TEST0] = SITAR_A_INTR_TEST0__POR,
+ [SITAR_A_INTR_TEST1] = SITAR_A_INTR_TEST1__POR,
+ [SITAR_A_INTR_TEST2] = SITAR_A_INTR_TEST2__POR,
+ [SITAR_A_INTR_SET0] = SITAR_A_INTR_SET0__POR,
+ [SITAR_A_INTR_SET1] = SITAR_A_INTR_SET1__POR,
+ [SITAR_A_INTR_SET2] = SITAR_A_INTR_SET2__POR,
+ [SITAR_A_CDC_TX_I2S_SCK_MODE] = SITAR_A_CDC_TX_I2S_SCK_MODE__POR,
+ [SITAR_A_CDC_TX_I2S_WS_MODE] = SITAR_A_CDC_TX_I2S_WS_MODE__POR,
+ [SITAR_A_CDC_DMIC_DATA0_MODE] = SITAR_A_CDC_DMIC_DATA0_MODE__POR,
+ [SITAR_A_CDC_DMIC_CLK0_MODE] = SITAR_A_CDC_DMIC_CLK0_MODE__POR,
+ [SITAR_A_CDC_DMIC_DATA1_MODE] = SITAR_A_CDC_DMIC_DATA1_MODE__POR,
+ [SITAR_A_CDC_DMIC_CLK1_MODE] = SITAR_A_CDC_DMIC_CLK1_MODE__POR,
+ [SITAR_A_CDC_TX_I2S_SD0_MODE] = SITAR_A_CDC_TX_I2S_SD0_MODE__POR,
+ [SITAR_A_CDC_INTR_MODE] = SITAR_A_CDC_INTR_MODE__POR,
+ [SITAR_A_CDC_RX_I2S_SD0_MODE] = SITAR_A_CDC_RX_I2S_SD0_MODE__POR,
+ [SITAR_A_CDC_RX_I2S_SD1_MODE] = SITAR_A_CDC_RX_I2S_SD1_MODE__POR,
+ [SITAR_A_BIAS_REF_CTL] = SITAR_A_BIAS_REF_CTL__POR,
+ [SITAR_A_BIAS_CENTRAL_BG_CTL] = SITAR_A_BIAS_CENTRAL_BG_CTL__POR,
+ [SITAR_A_BIAS_PRECHRG_CTL] = SITAR_A_BIAS_PRECHRG_CTL__POR,
+ [SITAR_A_BIAS_CURR_CTL_1] = SITAR_A_BIAS_CURR_CTL_1__POR,
+ [SITAR_A_BIAS_CURR_CTL_2] = SITAR_A_BIAS_CURR_CTL_2__POR,
+ [SITAR_A_BIAS_OSC_BG_CTL] = SITAR_A_BIAS_OSC_BG_CTL__POR,
+ [SITAR_A_CLK_BUFF_EN1] = SITAR_A_CLK_BUFF_EN1__POR,
+ [SITAR_A_CLK_BUFF_EN2] = SITAR_A_CLK_BUFF_EN2__POR,
+ [SITAR_A_LDO_H_MODE_1] = SITAR_A_LDO_H_MODE_1__POR,
+ [SITAR_A_LDO_H_MODE_2] = SITAR_A_LDO_H_MODE_2__POR,
+ [SITAR_A_LDO_H_LOOP_CTL] = SITAR_A_LDO_H_LOOP_CTL__POR,
+ [SITAR_A_LDO_H_COMP_1] = SITAR_A_LDO_H_COMP_1__POR,
+ [SITAR_A_LDO_H_COMP_2] = SITAR_A_LDO_H_COMP_2__POR,
+ [SITAR_A_LDO_H_BIAS_1] = SITAR_A_LDO_H_BIAS_1__POR,
+ [SITAR_A_LDO_H_BIAS_2] = SITAR_A_LDO_H_BIAS_2__POR,
+ [SITAR_A_LDO_H_BIAS_3] = SITAR_A_LDO_H_BIAS_3__POR,
+ [SITAR_A_MICB_CFILT_1_CTL] = SITAR_A_MICB_CFILT_1_CTL__POR,
+ [SITAR_A_MICB_CFILT_1_VAL] = SITAR_A_MICB_CFILT_1_VAL__POR,
+ [SITAR_A_MICB_CFILT_1_PRECHRG] = SITAR_A_MICB_CFILT_1_PRECHRG__POR,
+ [SITAR_A_MICB_1_CTL] = SITAR_A_MICB_1_CTL__POR,
+ [SITAR_A_MICB_1_INT_RBIAS] = SITAR_A_MICB_1_INT_RBIAS__POR,
+ [SITAR_A_MICB_1_MBHC] = SITAR_A_MICB_1_MBHC__POR,
+ [SITAR_A_MICB_CFILT_2_CTL] = SITAR_A_MICB_CFILT_2_CTL__POR,
+ [SITAR_A_MICB_CFILT_2_VAL] = SITAR_A_MICB_CFILT_2_VAL__POR,
+ [SITAR_A_MICB_CFILT_2_PRECHRG] = SITAR_A_MICB_CFILT_2_PRECHRG__POR,
+ [SITAR_A_MICB_2_CTL] = SITAR_A_MICB_2_CTL__POR,
+ [SITAR_A_MICB_2_INT_RBIAS] = SITAR_A_MICB_2_INT_RBIAS__POR,
+ [SITAR_A_MICB_2_MBHC] = SITAR_A_MICB_2_MBHC__POR,
+ [SITAR_A_TX_COM_BIAS] = SITAR_A_TX_COM_BIAS__POR,
+ [SITAR_A_MBHC_SCALING_MUX_1] = SITAR_A_MBHC_SCALING_MUX_1__POR,
+ [SITAR_A_MBHC_SCALING_MUX_2] = SITAR_A_MBHC_SCALING_MUX_2__POR,
+ [SITAR_A_TX_SUP_SWITCH_CTRL_1] = SITAR_A_TX_SUP_SWITCH_CTRL_1__POR,
+ [SITAR_A_TX_SUP_SWITCH_CTRL_2] = SITAR_A_TX_SUP_SWITCH_CTRL_2__POR,
+ [SITAR_A_TX_1_2_EN] = SITAR_A_TX_1_2_EN__POR,
+ [SITAR_A_TX_1_2_TEST_EN] = SITAR_A_TX_1_2_TEST_EN__POR,
+ [SITAR_A_TX_1_2_ADC_CH1] = SITAR_A_TX_1_2_ADC_CH1__POR,
+ [SITAR_A_TX_1_2_ADC_CH2] = SITAR_A_TX_1_2_ADC_CH2__POR,
+ [SITAR_A_TX_1_2_ATEST_REFCTRL] = SITAR_A_TX_1_2_ATEST_REFCTRL__POR,
+ [SITAR_A_TX_1_2_TEST_CTL] = SITAR_A_TX_1_2_TEST_CTL__POR,
+ [SITAR_A_TX_1_2_TEST_BLOCK_EN] = SITAR_A_TX_1_2_TEST_BLOCK_EN__POR,
+ [SITAR_A_TX_1_2_TXFE_CLKDIV] = SITAR_A_TX_1_2_TXFE_CLKDIV__POR,
+ [SITAR_A_TX_1_2_SAR_ERR_CH1] = SITAR_A_TX_1_2_SAR_ERR_CH1__POR,
+ [SITAR_A_TX_1_2_SAR_ERR_CH2] = SITAR_A_TX_1_2_SAR_ERR_CH2__POR,
+ [SITAR_A_TX_3_EN] = SITAR_A_TX_3_EN__POR,
+ [SITAR_A_TX_3_TEST_EN] = SITAR_A_TX_3_TEST_EN__POR,
+ [SITAR_A_TX_3_ADC] = SITAR_A_TX_3_ADC__POR,
+ [SITAR_A_TX_3_MBHC_ATEST_REFCTRL] =
+ SITAR_A_TX_3_MBHC_ATEST_REFCTRL__POR,
+ [SITAR_A_TX_3_TEST_CTL] = SITAR_A_TX_3_TEST_CTL__POR,
+ [SITAR_A_TX_3_TEST_BLOCK_EN] = SITAR_A_TX_3_TEST_BLOCK_EN__POR,
+ [SITAR_A_TX_3_TXFE_CKDIV] = SITAR_A_TX_3_TXFE_CKDIV__POR,
+ [SITAR_A_TX_3_SAR_ERR] = SITAR_A_TX_3_SAR_ERR__POR,
+ [SITAR_A_TX_4_MBHC_EN] = SITAR_A_TX_4_MBHC_EN__POR,
+ [SITAR_A_TX_4_MBHC_ADC] = SITAR_A_TX_4_MBHC_ADC__POR,
+ [SITAR_A_TX_4_MBHC_TEST_CTL] = SITAR_A_TX_4_MBHC_TEST_CTL__POR,
+ [SITAR_A_TX_4_MBHC_SAR_ERR] = SITAR_A_TX_4_MBHC_SAR_ERR__POR,
+ [SITAR_A_TX_4_TXFE_CLKDIV] = SITAR_A_TX_4_TXFE_CLKDIV__POR,
+ [SITAR_A_AUX_COM_CTL] = SITAR_A_AUX_COM_CTL__POR,
+ [SITAR_A_AUX_COM_ATEST] = SITAR_A_AUX_COM_ATEST__POR,
+ [SITAR_A_AUX_L_EN] = SITAR_A_AUX_L_EN__POR,
+ [SITAR_A_AUX_L_GAIN] = SITAR_A_AUX_L_GAIN__POR,
+ [SITAR_A_AUX_L_PA_CONN] = SITAR_A_AUX_L_PA_CONN__POR,
+ [SITAR_A_AUX_L_PA_CONN_INV] = SITAR_A_AUX_L_PA_CONN_INV__POR,
+ [SITAR_A_AUX_R_EN] = SITAR_A_AUX_R_EN__POR,
+ [SITAR_A_AUX_R_GAIN] = SITAR_A_AUX_R_GAIN__POR,
+ [SITAR_A_AUX_R_PA_CONN] = SITAR_A_AUX_R_PA_CONN__POR,
+ [SITAR_A_AUX_R_PA_CONN_INV] = SITAR_A_AUX_R_PA_CONN_INV__POR,
+ [SITAR_A_CP_EN] = SITAR_A_CP_EN__POR,
+ [SITAR_A_CP_CLK] = SITAR_A_CP_CLK__POR,
+ [SITAR_A_CP_STATIC] = SITAR_A_CP_STATIC__POR,
+ [SITAR_A_CP_DCC1] = SITAR_A_CP_DCC1__POR,
+ [SITAR_A_CP_DCC3] = SITAR_A_CP_DCC3__POR,
+ [SITAR_A_CP_ATEST] = SITAR_A_CP_ATEST__POR,
+ [SITAR_A_CP_DTEST] = SITAR_A_CP_DTEST__POR,
+ [SITAR_A_RX_COM_TIMER_DIV] = SITAR_A_RX_COM_TIMER_DIV__POR,
+ [SITAR_A_RX_COM_OCP_CTL] = SITAR_A_RX_COM_OCP_CTL__POR,
+ [SITAR_A_RX_COM_OCP_COUNT] = SITAR_A_RX_COM_OCP_COUNT__POR,
+ [SITAR_A_RX_COM_DAC_CTL] = SITAR_A_RX_COM_DAC_CTL__POR,
+ [SITAR_A_RX_COM_BIAS] = SITAR_A_RX_COM_BIAS__POR,
+ [SITAR_A_RX_HPH_BIAS_PA] = SITAR_A_RX_HPH_BIAS_PA__POR,
+ [SITAR_A_RX_HPH_BIAS_LDO] = SITAR_A_RX_HPH_BIAS_LDO__POR,
+ [SITAR_A_RX_HPH_BIAS_CNP] = SITAR_A_RX_HPH_BIAS_CNP__POR,
+ [SITAR_A_RX_HPH_BIAS_WG] = SITAR_A_RX_HPH_BIAS_WG__POR,
+ [SITAR_A_RX_HPH_OCP_CTL] = SITAR_A_RX_HPH_OCP_CTL__POR,
+ [SITAR_A_RX_HPH_CNP_EN] = SITAR_A_RX_HPH_CNP_EN__POR,
+ [SITAR_A_RX_HPH_CNP_WG_CTL] = SITAR_A_RX_HPH_CNP_WG_CTL__POR,
+ [SITAR_A_RX_HPH_CNP_WG_TIME] = SITAR_A_RX_HPH_CNP_WG_TIME__POR,
+ [SITAR_A_RX_HPH_L_GAIN] = SITAR_A_RX_HPH_L_GAIN__POR,
+ [SITAR_A_RX_HPH_L_TEST] = SITAR_A_RX_HPH_L_TEST__POR,
+ [SITAR_A_RX_HPH_L_PA_CTL] = SITAR_A_RX_HPH_L_PA_CTL__POR,
+ [SITAR_A_RX_HPH_L_DAC_CTL] = SITAR_A_RX_HPH_L_DAC_CTL__POR,
+ [SITAR_A_RX_HPH_L_ATEST] = SITAR_A_RX_HPH_L_ATEST__POR,
+ [SITAR_A_RX_HPH_L_STATUS] = SITAR_A_RX_HPH_L_STATUS__POR,
+ [SITAR_A_RX_HPH_R_GAIN] = SITAR_A_RX_HPH_R_GAIN__POR,
+ [SITAR_A_RX_HPH_R_TEST] = SITAR_A_RX_HPH_R_TEST__POR,
+ [SITAR_A_RX_HPH_R_PA_CTL] = SITAR_A_RX_HPH_R_PA_CTL__POR,
+ [SITAR_A_RX_HPH_R_DAC_CTL] = SITAR_A_RX_HPH_R_DAC_CTL__POR,
+ [SITAR_A_RX_HPH_R_ATEST] = SITAR_A_RX_HPH_R_ATEST__POR,
+ [SITAR_A_RX_HPH_R_STATUS] = SITAR_A_RX_HPH_R_STATUS__POR,
+ [SITAR_A_RX_EAR_BIAS_PA] = SITAR_A_RX_EAR_BIAS_PA__POR,
+ [SITAR_A_RX_EAR_BIAS_CMBUFF] = SITAR_A_RX_EAR_BIAS_CMBUFF__POR,
+ [SITAR_A_RX_EAR_EN] = SITAR_A_RX_EAR_EN__POR,
+ [SITAR_A_RX_EAR_GAIN] = SITAR_A_RX_EAR_GAIN__POR,
+ [SITAR_A_RX_EAR_CMBUFF] = SITAR_A_RX_EAR_CMBUFF__POR,
+ [SITAR_A_RX_EAR_ICTL] = SITAR_A_RX_EAR_ICTL__POR,
+ [SITAR_A_RX_EAR_CCOMP] = SITAR_A_RX_EAR_CCOMP__POR,
+ [SITAR_A_RX_EAR_VCM] = SITAR_A_RX_EAR_VCM__POR,
+ [SITAR_A_RX_EAR_CNP] = SITAR_A_RX_EAR_CNP__POR,
+ [SITAR_A_RX_EAR_ATEST] = SITAR_A_RX_EAR_ATEST__POR,
+ [SITAR_A_RX_EAR_STATUS] = SITAR_A_RX_EAR_STATUS__POR,
+ [SITAR_A_RX_LINE_BIAS_PA] = SITAR_A_RX_LINE_BIAS_PA__POR,
+ [SITAR_A_RX_LINE_BIAS_LDO] = SITAR_A_RX_LINE_BIAS_LDO__POR,
+ [SITAR_A_RX_LINE_BIAS_CNP1] = SITAR_A_RX_LINE_BIAS_CNP1__POR,
+ [SITAR_A_RX_LINE_COM] = SITAR_A_RX_LINE_COM__POR,
+ [SITAR_A_RX_LINE_CNP_EN] = SITAR_A_RX_LINE_CNP_EN__POR,
+ [SITAR_A_RX_LINE_CNP_WG_CTL] = SITAR_A_RX_LINE_CNP_WG_CTL__POR,
+ [SITAR_A_RX_LINE_CNP_WG_TIME] = SITAR_A_RX_LINE_CNP_WG_TIME__POR,
+ [SITAR_A_RX_LINE_1_GAIN] = SITAR_A_RX_LINE_1_GAIN__POR,
+ [SITAR_A_RX_LINE_1_TEST] = SITAR_A_RX_LINE_1_TEST__POR,
+ [SITAR_A_RX_LINE_1_DAC_CTL] = SITAR_A_RX_LINE_1_DAC_CTL__POR,
+ [SITAR_A_RX_LINE_1_STATUS] = SITAR_A_RX_LINE_1_STATUS__POR,
+ [SITAR_A_RX_LINE_2_GAIN] = SITAR_A_RX_LINE_2_GAIN__POR,
+ [SITAR_A_RX_LINE_2_TEST] = SITAR_A_RX_LINE_2_TEST__POR,
+ [SITAR_A_RX_LINE_2_DAC_CTL] = SITAR_A_RX_LINE_2_DAC_CTL__POR,
+ [SITAR_A_RX_LINE_2_STATUS] = SITAR_A_RX_LINE_2_STATUS__POR,
+ [SITAR_A_RX_LINE_BIAS_CNP2] = SITAR_A_RX_LINE_BIAS_CNP2__POR,
+ [SITAR_A_RX_LINE_OCP_CTL] = SITAR_A_RX_LINE_OCP_CTL__POR,
+ [SITAR_A_RX_LINE_1_PA_CTL] = SITAR_A_RX_LINE_1_PA_CTL__POR,
+ [SITAR_A_RX_LINE_2_PA_CTL] = SITAR_A_RX_LINE_2_PA_CTL__POR,
+ [SITAR_A_RX_LINE_CNP_DBG] = SITAR_A_RX_LINE_CNP_DBG__POR,
+ [SITAR_A_MBHC_HPH] = SITAR_A_MBHC_HPH__POR,
+ [SITAR_A_RC_OSC_FREQ] = SITAR_A_RC_OSC_FREQ__POR,
+ [SITAR_A_RC_OSC_TEST] = SITAR_A_RC_OSC_TEST__POR,
+ [SITAR_A_RC_OSC_STATUS] = SITAR_A_RC_OSC_STATUS__POR,
+ [SITAR_A_RC_OSC_TUNER] = SITAR_A_RC_OSC_TUNER__POR,
+ [SITAR_A_CDC_ANC1_CTL] = SITAR_A_CDC_ANC1_CTL__POR,
+ [SITAR_A_CDC_ANC1_SHIFT] = SITAR_A_CDC_ANC1_SHIFT__POR,
+ [SITAR_A_CDC_ANC1_IIR_B1_CTL] = SITAR_A_CDC_ANC1_IIR_B1_CTL__POR,
+ [SITAR_A_CDC_ANC1_IIR_B2_CTL] = SITAR_A_CDC_ANC1_IIR_B2_CTL__POR,
+ [SITAR_A_CDC_ANC1_IIR_B3_CTL] = SITAR_A_CDC_ANC1_IIR_B3_CTL__POR,
+ [SITAR_A_CDC_ANC1_IIR_B4_CTL] = SITAR_A_CDC_ANC1_IIR_B4_CTL__POR,
+ [SITAR_A_CDC_ANC1_LPF_B1_CTL] = SITAR_A_CDC_ANC1_LPF_B1_CTL__POR,
+ [SITAR_A_CDC_ANC1_LPF_B2_CTL] = SITAR_A_CDC_ANC1_LPF_B2_CTL__POR,
+ [SITAR_A_CDC_ANC1_LPF_B3_CTL] = SITAR_A_CDC_ANC1_LPF_B3_CTL__POR,
+ [SITAR_A_CDC_ANC1_SPARE] = SITAR_A_CDC_ANC1_SPARE__POR,
+ [SITAR_A_CDC_ANC1_SMLPF_CTL] = SITAR_A_CDC_ANC1_SMLPF_CTL__POR,
+ [SITAR_A_CDC_ANC1_DCFLT_CTL] = SITAR_A_CDC_ANC1_DCFLT_CTL__POR,
+ [SITAR_A_CDC_TX1_VOL_CTL_TIMER] = SITAR_A_CDC_TX1_VOL_CTL_TIMER__POR,
+ [SITAR_A_CDC_TX1_VOL_CTL_GAIN] = SITAR_A_CDC_TX1_VOL_CTL_GAIN__POR,
+ [SITAR_A_CDC_TX1_VOL_CTL_CFG] = SITAR_A_CDC_TX1_VOL_CTL_CFG__POR,
+ [SITAR_A_CDC_TX1_MUX_CTL] = SITAR_A_CDC_TX1_MUX_CTL__POR,
+ [SITAR_A_CDC_TX1_CLK_FS_CTL] = SITAR_A_CDC_TX1_CLK_FS_CTL__POR,
+ [SITAR_A_CDC_TX1_DMIC_CTL] = SITAR_A_CDC_TX1_DMIC_CTL__POR,
+ [SITAR_A_CDC_SRC1_PDA_CFG] = SITAR_A_CDC_SRC1_PDA_CFG__POR,
+ [SITAR_A_CDC_SRC1_FS_CTL] = SITAR_A_CDC_SRC1_FS_CTL__POR,
+ [SITAR_A_CDC_RX1_B1_CTL] = SITAR_A_CDC_RX1_B1_CTL__POR,
+ [SITAR_A_CDC_RX1_B2_CTL] = SITAR_A_CDC_RX1_B2_CTL__POR,
+ [SITAR_A_CDC_RX1_B3_CTL] = SITAR_A_CDC_RX1_B3_CTL__POR,
+ [SITAR_A_CDC_RX1_B4_CTL] = SITAR_A_CDC_RX1_B4_CTL__POR,
+ [SITAR_A_CDC_RX1_B5_CTL] = SITAR_A_CDC_RX1_B5_CTL__POR,
+ [SITAR_A_CDC_RX1_B6_CTL] = SITAR_A_CDC_RX1_B6_CTL__POR,
+ [SITAR_A_CDC_RX1_VOL_CTL_B1_CTL] = SITAR_A_CDC_RX1_VOL_CTL_B1_CTL__POR,
+ [SITAR_A_CDC_RX1_VOL_CTL_B2_CTL] = SITAR_A_CDC_RX1_VOL_CTL_B2_CTL__POR,
+ [SITAR_A_CDC_CLK_ANC_RESET_CTL] = SITAR_A_CDC_CLK_ANC_RESET_CTL__POR,
+ [SITAR_A_CDC_CLK_RX_RESET_CTL] = SITAR_A_CDC_CLK_RX_RESET_CTL__POR,
+ [SITAR_A_CDC_CLK_TX_RESET_B1_CTL] =
+ SITAR_A_CDC_CLK_TX_RESET_B1_CTL__POR,
+ [SITAR_A_CDC_CLK_TX_RESET_B2_CTL] =
+ SITAR_A_CDC_CLK_TX_RESET_B2_CTL__POR,
+ [SITAR_A_CDC_CLK_DMIC_CTL] = SITAR_A_CDC_CLK_DMIC_CTL__POR,
+ [SITAR_A_CDC_CLK_RX_I2S_CTL] = SITAR_A_CDC_CLK_RX_I2S_CTL__POR,
+ [SITAR_A_CDC_CLK_TX_I2S_CTL] = SITAR_A_CDC_CLK_TX_I2S_CTL__POR,
+ [SITAR_A_CDC_CLK_OTHR_RESET_CTL] = SITAR_A_CDC_CLK_OTHR_RESET_CTL__POR,
+ [SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL] =
+ SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL__POR,
+ [SITAR_A_CDC_CLK_OTHR_CTL] = SITAR_A_CDC_CLK_OTHR_CTL__POR,
+ [SITAR_A_CDC_CLK_RDAC_CLK_EN_CTL] =
+ SITAR_A_CDC_CLK_RDAC_CLK_EN_CTL__POR,
+ [SITAR_A_CDC_CLK_ANC_CLK_EN_CTL] = SITAR_A_CDC_CLK_ANC_CLK_EN_CTL__POR,
+ [SITAR_A_CDC_CLK_RX_B1_CTL] = SITAR_A_CDC_CLK_RX_B1_CTL__POR,
+ [SITAR_A_CDC_CLK_RX_B2_CTL] = SITAR_A_CDC_CLK_RX_B2_CTL__POR,
+ [SITAR_A_CDC_CLK_MCLK_CTL] = SITAR_A_CDC_CLK_MCLK_CTL__POR,
+ [SITAR_A_CDC_CLK_PDM_CTL] = SITAR_A_CDC_CLK_PDM_CTL__POR,
+ [SITAR_A_CDC_CLK_SD_CTL] = SITAR_A_CDC_CLK_SD_CTL__POR,
+ [SITAR_A_CDC_CLK_LP_CTL] = SITAR_A_CDC_CLK_LP_CTL__POR,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B1_CTL] =
+ SITAR_A_CDC_CLSG_FREQ_THRESH_B1_CTL__POR,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B2_CTL] =
+ SITAR_A_CDC_CLSG_FREQ_THRESH_B2_CTL__POR,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B3_CTL] =
+ SITAR_A_CDC_CLSG_FREQ_THRESH_B3_CTL__POR,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B4_CTL] =
+ SITAR_A_CDC_CLSG_FREQ_THRESH_B4_CTL__POR,
+ [SITAR_A_CDC_CLSG_GAIN_THRESH_CTL] =
+ SITAR_A_CDC_CLSG_GAIN_THRESH_CTL__POR,
+ [SITAR_A_CDC_CLSG_TIMER_B1_CFG] = SITAR_A_CDC_CLSG_TIMER_B1_CFG__POR,
+ [SITAR_A_CDC_CLSG_TIMER_B2_CFG] = SITAR_A_CDC_CLSG_TIMER_B2_CFG__POR,
+ [SITAR_A_CDC_CLSG_CTL] = SITAR_A_CDC_CLSG_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B1_CTL] = SITAR_A_CDC_IIR1_GAIN_B1_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B2_CTL] = SITAR_A_CDC_IIR1_GAIN_B2_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B3_CTL] = SITAR_A_CDC_IIR1_GAIN_B3_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B4_CTL] = SITAR_A_CDC_IIR1_GAIN_B4_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B5_CTL] = SITAR_A_CDC_IIR1_GAIN_B5_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B6_CTL] = SITAR_A_CDC_IIR1_GAIN_B6_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B7_CTL] = SITAR_A_CDC_IIR1_GAIN_B7_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_B8_CTL] = SITAR_A_CDC_IIR1_GAIN_B8_CTL__POR,
+ [SITAR_A_CDC_IIR1_CTL] = SITAR_A_CDC_IIR1_CTL__POR,
+ [SITAR_A_CDC_IIR1_GAIN_TIMER_CTL] =
+ SITAR_A_CDC_IIR1_GAIN_TIMER_CTL__POR,
+ [SITAR_A_CDC_IIR1_COEF_B1_CTL] = SITAR_A_CDC_IIR1_COEF_B1_CTL__POR,
+ [SITAR_A_CDC_IIR1_COEF_B2_CTL] = SITAR_A_CDC_IIR1_COEF_B2_CTL__POR,
+ [SITAR_A_CDC_IIR1_COEF_B3_CTL] = SITAR_A_CDC_IIR1_COEF_B3_CTL__POR,
+ [SITAR_A_CDC_IIR1_COEF_B4_CTL] = SITAR_A_CDC_IIR1_COEF_B4_CTL__POR,
+ [SITAR_A_CDC_IIR1_COEF_B5_CTL] = SITAR_A_CDC_IIR1_COEF_B5_CTL__POR,
+ [SITAR_A_CDC_TOP_GAIN_UPDATE] = SITAR_A_CDC_TOP_GAIN_UPDATE__POR,
+ [SITAR_A_CDC_TOP_RDAC_DOUT_CTL] = SITAR_A_CDC_TOP_RDAC_DOUT_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B1_CTL] = SITAR_A_CDC_DEBUG_B1_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B2_CTL] = SITAR_A_CDC_DEBUG_B2_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B3_CTL] = SITAR_A_CDC_DEBUG_B3_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B4_CTL] = SITAR_A_CDC_DEBUG_B4_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B5_CTL] = SITAR_A_CDC_DEBUG_B5_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B6_CTL] = SITAR_A_CDC_DEBUG_B6_CTL__POR,
+ [SITAR_A_CDC_DEBUG_B7_CTL] = SITAR_A_CDC_DEBUG_B7_CTL__POR,
+ [SITAR_A_CDC_COMP1_B1_CTL] = SITAR_A_CDC_COMP1_B1_CTL__POR,
+ [SITAR_A_CDC_COMP1_B2_CTL] = SITAR_A_CDC_COMP1_B2_CTL__POR,
+ [SITAR_A_CDC_COMP1_B3_CTL] = SITAR_A_CDC_COMP1_B3_CTL__POR,
+ [SITAR_A_CDC_COMP1_B4_CTL] = SITAR_A_CDC_COMP1_B4_CTL__POR,
+ [SITAR_A_CDC_COMP1_B5_CTL] = SITAR_A_CDC_COMP1_B5_CTL__POR,
+ [SITAR_A_CDC_COMP1_B6_CTL] = SITAR_A_CDC_COMP1_B6_CTL__POR,
+ [SITAR_A_CDC_COMP1_SHUT_DOWN_STATUS] =
+ SITAR_A_CDC_COMP1_SHUT_DOWN_STATUS__POR,
+ [SITAR_A_CDC_COMP1_FS_CFG] = SITAR_A_CDC_COMP1_FS_CFG__POR,
+ [SITAR_A_CDC_CONN_RX1_B1_CTL] = SITAR_A_CDC_CONN_RX1_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_RX1_B2_CTL] = SITAR_A_CDC_CONN_RX1_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_RX1_B3_CTL] = SITAR_A_CDC_CONN_RX1_B3_CTL__POR,
+ [SITAR_A_CDC_CONN_RX2_B1_CTL] = SITAR_A_CDC_CONN_RX2_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_RX2_B2_CTL] = SITAR_A_CDC_CONN_RX2_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_RX2_B3_CTL] = SITAR_A_CDC_CONN_RX2_B3_CTL__POR,
+ [SITAR_A_CDC_CONN_RX3_B1_CTL] = SITAR_A_CDC_CONN_RX3_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_RX3_B2_CTL] = SITAR_A_CDC_CONN_RX3_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_RX3_B3_CTL] = SITAR_A_CDC_CONN_RX3_B3_CTL__POR,
+ [SITAR_A_CDC_CONN_ANC_B1_CTL] = SITAR_A_CDC_CONN_ANC_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_ANC_B2_CTL] = SITAR_A_CDC_CONN_ANC_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_B1_CTL] = SITAR_A_CDC_CONN_TX_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_B2_CTL] = SITAR_A_CDC_CONN_TX_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ1_B1_CTL] = SITAR_A_CDC_CONN_EQ1_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ1_B2_CTL] = SITAR_A_CDC_CONN_EQ1_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ1_B3_CTL] = SITAR_A_CDC_CONN_EQ1_B3_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ1_B4_CTL] = SITAR_A_CDC_CONN_EQ1_B4_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ2_B1_CTL] = SITAR_A_CDC_CONN_EQ2_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ2_B2_CTL] = SITAR_A_CDC_CONN_EQ2_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ2_B3_CTL] = SITAR_A_CDC_CONN_EQ2_B3_CTL__POR,
+ [SITAR_A_CDC_CONN_EQ2_B4_CTL] = SITAR_A_CDC_CONN_EQ2_B4_CTL__POR,
+ [SITAR_A_CDC_CONN_SRC1_B1_CTL] = SITAR_A_CDC_CONN_SRC1_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_SRC1_B2_CTL] = SITAR_A_CDC_CONN_SRC1_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_SRC2_B1_CTL] = SITAR_A_CDC_CONN_SRC2_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_SRC2_B2_CTL] = SITAR_A_CDC_CONN_SRC2_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_SB_B1_CTL] = SITAR_A_CDC_CONN_TX_SB_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_SB_B2_CTL] = SITAR_A_CDC_CONN_TX_SB_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_SB_B3_CTL] = SITAR_A_CDC_CONN_TX_SB_B3_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_SB_B4_CTL] = SITAR_A_CDC_CONN_TX_SB_B4_CTL__POR,
+ [SITAR_A_CDC_CONN_TX_SB_B5_CTL] = SITAR_A_CDC_CONN_TX_SB_B5_CTL__POR,
+ [SITAR_A_CDC_CONN_RX_SB_B1_CTL] = SITAR_A_CDC_CONN_RX_SB_B1_CTL__POR,
+ [SITAR_A_CDC_CONN_RX_SB_B2_CTL] = SITAR_A_CDC_CONN_RX_SB_B2_CTL__POR,
+ [SITAR_A_CDC_CONN_CLSG_CTL] = SITAR_A_CDC_CONN_CLSG_CTL__POR,
+ [SITAR_A_CDC_CONN_SPARE] = SITAR_A_CDC_CONN_SPARE__POR,
+ [SITAR_A_CDC_MBHC_EN_CTL] = SITAR_A_CDC_MBHC_EN_CTL__POR,
+ [SITAR_A_CDC_MBHC_FIR_B1_CFG] = SITAR_A_CDC_MBHC_FIR_B1_CFG__POR,
+ [SITAR_A_CDC_MBHC_FIR_B2_CFG] = SITAR_A_CDC_MBHC_FIR_B2_CFG__POR,
+ [SITAR_A_CDC_MBHC_TIMER_B1_CTL] = SITAR_A_CDC_MBHC_TIMER_B1_CTL__POR,
+ [SITAR_A_CDC_MBHC_TIMER_B2_CTL] = SITAR_A_CDC_MBHC_TIMER_B2_CTL__POR,
+ [SITAR_A_CDC_MBHC_TIMER_B3_CTL] = SITAR_A_CDC_MBHC_TIMER_B3_CTL__POR,
+ [SITAR_A_CDC_MBHC_TIMER_B4_CTL] = SITAR_A_CDC_MBHC_TIMER_B4_CTL__POR,
+ [SITAR_A_CDC_MBHC_TIMER_B5_CTL] = SITAR_A_CDC_MBHC_TIMER_B5_CTL__POR,
+ [SITAR_A_CDC_MBHC_TIMER_B6_CTL] = SITAR_A_CDC_MBHC_TIMER_B6_CTL__POR,
+ [SITAR_A_CDC_MBHC_B1_STATUS] = SITAR_A_CDC_MBHC_B1_STATUS__POR,
+ [SITAR_A_CDC_MBHC_B2_STATUS] = SITAR_A_CDC_MBHC_B2_STATUS__POR,
+ [SITAR_A_CDC_MBHC_B3_STATUS] = SITAR_A_CDC_MBHC_B3_STATUS__POR,
+ [SITAR_A_CDC_MBHC_B4_STATUS] = SITAR_A_CDC_MBHC_B4_STATUS__POR,
+ [SITAR_A_CDC_MBHC_B5_STATUS] = SITAR_A_CDC_MBHC_B5_STATUS__POR,
+ [SITAR_A_CDC_MBHC_B1_CTL] = SITAR_A_CDC_MBHC_B1_CTL__POR,
+ [SITAR_A_CDC_MBHC_B2_CTL] = SITAR_A_CDC_MBHC_B2_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B1_CTL] = SITAR_A_CDC_MBHC_VOLT_B1_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B2_CTL] = SITAR_A_CDC_MBHC_VOLT_B2_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B3_CTL] = SITAR_A_CDC_MBHC_VOLT_B3_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B4_CTL] = SITAR_A_CDC_MBHC_VOLT_B4_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B5_CTL] = SITAR_A_CDC_MBHC_VOLT_B5_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B6_CTL] = SITAR_A_CDC_MBHC_VOLT_B6_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B7_CTL] = SITAR_A_CDC_MBHC_VOLT_B7_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B8_CTL] = SITAR_A_CDC_MBHC_VOLT_B8_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B9_CTL] = SITAR_A_CDC_MBHC_VOLT_B9_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B10_CTL] = SITAR_A_CDC_MBHC_VOLT_B10_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B11_CTL] = SITAR_A_CDC_MBHC_VOLT_B11_CTL__POR,
+ [SITAR_A_CDC_MBHC_VOLT_B12_CTL] = SITAR_A_CDC_MBHC_VOLT_B12_CTL__POR,
+ [SITAR_A_CDC_MBHC_CLK_CTL] = SITAR_A_CDC_MBHC_CLK_CTL__POR,
+ [SITAR_A_CDC_MBHC_INT_CTL] = SITAR_A_CDC_MBHC_INT_CTL__POR,
+ [SITAR_A_CDC_MBHC_DEBUG_CTL] = SITAR_A_CDC_MBHC_DEBUG_CTL__POR,
+ [SITAR_A_CDC_MBHC_SPARE] = SITAR_A_CDC_MBHC_SPARE__POR,
+};
+
+const u8 sitar_reg_readable[SITAR_CACHE_SIZE] = {
+ [WCD9XXX_A_CHIP_CTL] = 1,
+ [WCD9XXX_A_CHIP_STATUS] = 1,
+ [WCD9XXX_A_CHIP_ID_BYTE_0] = 1,
+ [WCD9XXX_A_CHIP_ID_BYTE_1] = 1,
+ [WCD9XXX_A_CHIP_ID_BYTE_2] = 1,
+ [WCD9XXX_A_CHIP_ID_BYTE_3] = 1,
+ [WCD9XXX_A_CHIP_VERSION] = 1,
+ [WCD9XXX_A_SB_VERSION] = 1,
+ [WCD9XXX_A_SLAVE_ID_1] = 1,
+ [WCD9XXX_A_SLAVE_ID_2] = 1,
+ [WCD9XXX_A_SLAVE_ID_3] = 1,
+ [SITAR_A_PIN_CTL_OE0] = 1,
+ [SITAR_A_PIN_CTL_OE1] = 1,
+ [SITAR_A_PIN_CTL_DATA0] = 1,
+ [SITAR_A_PIN_CTL_DATA1] = 1,
+ [SITAR_A_HDRIVE_GENERIC] = 1,
+ [SITAR_A_HDRIVE_OVERRIDE] = 1,
+ [SITAR_A_ANA_CSR_WAIT_STATE] = 1,
+ [SITAR_A_PROCESS_MONITOR_CTL0] = 1,
+ [SITAR_A_PROCESS_MONITOR_CTL1] = 1,
+ [SITAR_A_PROCESS_MONITOR_CTL2] = 1,
+ [SITAR_A_PROCESS_MONITOR_CTL3] = 1,
+ [SITAR_A_QFUSE_CTL] = 1,
+ [SITAR_A_QFUSE_STATUS] = 1,
+ [SITAR_A_QFUSE_DATA_OUT0] = 1,
+ [SITAR_A_QFUSE_DATA_OUT1] = 1,
+ [SITAR_A_QFUSE_DATA_OUT2] = 1,
+ [SITAR_A_QFUSE_DATA_OUT3] = 1,
+ [SITAR_A_CDC_CTL] = 1,
+ [SITAR_A_LEAKAGE_CTL] = 1,
+ [SITAR_A_INTR_MODE] = 1,
+ [SITAR_A_INTR_MASK0] = 1,
+ [SITAR_A_INTR_MASK1] = 1,
+ [SITAR_A_INTR_MASK2] = 1,
+ [SITAR_A_INTR_STATUS0] = 1,
+ [SITAR_A_INTR_STATUS1] = 1,
+ [SITAR_A_INTR_STATUS2] = 1,
+ [SITAR_A_INTR_LEVEL0] = 1,
+ [SITAR_A_INTR_LEVEL1] = 1,
+ [SITAR_A_INTR_LEVEL2] = 1,
+ [SITAR_A_INTR_TEST0] = 1,
+ [SITAR_A_INTR_TEST1] = 1,
+ [SITAR_A_INTR_TEST2] = 1,
+ [SITAR_A_INTR_SET0] = 1,
+ [SITAR_A_INTR_SET1] = 1,
+ [SITAR_A_INTR_SET2] = 1,
+ [SITAR_A_CDC_TX_I2S_SCK_MODE] = 1,
+ [SITAR_A_CDC_TX_I2S_WS_MODE] = 1,
+ [SITAR_A_CDC_DMIC_DATA0_MODE] = 1,
+ [SITAR_A_CDC_DMIC_CLK0_MODE] = 1,
+ [SITAR_A_CDC_DMIC_DATA1_MODE] = 1,
+ [SITAR_A_CDC_DMIC_CLK1_MODE] = 1,
+ [SITAR_A_CDC_TX_I2S_SD0_MODE] = 1,
+ [SITAR_A_CDC_INTR_MODE] = 1,
+ [SITAR_A_CDC_RX_I2S_SD0_MODE] = 1,
+ [SITAR_A_CDC_RX_I2S_SD1_MODE] = 1,
+ [SITAR_A_BIAS_REF_CTL] = 1,
+ [SITAR_A_BIAS_CENTRAL_BG_CTL] = 1,
+ [SITAR_A_BIAS_PRECHRG_CTL] = 1,
+ [SITAR_A_BIAS_CURR_CTL_1] = 1,
+ [SITAR_A_BIAS_CURR_CTL_2] = 1,
+ [SITAR_A_BIAS_OSC_BG_CTL] = 1,
+ [SITAR_A_CLK_BUFF_EN1] = 1,
+ [SITAR_A_CLK_BUFF_EN2] = 1,
+ [SITAR_A_LDO_H_MODE_1] = 1,
+ [SITAR_A_LDO_H_MODE_2] = 1,
+ [SITAR_A_LDO_H_LOOP_CTL] = 1,
+ [SITAR_A_LDO_H_COMP_1] = 1,
+ [SITAR_A_LDO_H_COMP_2] = 1,
+ [SITAR_A_LDO_H_BIAS_1] = 1,
+ [SITAR_A_LDO_H_BIAS_2] = 1,
+ [SITAR_A_LDO_H_BIAS_3] = 1,
+ [SITAR_A_MICB_CFILT_1_CTL] = 1,
+ [SITAR_A_MICB_CFILT_1_VAL] = 1,
+ [SITAR_A_MICB_CFILT_1_PRECHRG] = 1,
+ [SITAR_A_MICB_1_CTL] = 1,
+ [SITAR_A_MICB_1_INT_RBIAS] = 1,
+ [SITAR_A_MICB_1_MBHC] = 1,
+ [SITAR_A_MICB_CFILT_2_CTL] = 1,
+ [SITAR_A_MICB_CFILT_2_VAL] = 1,
+ [SITAR_A_MICB_CFILT_2_PRECHRG] = 1,
+ [SITAR_A_MICB_2_CTL] = 1,
+ [SITAR_A_MICB_2_INT_RBIAS] = 1,
+ [SITAR_A_MICB_2_MBHC] = 1,
+ [SITAR_A_TX_COM_BIAS] = 1,
+ [SITAR_A_MBHC_SCALING_MUX_1] = 1,
+ [SITAR_A_MBHC_SCALING_MUX_2] = 1,
+ [SITAR_A_TX_SUP_SWITCH_CTRL_1] = 1,
+ [SITAR_A_TX_SUP_SWITCH_CTRL_2] = 1,
+ [SITAR_A_TX_1_2_EN] = 1,
+ [SITAR_A_TX_1_2_TEST_EN] = 1,
+ [SITAR_A_TX_1_2_ADC_CH1] = 1,
+ [SITAR_A_TX_1_2_ADC_CH2] = 1,
+ [SITAR_A_TX_1_2_ATEST_REFCTRL] = 1,
+ [SITAR_A_TX_1_2_TEST_CTL] = 1,
+ [SITAR_A_TX_1_2_TEST_BLOCK_EN] = 1,
+ [SITAR_A_TX_1_2_TXFE_CLKDIV] = 1,
+ [SITAR_A_TX_1_2_SAR_ERR_CH1] = 1,
+ [SITAR_A_TX_1_2_SAR_ERR_CH2] = 1,
+ [SITAR_A_TX_3_EN] = 1,
+ [SITAR_A_TX_3_TEST_EN] = 1,
+ [SITAR_A_TX_3_ADC] = 1,
+ [SITAR_A_TX_3_MBHC_ATEST_REFCTRL] = 1,
+ [SITAR_A_TX_3_TEST_CTL] = 1,
+ [SITAR_A_TX_3_TEST_BLOCK_EN] = 1,
+ [SITAR_A_TX_3_TXFE_CKDIV] = 1,
+ [SITAR_A_TX_3_SAR_ERR] = 1,
+ [SITAR_A_TX_4_MBHC_EN] = 1,
+ [SITAR_A_TX_4_MBHC_ADC] = 1,
+ [SITAR_A_TX_4_MBHC_TEST_CTL] = 1,
+ [SITAR_A_TX_4_MBHC_SAR_ERR] = 1,
+ [SITAR_A_TX_4_TXFE_CLKDIV] = 1,
+ [SITAR_A_AUX_COM_CTL] = 1,
+ [SITAR_A_AUX_COM_ATEST] = 1,
+ [SITAR_A_AUX_L_EN] = 1,
+ [SITAR_A_AUX_L_GAIN] = 1,
+ [SITAR_A_AUX_L_PA_CONN] = 1,
+ [SITAR_A_AUX_L_PA_CONN_INV] = 1,
+ [SITAR_A_AUX_R_EN] = 1,
+ [SITAR_A_AUX_R_GAIN] = 1,
+ [SITAR_A_AUX_R_PA_CONN] = 1,
+ [SITAR_A_AUX_R_PA_CONN_INV] = 1,
+ [SITAR_A_CP_EN] = 1,
+ [SITAR_A_CP_CLK] = 1,
+ [SITAR_A_CP_STATIC] = 1,
+ [SITAR_A_CP_DCC1] = 1,
+ [SITAR_A_CP_DCC3] = 1,
+ [SITAR_A_CP_ATEST] = 1,
+ [SITAR_A_CP_DTEST] = 1,
+ [SITAR_A_RX_COM_TIMER_DIV] = 1,
+ [SITAR_A_RX_COM_OCP_CTL] = 1,
+ [SITAR_A_RX_COM_OCP_COUNT] = 1,
+ [SITAR_A_RX_COM_DAC_CTL] = 1,
+ [SITAR_A_RX_COM_BIAS] = 1,
+ [SITAR_A_RX_HPH_BIAS_PA] = 1,
+ [SITAR_A_RX_HPH_BIAS_LDO] = 1,
+ [SITAR_A_RX_HPH_BIAS_CNP] = 1,
+ [SITAR_A_RX_HPH_BIAS_WG] = 1,
+ [SITAR_A_RX_HPH_OCP_CTL] = 1,
+ [SITAR_A_RX_HPH_CNP_EN] = 1,
+ [SITAR_A_RX_HPH_CNP_WG_CTL] = 1,
+ [SITAR_A_RX_HPH_CNP_WG_TIME] = 1,
+ [SITAR_A_RX_HPH_L_GAIN] = 1,
+ [SITAR_A_RX_HPH_L_TEST] = 1,
+ [SITAR_A_RX_HPH_L_PA_CTL] = 1,
+ [SITAR_A_RX_HPH_L_DAC_CTL] = 1,
+ [SITAR_A_RX_HPH_L_ATEST] = 1,
+ [SITAR_A_RX_HPH_L_STATUS] = 1,
+ [SITAR_A_RX_HPH_R_GAIN] = 1,
+ [SITAR_A_RX_HPH_R_TEST] = 1,
+ [SITAR_A_RX_HPH_R_PA_CTL] = 1,
+ [SITAR_A_RX_HPH_R_DAC_CTL] = 1,
+ [SITAR_A_RX_HPH_R_ATEST] = 1,
+ [SITAR_A_RX_HPH_R_STATUS] = 1,
+ [SITAR_A_RX_EAR_BIAS_PA] = 1,
+ [SITAR_A_RX_EAR_BIAS_CMBUFF] = 1,
+ [SITAR_A_RX_EAR_EN] = 1,
+ [SITAR_A_RX_EAR_GAIN] = 1,
+ [SITAR_A_RX_EAR_CMBUFF] = 1,
+ [SITAR_A_RX_EAR_ICTL] = 1,
+ [SITAR_A_RX_EAR_CCOMP] = 1,
+ [SITAR_A_RX_EAR_VCM] = 1,
+ [SITAR_A_RX_EAR_CNP] = 1,
+ [SITAR_A_RX_EAR_ATEST] = 1,
+ [SITAR_A_RX_EAR_STATUS] = 1,
+ [SITAR_A_RX_LINE_BIAS_PA] = 1,
+ [SITAR_A_RX_LINE_BIAS_LDO] = 1,
+ [SITAR_A_RX_LINE_BIAS_CNP1] = 1,
+ [SITAR_A_RX_LINE_COM] = 1,
+ [SITAR_A_RX_LINE_CNP_EN] = 1,
+ [SITAR_A_RX_LINE_CNP_WG_CTL] = 1,
+ [SITAR_A_RX_LINE_CNP_WG_TIME] = 1,
+ [SITAR_A_RX_LINE_1_GAIN] = 1,
+ [SITAR_A_RX_LINE_1_TEST] = 1,
+ [SITAR_A_RX_LINE_1_DAC_CTL] = 1,
+ [SITAR_A_RX_LINE_1_STATUS] = 1,
+ [SITAR_A_RX_LINE_2_GAIN] = 1,
+ [SITAR_A_RX_LINE_2_TEST] = 1,
+ [SITAR_A_RX_LINE_2_DAC_CTL] = 1,
+ [SITAR_A_RX_LINE_2_STATUS] = 1,
+ [SITAR_A_RX_LINE_BIAS_CNP2] = 1,
+ [SITAR_A_RX_LINE_OCP_CTL] = 1,
+ [SITAR_A_RX_LINE_1_PA_CTL] = 1,
+ [SITAR_A_RX_LINE_2_PA_CTL] = 1,
+ [SITAR_A_RX_LINE_CNP_DBG] = 1,
+ [SITAR_A_MBHC_HPH] = 1,
+ [SITAR_A_RC_OSC_FREQ] = 1,
+ [SITAR_A_RC_OSC_TEST] = 1,
+ [SITAR_A_RC_OSC_STATUS] = 1,
+ [SITAR_A_RC_OSC_TUNER] = 1,
+ [SITAR_A_CDC_ANC1_CTL] = 1,
+ [SITAR_A_CDC_ANC1_SHIFT] = 1,
+ [SITAR_A_CDC_ANC1_IIR_B1_CTL] = 1,
+ [SITAR_A_CDC_ANC1_IIR_B2_CTL] = 1,
+ [SITAR_A_CDC_ANC1_IIR_B3_CTL] = 1,
+ [SITAR_A_CDC_ANC1_IIR_B4_CTL] = 1,
+ [SITAR_A_CDC_ANC1_LPF_B1_CTL] = 1,
+ [SITAR_A_CDC_ANC1_LPF_B2_CTL] = 1,
+ [SITAR_A_CDC_ANC1_LPF_B3_CTL] = 1,
+ [SITAR_A_CDC_ANC1_SPARE] = 1,
+ [SITAR_A_CDC_ANC1_SMLPF_CTL] = 1,
+ [SITAR_A_CDC_ANC1_DCFLT_CTL] = 1,
+ [SITAR_A_CDC_TX1_VOL_CTL_TIMER] = 1,
+ [SITAR_A_CDC_TX1_VOL_CTL_GAIN] = 1,
+ [SITAR_A_CDC_TX1_VOL_CTL_CFG] = 1,
+ [SITAR_A_CDC_TX1_MUX_CTL] = 1,
+ [SITAR_A_CDC_TX1_CLK_FS_CTL] = 1,
+ [SITAR_A_CDC_TX1_DMIC_CTL] = 1,
+ [SITAR_A_CDC_SRC1_PDA_CFG] = 1,
+ [SITAR_A_CDC_SRC1_FS_CTL] = 1,
+ [SITAR_A_CDC_RX1_B1_CTL] = 1,
+ [SITAR_A_CDC_RX1_B2_CTL] = 1,
+ [SITAR_A_CDC_RX1_B3_CTL] = 1,
+ [SITAR_A_CDC_RX1_B4_CTL] = 1,
+ [SITAR_A_CDC_RX1_B5_CTL] = 1,
+ [SITAR_A_CDC_RX2_B5_CTL] = 1,
+ [SITAR_A_CDC_RX3_B5_CTL] = 1,
+ [SITAR_A_CDC_RX1_B6_CTL] = 1,
+ [SITAR_A_CDC_RX1_VOL_CTL_B1_CTL] = 1,
+ [SITAR_A_CDC_RX1_VOL_CTL_B2_CTL] = 1,
+ [SITAR_A_CDC_CLK_ANC_RESET_CTL] = 1,
+ [SITAR_A_CDC_CLK_RX_RESET_CTL] = 1,
+ [SITAR_A_CDC_CLK_TX_RESET_B1_CTL] = 1,
+ [SITAR_A_CDC_CLK_TX_RESET_B2_CTL] = 1,
+ [SITAR_A_CDC_CLK_DMIC_CTL] = 1,
+ [SITAR_A_CDC_CLK_RX_I2S_CTL] = 1,
+ [SITAR_A_CDC_CLK_TX_I2S_CTL] = 1,
+ [SITAR_A_CDC_CLK_OTHR_RESET_CTL] = 1,
+ [SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL] = 1,
+ [SITAR_A_CDC_CLK_OTHR_CTL] = 1,
+ [SITAR_A_CDC_CLK_RDAC_CLK_EN_CTL] = 1,
+ [SITAR_A_CDC_CLK_ANC_CLK_EN_CTL] = 1,
+ [SITAR_A_CDC_CLK_RX_B1_CTL] = 1,
+ [SITAR_A_CDC_CLK_RX_B2_CTL] = 1,
+ [SITAR_A_CDC_CLK_MCLK_CTL] = 1,
+ [SITAR_A_CDC_CLK_PDM_CTL] = 1,
+ [SITAR_A_CDC_CLK_SD_CTL] = 1,
+ [SITAR_A_CDC_CLK_LP_CTL] = 1,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B1_CTL] = 1,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B2_CTL] = 1,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B3_CTL] = 1,
+ [SITAR_A_CDC_CLSG_FREQ_THRESH_B4_CTL] = 1,
+ [SITAR_A_CDC_CLSG_GAIN_THRESH_CTL] = 1,
+ [SITAR_A_CDC_CLSG_TIMER_B1_CFG] = 1,
+ [SITAR_A_CDC_CLSG_TIMER_B2_CFG] = 1,
+ [SITAR_A_CDC_CLSG_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B1_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B2_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B3_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B4_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B5_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B6_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B7_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_B8_CTL] = 1,
+ [SITAR_A_CDC_IIR1_CTL] = 1,
+ [SITAR_A_CDC_IIR1_GAIN_TIMER_CTL] = 1,
+ [SITAR_A_CDC_IIR1_COEF_B1_CTL] = 1,
+ [SITAR_A_CDC_IIR1_COEF_B2_CTL] = 1,
+ [SITAR_A_CDC_IIR1_COEF_B3_CTL] = 1,
+ [SITAR_A_CDC_IIR1_COEF_B4_CTL] = 1,
+ [SITAR_A_CDC_IIR1_COEF_B5_CTL] = 1,
+ [SITAR_A_CDC_TOP_GAIN_UPDATE] = 1,
+ [SITAR_A_CDC_TOP_RDAC_DOUT_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B1_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B2_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B3_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B4_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B5_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B6_CTL] = 1,
+ [SITAR_A_CDC_DEBUG_B7_CTL] = 1,
+ [SITAR_A_CDC_COMP1_B1_CTL] = 1,
+ [SITAR_A_CDC_COMP1_B2_CTL] = 1,
+ [SITAR_A_CDC_COMP1_B3_CTL] = 1,
+ [SITAR_A_CDC_COMP1_B4_CTL] = 1,
+ [SITAR_A_CDC_COMP1_B5_CTL] = 1,
+ [SITAR_A_CDC_COMP1_B6_CTL] = 1,
+ [SITAR_A_CDC_COMP1_SHUT_DOWN_STATUS] = 1,
+ [SITAR_A_CDC_COMP1_FS_CFG] = 1,
+ [SITAR_A_CDC_CONN_RX1_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX1_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX1_B3_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX2_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX2_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX2_B3_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX3_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX3_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX3_B3_CTL] = 1,
+ [SITAR_A_CDC_CONN_ANC_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_ANC_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ1_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ1_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ1_B3_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ1_B4_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ2_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ2_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ2_B3_CTL] = 1,
+ [SITAR_A_CDC_CONN_EQ2_B4_CTL] = 1,
+ [SITAR_A_CDC_CONN_SRC1_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_SRC1_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_SRC2_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_SRC2_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_SB_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_SB_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_SB_B3_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_SB_B4_CTL] = 1,
+ [SITAR_A_CDC_CONN_TX_SB_B5_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX_SB_B1_CTL] = 1,
+ [SITAR_A_CDC_CONN_RX_SB_B2_CTL] = 1,
+ [SITAR_A_CDC_CONN_CLSG_CTL] = 1,
+ [SITAR_A_CDC_CONN_SPARE] = 1,
+ [SITAR_A_CDC_MBHC_EN_CTL] = 1,
+ [SITAR_A_CDC_MBHC_FIR_B1_CFG] = 1,
+ [SITAR_A_CDC_MBHC_FIR_B2_CFG] = 1,
+ [SITAR_A_CDC_MBHC_TIMER_B1_CTL] = 1,
+ [SITAR_A_CDC_MBHC_TIMER_B2_CTL] = 1,
+ [SITAR_A_CDC_MBHC_TIMER_B3_CTL] = 1,
+ [SITAR_A_CDC_MBHC_TIMER_B4_CTL] = 1,
+ [SITAR_A_CDC_MBHC_TIMER_B5_CTL] = 1,
+ [SITAR_A_CDC_MBHC_TIMER_B6_CTL] = 1,
+ [SITAR_A_CDC_MBHC_B1_STATUS] = 1,
+ [SITAR_A_CDC_MBHC_B2_STATUS] = 1,
+ [SITAR_A_CDC_MBHC_B3_STATUS] = 1,
+ [SITAR_A_CDC_MBHC_B4_STATUS] = 1,
+ [SITAR_A_CDC_MBHC_B5_STATUS] = 1,
+ [SITAR_A_CDC_MBHC_B1_CTL] = 1,
+ [SITAR_A_CDC_MBHC_B2_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B1_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B2_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B3_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B4_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B5_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B6_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B7_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B8_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B9_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B10_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B11_CTL] = 1,
+ [SITAR_A_CDC_MBHC_VOLT_B12_CTL] = 1,
+ [SITAR_A_CDC_MBHC_CLK_CTL] = 1,
+ [SITAR_A_CDC_MBHC_INT_CTL] = 1,
+ [SITAR_A_CDC_MBHC_DEBUG_CTL] = 1,
+ [SITAR_A_CDC_MBHC_SPARE] = 1,
+};
diff --git a/sound/soc/codecs/wcd9304.c b/sound/soc/codecs/wcd9304.c
new file mode 100644
index 0000000..c5dbdae
--- /dev/null
+++ b/sound/soc/codecs/wcd9304.c
@@ -0,0 +1,4817 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/firmware.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/printk.h>
+#include <linux/ratelimit.h>
+#include <linux/debugfs.h>
+#include <linux/mfd/wcd9xxx/core.h>
+#include <linux/mfd/wcd9xxx/wcd9xxx_registers.h>
+#include <linux/mfd/wcd9xxx/wcd9304_registers.h>
+#include <linux/mfd/wcd9xxx/pdata.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <linux/bitops.h>
+#include <linux/delay.h>
+#include <linux/pm_runtime.h>
+#include <linux/kernel.h>
+#include <linux/gpio.h>
+#include "wcd9304.h"
+
+#define WCD9304_RATES (SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|\
+ SNDRV_PCM_RATE_32000|SNDRV_PCM_RATE_48000)
+
+#define NUM_DECIMATORS 4
+#define NUM_INTERPOLATORS 3
+#define BITS_PER_REG 8
+#define SITAR_CFILT_FAST_MODE 0x00
+#define SITAR_CFILT_SLOW_MODE 0x40
+#define MBHC_FW_READ_ATTEMPTS 15
+#define MBHC_FW_READ_TIMEOUT 2000000
+
+#define SITAR_JACK_MASK (SND_JACK_HEADSET | SND_JACK_OC_HPHL | SND_JACK_OC_HPHR)
+
+#define SITAR_I2S_MASTER_MODE_MASK 0x08
+
+#define SITAR_OCP_ATTEMPT 1
+
+#define AIF1_PB 1
+#define AIF1_CAP 2
+#define NUM_CODEC_DAIS 2
+
+struct sitar_codec_dai_data {
+ u32 rate;
+ u32 *ch_num;
+ u32 ch_act;
+ u32 ch_tot;
+};
+
+#define SITAR_MCLK_RATE_12288KHZ 12288000
+#define SITAR_MCLK_RATE_9600KHZ 9600000
+
+#define SITAR_FAKE_INS_THRESHOLD_MS 2500
+#define SITAR_FAKE_REMOVAL_MIN_PERIOD_MS 50
+#define SITAR_MBHC_BUTTON_MIN 0x8000
+#define SITAR_GPIO_IRQ_DEBOUNCE_TIME_US 5000
+
+#define SITAR_ACQUIRE_LOCK(x) do { mutex_lock(&x); } while (0)
+#define SITAR_RELEASE_LOCK(x) do { mutex_unlock(&x); } while (0)
+
+#define MBHC_NUM_DCE_PLUG_DETECT 3
+#define SITAR_MBHC_FAKE_INSERT_LOW 10
+#define SITAR_MBHC_FAKE_INSERT_HIGH 80
+#define SITAR_MBHC_FAKE_INSERT_VOLT_DELTA_MV 500
+#define SITAR_HS_DETECT_PLUG_TIME_MS (5 * 1000)
+#define SITAR_HS_DETECT_PLUG_INERVAL_MS 100
+#define NUM_ATTEMPTS_TO_REPORT 5
+#define SITAR_MBHC_STATUS_REL_DETECTION 0x0C
+#define SITAR_MBHC_GPIO_REL_DEBOUNCE_TIME_MS 200
+
+static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
+static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
+static struct snd_soc_dai_driver sitar_dai[];
+static int sitar_codec_enable_slimtx(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
+static int sitar_codec_enable_slimrx(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
+
+enum sitar_bandgap_type {
+ SITAR_BANDGAP_OFF = 0,
+ SITAR_BANDGAP_AUDIO_MODE,
+ SITAR_BANDGAP_MBHC_MODE,
+};
+
+struct mbhc_micbias_regs {
+ u16 cfilt_val;
+ u16 cfilt_ctl;
+ u16 mbhc_reg;
+ u16 int_rbias;
+ u16 ctl_reg;
+ u8 cfilt_sel;
+};
+
+/* Codec supports 2 IIR filters */
+enum {
+ IIR1 = 0,
+ IIR2,
+ IIR_MAX,
+};
+/* Codec supports 5 bands */
+enum {
+ BAND1 = 0,
+ BAND2,
+ BAND3,
+ BAND4,
+ BAND5,
+ BAND_MAX,
+};
+
+/* Flags to track of PA and DAC state.
+ * PA and DAC should be tracked separately as AUXPGA loopback requires
+ * only PA to be turned on without DAC being on. */
+enum sitar_priv_ack_flags {
+ SITAR_HPHL_PA_OFF_ACK = 0,
+ SITAR_HPHR_PA_OFF_ACK,
+ SITAR_HPHL_DAC_OFF_ACK,
+ SITAR_HPHR_DAC_OFF_ACK
+};
+
+/* Data used by MBHC */
+struct mbhc_internal_cal_data {
+ u16 dce_z;
+ u16 dce_mb;
+ u16 sta_z;
+ u16 sta_mb;
+ u32 t_sta_dce;
+ u32 t_dce;
+ u32 t_sta;
+ u32 micb_mv;
+ u16 v_ins_hu;
+ u16 v_ins_h;
+ u16 v_b1_hu;
+ u16 v_b1_h;
+ u16 v_b1_huc;
+ u16 v_brh;
+ u16 v_brl;
+ u16 v_no_mic;
+ u8 npoll;
+ u8 nbounce_wait;
+};
+
+enum sitar_mbhc_plug_type {
+ PLUG_TYPE_INVALID = -1,
+ PLUG_TYPE_NONE,
+ PLUG_TYPE_HEADSET,
+ PLUG_TYPE_HEADPHONE,
+ PLUG_TYPE_HIGH_HPH,
+};
+
+enum sitar_mbhc_state {
+ MBHC_STATE_NONE = -1,
+ MBHC_STATE_POTENTIAL,
+ MBHC_STATE_POTENTIAL_RECOVERY,
+ MBHC_STATE_RELEASE,
+};
+
+struct sitar_priv {
+ struct snd_soc_codec *codec;
+ u32 mclk_freq;
+ u32 adc_count;
+ u32 cfilt1_cnt;
+ u32 cfilt2_cnt;
+ u32 cfilt3_cnt;
+ u32 rx_bias_count;
+ enum sitar_bandgap_type bandgap_type;
+ bool mclk_enabled;
+ bool clock_active;
+ bool config_mode_active;
+ bool mbhc_polling_active;
+ unsigned long mbhc_fake_ins_start;
+ int buttons_pressed;
+
+ enum sitar_micbias_num micbias;
+ /* void* calibration contains:
+ * struct sitar_mbhc_general_cfg generic;
+ * struct sitar_mbhc_plug_detect_cfg plug_det;
+ * struct sitar_mbhc_plug_type_cfg plug_type;
+ * struct sitar_mbhc_btn_detect_cfg btn_det;
+ * struct sitar_mbhc_imped_detect_cfg imped_det;
+ * Note: various size depends on btn_det->num_btn
+ */
+ void *calibration;
+ struct mbhc_internal_cal_data mbhc_data;
+
+ struct wcd9xxx_pdata *pdata;
+ u32 anc_slot;
+
+ bool no_mic_headset_override;
+
+ struct mbhc_micbias_regs mbhc_bias_regs;
+ u8 cfilt_k_value;
+ bool mbhc_micbias_switched;
+
+ /* track PA/DAC state */
+ unsigned long hph_pa_dac_state;
+
+ /*track sitar interface type*/
+ u8 intf_type;
+
+ u32 hph_status; /* track headhpone status */
+ /* define separate work for left and right headphone OCP to avoid
+ * additional checking on which OCP event to report so no locking
+ * to ensure synchronization is required
+ */
+ struct work_struct hphlocp_work; /* reporting left hph ocp off */
+ struct work_struct hphrocp_work; /* reporting right hph ocp off */
+
+ u8 hphlocp_cnt; /* headphone left ocp retry */
+ u8 hphrocp_cnt; /* headphone right ocp retry */
+
+ /* Callback function to enable MCLK */
+ int (*mclk_cb) (struct snd_soc_codec*, int);
+
+ /* Work to perform MBHC Firmware Read */
+ struct delayed_work mbhc_firmware_dwork;
+ const struct firmware *mbhc_fw;
+
+ /* num of slim ports required */
+ struct sitar_codec_dai_data dai[NUM_CODEC_DAIS];
+
+ /* Currently, only used for mbhc purpose, to protect
+ * concurrent execution of mbhc threaded irq handlers and
+ * kill race between DAPM and MBHC.But can serve as a
+ * general lock to protect codec resource
+ */
+ struct mutex codec_resource_lock;
+
+ struct sitar_mbhc_config mbhc_cfg;
+ bool in_gpio_handler;
+ u8 current_plug;
+ bool lpi_enabled;
+ enum sitar_mbhc_state mbhc_state;
+ struct work_struct hs_correct_plug_work;
+ bool hs_detect_work_stop;
+ struct delayed_work mbhc_btn_dwork;
+ unsigned long mbhc_last_resume; /* in jiffies */
+};
+
+#ifdef CONFIG_DEBUG_FS
+struct sitar_priv *debug_sitar_priv;
+#endif
+
+
+static int sitar_pa_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 ear_pa_gain;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ ear_pa_gain = snd_soc_read(codec, SITAR_A_RX_EAR_GAIN);
+
+ ear_pa_gain = ear_pa_gain >> 5;
+
+ if (ear_pa_gain == 0x00) {
+ ucontrol->value.integer.value[0] = 0;
+ } else if (ear_pa_gain == 0x04) {
+ ucontrol->value.integer.value[0] = 1;
+ } else {
+ pr_err("%s: ERROR: Unsupported Ear Gain = 0x%x\n",
+ __func__, ear_pa_gain);
+ return -EINVAL;
+ }
+
+ pr_debug("%s: ear_pa_gain = 0x%x\n", __func__, ear_pa_gain);
+
+ return 0;
+}
+
+static int sitar_pa_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 ear_pa_gain;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s: ucontrol->value.integer.value[0] = %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ ear_pa_gain = 0x00;
+ break;
+ case 1:
+ ear_pa_gain = 0x80;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, SITAR_A_RX_EAR_GAIN, ear_pa_gain);
+ return 0;
+}
+
+static int sitar_get_iir_enable_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ ucontrol->value.integer.value[0] =
+ snd_soc_read(codec, (SITAR_A_CDC_IIR1_CTL + 16 * iir_idx)) &
+ (1 << band_idx);
+
+ pr_debug("%s: IIR #%d band #%d enable %d\n", __func__,
+ iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int sitar_put_iir_enable_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+ int value = ucontrol->value.integer.value[0];
+
+ /* Mask first 5 bits, 6-8 are reserved */
+ snd_soc_update_bits(codec, (SITAR_A_CDC_IIR1_CTL + 16 * iir_idx),
+ (1 << band_idx), (value << band_idx));
+
+ pr_debug("%s: IIR #%d band #%d enable %d\n", __func__,
+ iir_idx, band_idx, value);
+ return 0;
+}
+static uint32_t get_iir_band_coeff(struct snd_soc_codec *codec,
+ int iir_idx, int band_idx,
+ int coeff_idx)
+{
+ /* Address does not automatically update if reading */
+ snd_soc_update_bits(codec,
+ (SITAR_A_CDC_IIR1_COEF_B1_CTL + 16 * iir_idx),
+ 0x1F, band_idx * BAND_MAX + coeff_idx);
+
+ /* Mask bits top 2 bits since they are reserved */
+ return ((snd_soc_read(codec,
+ (SITAR_A_CDC_IIR1_COEF_B2_CTL + 16 * iir_idx)) << 24) |
+ (snd_soc_read(codec,
+ (SITAR_A_CDC_IIR1_COEF_B3_CTL + 16 * iir_idx)) << 16) |
+ (snd_soc_read(codec,
+ (SITAR_A_CDC_IIR1_COEF_B4_CTL + 16 * iir_idx)) << 8) |
+ (snd_soc_read(codec,
+ (SITAR_A_CDC_IIR1_COEF_B5_CTL + 16 * iir_idx)))) &
+ 0x3FFFFFFF;
+}
+
+static int sitar_get_iir_band_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ ucontrol->value.integer.value[0] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 0);
+ ucontrol->value.integer.value[1] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 1);
+ ucontrol->value.integer.value[2] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 2);
+ ucontrol->value.integer.value[3] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 3);
+ ucontrol->value.integer.value[4] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 4);
+
+ pr_debug("%s: IIR #%d band #%d b0 = 0x%x\n"
+ "%s: IIR #%d band #%d b1 = 0x%x\n"
+ "%s: IIR #%d band #%d b2 = 0x%x\n"
+ "%s: IIR #%d band #%d a1 = 0x%x\n"
+ "%s: IIR #%d band #%d a2 = 0x%x\n",
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[0],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[1],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[2],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[3],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[4]);
+ return 0;
+}
+
+static void set_iir_band_coeff(struct snd_soc_codec *codec,
+ int iir_idx, int band_idx,
+ int coeff_idx, uint32_t value)
+{
+ /* Mask top 3 bits, 6-8 are reserved */
+ /* Update address manually each time */
+ snd_soc_update_bits(codec,
+ (SITAR_A_CDC_IIR1_COEF_B1_CTL + 16 * iir_idx),
+ 0x1F, band_idx * BAND_MAX + coeff_idx);
+
+ /* Mask top 2 bits, 7-8 are reserved */
+ snd_soc_update_bits(codec,
+ (SITAR_A_CDC_IIR1_COEF_B2_CTL + 16 * iir_idx),
+ 0x3F, (value >> 24) & 0x3F);
+
+ /* Isolate 8bits at a time */
+ snd_soc_update_bits(codec,
+ (SITAR_A_CDC_IIR1_COEF_B3_CTL + 16 * iir_idx),
+ 0xFF, (value >> 16) & 0xFF);
+
+ snd_soc_update_bits(codec,
+ (SITAR_A_CDC_IIR1_COEF_B4_CTL + 16 * iir_idx),
+ 0xFF, (value >> 8) & 0xFF);
+
+ snd_soc_update_bits(codec,
+ (SITAR_A_CDC_IIR1_COEF_B5_CTL + 16 * iir_idx),
+ 0xFF, value & 0xFF);
+}
+
+static int sitar_put_iir_band_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ set_iir_band_coeff(codec, iir_idx, band_idx, 0,
+ ucontrol->value.integer.value[0]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 1,
+ ucontrol->value.integer.value[1]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 2,
+ ucontrol->value.integer.value[2]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 3,
+ ucontrol->value.integer.value[3]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 4,
+ ucontrol->value.integer.value[4]);
+
+ pr_debug("%s: IIR #%d band #%d b0 = 0x%x\n"
+ "%s: IIR #%d band #%d b1 = 0x%x\n"
+ "%s: IIR #%d band #%d b2 = 0x%x\n"
+ "%s: IIR #%d band #%d a1 = 0x%x\n"
+ "%s: IIR #%d band #%d a2 = 0x%x\n",
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 0),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 1),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 2),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 3),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 4));
+ return 0;
+}
+
+static const char *sitar_ear_pa_gain_text[] = {"POS_6_DB", "POS_2_DB"};
+static const struct soc_enum sitar_ear_pa_gain_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, sitar_ear_pa_gain_text),
+};
+
+/*cut of frequency for high pass filter*/
+static const char *cf_text[] = {
+ "MIN_3DB_4Hz", "MIN_3DB_75Hz", "MIN_3DB_150Hz"
+};
+
+static const struct soc_enum cf_dec1_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_TX1_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_rxmix1_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_RX1_B4_CTL, 1, 3, cf_text);
+
+static const struct snd_kcontrol_new sitar_snd_controls[] = {
+
+ SOC_ENUM_EXT("EAR PA Gain", sitar_ear_pa_gain_enum[0],
+ sitar_pa_gain_get, sitar_pa_gain_put),
+
+ SOC_SINGLE_TLV("LINEOUT1 Volume", SITAR_A_RX_LINE_1_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("LINEOUT2 Volume", SITAR_A_RX_LINE_2_GAIN, 0, 12, 1,
+ line_gain),
+
+ SOC_SINGLE_TLV("HPHL Volume", SITAR_A_RX_HPH_L_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("HPHR Volume", SITAR_A_RX_HPH_R_GAIN, 0, 12, 1,
+ line_gain),
+
+ SOC_SINGLE_S8_TLV("RX1 Digital Volume", SITAR_A_CDC_RX1_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX2 Digital Volume", SITAR_A_CDC_RX2_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX3 Digital Volume", SITAR_A_CDC_RX3_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+
+ SOC_SINGLE_S8_TLV("DEC1 Volume", SITAR_A_CDC_TX1_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC2 Volume", SITAR_A_CDC_TX2_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC3 Volume", SITAR_A_CDC_TX3_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC4 Volume", SITAR_A_CDC_TX4_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+
+ SOC_SINGLE_S8_TLV("IIR1 INP1 Volume", SITAR_A_CDC_IIR1_GAIN_B1_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP2 Volume", SITAR_A_CDC_IIR1_GAIN_B2_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP3 Volume", SITAR_A_CDC_IIR1_GAIN_B3_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP4 Volume", SITAR_A_CDC_IIR1_GAIN_B4_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_TLV("ADC1 Volume", SITAR_A_TX_1_2_EN, 5, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC2 Volume", SITAR_A_TX_1_2_EN, 1, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC3 Volume", SITAR_A_TX_3_EN, 5, 3, 0, analog_gain),
+
+ SOC_SINGLE("MICBIAS1 CAPLESS Switch", SITAR_A_MICB_1_CTL, 4, 1, 1),
+ SOC_SINGLE("MICBIAS2 CAPLESS Switch", SITAR_A_MICB_2_CTL, 4, 1, 1),
+
+ SOC_ENUM("TX1 HPF cut off", cf_dec1_enum),
+
+ SOC_SINGLE("TX1 HPF Switch", SITAR_A_CDC_TX1_MUX_CTL, 3, 1, 0),
+
+ SOC_SINGLE("RX1 HPF Switch", SITAR_A_CDC_RX1_B5_CTL, 2, 1, 0),
+
+ SOC_ENUM("RX1 HPF cut off", cf_rxmix1_enum),
+
+ SOC_SINGLE_EXT("IIR1 Enable Band1", IIR1, BAND1, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band2", IIR1, BAND2, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band3", IIR1, BAND3, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band4", IIR1, BAND4, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band5", IIR1, BAND5, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band1", IIR2, BAND1, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band2", IIR2, BAND2, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band3", IIR2, BAND3, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band4", IIR2, BAND4, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band5", IIR2, BAND5, 1, 0,
+ sitar_get_iir_enable_audio_mixer, sitar_put_iir_enable_audio_mixer),
+
+ SOC_SINGLE_MULTI_EXT("IIR1 Band1", IIR1, BAND1, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band2", IIR1, BAND2, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band3", IIR1, BAND3, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band4", IIR1, BAND4, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band5", IIR1, BAND5, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band1", IIR2, BAND1, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band2", IIR2, BAND2, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band3", IIR2, BAND3, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band4", IIR2, BAND4, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band5", IIR2, BAND5, 255, 0, 5,
+ sitar_get_iir_band_audio_mixer, sitar_put_iir_band_audio_mixer),
+};
+
+static const char *rx_mix1_text[] = {
+ "ZERO", "SRC1", "SRC2", "IIR1", "IIR2", "RX1", "RX2", "RX3", "RX4",
+ "RX5"
+};
+
+static const char *rx_dac1_text[] = {
+ "ZERO", "RX1", "RX2"
+};
+
+static const char *rx_dac2_text[] = {
+ "ZERO", "RX1",
+};
+
+static const char *rx_dac3_text[] = {
+ "ZERO", "RX1", "INV_RX1", "RX2"
+};
+
+static const char *rx_dac4_text[] = {
+ "ZERO", "ON"
+};
+
+static const char *sb_tx1_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC1"
+};
+
+static const char *sb_tx2_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC2"
+};
+
+static const char *sb_tx3_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC3"
+};
+
+static const char *sb_tx4_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC4"
+};
+
+static const char *sb_tx5_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "DEC1", "DEC2", "DEC3", "DEC4"
+};
+
+static const char *dec1_mux_text[] = {
+ "ZERO", "DMIC1", "ADC1", "ADC2", "ADC3", "MBADC", "DMIC4", "ANCFB1",
+};
+
+static const char *dec2_mux_text[] = {
+ "ZERO", "DMIC2", "ADC1", "ADC2", "ADC3", "MBADC", "DMIC3", "ANCFB2",
+};
+
+static const char *dec3_mux_text[] = {
+ "ZERO", "DMIC3", "ADC1", "ADC2", "ADC3", "MBADC", "DMIC2", "DMIC4"
+};
+
+static const char *dec4_mux_text[] = {
+ "ZERO", "DMIC4", "ADC1", "ADC2", "ADC3", "DMIC3", "DMIC2", "DMIC1"
+};
+
+static const char *iir1_inp1_text[] = {
+ "ZERO", "DEC1", "DEC2", "DEC3", "DEC4", "ZERO", "ZERO", "ZERO",
+ "ZERO", "ZERO", "ZERO", "RX1", "RX2", "RX3", "RX4", "RX5",
+};
+
+static const struct soc_enum rx_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_RX1_B1_CTL, 0, 10, rx_mix1_text);
+
+static const struct soc_enum rx_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_RX1_B1_CTL, 4, 10, rx_mix1_text);
+
+static const struct soc_enum rx2_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_RX2_B1_CTL, 0, 10, rx_mix1_text);
+
+static const struct soc_enum rx2_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_RX2_B1_CTL, 4, 10, rx_mix1_text);
+
+static const struct soc_enum rx3_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_RX3_B1_CTL, 0, 10, rx_mix1_text);
+
+static const struct soc_enum rx3_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_RX3_B1_CTL, 4, 10, rx_mix1_text);
+
+static const struct soc_enum rx_dac1_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_TOP_RDAC_DOUT_CTL, 6, 3, rx_dac1_text);
+
+static const struct soc_enum rx_dac2_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_TOP_RDAC_DOUT_CTL, 4, 2, rx_dac2_text);
+
+static const struct soc_enum rx_dac3_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_TOP_RDAC_DOUT_CTL, 2, 4, rx_dac3_text);
+
+static const struct soc_enum rx_dac4_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_TOP_RDAC_DOUT_CTL, 0, 2, rx_dac4_text);
+
+static const struct soc_enum sb_tx5_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_SB_B5_CTL, 0, 9, sb_tx5_mux_text);
+
+static const struct soc_enum sb_tx4_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_SB_B4_CTL, 0, 9, sb_tx4_mux_text);
+
+static const struct soc_enum sb_tx3_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_SB_B3_CTL, 0, 9, sb_tx3_mux_text);
+
+static const struct soc_enum sb_tx2_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_SB_B2_CTL, 0, 9, sb_tx2_mux_text);
+
+static const struct soc_enum sb_tx1_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_SB_B1_CTL, 0, 9, sb_tx1_mux_text);
+
+static const struct soc_enum dec1_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_B1_CTL, 0, 8, dec1_mux_text);
+
+static const struct soc_enum dec2_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_B1_CTL, 3, 8, dec2_mux_text);
+
+static const struct soc_enum dec3_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_B2_CTL, 0, 8, dec3_mux_text);
+
+static const struct soc_enum dec4_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_TX_B2_CTL, 3, 8, dec4_mux_text);
+
+static const struct soc_enum iir1_inp1_mux_enum =
+ SOC_ENUM_SINGLE(SITAR_A_CDC_CONN_EQ1_B1_CTL, 0, 16, iir1_inp1_text);
+
+static const struct snd_kcontrol_new rx_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX1 MIX1 INP1 Mux", rx_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX1 MIX1 INP2 Mux", rx_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx2_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX2 MIX1 INP1 Mux", rx2_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx2_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX2 MIX1 INP2 Mux", rx2_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx3_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX3 MIX1 INP1 Mux", rx3_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx3_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX3 MIX1 INP2 Mux", rx3_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_dac1_mux =
+ SOC_DAPM_ENUM("RX DAC1 Mux", rx_dac1_enum);
+
+static const struct snd_kcontrol_new rx_dac2_mux =
+ SOC_DAPM_ENUM("RX DAC2 Mux", rx_dac2_enum);
+
+static const struct snd_kcontrol_new rx_dac3_mux =
+ SOC_DAPM_ENUM("RX DAC3 Mux", rx_dac3_enum);
+
+static const struct snd_kcontrol_new rx_dac4_mux =
+ SOC_DAPM_ENUM("RX DAC4 Mux", rx_dac4_enum);
+
+static const struct snd_kcontrol_new sb_tx5_mux =
+ SOC_DAPM_ENUM("SLIM TX5 MUX Mux", sb_tx5_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx4_mux =
+ SOC_DAPM_ENUM("SLIM TX4 MUX Mux", sb_tx4_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx3_mux =
+ SOC_DAPM_ENUM("SLIM TX3 MUX Mux", sb_tx3_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx2_mux =
+ SOC_DAPM_ENUM("SLIM TX2 MUX Mux", sb_tx2_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx1_mux =
+ SOC_DAPM_ENUM("SLIM TX1 MUX Mux", sb_tx1_mux_enum);
+
+static const struct snd_kcontrol_new dec1_mux =
+ SOC_DAPM_ENUM("DEC1 MUX Mux", dec1_mux_enum);
+
+static const struct snd_kcontrol_new dec2_mux =
+ SOC_DAPM_ENUM("DEC2 MUX Mux", dec2_mux_enum);
+
+static const struct snd_kcontrol_new dec3_mux =
+ SOC_DAPM_ENUM("DEC3 MUX Mux", dec3_mux_enum);
+
+static const struct snd_kcontrol_new dec4_mux =
+ SOC_DAPM_ENUM("DEC4 MUX Mux", dec4_mux_enum);
+
+static const struct snd_kcontrol_new iir1_inp1_mux =
+ SOC_DAPM_ENUM("IIR1 INP1 Mux", iir1_inp1_mux_enum);
+
+static const struct snd_kcontrol_new dac1_switch[] = {
+ SOC_DAPM_SINGLE("Switch", SITAR_A_RX_EAR_EN, 5, 1, 0),
+};
+
+static void sitar_codec_enable_adc_block(struct snd_soc_codec *codec,
+ int enable)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s %d\n", __func__, enable);
+
+ if (enable) {
+ sitar->adc_count++;
+ snd_soc_update_bits(codec, SITAR_A_TX_COM_BIAS, 0xE0, 0xE0);
+
+ } else {
+ sitar->adc_count--;
+ if (!sitar->adc_count) {
+ if (!sitar->mbhc_polling_active)
+ snd_soc_update_bits(codec, SITAR_A_TX_COM_BIAS,
+ 0xE0, 0x0);
+ }
+ }
+}
+
+static int sitar_codec_enable_adc(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 adc_reg;
+
+ pr_debug("%s %d\n", __func__, event);
+
+ if (w->reg == SITAR_A_TX_1_2_EN)
+ adc_reg = SITAR_A_TX_1_2_TEST_CTL;
+ else {
+ pr_err("%s: Error, invalid adc register\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ sitar_codec_enable_adc_block(codec, 1);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, adc_reg, 1 << w->shift,
+ 1 << w->shift);
+ usleep_range(1000, 1000);
+ snd_soc_update_bits(codec, adc_reg, 1 << w->shift, 0x00);
+ usleep_range(1000, 1000);
+ snd_soc_update_bits(codec, adc_reg, 0x08, 0x08);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ sitar_codec_enable_adc_block(codec, 0);
+ break;
+ }
+ return 0;
+}
+
+static int sitar_lineout_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %s %d\n", __func__, w->name, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x40);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static int sitar_codec_enable_lineout(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 lineout_gain_reg;
+
+ pr_debug("%s %d %s\n", __func__, event, w->name);
+
+ switch (w->shift) {
+ case 0:
+ lineout_gain_reg = SITAR_A_RX_LINE_1_GAIN;
+ break;
+ case 1:
+ lineout_gain_reg = SITAR_A_RX_LINE_2_GAIN;
+ break;
+ default:
+ pr_err("%s: Error, incorrect lineout register value\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, lineout_gain_reg, 0x10, 0x10);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ pr_debug("%s: sleeping 16 ms after %s PA turn on\n",
+ __func__, w->name);
+ usleep_range(16000, 16000);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, lineout_gain_reg, 0x10, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static int sitar_codec_enable_dmic(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 tx_dmic_ctl_reg, tx_mux_ctl_reg;
+ u8 dmic_clk_sel, dmic_clk_en;
+ unsigned int dmic;
+ int ret;
+
+ ret = kstrtouint(strpbrk(w->name, "1234"), 10, &dmic);
+ if (ret < 0) {
+ pr_err("%s: Invalid DMIC line on the codec\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (dmic) {
+ case 1:
+ case 2:
+ dmic_clk_sel = 0x02;
+ dmic_clk_en = 0x01;
+ break;
+ case 3:
+ case 4:
+ dmic_clk_sel = 0x08;
+ dmic_clk_en = 0x04;
+ break;
+
+ break;
+
+ default:
+ pr_err("%s: Invalid DMIC Selection\n", __func__);
+ return -EINVAL;
+ }
+
+ tx_mux_ctl_reg = SITAR_A_CDC_TX1_MUX_CTL + 8 * (dmic - 1);
+ tx_dmic_ctl_reg = SITAR_A_CDC_TX1_DMIC_CTL + 8 * (dmic - 1);
+
+ pr_debug("%s %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x01, 0x01);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_DMIC_CTL,
+ dmic_clk_sel, dmic_clk_sel);
+
+ snd_soc_update_bits(codec, tx_dmic_ctl_reg, 0x1, 0x1);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_DMIC_CTL,
+ dmic_clk_en, dmic_clk_en);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_DMIC_CTL,
+ dmic_clk_en, 0);
+
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x01, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static void sitar_codec_start_hs_polling(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ int mbhc_state = sitar->mbhc_state;
+
+ pr_debug("%s: enter\n", __func__);
+ if (!sitar->mbhc_polling_active) {
+ pr_debug("Polling is not active, do not start polling\n");
+ return;
+ }
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x84);
+
+
+ if (!sitar->no_mic_headset_override) {
+ if (mbhc_state == MBHC_STATE_POTENTIAL) {
+ pr_debug("%s recovering MBHC state macine\n", __func__);
+ sitar->mbhc_state = MBHC_STATE_POTENTIAL_RECOVERY;
+ /* set to max button press threshold */
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B2_CTL,
+ 0x7F);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B1_CTL,
+ 0xFF);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B4_CTL,
+ 0x7F);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B3_CTL,
+ 0xFF);
+ /* set to max */
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B6_CTL,
+ 0x7F);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B5_CTL,
+ 0xFF);
+ }
+ }
+
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x84);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x1);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x0);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x1);
+}
+
+static void sitar_codec_pause_hs_polling(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: enter\n", __func__);
+ if (!sitar->mbhc_polling_active) {
+ pr_debug("polling not active, nothing to pause\n");
+ return;
+ }
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ pr_debug("%s: leave\n", __func__);
+
+}
+
+static void sitar_codec_switch_cfilt_mode(struct snd_soc_codec *codec,
+ int mode)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ u8 reg_mode_val, cur_mode_val;
+ bool mbhc_was_polling = false;
+
+ if (mode)
+ reg_mode_val = SITAR_CFILT_FAST_MODE;
+ else
+ reg_mode_val = SITAR_CFILT_SLOW_MODE;
+
+ cur_mode_val = snd_soc_read(codec,
+ sitar->mbhc_bias_regs.cfilt_ctl) & 0x40;
+
+ if (cur_mode_val != reg_mode_val) {
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ if (sitar->mbhc_polling_active) {
+ sitar_codec_pause_hs_polling(codec);
+ mbhc_was_polling = true;
+ }
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.cfilt_ctl, 0x40, reg_mode_val);
+ if (mbhc_was_polling)
+ sitar_codec_start_hs_polling(codec);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ pr_debug("%s: CFILT mode change (%x to %x)\n", __func__,
+ cur_mode_val, reg_mode_val);
+ } else {
+ pr_err("%s: CFILT Value is already %x\n",
+ __func__, cur_mode_val);
+ }
+}
+
+static void sitar_codec_update_cfilt_usage(struct snd_soc_codec *codec,
+ u8 cfilt_sel, int inc)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ u32 *cfilt_cnt_ptr = NULL;
+ u16 micb_cfilt_reg;
+
+ switch (cfilt_sel) {
+ case SITAR_CFILT1_SEL:
+ cfilt_cnt_ptr = &sitar->cfilt1_cnt;
+ micb_cfilt_reg = SITAR_A_MICB_CFILT_1_CTL;
+ break;
+ case SITAR_CFILT2_SEL:
+ cfilt_cnt_ptr = &sitar->cfilt2_cnt;
+ micb_cfilt_reg = SITAR_A_MICB_CFILT_2_CTL;
+ break;
+ default:
+ return; /* should not happen */
+ }
+
+ if (inc) {
+ if (!(*cfilt_cnt_ptr)++) {
+ /* Switch CFILT to slow mode if MBHC CFILT being used */
+ if (cfilt_sel == sitar->mbhc_bias_regs.cfilt_sel)
+ sitar_codec_switch_cfilt_mode(codec, 0);
+
+ snd_soc_update_bits(codec, micb_cfilt_reg, 0x80, 0x80);
+ }
+ } else {
+ /* check if count not zero, decrement
+ * then check if zero, go ahead disable cfilter
+ */
+ if ((*cfilt_cnt_ptr) && !--(*cfilt_cnt_ptr)) {
+ snd_soc_update_bits(codec, micb_cfilt_reg, 0x80, 0);
+
+ /* Switch CFILT to fast mode if MBHC CFILT being used */
+ if (cfilt_sel == sitar->mbhc_bias_regs.cfilt_sel)
+ sitar_codec_switch_cfilt_mode(codec, 1);
+ }
+ }
+}
+
+static int sitar_find_k_value(unsigned int ldoh_v, unsigned int cfilt_mv)
+{
+ int rc = -EINVAL;
+ unsigned min_mv, max_mv;
+
+ switch (ldoh_v) {
+ case SITAR_LDOH_1P95_V:
+ min_mv = 160;
+ max_mv = 1800;
+ break;
+ case SITAR_LDOH_2P35_V:
+ min_mv = 200;
+ max_mv = 2200;
+ break;
+ case SITAR_LDOH_2P75_V:
+ min_mv = 240;
+ max_mv = 2600;
+ break;
+ case SITAR_LDOH_2P85_V:
+ min_mv = 250;
+ max_mv = 2700;
+ break;
+ default:
+ goto done;
+ }
+
+ if (cfilt_mv < min_mv || cfilt_mv > max_mv)
+ goto done;
+
+ for (rc = 4; rc <= 44; rc++) {
+ min_mv = max_mv * (rc) / 44;
+ if (min_mv >= cfilt_mv) {
+ rc -= 4;
+ break;
+ }
+ }
+done:
+ return rc;
+}
+
+static bool sitar_is_hph_pa_on(struct snd_soc_codec *codec)
+{
+ u8 hph_reg_val = 0;
+ hph_reg_val = snd_soc_read(codec, SITAR_A_RX_HPH_CNP_EN);
+
+ return (hph_reg_val & 0x30) ? true : false;
+}
+
+static bool sitar_is_hph_dac_on(struct snd_soc_codec *codec, int left)
+{
+ u8 hph_reg_val = 0;
+ if (left)
+ hph_reg_val = snd_soc_read(codec,
+ SITAR_A_RX_HPH_L_DAC_CTL);
+ else
+ hph_reg_val = snd_soc_read(codec,
+ SITAR_A_RX_HPH_R_DAC_CTL);
+
+ return (hph_reg_val & 0xC0) ? true : false;
+}
+
+static void sitar_codec_switch_micbias(struct snd_soc_codec *codec,
+ int vddio_switch)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ int cfilt_k_val;
+ bool mbhc_was_polling = false;
+
+ switch (vddio_switch) {
+ case 1:
+ if (sitar->mbhc_micbias_switched == 0 &&
+ sitar->mbhc_polling_active) {
+
+ sitar_codec_pause_hs_polling(codec);
+ /* Enable Mic Bias switch to VDDIO */
+ sitar->cfilt_k_value = snd_soc_read(codec,
+ sitar->mbhc_bias_regs.cfilt_val);
+ cfilt_k_val = sitar_find_k_value(
+ sitar->pdata->micbias.ldoh_v, 1800);
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.cfilt_val,
+ 0xFC, (cfilt_k_val << 2));
+
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.mbhc_reg, 0x80, 0x80);
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.mbhc_reg, 0x10, 0x00);
+ sitar_codec_start_hs_polling(codec);
+
+ sitar->mbhc_micbias_switched = true;
+ pr_debug("%s: Enabled MBHC Mic bias to VDDIO Switch\n",
+ __func__);
+ }
+ break;
+
+ case 0:
+ if (sitar->mbhc_micbias_switched) {
+ if (sitar->mbhc_polling_active) {
+ sitar_codec_pause_hs_polling(codec);
+ mbhc_was_polling = true;
+ }
+ /* Disable Mic Bias switch to VDDIO */
+ if (sitar->cfilt_k_value != 0)
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.cfilt_val, 0XFC,
+ sitar->cfilt_k_value);
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.mbhc_reg, 0x80, 0x00);
+ snd_soc_update_bits(codec,
+ sitar->mbhc_bias_regs.mbhc_reg, 0x10, 0x00);
+
+ if (mbhc_was_polling)
+ sitar_codec_start_hs_polling(codec);
+
+ sitar->mbhc_micbias_switched = false;
+ pr_debug("%s: Disabled MBHC Mic bias to VDDIO Switch\n",
+ __func__);
+ }
+ break;
+ }
+}
+
+static int sitar_codec_enable_micbias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ u16 micb_int_reg;
+ int micb_line;
+ u8 cfilt_sel_val = 0;
+ char *internal1_text = "Internal1";
+ char *internal2_text = "Internal2";
+
+ pr_debug("%s %d\n", __func__, event);
+ switch (w->reg) {
+ case SITAR_A_MICB_1_CTL:
+ micb_int_reg = SITAR_A_MICB_1_INT_RBIAS;
+ cfilt_sel_val = sitar->pdata->micbias.bias1_cfilt_sel;
+ micb_line = SITAR_MICBIAS1;
+ break;
+ case SITAR_A_MICB_2_CTL:
+ micb_int_reg = SITAR_A_MICB_2_INT_RBIAS;
+ cfilt_sel_val = sitar->pdata->micbias.bias2_cfilt_sel;
+ micb_line = SITAR_MICBIAS2;
+ break;
+ default:
+ pr_err("%s: Error, invalid micbias register\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Decide whether to switch the micbias for MBHC */
+ if (w->reg == sitar->mbhc_bias_regs.ctl_reg) {
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ sitar_codec_switch_micbias(codec, 0);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ }
+
+ snd_soc_update_bits(codec, w->reg, 0x1E, 0x00);
+ sitar_codec_update_cfilt_usage(codec, cfilt_sel_val, 1);
+
+ if (strnstr(w->name, internal1_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0xFF, 0xA4);
+ else if (strnstr(w->name, internal2_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x1C, 0x1C);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ if (sitar->mbhc_polling_active &&
+ sitar->mbhc_cfg.micbias == micb_line) {
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ sitar_codec_pause_hs_polling(codec);
+ sitar_codec_start_hs_polling(codec);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+
+ if ((w->reg == sitar->mbhc_bias_regs.ctl_reg)
+ && sitar_is_hph_pa_on(codec))
+ sitar_codec_switch_micbias(codec, 1);
+
+ if (strnstr(w->name, internal1_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x80, 0x00);
+ else if (strnstr(w->name, internal2_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x10, 0x00);
+ sitar_codec_update_cfilt_usage(codec, cfilt_sel_val, 0);
+ break;
+ }
+
+ return 0;
+}
+
+static int sitar_codec_enable_dec(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 dec_reset_reg;
+
+ pr_debug("%s %d\n", __func__, event);
+
+ if (w->reg == SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL)
+ dec_reset_reg = SITAR_A_CDC_CLK_TX_RESET_B1_CTL;
+ else {
+ pr_err("%s: Error, incorrect dec\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, dec_reset_reg, 1 << w->shift,
+ 1 << w->shift);
+ snd_soc_update_bits(codec, dec_reset_reg, 1 << w->shift, 0x0);
+ break;
+ }
+ return 0;
+}
+
+static int sitar_codec_reset_interpolator(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %d %s\n", __func__, event, w->name);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_RX_RESET_CTL,
+ 1 << w->shift, 1 << w->shift);
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_RX_RESET_CTL,
+ 1 << w->shift, 0x0);
+ break;
+ }
+ return 0;
+}
+
+static int sitar_codec_enable_ldo_h(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ case SND_SOC_DAPM_POST_PMD:
+ usleep_range(1000, 1000);
+ pr_debug("LDO_H\n");
+ break;
+ }
+ return 0;
+}
+
+static void sitar_enable_rx_bias(struct snd_soc_codec *codec, u32 enable)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ if (enable) {
+ sitar->rx_bias_count++;
+ if (sitar->rx_bias_count == 1)
+ snd_soc_update_bits(codec, SITAR_A_RX_COM_BIAS,
+ 0x80, 0x80);
+ } else {
+ sitar->rx_bias_count--;
+ if (!sitar->rx_bias_count)
+ snd_soc_update_bits(codec, SITAR_A_RX_COM_BIAS,
+ 0x80, 0x00);
+ }
+}
+
+static int sitar_codec_enable_rx_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ sitar_enable_rx_bias(codec, 1);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ sitar_enable_rx_bias(codec, 0);
+ break;
+ }
+ return 0;
+}
+static int sitar_hph_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %s %d\n", __func__, w->name, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x40);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static void sitar_snd_soc_jack_report(struct sitar_priv *sitar,
+ struct snd_soc_jack *jack, int status,
+ int mask)
+{
+ /* XXX: wake_lock_timeout()? */
+ snd_soc_jack_report(jack, status, mask);
+}
+
+static void hphocp_off_report(struct sitar_priv *sitar,
+ u32 jack_status, int irq)
+{
+ struct snd_soc_codec *codec;
+
+ if (!sitar) {
+ pr_err("%s: Bad sitar private data\n", __func__);
+ return;
+ }
+
+ pr_info("%s: clear ocp status %x\n", __func__, jack_status);
+ codec = sitar->codec;
+ if (sitar->hph_status & jack_status) {
+ sitar->hph_status &= ~jack_status;
+ if (sitar->mbhc_cfg.headset_jack)
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.headset_jack,
+ sitar->hph_status,
+ SITAR_JACK_MASK);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL, 0x10, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL, 0x10, 0x10);
+ /* reset retry counter as PA is turned off signifying
+ * start of new OCP detection session
+ */
+ if (SITAR_IRQ_HPH_PA_OCPL_FAULT)
+ sitar->hphlocp_cnt = 0;
+ else
+ sitar->hphrocp_cnt = 0;
+ wcd9xxx_enable_irq(codec->control_data, irq);
+ }
+}
+
+static void hphlocp_off_report(struct work_struct *work)
+{
+ struct sitar_priv *sitar = container_of(work, struct sitar_priv,
+ hphlocp_work);
+ hphocp_off_report(sitar, SND_JACK_OC_HPHL, SITAR_IRQ_HPH_PA_OCPL_FAULT);
+}
+
+static void hphrocp_off_report(struct work_struct *work)
+{
+ struct sitar_priv *sitar = container_of(work, struct sitar_priv,
+ hphrocp_work);
+ hphocp_off_report(sitar, SND_JACK_OC_HPHR, SITAR_IRQ_HPH_PA_OCPR_FAULT);
+}
+
+static int sitar_hph_pa_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ u8 mbhc_micb_ctl_val;
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ mbhc_micb_ctl_val = snd_soc_read(codec,
+ sitar->mbhc_bias_regs.ctl_reg);
+
+ if (!(mbhc_micb_ctl_val & 0x80)) {
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ sitar_codec_switch_micbias(codec, 1);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ }
+
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ /* schedule work is required because at the time HPH PA DAPM
+ * event callback is called by DAPM framework, CODEC dapm mutex
+ * would have been locked while snd_soc_jack_report also
+ * attempts to acquire same lock.
+ */
+ if (w->shift == 5) {
+ clear_bit(SITAR_HPHL_PA_OFF_ACK,
+ &sitar->hph_pa_dac_state);
+ clear_bit(SITAR_HPHL_DAC_OFF_ACK,
+ &sitar->hph_pa_dac_state);
+ if (sitar->hph_status & SND_JACK_OC_HPHL)
+ schedule_work(&sitar->hphlocp_work);
+ } else if (w->shift == 4) {
+ clear_bit(SITAR_HPHR_PA_OFF_ACK,
+ &sitar->hph_pa_dac_state);
+ clear_bit(SITAR_HPHR_DAC_OFF_ACK,
+ &sitar->hph_pa_dac_state);
+ if (sitar->hph_status & SND_JACK_OC_HPHR)
+ schedule_work(&sitar->hphrocp_work);
+ }
+
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ sitar_codec_switch_micbias(codec, 0);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+
+ pr_debug("%s: sleep 10 ms after %s PA disable.\n", __func__,
+ w->name);
+ usleep_range(10000, 10000);
+
+ break;
+ }
+ return 0;
+}
+
+static void sitar_get_mbhc_micbias_regs(struct snd_soc_codec *codec,
+ struct mbhc_micbias_regs *micbias_regs)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ unsigned int cfilt;
+
+ switch (sitar->mbhc_cfg.micbias) {
+ case SITAR_MICBIAS1:
+ cfilt = sitar->pdata->micbias.bias1_cfilt_sel;
+ micbias_regs->mbhc_reg = SITAR_A_MICB_1_MBHC;
+ micbias_regs->int_rbias = SITAR_A_MICB_1_INT_RBIAS;
+ micbias_regs->ctl_reg = SITAR_A_MICB_1_CTL;
+ break;
+ case SITAR_MICBIAS2:
+ cfilt = sitar->pdata->micbias.bias2_cfilt_sel;
+ micbias_regs->mbhc_reg = SITAR_A_MICB_2_MBHC;
+ micbias_regs->int_rbias = SITAR_A_MICB_2_INT_RBIAS;
+ micbias_regs->ctl_reg = SITAR_A_MICB_2_CTL;
+ break;
+ default:
+ /* Should never reach here */
+ pr_err("%s: Invalid MIC BIAS for MBHC\n", __func__);
+ return;
+ }
+
+ micbias_regs->cfilt_sel = cfilt;
+
+ switch (cfilt) {
+ case SITAR_CFILT1_SEL:
+ micbias_regs->cfilt_val = SITAR_A_MICB_CFILT_1_VAL;
+ micbias_regs->cfilt_ctl = SITAR_A_MICB_CFILT_1_CTL;
+ sitar->mbhc_data.micb_mv = sitar->pdata->micbias.cfilt1_mv;
+ break;
+ case SITAR_CFILT2_SEL:
+ micbias_regs->cfilt_val = SITAR_A_MICB_CFILT_2_VAL;
+ micbias_regs->cfilt_ctl = SITAR_A_MICB_CFILT_2_CTL;
+ sitar->mbhc_data.micb_mv = sitar->pdata->micbias.cfilt2_mv;
+ break;
+ }
+}
+
+static int sitar_codec_enable_charge_pump(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %d\n", __func__, event);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_OTHR_RESET_CTL, 0x10,
+ 0x00);
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_OTHR_CTL, 0x01,
+ 0x01);
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLSG_CTL, 0x08, 0x08);
+ usleep_range(200, 200);
+ snd_soc_update_bits(codec, SITAR_A_CP_STATIC, 0x10, 0x00);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_OTHR_RESET_CTL, 0x10,
+ 0x10);
+ usleep_range(20, 20);
+ snd_soc_update_bits(codec, SITAR_A_CP_STATIC, 0x08, 0x08);
+ snd_soc_update_bits(codec, SITAR_A_CP_STATIC, 0x10, 0x10);
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLSG_CTL, 0x08, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_OTHR_CTL, 0x01,
+ 0x00);
+ snd_soc_update_bits(codec, SITAR_A_CP_STATIC, 0x08, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget sitar_dapm_i2s_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("RX_I2S_CLK", SITAR_A_CDC_CLK_RX_I2S_CTL,
+ 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("TX_I2S_CLK", SITAR_A_CDC_CLK_TX_I2S_CTL, 4,
+ 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget sitar_dapm_widgets[] = {
+ /*RX stuff */
+ SND_SOC_DAPM_OUTPUT("EAR"),
+
+ SND_SOC_DAPM_PGA("EAR PA", SITAR_A_RX_EAR_EN, 4, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("DAC1", SITAR_A_RX_EAR_EN, 6, 0, dac1_switch,
+ ARRAY_SIZE(dac1_switch)),
+ SND_SOC_DAPM_SUPPLY("EAR DRIVER", SITAR_A_RX_EAR_EN, 3, 0, NULL, 0),
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX1", "AIF1 Playback", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX2", "AIF1 Playback", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN("SLIM RX3", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIM RX4", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIM RX5", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Headphone */
+ SND_SOC_DAPM_OUTPUT("HEADPHONE"),
+ SND_SOC_DAPM_PGA_E("HPHL", SITAR_A_RX_HPH_CNP_EN, 5, 0, NULL, 0,
+ sitar_hph_pa_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_PGA_E("HPHR", SITAR_A_RX_HPH_CNP_EN, 4, 0, NULL, 0,
+ sitar_hph_pa_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_DAC_E("HPHL DAC", NULL, SITAR_A_RX_HPH_L_DAC_CTL, 7, 0,
+ sitar_hph_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("HPHR DAC", NULL, SITAR_A_RX_HPH_R_DAC_CTL, 7, 0,
+ sitar_hph_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Speaker */
+ SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+
+ SND_SOC_DAPM_DAC_E("LINEOUT1 DAC", NULL, SITAR_A_RX_LINE_1_DAC_CTL, 7, 0
+ , sitar_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("LINEOUT2 DAC", NULL, SITAR_A_RX_LINE_2_DAC_CTL, 7, 0
+ , sitar_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_PGA_E("LINEOUT1 PA", SITAR_A_RX_LINE_CNP_EN, 0, 0, NULL,
+ 0, sitar_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LINEOUT2 PA", SITAR_A_RX_LINE_CNP_EN, 1, 0, NULL,
+ 0, sitar_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MIXER_E("RX1 MIX1", SITAR_A_CDC_CLK_RX_B1_CTL, 0, 0, NULL,
+ 0, sitar_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MIXER_E("RX2 MIX1", SITAR_A_CDC_CLK_RX_B1_CTL, 1, 0, NULL,
+ 0, sitar_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MIXER_E("RX3 MIX1", SITAR_A_CDC_CLK_RX_B1_CTL, 2, 0, NULL,
+ 0, sitar_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU),
+
+ SND_SOC_DAPM_MUX("DAC1 MUX", SND_SOC_NOPM, 0, 0,
+ &rx_dac1_mux),
+ SND_SOC_DAPM_MUX("DAC2 MUX", SND_SOC_NOPM, 0, 0,
+ &rx_dac2_mux),
+ SND_SOC_DAPM_MUX("DAC3 MUX", SND_SOC_NOPM, 0, 0,
+ &rx_dac3_mux),
+ SND_SOC_DAPM_MUX("DAC4 MUX", SND_SOC_NOPM, 0, 0,
+ &rx_dac4_mux),
+
+ SND_SOC_DAPM_MIXER("RX1 CHAIN", SITAR_A_CDC_RX1_B6_CTL, 5, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX2 CHAIN", SITAR_A_CDC_RX2_B6_CTL, 5, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX3 CHAIN", SITAR_A_CDC_RX3_B6_CTL, 5, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("RX1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx2_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx2_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX3 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx3_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX3 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx3_mix1_inp2_mux),
+
+ SND_SOC_DAPM_SUPPLY("CP", SITAR_A_CP_EN, 0, 0,
+ sitar_codec_enable_charge_pump, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_SUPPLY("RX_BIAS", SND_SOC_NOPM, 0, 0,
+ sitar_codec_enable_rx_bias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SUPPLY("LDO_H", SITAR_A_LDO_H_MODE_1, 7, 0,
+ sitar_codec_enable_ldo_h, SND_SOC_DAPM_POST_PMU),
+ /* TX */
+
+ SND_SOC_DAPM_SUPPLY("CDC_CONN", SITAR_A_CDC_CLK_OTHR_CTL, 2, 0, NULL,
+ 0),
+ SND_SOC_DAPM_INPUT("AMIC1"),
+ SND_SOC_DAPM_INPUT("AMIC2"),
+ SND_SOC_DAPM_INPUT("AMIC3"),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS1 External", SITAR_A_MICB_1_CTL, 7, 0,
+ sitar_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS1 Internal1", SITAR_A_MICB_1_CTL, 7, 0,
+ sitar_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 External", SITAR_A_MICB_2_CTL, 7, 0,
+ sitar_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 Internal1", SITAR_A_MICB_2_CTL, 7, 0,
+ sitar_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 Internal2", SITAR_A_MICB_2_CTL, 7, 0,
+ sitar_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC_E("ADC1", NULL, SITAR_A_TX_1_2_EN, 7, 0,
+ sitar_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("ADC2", NULL, SITAR_A_TX_1_2_EN, 3, 0,
+ sitar_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("ADC3", NULL, SITAR_A_TX_3_EN, 7, 0,
+ sitar_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC1 MUX", SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL, 0, 0,
+ &dec1_mux, sitar_codec_enable_dec, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MUX_E("DEC2 MUX", SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL, 1, 0,
+ &dec2_mux, sitar_codec_enable_dec, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MUX_E("DEC3 MUX", SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL, 2, 0,
+ &dec3_mux, sitar_codec_enable_dec, SND_SOC_DAPM_PRE_PMU),
+ SND_SOC_DAPM_MUX_E("DEC4 MUX", SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL, 3, 0,
+ &dec4_mux, sitar_codec_enable_dec, SND_SOC_DAPM_PRE_PMU),
+
+ SND_SOC_DAPM_MUX("SLIM TX1 MUX", SND_SOC_NOPM, 0, 0, &sb_tx1_mux),
+ SND_SOC_DAPM_MUX("SLIM TX2 MUX", SND_SOC_NOPM, 0, 0, &sb_tx2_mux),
+ SND_SOC_DAPM_MUX("SLIM TX3 MUX", SND_SOC_NOPM, 0, 0, &sb_tx3_mux),
+ SND_SOC_DAPM_MUX("SLIM TX4 MUX", SND_SOC_NOPM, 0, 0, &sb_tx4_mux),
+ SND_SOC_DAPM_MUX("SLIM TX5 MUX", SND_SOC_NOPM, 0, 0, &sb_tx5_mux),
+
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX1", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX2", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX3", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX4", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX5", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, sitar_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Digital Mic Inputs */
+ SND_SOC_DAPM_ADC_E("DMIC1", NULL, SND_SOC_NOPM, 0, 0,
+ sitar_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC2", NULL, SND_SOC_NOPM, 0, 0,
+ sitar_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC3", NULL, SND_SOC_NOPM, 0, 0,
+ sitar_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC4", NULL, SND_SOC_NOPM, 0, 0,
+ sitar_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ /* Sidetone */
+ SND_SOC_DAPM_MUX("IIR1 INP1 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp1_mux),
+ SND_SOC_DAPM_PGA("IIR1", SITAR_A_CDC_CLK_SD_CTL, 0, 0, NULL, 0),
+
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Earpiece (RX MIX1) */
+ {"EAR", NULL, "EAR PA"},
+ {"EAR PA", "NULL", "DAC1"},
+ {"DAC1", "Switch", "DAC1 MUX"},
+ {"DAC1", NULL, "CP"},
+ {"DAC1", NULL, "EAR DRIVER"},
+
+ {"CP", NULL, "RX_BIAS"},
+
+ {"LINEOUT1 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT2 DAC", NULL, "RX_BIAS"},
+
+ {"LINEOUT2", NULL, "LINEOUT2 PA"},
+ {"LINEOUT2 PA", NULL, "LINEOUT2 DAC"},
+ {"LINEOUT2 DAC", NULL, "DAC3 MUX"},
+
+ {"LINEOUT1", NULL, "LINEOUT1 PA"},
+ {"LINEOUT1 PA", NULL, "LINEOUT1 DAC"},
+ {"LINEOUT1 DAC", NULL, "DAC2 MUX"},
+
+
+ /* Headset (RX MIX1 and RX MIX2) */
+ {"HEADPHONE", NULL, "HPHL"},
+ {"HEADPHONE", NULL, "HPHR"},
+
+ {"HPHL DAC", NULL, "CP"},
+ {"HPHR DAC", NULL, "CP"},
+
+ {"HPHL", NULL, "HPHL DAC"},
+ {"HPHL DAC", "NULL", "DAC4 MUX"},
+ {"HPHR", NULL, "HPHR DAC"},
+ {"HPHR DAC", NULL, "RX3 MIX1"},
+
+ {"DAC1 MUX", "RX1", "RX1 CHAIN"},
+ {"DAC2 MUX", "RX1", "RX1 CHAIN"},
+
+ {"DAC3 MUX", "RX1", "RX1 CHAIN"},
+ {"DAC3 MUX", "INV_RX1", "RX1 CHAIN"},
+ {"DAC3 MUX", "RX2", "RX2 MIX1"},
+
+ {"DAC4 MUX", "ON", "RX2 MIX1"},
+
+ {"RX1 CHAIN", NULL, "RX1 MIX1"},
+
+ {"RX1 MIX1", NULL, "RX1 MIX1 INP1"},
+ {"RX1 MIX1", NULL, "RX1 MIX1 INP2"},
+ {"RX2 MIX1", NULL, "RX2 MIX1 INP1"},
+ {"RX2 MIX1", NULL, "RX2 MIX1 INP2"},
+ {"RX3 MIX1", NULL, "RX3 MIX1 INP1"},
+ {"RX3 MIX1", NULL, "RX3 MIX1 INP2"},
+
+ /* SLIMBUS Connections */
+
+ /* Slimbus port 5 is non functional in Sitar 1.0 */
+ {"RX1 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX1 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX1 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX1 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX1 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX1 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX1 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX1 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX1 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX1 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX2 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX2 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX2 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX2 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX2 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX2 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX2 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX2 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX2 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX2 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX3 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX3 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX3 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX3 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX3 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX3 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX3 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX3 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX3 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX3 MIX1 INP2", "IIR1", "IIR1"},
+
+
+ /* TX */
+ {"SLIM TX1", NULL, "SLIM TX1 MUX"},
+ {"SLIM TX2", NULL, "SLIM TX2 MUX"},
+ {"SLIM TX3", NULL, "SLIM TX3 MUX"},
+ {"SLIM TX4", NULL, "SLIM TX4 MUX"},
+ {"SLIM TX5", NULL, "SLIM TX5 MUX"},
+
+ {"SLIM TX1 MUX", "DEC1", "DEC1 MUX"},
+ {"SLIM TX2 MUX", "DEC2", "DEC2 MUX"},
+ {"SLIM TX3 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX4 MUX", "DEC4", "DEC4 MUX"},
+ {"SLIM TX5 MUX", "DEC1", "DEC1 MUX"},
+ {"SLIM TX5 MUX", "DEC2", "DEC2 MUX"},
+ {"SLIM TX5 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX5 MUX", "DEC4", "DEC4 MUX"},
+
+ /* Decimator Inputs */
+ {"DEC1 MUX", "DMIC1", "DMIC1"},
+ {"DEC1 MUX", "DMIC4", "DMIC4"},
+ {"DEC1 MUX", "ADC1", "ADC1"},
+ {"DEC1 MUX", "ADC2", "ADC2"},
+ {"DEC1 MUX", "ADC3", "ADC3"},
+ {"DEC1 MUX", NULL, "CDC_CONN"},
+ {"DEC2 MUX", "DMIC2", "DMIC2"},
+ {"DEC2 MUX", "DMIC3", "DMIC3"},
+ {"DEC2 MUX", "ADC1", "ADC1"},
+ {"DEC2 MUX", "ADC2", "ADC2"},
+ {"DEC2 MUX", "ADC3", "ADC3"},
+ {"DEC2 MUX", NULL, "CDC_CONN"},
+ {"DEC3 MUX", "DMIC3", "DMIC3"},
+ {"DEC3 MUX", "ADC1", "ADC1"},
+ {"DEC3 MUX", "ADC2", "ADC2"},
+ {"DEC3 MUX", "ADC3", "ADC3"},
+ {"DEC3 MUX", "DMIC2", "DMIC2"},
+ {"DEC3 MUX", "DMIC4", "DMIC4"},
+ {"DEC3 MUX", NULL, "CDC_CONN"},
+ {"DEC4 MUX", "DMIC4", "DMIC4"},
+ {"DEC4 MUX", "ADC1", "ADC1"},
+ {"DEC4 MUX", "ADC2", "ADC2"},
+ {"DEC4 MUX", "ADC3", "ADC3"},
+ {"DEC4 MUX", "DMIC3", "DMIC3"},
+ {"DEC4 MUX", "DMIC2", "DMIC2"},
+ {"DEC4 MUX", "DMIC1", "DMIC1"},
+ {"DEC4 MUX", NULL, "CDC_CONN"},
+
+ /* ADC Connections */
+ {"ADC1", NULL, "AMIC1"},
+ {"ADC2", NULL, "AMIC2"},
+ {"ADC3", NULL, "AMIC3"},
+
+ /* IIR */
+ {"IIR1", NULL, "IIR1 INP1 MUX"},
+ {"IIR1 INP1 MUX", "DEC1", "DEC1 MUX"},
+ {"MIC BIAS1 Internal1", NULL, "LDO_H"},
+ {"MIC BIAS1 External", NULL, "LDO_H"},
+ {"MIC BIAS2 Internal1", NULL, "LDO_H"},
+ {"MIC BIAS2 External", NULL, "LDO_H"},
+};
+
+static int sitar_readable(struct snd_soc_codec *ssc, unsigned int reg)
+{
+ return sitar_reg_readable[reg];
+}
+
+static int sitar_volatile(struct snd_soc_codec *ssc, unsigned int reg)
+{
+ /* Registers lower than 0x100 are top level registers which can be
+ * written by the Sitar core driver.
+ */
+
+ if ((reg >= SITAR_A_CDC_MBHC_EN_CTL) || (reg < 0x100))
+ return 1;
+
+ /* IIR Coeff registers are not cacheable */
+ if ((reg >= SITAR_A_CDC_IIR1_COEF_B1_CTL) &&
+ (reg <= SITAR_A_CDC_IIR1_COEF_B5_CTL))
+ return 1;
+
+ return 0;
+}
+
+#define SITAR_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+static int sitar_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+
+ BUG_ON(reg > SITAR_MAX_REGISTER);
+
+ if (!sitar_volatile(codec, reg)) {
+ ret = snd_soc_cache_write(codec, reg, value);
+ if (ret != 0)
+ dev_err(codec->dev, "Cache write to %x failed: %d\n",
+ reg, ret);
+ }
+
+ return wcd9xxx_reg_write(codec->control_data, reg, value);
+}
+static unsigned int sitar_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ unsigned int val;
+ int ret;
+
+ BUG_ON(reg > SITAR_MAX_REGISTER);
+
+ if (!sitar_volatile(codec, reg) && sitar_readable(codec, reg) &&
+ reg < codec->driver->reg_cache_size) {
+ ret = snd_soc_cache_read(codec, reg, &val);
+ if (ret >= 0) {
+ return val;
+ } else
+ dev_err(codec->dev, "Cache read from %x failed: %d\n",
+ reg, ret);
+ }
+
+ val = wcd9xxx_reg_read(codec->control_data, reg);
+ return val;
+}
+
+static void sitar_codec_enable_audio_mode_bandgap(struct snd_soc_codec *codec)
+{
+ struct wcd9xxx *sitar_core = dev_get_drvdata(codec->dev->parent);
+
+ if (SITAR_IS_1P0(sitar_core->version))
+ snd_soc_update_bits(codec, SITAR_A_LDO_H_MODE_1, 0x80, 0x80);
+
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CURR_CTL_2, 0x0C, 0x08);
+ usleep_range(1000, 1000);
+ snd_soc_write(codec, SITAR_A_BIAS_REF_CTL, 0x1C);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x80);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x04,
+ 0x04);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x01,
+ 0x01);
+ usleep_range(1000, 1000);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x00);
+}
+
+static void sitar_codec_enable_bandgap(struct snd_soc_codec *codec,
+ enum sitar_bandgap_type choice)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ struct wcd9xxx *sitar_core = dev_get_drvdata(codec->dev->parent);
+
+ /* TODO lock resources accessed by audio streams and threaded
+ * interrupt handlers
+ */
+
+ pr_debug("%s, choice is %d, current is %d\n", __func__, choice,
+ sitar->bandgap_type);
+
+ if (sitar->bandgap_type == choice)
+ return;
+
+ if ((sitar->bandgap_type == SITAR_BANDGAP_OFF) &&
+ (choice == SITAR_BANDGAP_AUDIO_MODE)) {
+ sitar_codec_enable_audio_mode_bandgap(codec);
+ } else if (choice == SITAR_BANDGAP_MBHC_MODE) {
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CURR_CTL_2, 0x0C, 0x08);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x2,
+ 0x2);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x80);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x4,
+ 0x4);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x01,
+ 0x1);
+ usleep_range(1000, 1000);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x00);
+ } else if ((sitar->bandgap_type == SITAR_BANDGAP_MBHC_MODE) &&
+ (choice == SITAR_BANDGAP_AUDIO_MODE)) {
+ snd_soc_write(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x50);
+ usleep_range(100, 100);
+ sitar_codec_enable_audio_mode_bandgap(codec);
+ } else if (choice == SITAR_BANDGAP_OFF) {
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CURR_CTL_2, 0x0C, 0x00);
+ snd_soc_write(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x50);
+ if (SITAR_IS_1P0(sitar_core->version))
+ snd_soc_update_bits(codec, SITAR_A_LDO_H_MODE_1,
+ 0xFF, 0x65);
+ usleep_range(1000, 1000);
+ } else {
+ pr_err("%s: Error, Invalid bandgap settings\n", __func__);
+ }
+ sitar->bandgap_type = choice;
+}
+
+static int sitar_codec_enable_config_mode(struct snd_soc_codec *codec,
+ int enable)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ if (enable) {
+ snd_soc_update_bits(codec, SITAR_A_RC_OSC_FREQ, 0x10, 0);
+ snd_soc_write(codec, SITAR_A_BIAS_OSC_BG_CTL, 0x17);
+ usleep_range(5, 5);
+ snd_soc_update_bits(codec, SITAR_A_RC_OSC_FREQ, 0x80,
+ 0x80);
+ snd_soc_update_bits(codec, SITAR_A_RC_OSC_TEST, 0x80,
+ 0x80);
+ usleep_range(10, 10);
+ snd_soc_update_bits(codec, SITAR_A_RC_OSC_TEST, 0x80, 0);
+ usleep_range(20, 20);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1, 0x08, 0x08);
+ } else {
+ snd_soc_update_bits(codec, SITAR_A_BIAS_OSC_BG_CTL, 0x1,
+ 0);
+ snd_soc_update_bits(codec, SITAR_A_RC_OSC_FREQ, 0x80, 0);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1, 0x08, 0x00);
+ }
+ sitar->config_mode_active = enable ? true : false;
+
+ return 0;
+}
+
+static int sitar_codec_enable_clock_block(struct snd_soc_codec *codec,
+ int config_mode)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s\n", __func__);
+
+ if (config_mode) {
+ sitar_codec_enable_config_mode(codec, 1);
+ snd_soc_write(codec, SITAR_A_CLK_BUFF_EN2, 0x00);
+ snd_soc_write(codec, SITAR_A_CLK_BUFF_EN2, 0x02);
+ snd_soc_write(codec, SITAR_A_CLK_BUFF_EN1, 0x0D);
+ usleep_range(1000, 1000);
+ } else
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1, 0x08, 0x00);
+
+ if (!config_mode && sitar->mbhc_polling_active) {
+ snd_soc_write(codec, SITAR_A_CLK_BUFF_EN2, 0x02);
+ sitar_codec_enable_config_mode(codec, 0);
+
+ }
+
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1, 0x05, 0x05);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN2, 0x02, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN2, 0x04, 0x04);
+ usleep_range(50, 50);
+ sitar->clock_active = true;
+ return 0;
+}
+static void sitar_codec_disable_clock_block(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ pr_debug("%s\n", __func__);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN2, 0x04, 0x00);
+ ndelay(160);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN2, 0x02, 0x02);
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1, 0x05, 0x00);
+ sitar->clock_active = false;
+}
+
+static int sitar_codec_mclk_index(const struct sitar_priv *sitar)
+{
+ if (sitar->mbhc_cfg.mclk_rate == SITAR_MCLK_RATE_12288KHZ)
+ return 0;
+ else if (sitar->mbhc_cfg.mclk_rate == SITAR_MCLK_RATE_9600KHZ)
+ return 1;
+ else {
+ BUG_ON(1);
+ return -EINVAL;
+ }
+}
+
+static void sitar_codec_calibrate_hs_polling(struct snd_soc_codec *codec)
+{
+ u8 *n_ready, *n_cic;
+ struct sitar_mbhc_btn_detect_cfg *btn_det;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ btn_det = SITAR_MBHC_CAL_BTN_DET_PTR(sitar->mbhc_cfg.calibration);
+
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B1_CTL,
+ sitar->mbhc_data.v_ins_hu & 0xFF);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B2_CTL,
+ (sitar->mbhc_data.v_ins_hu >> 8) & 0xFF);
+
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B3_CTL,
+ sitar->mbhc_data.v_b1_hu & 0xFF);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B4_CTL,
+ (sitar->mbhc_data.v_b1_hu >> 8) & 0xFF);
+
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B5_CTL,
+ sitar->mbhc_data.v_b1_h & 0xFF);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B6_CTL,
+ (sitar->mbhc_data.v_b1_h >> 8) & 0xFF);
+
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B9_CTL,
+ sitar->mbhc_data.v_brh & 0xFF);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B10_CTL,
+ (sitar->mbhc_data.v_brh >> 8) & 0xFF);
+
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B11_CTL,
+ sitar->mbhc_data.v_brl & 0xFF);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_VOLT_B12_CTL,
+ (sitar->mbhc_data.v_brl >> 8) & 0xFF);
+
+ n_ready = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_N_READY);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_TIMER_B1_CTL,
+ n_ready[sitar_codec_mclk_index(sitar)]);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_TIMER_B2_CTL,
+ sitar->mbhc_data.npoll);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_TIMER_B3_CTL,
+ sitar->mbhc_data.nbounce_wait);
+ n_cic = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_N_CIC);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_TIMER_B6_CTL,
+ n_cic[sitar_codec_mclk_index(sitar)]);
+}
+
+static int sitar_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct wcd9xxx *wcd9xxx = dev_get_drvdata(dai->codec->dev->parent);
+ if ((wcd9xxx != NULL) && (wcd9xxx->dev != NULL) &&
+ (wcd9xxx->dev->parent != NULL))
+ pm_runtime_get_sync(wcd9xxx->dev->parent);
+ pr_debug("%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+
+ return 0;
+}
+
+static void sitar_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct wcd9xxx *wcd9xxx = dev_get_drvdata(dai->codec->dev->parent);
+ if ((wcd9xxx != NULL) && (wcd9xxx->dev != NULL) &&
+ (wcd9xxx->dev->parent != NULL)) {
+ pm_runtime_mark_last_busy(wcd9xxx->dev->parent);
+ pm_runtime_put(wcd9xxx->dev->parent);
+ }
+ pr_debug("%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+}
+
+int sitar_mclk_enable(struct snd_soc_codec *codec, int mclk_enable, bool dapm)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s() mclk_enable = %u\n", __func__, mclk_enable);
+
+ if (dapm)
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ if (mclk_enable) {
+ sitar->mclk_enabled = true;
+
+ if (sitar->mbhc_polling_active && (sitar->mclk_enabled)) {
+ sitar_codec_pause_hs_polling(codec);
+ sitar_codec_enable_bandgap(codec,
+ SITAR_BANDGAP_AUDIO_MODE);
+ sitar_codec_enable_clock_block(codec, 0);
+ sitar_codec_calibrate_hs_polling(codec);
+ sitar_codec_start_hs_polling(codec);
+ } else {
+ sitar_codec_enable_bandgap(codec,
+ SITAR_BANDGAP_AUDIO_MODE);
+ sitar_codec_enable_clock_block(codec, 0);
+ }
+ } else {
+
+ if (!sitar->mclk_enabled) {
+ if (dapm)
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ pr_err("Error, MCLK already diabled\n");
+ return -EINVAL;
+ }
+ sitar->mclk_enabled = false;
+
+ if (sitar->mbhc_polling_active) {
+ if (!sitar->mclk_enabled) {
+ sitar_codec_pause_hs_polling(codec);
+ sitar_codec_enable_bandgap(codec,
+ SITAR_BANDGAP_MBHC_MODE);
+ sitar_enable_rx_bias(codec, 1);
+ sitar_codec_enable_clock_block(codec, 1);
+ sitar_codec_calibrate_hs_polling(codec);
+ sitar_codec_start_hs_polling(codec);
+ }
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1,
+ 0x05, 0x01);
+ } else {
+ sitar_codec_disable_clock_block(codec);
+ sitar_codec_enable_bandgap(codec,
+ SITAR_BANDGAP_OFF);
+ }
+ }
+ if (dapm)
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ return 0;
+}
+
+static int sitar_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ pr_debug("%s\n", __func__);
+ return 0;
+}
+
+static int sitar_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ u8 val = 0;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(dai->codec);
+
+ pr_debug("%s\n", __func__);
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ /* CPU is master */
+ if (sitar->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ if (dai->id == AIF1_CAP)
+ snd_soc_update_bits(dai->codec,
+ SITAR_A_CDC_CLK_TX_I2S_CTL,
+ SITAR_I2S_MASTER_MODE_MASK, 0);
+ else if (dai->id == AIF1_PB)
+ snd_soc_update_bits(dai->codec,
+ SITAR_A_CDC_CLK_RX_I2S_CTL,
+ SITAR_I2S_MASTER_MODE_MASK, 0);
+ }
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* CPU is slave */
+ if (sitar->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ val = SITAR_I2S_MASTER_MODE_MASK;
+ if (dai->id == AIF1_CAP)
+ snd_soc_update_bits(dai->codec,
+ SITAR_A_CDC_CLK_TX_I2S_CTL, val, val);
+ else if (dai->id == AIF1_PB)
+ snd_soc_update_bits(dai->codec,
+ SITAR_A_CDC_CLK_RX_I2S_CTL, val, val);
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+static int sitar_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(dai->codec);
+ u32 i = 0;
+ if (!tx_slot && !rx_slot) {
+ pr_err("%s: Invalid\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s: DAI-ID %x %d %d\n", __func__, dai->id, tx_num, rx_num);
+
+ if (dai->id == AIF1_PB) {
+ for (i = 0; i < rx_num; i++) {
+ sitar->dai[dai->id - 1].ch_num[i] = rx_slot[i];
+ sitar->dai[dai->id - 1].ch_act = 0;
+ sitar->dai[dai->id - 1].ch_tot = rx_num;
+ }
+ } else if (dai->id == AIF1_CAP) {
+ for (i = 0; i < tx_num; i++) {
+ sitar->dai[dai->id - 1].ch_num[i] = tx_slot[i];
+ sitar->dai[dai->id - 1].ch_act = 0;
+ sitar->dai[dai->id - 1].ch_tot = tx_num;
+ }
+ }
+ return 0;
+}
+
+static int sitar_get_channel_map(struct snd_soc_dai *dai,
+ unsigned int *tx_num, unsigned int *tx_slot,
+ unsigned int *rx_num, unsigned int *rx_slot)
+
+{
+ struct wcd9xxx *sitar = dev_get_drvdata(dai->codec->control_data);
+
+ u32 cnt = 0;
+ u32 tx_ch[SLIM_MAX_TX_PORTS];
+ u32 rx_ch[SLIM_MAX_RX_PORTS];
+
+ if (!rx_slot && !tx_slot) {
+ pr_err("%s: Invalid\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s: DAI-ID %x\n", __func__, dai->id);
+ /* for virtual port, codec driver needs to do
+ * housekeeping, for now should be ok
+ */
+ wcd9xxx_get_channel(sitar, rx_ch, tx_ch);
+ if (dai->id == AIF1_PB) {
+ *rx_num = sitar_dai[dai->id - 1].playback.channels_max;
+ while (cnt < *rx_num) {
+ rx_slot[cnt] = rx_ch[cnt];
+ cnt++;
+ }
+ } else if (dai->id == AIF1_CAP) {
+ *tx_num = sitar_dai[dai->id - 1].capture.channels_max;
+ tx_slot[0] = tx_ch[cnt];
+ tx_slot[1] = tx_ch[4 + cnt];
+ tx_slot[2] = tx_ch[2 + cnt];
+ tx_slot[3] = tx_ch[3 + cnt];
+ }
+ return 0;
+}
+
+static int sitar_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(dai->codec);
+ u8 path, shift;
+ u16 tx_fs_reg, rx_fs_reg;
+ u8 tx_fs_rate, rx_fs_rate, rx_state, tx_state;
+
+ pr_debug("%s: DAI-ID %x\n", __func__, dai->id);
+
+ switch (params_rate(params)) {
+ case 8000:
+ tx_fs_rate = 0x00;
+ rx_fs_rate = 0x00;
+ break;
+ case 16000:
+ tx_fs_rate = 0x01;
+ rx_fs_rate = 0x20;
+ break;
+ case 32000:
+ tx_fs_rate = 0x02;
+ rx_fs_rate = 0x40;
+ break;
+ case 48000:
+ tx_fs_rate = 0x03;
+ rx_fs_rate = 0x60;
+ break;
+ default:
+ pr_err("%s: Invalid sampling rate %d\n", __func__,
+ params_rate(params));
+ return -EINVAL;
+ }
+
+
+ /**
+ * If current dai is a tx dai, set sample rate to
+ * all the txfe paths that are currently not active
+ */
+ if (dai->id == AIF1_CAP) {
+
+ tx_state = snd_soc_read(codec,
+ SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL);
+
+ for (path = 1, shift = 0;
+ path <= NUM_DECIMATORS; path++, shift++) {
+
+ if (!(tx_state & (1 << shift))) {
+ tx_fs_reg = SITAR_A_CDC_TX1_CLK_FS_CTL
+ + (BITS_PER_REG*(path-1));
+ snd_soc_update_bits(codec, tx_fs_reg,
+ 0x03, tx_fs_rate);
+ }
+ }
+ if (sitar->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ snd_soc_update_bits(codec,
+ SITAR_A_CDC_CLK_TX_I2S_CTL,
+ 0x20, 0x20);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ snd_soc_update_bits(codec,
+ SITAR_A_CDC_CLK_TX_I2S_CTL,
+ 0x20, 0x00);
+ break;
+ default:
+ pr_err("invalid format\n");
+ break;
+ }
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_TX_I2S_CTL,
+ 0x03, tx_fs_rate);
+ }
+ } else {
+ sitar->dai[dai->id - 1].rate = params_rate(params);
+ }
+
+ /**
+ * TODO: Need to handle case where same RX chain takes 2 or more inputs
+ * with varying sample rates
+ */
+
+ /**
+ * If current dai is a rx dai, set sample rate to
+ * all the rx paths that are currently not active
+ */
+ if (dai->id == AIF1_PB) {
+
+ rx_state = snd_soc_read(codec,
+ SITAR_A_CDC_CLK_RX_B1_CTL);
+
+ for (path = 1, shift = 0;
+ path <= NUM_INTERPOLATORS; path++, shift++) {
+
+ if (!(rx_state & (1 << shift))) {
+ rx_fs_reg = SITAR_A_CDC_RX1_B5_CTL
+ + (BITS_PER_REG*(path-1));
+ snd_soc_update_bits(codec, rx_fs_reg,
+ 0xE0, rx_fs_rate);
+ }
+ }
+ if (sitar->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ snd_soc_update_bits(codec,
+ SITAR_A_CDC_CLK_RX_I2S_CTL,
+ 0x20, 0x20);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ snd_soc_update_bits(codec,
+ SITAR_A_CDC_CLK_RX_I2S_CTL,
+ 0x20, 0x00);
+ break;
+ default:
+ pr_err("invalid format\n");
+ break;
+ }
+ snd_soc_update_bits(codec, SITAR_A_CDC_CLK_RX_I2S_CTL,
+ 0x03, (rx_fs_rate >> 0x05));
+ }
+ } else {
+ sitar->dai[dai->id - 1].rate = params_rate(params);
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops sitar_dai_ops = {
+ .startup = sitar_startup,
+ .shutdown = sitar_shutdown,
+ .hw_params = sitar_hw_params,
+ .set_sysclk = sitar_set_dai_sysclk,
+ .set_fmt = sitar_set_dai_fmt,
+ .set_channel_map = sitar_set_channel_map,
+ .get_channel_map = sitar_get_channel_map,
+};
+
+static struct snd_soc_dai_driver sitar_dai[] = {
+ {
+ .name = "sitar_rx1",
+ .id = AIF1_PB,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .rates = WCD9304_RATES,
+ .formats = SITAR_FORMATS,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &sitar_dai_ops,
+ },
+ {
+ .name = "sitar_tx1",
+ .id = AIF1_CAP,
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .rates = WCD9304_RATES,
+ .formats = SITAR_FORMATS,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &sitar_dai_ops,
+ },
+};
+
+static int sitar_codec_enable_slimrx(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct wcd9xxx *sitar;
+ struct snd_soc_codec *codec = w->codec;
+ struct sitar_priv *sitar_p = snd_soc_codec_get_drvdata(codec);
+ u32 j = 0;
+ codec->control_data = dev_get_drvdata(codec->dev->parent);
+ sitar = codec->control_data;
+ /* Execute the callback only if interface type is slimbus */
+ if (sitar_p->intf_type != WCD9XXX_INTERFACE_TYPE_SLIMBUS)
+ return 0;
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ for (j = 0; j < ARRAY_SIZE(sitar_dai); j++) {
+ if (sitar_dai[j].id == AIF1_CAP)
+ continue;
+ if (!strncmp(w->sname,
+ sitar_dai[j].playback.stream_name, 13)) {
+ ++sitar_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (sitar_p->dai[j].ch_act == sitar_p->dai[j].ch_tot)
+ wcd9xxx_cfg_slim_sch_rx(sitar,
+ sitar_p->dai[j].ch_num,
+ sitar_p->dai[j].ch_tot,
+ sitar_p->dai[j].rate);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ for (j = 0; j < ARRAY_SIZE(sitar_dai); j++) {
+ if (sitar_dai[j].id == AIF1_CAP)
+ continue;
+ if (!strncmp(w->sname,
+ sitar_dai[j].playback.stream_name, 13)) {
+ --sitar_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (!sitar_p->dai[j].ch_act) {
+ wcd9xxx_close_slim_sch_rx(sitar,
+ sitar_p->dai[j].ch_num,
+ sitar_p->dai[j].ch_tot);
+ /* Wait for remove channel to complete
+ * before derouting Rx path
+ */
+ usleep_range(15000, 15000);
+ sitar_p->dai[j].rate = 0;
+ memset(sitar_p->dai[j].ch_num, 0, (sizeof(u32)*
+ sitar_p->dai[j].ch_tot));
+ sitar_p->dai[j].ch_tot = 0;
+ }
+ }
+ return 0;
+}
+
+static int sitar_codec_enable_slimtx(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct wcd9xxx *sitar;
+ struct snd_soc_codec *codec = w->codec;
+ struct sitar_priv *sitar_p = snd_soc_codec_get_drvdata(codec);
+ /* index to the DAI ID, for now hardcoding */
+ u32 j = 0;
+
+ codec->control_data = dev_get_drvdata(codec->dev->parent);
+ sitar = codec->control_data;
+
+ /* Execute the callback only if interface type is slimbus */
+ if (sitar_p->intf_type != WCD9XXX_INTERFACE_TYPE_SLIMBUS)
+ return 0;
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ for (j = 0; j < ARRAY_SIZE(sitar_dai); j++) {
+ if (sitar_dai[j].id == AIF1_PB)
+ continue;
+ if (!strncmp(w->sname,
+ sitar_dai[j].capture.stream_name, 13)) {
+ ++sitar_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (sitar_p->dai[j].ch_act == sitar_p->dai[j].ch_tot)
+ wcd9xxx_cfg_slim_sch_tx(sitar,
+ sitar_p->dai[j].ch_num,
+ sitar_p->dai[j].ch_tot,
+ sitar_p->dai[j].rate);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ for (j = 0; j < ARRAY_SIZE(sitar_dai); j++) {
+ if (sitar_dai[j].id == AIF1_PB)
+ continue;
+ if (!strncmp(w->sname,
+ sitar_dai[j].capture.stream_name, 13)) {
+ --sitar_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (!sitar_p->dai[j].ch_act) {
+ wcd9xxx_close_slim_sch_tx(sitar,
+ sitar_p->dai[j].ch_num,
+ sitar_p->dai[j].ch_tot);
+ sitar_p->dai[j].rate = 0;
+ memset(sitar_p->dai[j].ch_num, 0, (sizeof(u32)*
+ sitar_p->dai[j].ch_tot));
+ sitar_p->dai[j].ch_tot = 0;
+ }
+ }
+ return 0;
+}
+
+
+static short sitar_codec_read_sta_result(struct snd_soc_codec *codec)
+{
+ u8 bias_msb, bias_lsb;
+ short bias_value;
+
+ bias_msb = snd_soc_read(codec, SITAR_A_CDC_MBHC_B3_STATUS);
+ bias_lsb = snd_soc_read(codec, SITAR_A_CDC_MBHC_B2_STATUS);
+ bias_value = (bias_msb << 8) | bias_lsb;
+ return bias_value;
+}
+
+static short sitar_codec_read_dce_result(struct snd_soc_codec *codec)
+{
+ u8 bias_msb, bias_lsb;
+ short bias_value;
+
+ bias_msb = snd_soc_read(codec, SITAR_A_CDC_MBHC_B5_STATUS);
+ bias_lsb = snd_soc_read(codec, SITAR_A_CDC_MBHC_B4_STATUS);
+ bias_value = (bias_msb << 8) | bias_lsb;
+ return bias_value;
+}
+
+static void sitar_turn_onoff_rel_detection(struct snd_soc_codec *codec,
+ bool on)
+{
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x02, on << 1);
+}
+
+static short __sitar_codec_sta_dce(struct snd_soc_codec *codec, int dce,
+ bool override_bypass, bool noreldetection)
+{
+ short bias_value;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ wcd9xxx_disable_irq(codec->control_data, SITAR_IRQ_MBHC_POTENTIAL);
+ if (noreldetection)
+ sitar_turn_onoff_rel_detection(codec, false);
+
+ /* Turn on the override */
+ if (!override_bypass)
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x4, 0x4);
+ if (dce) {
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x4);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x0);
+ usleep_range(sitar->mbhc_data.t_sta_dce,
+ sitar->mbhc_data.t_sta_dce);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x4);
+ usleep_range(sitar->mbhc_data.t_dce,
+ sitar->mbhc_data.t_dce);
+ bias_value = sitar_codec_read_dce_result(codec);
+ } else {
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x2);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x0);
+ usleep_range(sitar->mbhc_data.t_sta_dce,
+ sitar->mbhc_data.t_sta_dce);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x2);
+ usleep_range(sitar->mbhc_data.t_sta,
+ sitar->mbhc_data.t_sta);
+ bias_value = sitar_codec_read_sta_result(codec);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x0);
+ }
+ /* Turn off the override after measuring mic voltage */
+ if (!override_bypass)
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x04, 0x00);
+
+ if (noreldetection)
+ sitar_turn_onoff_rel_detection(codec, true);
+ wcd9xxx_enable_irq(codec->control_data, SITAR_IRQ_MBHC_POTENTIAL);
+
+ return bias_value;
+}
+
+static short sitar_codec_sta_dce(struct snd_soc_codec *codec, int dce,
+ bool norel)
+{
+ return __sitar_codec_sta_dce(codec, dce, false, norel);
+}
+
+static void sitar_codec_shutdown_hs_removal_detect(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ const struct sitar_mbhc_general_cfg *generic =
+ SITAR_MBHC_CAL_GENERAL_PTR(sitar->mbhc_cfg.calibration);
+
+ if (!sitar->mclk_enabled && !sitar->mbhc_polling_active)
+ sitar_codec_enable_config_mode(codec, 1);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x2, 0x2);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x6, 0x0);
+
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.mbhc_reg, 0x80, 0x00);
+
+ usleep_range(generic->t_shutdown_plug_rem,
+ generic->t_shutdown_plug_rem);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0xA, 0x8);
+ if (!sitar->mclk_enabled && !sitar->mbhc_polling_active)
+ sitar_codec_enable_config_mode(codec, 0);
+
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x00);
+}
+
+static void sitar_codec_cleanup_hs_polling(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ sitar_codec_shutdown_hs_removal_detect(codec);
+
+ if (!sitar->mclk_enabled) {
+ sitar_codec_disable_clock_block(codec);
+ sitar_codec_enable_bandgap(codec, SITAR_BANDGAP_OFF);
+ }
+
+ sitar->mbhc_polling_active = false;
+ sitar->mbhc_state = MBHC_STATE_NONE;
+}
+
+/* called only from interrupt which is under codec_resource_lock acquisition */
+static short sitar_codec_setup_hs_polling(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ short bias_value;
+ u8 cfilt_mode;
+
+ if (!sitar->mbhc_cfg.calibration) {
+ pr_err("Error, no sitar calibration\n");
+ return -ENODEV;
+ }
+
+ if (!sitar->mclk_enabled) {
+ sitar_codec_enable_bandgap(codec, SITAR_BANDGAP_MBHC_MODE);
+ sitar_enable_rx_bias(codec, 1);
+ sitar_codec_enable_clock_block(codec, 1);
+ }
+
+ snd_soc_update_bits(codec, SITAR_A_CLK_BUFF_EN1, 0x05, 0x01);
+
+ /* Make sure CFILT is in fast mode, save current mode */
+ cfilt_mode = snd_soc_read(codec, sitar->mbhc_bias_regs.cfilt_ctl);
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.cfilt_ctl, 0x70, 0x00);
+
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.ctl_reg, 0x1F, 0x16);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x2, 0x2);
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x84);
+
+ snd_soc_update_bits(codec, SITAR_A_TX_4_MBHC_EN, 0x80, 0x80);
+ snd_soc_update_bits(codec, SITAR_A_TX_4_MBHC_EN, 0x1F, 0x1C);
+ snd_soc_update_bits(codec, SITAR_A_TX_4_MBHC_TEST_CTL, 0x40, 0x40);
+
+ snd_soc_update_bits(codec, SITAR_A_TX_4_MBHC_EN, 0x80, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x00);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x2, 0x2);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+
+ sitar_codec_calibrate_hs_polling(codec);
+
+ /* don't flip override */
+ bias_value = __sitar_codec_sta_dce(codec, 1, true, true);
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.cfilt_ctl, 0x40,
+ cfilt_mode);
+ snd_soc_update_bits(codec, SITAR_A_MBHC_HPH, 0x13, 0x00);
+
+ return bias_value;
+}
+
+static int sitar_cancel_btn_work(struct sitar_priv *sitar)
+{
+ int r = 0;
+ struct wcd9xxx *core = dev_get_drvdata(sitar->codec->dev->parent);
+
+ if (cancel_delayed_work_sync(&sitar->mbhc_btn_dwork)) {
+ /* if scheduled mbhc_btn_dwork is canceled from here,
+ * we have to unlock from here instead btn_work */
+ wcd9xxx_unlock_sleep(core);
+ r = 1;
+ }
+ return r;
+}
+
+
+static u16 sitar_codec_v_sta_dce(struct snd_soc_codec *codec, bool dce,
+ s16 vin_mv)
+{
+ short diff, zero;
+ struct sitar_priv *sitar;
+ u32 mb_mv, in;
+
+ sitar = snd_soc_codec_get_drvdata(codec);
+ mb_mv = sitar->mbhc_data.micb_mv;
+
+ if (mb_mv == 0) {
+ pr_err("%s: Mic Bias voltage is set to zero\n", __func__);
+ return -EINVAL;
+ }
+
+ if (dce) {
+ diff = sitar->mbhc_data.dce_mb - sitar->mbhc_data.dce_z;
+ zero = sitar->mbhc_data.dce_z;
+ } else {
+ diff = sitar->mbhc_data.sta_mb - sitar->mbhc_data.sta_z;
+ zero = sitar->mbhc_data.sta_z;
+ }
+ in = (u32) diff * vin_mv;
+
+ return (u16) (in / mb_mv) + zero;
+}
+
+static s32 sitar_codec_sta_dce_v(struct snd_soc_codec *codec, s8 dce,
+ u16 bias_value)
+{
+ struct sitar_priv *sitar;
+ s16 value, z, mb;
+ s32 mv;
+
+ sitar = snd_soc_codec_get_drvdata(codec);
+ value = bias_value;
+
+ if (dce) {
+ z = (sitar->mbhc_data.dce_z);
+ mb = (sitar->mbhc_data.dce_mb);
+ mv = (value - z) * (s32)sitar->mbhc_data.micb_mv / (mb - z);
+ } else {
+ z = (sitar->mbhc_data.sta_z);
+ mb = (sitar->mbhc_data.sta_mb);
+ mv = (value - z) * (s32)sitar->mbhc_data.micb_mv / (mb - z);
+ }
+
+ return mv;
+}
+
+static void btn_lpress_fn(struct work_struct *work)
+{
+ struct delayed_work *delayed_work;
+ struct sitar_priv *sitar;
+ short bias_value;
+ int dce_mv, sta_mv;
+ struct wcd9xxx *core;
+
+ pr_debug("%s:\n", __func__);
+
+ delayed_work = to_delayed_work(work);
+ sitar = container_of(delayed_work, struct sitar_priv, mbhc_btn_dwork);
+ core = dev_get_drvdata(sitar->codec->dev->parent);
+
+ if (sitar) {
+ if (sitar->mbhc_cfg.button_jack) {
+ bias_value = sitar_codec_read_sta_result(sitar->codec);
+ sta_mv = sitar_codec_sta_dce_v(sitar->codec, 0,
+ bias_value);
+ bias_value = sitar_codec_read_dce_result(sitar->codec);
+ dce_mv = sitar_codec_sta_dce_v(sitar->codec, 1,
+ bias_value);
+ pr_debug("%s: Reporting long button press event"
+ " STA: %d, DCE: %d\n", __func__, sta_mv, dce_mv);
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.button_jack,
+ sitar->buttons_pressed,
+ sitar->buttons_pressed);
+ }
+ } else {
+ pr_err("%s: Bad sitar private data\n", __func__);
+ }
+
+ pr_debug("%s: leave\n", __func__);
+ wcd9xxx_unlock_sleep(core);
+}
+
+
+void sitar_mbhc_cal(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar;
+ struct sitar_mbhc_btn_detect_cfg *btn_det;
+ u8 cfilt_mode, bg_mode;
+ u8 ncic, nmeas, navg;
+ u32 mclk_rate;
+ u32 dce_wait, sta_wait;
+ u8 *n_cic;
+ void *calibration;
+
+ sitar = snd_soc_codec_get_drvdata(codec);
+ calibration = sitar->mbhc_cfg.calibration;
+
+ wcd9xxx_disable_irq(codec->control_data, SITAR_IRQ_MBHC_POTENTIAL);
+ sitar_turn_onoff_rel_detection(codec, false);
+
+ /* First compute the DCE / STA wait times
+ * depending on tunable parameters.
+ * The value is computed in microseconds
+ */
+ btn_det = SITAR_MBHC_CAL_BTN_DET_PTR(calibration);
+ n_cic = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_N_CIC);
+ ncic = n_cic[sitar_codec_mclk_index(sitar)];
+ nmeas = SITAR_MBHC_CAL_BTN_DET_PTR(calibration)->n_meas;
+ navg = SITAR_MBHC_CAL_GENERAL_PTR(calibration)->mbhc_navg;
+ mclk_rate = sitar->mbhc_cfg.mclk_rate;
+ dce_wait = (1000 * 512 * 60 * (nmeas + 1)) / (mclk_rate / 1000);
+ sta_wait = (1000 * 128 * (navg + 1)) / (mclk_rate / 1000);
+
+ sitar->mbhc_data.t_dce = DEFAULT_DCE_WAIT;
+ sitar->mbhc_data.t_sta = DEFAULT_STA_WAIT;
+
+ /* LDOH and CFILT are already configured during pdata handling.
+ * Only need to make sure CFILT and bandgap are in Fast mode.
+ * Need to restore defaults once calculation is done.
+ */
+ cfilt_mode = snd_soc_read(codec, sitar->mbhc_bias_regs.cfilt_ctl);
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.cfilt_ctl, 0x40, 0x00);
+ bg_mode = snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x02,
+ 0x02);
+
+ /* Micbias, CFILT, LDOH, MBHC MUX mode settings
+ * to perform ADC calibration
+ */
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.ctl_reg, 0x60,
+ sitar->mbhc_cfg.micbias << 5);
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.ctl_reg, 0x01, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_LDO_H_MODE_1, 0x60, 0x60);
+ snd_soc_write(codec, SITAR_A_TX_4_MBHC_TEST_CTL, 0x78);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x04, 0x04);
+
+ /* DCE measurement for 0 volts */
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x0A);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x04);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x02);
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x81);
+ usleep_range(100, 100);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x04);
+ usleep_range(sitar->mbhc_data.t_dce, sitar->mbhc_data.t_dce);
+ sitar->mbhc_data.dce_z = sitar_codec_read_dce_result(codec);
+
+ /* DCE measurment for MB voltage */
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x0A);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x02);
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x82);
+ usleep_range(100, 100);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x04);
+ usleep_range(sitar->mbhc_data.t_dce, sitar->mbhc_data.t_dce);
+ sitar->mbhc_data.dce_mb = sitar_codec_read_dce_result(codec);
+
+ /* Sta measuremnt for 0 volts */
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x0A);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x02);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_CLK_CTL, 0x02);
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x81);
+ usleep_range(100, 100);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x02);
+ usleep_range(sitar->mbhc_data.t_sta, sitar->mbhc_data.t_sta);
+ sitar->mbhc_data.sta_z = sitar_codec_read_sta_result(codec);
+
+ /* STA Measurement for MB Voltage */
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x82);
+ usleep_range(100, 100);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_EN_CTL, 0x02);
+ usleep_range(sitar->mbhc_data.t_sta, sitar->mbhc_data.t_sta);
+ sitar->mbhc_data.sta_mb = sitar_codec_read_sta_result(codec);
+
+ /* Restore default settings. */
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x04, 0x00);
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.cfilt_ctl, 0x40,
+ cfilt_mode);
+ snd_soc_update_bits(codec, SITAR_A_BIAS_CENTRAL_BG_CTL, 0x02, bg_mode);
+
+ snd_soc_write(codec, SITAR_A_MBHC_SCALING_MUX_1, 0x84);
+ usleep_range(100, 100);
+
+ wcd9xxx_enable_irq(codec->control_data, SITAR_IRQ_MBHC_POTENTIAL);
+ sitar_turn_onoff_rel_detection(codec, true);
+}
+
+void *sitar_mbhc_cal_btn_det_mp(const struct sitar_mbhc_btn_detect_cfg* btn_det,
+ const enum sitar_mbhc_btn_det_mem mem)
+{
+ void *ret = &btn_det->_v_btn_low;
+
+ switch (mem) {
+ case SITAR_BTN_DET_GAIN:
+ ret += sizeof(btn_det->_n_cic);
+ case SITAR_BTN_DET_N_CIC:
+ ret += sizeof(btn_det->_n_ready);
+ case SITAR_BTN_DET_N_READY:
+ ret += sizeof(btn_det->_v_btn_high[0]) * btn_det->num_btn;
+ case SITAR_BTN_DET_V_BTN_HIGH:
+ ret += sizeof(btn_det->_v_btn_low[0]) * btn_det->num_btn;
+ case SITAR_BTN_DET_V_BTN_LOW:
+ /* do nothing */
+ break;
+ default:
+ ret = NULL;
+ }
+
+ return ret;
+}
+
+static void sitar_mbhc_calc_thres(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar;
+ s16 btn_mv = 0, btn_delta_mv;
+ struct sitar_mbhc_btn_detect_cfg *btn_det;
+ struct sitar_mbhc_plug_type_cfg *plug_type;
+ u16 *btn_high;
+ u8 *n_ready;
+ int i;
+
+ sitar = snd_soc_codec_get_drvdata(codec);
+ btn_det = SITAR_MBHC_CAL_BTN_DET_PTR(sitar->mbhc_cfg.calibration);
+ plug_type = SITAR_MBHC_CAL_PLUG_TYPE_PTR(sitar->mbhc_cfg.calibration);
+
+ n_ready = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_N_READY);
+ if (sitar->mbhc_cfg.mclk_rate == SITAR_MCLK_RATE_12288KHZ) {
+ sitar->mbhc_data.npoll = 9;
+ sitar->mbhc_data.nbounce_wait = 30;
+ } else if (sitar->mbhc_cfg.mclk_rate == SITAR_MCLK_RATE_9600KHZ) {
+ sitar->mbhc_data.npoll = 7;
+ sitar->mbhc_data.nbounce_wait = 23;
+ }
+
+ sitar->mbhc_data.t_sta_dce = ((1000 * 256) /
+ (sitar->mbhc_cfg.mclk_rate / 1000) *
+ n_ready[sitar_codec_mclk_index(sitar)]) +
+ 10;
+ sitar->mbhc_data.v_ins_hu =
+ sitar_codec_v_sta_dce(codec, STA, plug_type->v_hs_max);
+ sitar->mbhc_data.v_ins_h =
+ sitar_codec_v_sta_dce(codec, DCE, plug_type->v_hs_max);
+
+ btn_high = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_V_BTN_HIGH);
+ for (i = 0; i < btn_det->num_btn; i++)
+ btn_mv = btn_high[i] > btn_mv ? btn_high[i] : btn_mv;
+
+ sitar->mbhc_data.v_b1_h = sitar_codec_v_sta_dce(codec, DCE, btn_mv);
+ btn_delta_mv = btn_mv + btn_det->v_btn_press_delta_sta;
+
+ sitar->mbhc_data.v_b1_hu =
+ sitar_codec_v_sta_dce(codec, STA, btn_delta_mv);
+
+ btn_delta_mv = btn_mv + btn_det->v_btn_press_delta_cic;
+
+ sitar->mbhc_data.v_b1_huc =
+ sitar_codec_v_sta_dce(codec, DCE, btn_delta_mv);
+
+ sitar->mbhc_data.v_brh = sitar->mbhc_data.v_b1_h;
+ sitar->mbhc_data.v_brl = SITAR_MBHC_BUTTON_MIN;
+
+ sitar->mbhc_data.v_no_mic =
+ sitar_codec_v_sta_dce(codec, STA, plug_type->v_no_mic);
+}
+
+void sitar_mbhc_init(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar;
+ struct sitar_mbhc_general_cfg *generic;
+ struct sitar_mbhc_btn_detect_cfg *btn_det;
+ int n;
+ u8 *n_cic, *gain;
+
+ pr_err("%s(): ENTER\n", __func__);
+ sitar = snd_soc_codec_get_drvdata(codec);
+ generic = SITAR_MBHC_CAL_GENERAL_PTR(sitar->mbhc_cfg.calibration);
+ btn_det = SITAR_MBHC_CAL_BTN_DET_PTR(sitar->mbhc_cfg.calibration);
+
+ for (n = 0; n < 8; n++) {
+ if (n != 7) {
+ snd_soc_update_bits(codec,
+ SITAR_A_CDC_MBHC_FIR_B1_CFG,
+ 0x07, n);
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_FIR_B2_CFG,
+ btn_det->c[n]);
+ }
+ }
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B2_CTL, 0x07,
+ btn_det->nc);
+
+ n_cic = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_N_CIC);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_TIMER_B6_CTL, 0xFF,
+ n_cic[sitar_codec_mclk_index(sitar)]);
+
+ gain = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_GAIN);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B2_CTL, 0x78,
+ gain[sitar_codec_mclk_index(sitar)] << 3);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_TIMER_B4_CTL, 0x70,
+ generic->mbhc_nsa << 4);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_TIMER_B4_CTL, 0x0F,
+ btn_det->n_meas);
+
+ snd_soc_write(codec, SITAR_A_CDC_MBHC_TIMER_B5_CTL, generic->mbhc_navg);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x80, 0x80);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x78,
+ btn_det->mbhc_nsc << 3);
+
+ snd_soc_update_bits(codec, SITAR_A_MICB_1_MBHC, 0x03,
+ sitar->mbhc_cfg.micbias);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x02, 0x02);
+
+ snd_soc_update_bits(codec, SITAR_A_MBHC_SCALING_MUX_2, 0xF0, 0xF0);
+
+}
+
+static bool sitar_mbhc_fw_validate(const struct firmware *fw)
+{
+ u32 cfg_offset;
+ struct sitar_mbhc_imped_detect_cfg *imped_cfg;
+ struct sitar_mbhc_btn_detect_cfg *btn_cfg;
+
+ if (fw->size < SITAR_MBHC_CAL_MIN_SIZE)
+ return false;
+
+ /* previous check guarantees that there is enough fw data up
+ * to num_btn
+ */
+ btn_cfg = SITAR_MBHC_CAL_BTN_DET_PTR(fw->data);
+ cfg_offset = (u32) ((void *) btn_cfg - (void *) fw->data);
+ if (fw->size < (cfg_offset + SITAR_MBHC_CAL_BTN_SZ(btn_cfg)))
+ return false;
+
+ /* previous check guarantees that there is enough fw data up
+ * to start of impedance detection configuration
+ */
+ imped_cfg = SITAR_MBHC_CAL_IMPED_DET_PTR(fw->data);
+ cfg_offset = (u32) ((void *) imped_cfg - (void *) fw->data);
+
+ if (fw->size < (cfg_offset + SITAR_MBHC_CAL_IMPED_MIN_SZ))
+ return false;
+
+ if (fw->size < (cfg_offset + SITAR_MBHC_CAL_IMPED_SZ(imped_cfg)))
+ return false;
+
+ return true;
+}
+
+
+static void sitar_turn_onoff_override(struct snd_soc_codec *codec, bool on)
+{
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_B1_CTL, 0x04, on << 2);
+}
+
+/* called under codec_resource_lock acquisition */
+void sitar_set_and_turnoff_hph_padac(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ u8 wg_time;
+
+ wg_time = snd_soc_read(codec, SITAR_A_RX_HPH_CNP_WG_TIME) ;
+ wg_time += 1;
+
+ /* If headphone PA is on, check if userspace receives
+ * removal event to sync-up PA's state */
+ if (sitar_is_hph_pa_on(codec)) {
+ pr_debug("%s PA is on, setting PA_OFF_ACK\n", __func__);
+ set_bit(SITAR_HPHL_PA_OFF_ACK, &sitar->hph_pa_dac_state);
+ set_bit(SITAR_HPHR_PA_OFF_ACK, &sitar->hph_pa_dac_state);
+ } else {
+ pr_debug("%s PA is off\n", __func__);
+ }
+
+ if (sitar_is_hph_dac_on(codec, 1))
+ set_bit(SITAR_HPHL_DAC_OFF_ACK, &sitar->hph_pa_dac_state);
+ if (sitar_is_hph_dac_on(codec, 0))
+ set_bit(SITAR_HPHR_DAC_OFF_ACK, &sitar->hph_pa_dac_state);
+
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_CNP_EN, 0x30, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_L_DAC_CTL,
+ 0xC0, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_R_DAC_CTL,
+ 0xC0, 0x00);
+ usleep_range(wg_time * 1000, wg_time * 1000);
+}
+
+static void sitar_clr_and_turnon_hph_padac(struct sitar_priv *sitar)
+{
+ bool pa_turned_on = false;
+ struct snd_soc_codec *codec = sitar->codec;
+ u8 wg_time;
+
+ wg_time = snd_soc_read(codec, SITAR_A_RX_HPH_CNP_WG_TIME) ;
+ wg_time += 1;
+
+ if (test_and_clear_bit(SITAR_HPHR_DAC_OFF_ACK,
+ &sitar->hph_pa_dac_state)) {
+ pr_debug("%s: HPHR clear flag and enable DAC\n", __func__);
+ snd_soc_update_bits(sitar->codec, SITAR_A_RX_HPH_R_DAC_CTL,
+ 0xC0, 0xC0);
+ }
+ if (test_and_clear_bit(SITAR_HPHL_DAC_OFF_ACK,
+ &sitar->hph_pa_dac_state)) {
+ pr_debug("%s: HPHL clear flag and enable DAC\n", __func__);
+ snd_soc_update_bits(sitar->codec, SITAR_A_RX_HPH_L_DAC_CTL,
+ 0xC0, 0xC0);
+ }
+
+ if (test_and_clear_bit(SITAR_HPHR_PA_OFF_ACK,
+ &sitar->hph_pa_dac_state)) {
+ pr_debug("%s: HPHR clear flag and enable PA\n", __func__);
+ snd_soc_update_bits(sitar->codec, SITAR_A_RX_HPH_CNP_EN, 0x10,
+ 1 << 4);
+ pa_turned_on = true;
+ }
+ if (test_and_clear_bit(SITAR_HPHL_PA_OFF_ACK,
+ &sitar->hph_pa_dac_state)) {
+ pr_debug("%s: HPHL clear flag and enable PA\n", __func__);
+ snd_soc_update_bits(sitar->codec, SITAR_A_RX_HPH_CNP_EN, 0x20,
+ 1 << 5);
+ pa_turned_on = true;
+ }
+
+ if (pa_turned_on) {
+ pr_debug("%s: PA was turned off by MBHC and not by DAPM\n",
+ __func__);
+ usleep_range(wg_time * 1000, wg_time * 1000);
+ }
+}
+
+static void sitar_codec_report_plug(struct snd_soc_codec *codec, int insertion,
+ enum snd_jack_types jack_type)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ if (!insertion) {
+ /* Report removal */
+ sitar->hph_status &= ~jack_type;
+ if (sitar->mbhc_cfg.headset_jack) {
+ /* cancel possibly scheduled btn work and
+ * report release if we reported button press */
+ if (sitar_cancel_btn_work(sitar)) {
+ pr_debug("%s: button press is canceled\n",
+ __func__);
+ } else if (sitar->buttons_pressed) {
+ pr_debug("%s: Reporting release for reported "
+ "button press %d\n", __func__,
+ jack_type);
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.button_jack, 0,
+ sitar->buttons_pressed);
+ sitar->buttons_pressed &=
+ ~SITAR_JACK_BUTTON_MASK;
+ }
+ pr_debug("%s: Reporting removal %d\n", __func__,
+ jack_type);
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.headset_jack,
+ sitar->hph_status,
+ SITAR_JACK_MASK);
+ }
+ sitar_set_and_turnoff_hph_padac(codec);
+ hphocp_off_report(sitar, SND_JACK_OC_HPHR,
+ SITAR_IRQ_HPH_PA_OCPR_FAULT);
+ hphocp_off_report(sitar, SND_JACK_OC_HPHL,
+ SITAR_IRQ_HPH_PA_OCPL_FAULT);
+ sitar->current_plug = PLUG_TYPE_NONE;
+ sitar->mbhc_polling_active = false;
+ } else {
+ /* Report insertion */
+ sitar->hph_status |= jack_type;
+
+ if (jack_type == SND_JACK_HEADPHONE)
+ sitar->current_plug = PLUG_TYPE_HEADPHONE;
+ else if (jack_type == SND_JACK_HEADSET) {
+ sitar->mbhc_polling_active = true;
+ sitar->current_plug = PLUG_TYPE_HEADSET;
+ }
+ if (sitar->mbhc_cfg.headset_jack) {
+ pr_debug("%s: Reporting insertion %d\n", __func__,
+ jack_type);
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.headset_jack,
+ sitar->hph_status,
+ SITAR_JACK_MASK);
+ }
+ sitar_clr_and_turnon_hph_padac(sitar);
+ }
+}
+
+
+static bool sitar_hs_gpio_level_remove(struct sitar_priv *sitar)
+{
+ return (gpio_get_value_cansleep(sitar->mbhc_cfg.gpio) !=
+ sitar->mbhc_cfg.gpio_level_insert);
+}
+
+static bool sitar_is_invalid_insert_delta(struct snd_soc_codec *codec,
+ int mic_volt, int mic_volt_prev)
+{
+ int delta = abs(mic_volt - mic_volt_prev);
+ if (delta > SITAR_MBHC_FAKE_INSERT_VOLT_DELTA_MV) {
+ pr_debug("%s: volt delta %dmv\n", __func__, delta);
+ return true;
+ }
+ return false;
+}
+
+static bool sitar_is_invalid_insertion_range(struct snd_soc_codec *codec,
+ s32 mic_volt)
+{
+ bool invalid = false;
+
+ if (mic_volt < SITAR_MBHC_FAKE_INSERT_HIGH
+ && (mic_volt > SITAR_MBHC_FAKE_INSERT_LOW)) {
+ invalid = true;
+ }
+
+ return invalid;
+}
+
+static bool sitar_codec_is_invalid_plug(struct snd_soc_codec *codec,
+ s32 mic_mv[MBHC_NUM_DCE_PLUG_DETECT],
+ enum sitar_mbhc_plug_type plug_type[MBHC_NUM_DCE_PLUG_DETECT])
+{
+ int i;
+ bool r = false;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ struct sitar_mbhc_plug_type_cfg *plug_type_ptr =
+ SITAR_MBHC_CAL_PLUG_TYPE_PTR(sitar->mbhc_cfg.calibration);
+
+ for (i = 0 ; i < MBHC_NUM_DCE_PLUG_DETECT && !r; i++) {
+ if (mic_mv[i] < plug_type_ptr->v_no_mic)
+ plug_type[i] = PLUG_TYPE_HEADPHONE;
+ else if (mic_mv[i] < plug_type_ptr->v_hs_max)
+ plug_type[i] = PLUG_TYPE_HEADSET;
+ else if (mic_mv[i] > plug_type_ptr->v_hs_max)
+ plug_type[i] = PLUG_TYPE_HIGH_HPH;
+
+ r = sitar_is_invalid_insertion_range(codec, mic_mv[i]);
+ if (!r && i > 0) {
+ if (plug_type[i-1] != plug_type[i])
+ r = true;
+ else
+ r = sitar_is_invalid_insert_delta(codec,
+ mic_mv[i],
+ mic_mv[i - 1]);
+ }
+ }
+
+ return r;
+}
+
+/* called under codec_resource_lock acquisition */
+void sitar_find_plug_and_report(struct snd_soc_codec *codec,
+ enum sitar_mbhc_plug_type plug_type)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ if (plug_type == PLUG_TYPE_HEADPHONE
+ && sitar->current_plug == PLUG_TYPE_NONE) {
+ /* Nothing was reported previously
+ * reporte a headphone
+ */
+ sitar_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+ sitar_codec_cleanup_hs_polling(codec);
+ } else if (plug_type == PLUG_TYPE_HEADSET) {
+ /* If Headphone was reported previously, this will
+ * only report the mic line
+ */
+ sitar_codec_report_plug(codec, 1, SND_JACK_HEADSET);
+ msleep(100);
+ sitar_codec_start_hs_polling(codec);
+ } else if (plug_type == PLUG_TYPE_HIGH_HPH) {
+ if (sitar->current_plug == PLUG_TYPE_NONE)
+ sitar_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+ sitar_codec_cleanup_hs_polling(codec);
+ pr_debug("setup mic trigger for further detection\n");
+ sitar->lpi_enabled = true;
+ /* TODO ::: sitar_codec_enable_hs_detect */
+ pr_err("%s(): High impedence hph not supported\n", __func__);
+ }
+}
+
+/* should be called under interrupt context that hold suspend */
+static void sitar_schedule_hs_detect_plug(struct sitar_priv *sitar)
+{
+ pr_debug("%s: scheduling sitar_hs_correct_gpio_plug\n", __func__);
+ sitar->hs_detect_work_stop = false;
+ wcd9xxx_lock_sleep(sitar->codec->control_data);
+ schedule_work(&sitar->hs_correct_plug_work);
+}
+
+/* called under codec_resource_lock acquisition */
+static void sitar_cancel_hs_detect_plug(struct sitar_priv *sitar)
+{
+ pr_debug("%s: canceling hs_correct_plug_work\n", __func__);
+ sitar->hs_detect_work_stop = true;
+ wmb();
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ if (cancel_work_sync(&sitar->hs_correct_plug_work)) {
+ pr_debug("%s: hs_correct_plug_work is canceled\n", __func__);
+ wcd9xxx_unlock_sleep(sitar->codec->control_data);
+ }
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+}
+
+static void sitar_hs_correct_gpio_plug(struct work_struct *work)
+{
+ struct sitar_priv *sitar;
+ struct snd_soc_codec *codec;
+ int retry = 0, i;
+ bool correction = false;
+ s32 mic_mv[MBHC_NUM_DCE_PLUG_DETECT];
+ short mb_v[MBHC_NUM_DCE_PLUG_DETECT];
+ enum sitar_mbhc_plug_type plug_type[MBHC_NUM_DCE_PLUG_DETECT];
+ unsigned long timeout;
+
+ sitar = container_of(work, struct sitar_priv, hs_correct_plug_work);
+ codec = sitar->codec;
+
+ pr_debug("%s: enter\n", __func__);
+ sitar->mbhc_cfg.mclk_cb_fn(codec, 1, false);
+
+ /* Keep override on during entire plug type correction work.
+ *
+ * This is okay under the assumption that any GPIO irqs which use
+ * MBHC block cancel and sync this work so override is off again
+ * prior to GPIO interrupt handler's MBHC block usage.
+ * Also while this correction work is running, we can guarantee
+ * DAPM doesn't use any MBHC block as this work only runs with
+ * headphone detection.
+ */
+ sitar_turn_onoff_override(codec, true);
+
+ timeout = jiffies + msecs_to_jiffies(SITAR_HS_DETECT_PLUG_TIME_MS);
+ while (!time_after(jiffies, timeout)) {
+ ++retry;
+ rmb();
+ if (sitar->hs_detect_work_stop) {
+ pr_debug("%s: stop requested\n", __func__);
+ break;
+ }
+
+ msleep(SITAR_HS_DETECT_PLUG_INERVAL_MS);
+ if (sitar_hs_gpio_level_remove(sitar)) {
+ pr_debug("%s: GPIO value is low\n", __func__);
+ break;
+ }
+
+ /* can race with removal interrupt */
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ for (i = 0; i < MBHC_NUM_DCE_PLUG_DETECT; i++) {
+ mb_v[i] = __sitar_codec_sta_dce(codec, 1, true, true);
+ mic_mv[i] = sitar_codec_sta_dce_v(codec, 1 , mb_v[i]);
+ pr_debug("%s : DCE run %d, mic_mv = %d(%x)\n",
+ __func__, retry, mic_mv[i], mb_v[i]);
+ }
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+
+ if (sitar_codec_is_invalid_plug(codec, mic_mv, plug_type)) {
+ pr_debug("Invalid plug in attempt # %d\n", retry);
+ if (retry == NUM_ATTEMPTS_TO_REPORT &&
+ sitar->current_plug == PLUG_TYPE_NONE) {
+ sitar_codec_report_plug(codec, 1,
+ SND_JACK_HEADPHONE);
+ }
+ } else if (!sitar_codec_is_invalid_plug(codec, mic_mv,
+ plug_type) &&
+ plug_type[0] == PLUG_TYPE_HEADPHONE) {
+ pr_debug("Good headphone detected, continue polling mic\n");
+ if (sitar->current_plug == PLUG_TYPE_NONE) {
+ sitar_codec_report_plug(codec, 1,
+ SND_JACK_HEADPHONE);
+ }
+ } else {
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ /* Turn off override */
+ sitar_turn_onoff_override(codec, false);
+ sitar_find_plug_and_report(codec, plug_type[0]);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ pr_debug("Attempt %d found correct plug %d\n", retry,
+ plug_type[0]);
+ correction = true;
+ break;
+ }
+ }
+
+ /* Turn off override */
+ if (!correction)
+ sitar_turn_onoff_override(codec, false);
+
+ sitar->mbhc_cfg.mclk_cb_fn(codec, 0, false);
+ pr_debug("%s: leave\n", __func__);
+ /* unlock sleep */
+ wcd9xxx_unlock_sleep(sitar->codec->control_data);
+}
+
+/* called under codec_resource_lock acquisition */
+static void sitar_codec_decide_gpio_plug(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ short mb_v[MBHC_NUM_DCE_PLUG_DETECT];
+ s32 mic_mv[MBHC_NUM_DCE_PLUG_DETECT];
+ enum sitar_mbhc_plug_type plug_type[MBHC_NUM_DCE_PLUG_DETECT];
+ int i;
+
+ pr_debug("%s: enter\n", __func__);
+
+ sitar_turn_onoff_override(codec, true);
+ mb_v[0] = sitar_codec_setup_hs_polling(codec);
+ mic_mv[0] = sitar_codec_sta_dce_v(codec, 1, mb_v[0]);
+ pr_debug("%s: DCE run 1, mic_mv = %d\n", __func__, mic_mv[0]);
+
+ for (i = 1; i < MBHC_NUM_DCE_PLUG_DETECT; i++) {
+ mb_v[i] = __sitar_codec_sta_dce(codec, 1, true, true);
+ mic_mv[i] = sitar_codec_sta_dce_v(codec, 1 , mb_v[i]);
+ pr_debug("%s: DCE run %d, mic_mv = %d\n", __func__, i + 1,
+ mic_mv[i]);
+ }
+ sitar_turn_onoff_override(codec, false);
+
+ if (sitar_hs_gpio_level_remove(sitar)) {
+ pr_debug("%s: GPIO value is low when determining plug\n",
+ __func__);
+ return;
+ }
+
+ if (sitar_codec_is_invalid_plug(codec, mic_mv, plug_type)) {
+ sitar_schedule_hs_detect_plug(sitar);
+ } else if (plug_type[0] == PLUG_TYPE_HEADPHONE) {
+ sitar_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+ sitar_schedule_hs_detect_plug(sitar);
+ } else if (plug_type[0] == PLUG_TYPE_HEADSET) {
+ pr_debug("%s: Valid plug found, determine plug type\n",
+ __func__);
+ sitar_find_plug_and_report(codec, plug_type[0]);
+ }
+
+}
+
+/* called under codec_resource_lock acquisition */
+static void sitar_codec_detect_plug_type(struct snd_soc_codec *codec)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ const struct sitar_mbhc_plug_detect_cfg *plug_det =
+ SITAR_MBHC_CAL_PLUG_DET_PTR(sitar->mbhc_cfg.calibration);
+
+ if (plug_det->t_ins_complete > 20)
+ msleep(plug_det->t_ins_complete);
+ else
+ usleep_range(plug_det->t_ins_complete * 1000,
+ plug_det->t_ins_complete * 1000);
+
+ if (sitar_hs_gpio_level_remove(sitar))
+ pr_debug("%s: GPIO value is low when determining "
+ "plug\n", __func__);
+ else
+ sitar_codec_decide_gpio_plug(codec);
+
+ return;
+}
+
+static void sitar_hs_gpio_handler(struct snd_soc_codec *codec)
+{
+ bool insert;
+ struct sitar_priv *priv = snd_soc_codec_get_drvdata(codec);
+ bool is_removed = false;
+
+ pr_debug("%s: enter\n", __func__);
+
+ priv->in_gpio_handler = true;
+ /* Wait here for debounce time */
+ usleep_range(SITAR_GPIO_IRQ_DEBOUNCE_TIME_US,
+ SITAR_GPIO_IRQ_DEBOUNCE_TIME_US);
+
+ SITAR_ACQUIRE_LOCK(priv->codec_resource_lock);
+
+ /* cancel pending button press */
+ if (sitar_cancel_btn_work(priv))
+ pr_debug("%s: button press is canceled\n", __func__);
+
+ insert = (gpio_get_value_cansleep(priv->mbhc_cfg.gpio) ==
+ priv->mbhc_cfg.gpio_level_insert);
+ if ((priv->current_plug == PLUG_TYPE_NONE) && insert) {
+ priv->lpi_enabled = false;
+ wmb();
+
+ /* cancel detect plug */
+ sitar_cancel_hs_detect_plug(priv);
+
+ /* Disable Mic Bias pull down and HPH Switch to GND */
+ snd_soc_update_bits(codec, priv->mbhc_bias_regs.ctl_reg, 0x01,
+ 0x00);
+ snd_soc_update_bits(codec, SITAR_A_MBHC_HPH, 0x01, 0x00);
+ sitar_codec_detect_plug_type(codec);
+ } else if ((priv->current_plug != PLUG_TYPE_NONE) && !insert) {
+ priv->lpi_enabled = false;
+ wmb();
+
+ /* cancel detect plug */
+ sitar_cancel_hs_detect_plug(priv);
+
+ if (priv->current_plug == PLUG_TYPE_HEADPHONE) {
+ sitar_codec_report_plug(codec, 0, SND_JACK_HEADPHONE);
+ is_removed = true;
+ } else if (priv->current_plug == PLUG_TYPE_HEADSET) {
+ sitar_codec_pause_hs_polling(codec);
+ sitar_codec_cleanup_hs_polling(codec);
+ sitar_codec_report_plug(codec, 0, SND_JACK_HEADSET);
+ is_removed = true;
+ }
+
+ if (is_removed) {
+ /* Enable Mic Bias pull down and HPH Switch to GND */
+ snd_soc_update_bits(codec,
+ priv->mbhc_bias_regs.ctl_reg, 0x01,
+ 0x01);
+ snd_soc_update_bits(codec, SITAR_A_MBHC_HPH, 0x01,
+ 0x01);
+ /* Make sure mic trigger is turned off */
+ snd_soc_update_bits(codec,
+ priv->mbhc_bias_regs.ctl_reg,
+ 0x01, 0x01);
+ snd_soc_update_bits(codec,
+ priv->mbhc_bias_regs.mbhc_reg,
+ 0x90, 0x00);
+ /* Reset MBHC State Machine */
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL,
+ 0x08, 0x08);
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_CLK_CTL,
+ 0x08, 0x00);
+ /* Turn off override */
+ sitar_turn_onoff_override(codec, false);
+ }
+ }
+
+ priv->in_gpio_handler = false;
+ SITAR_RELEASE_LOCK(priv->codec_resource_lock);
+ pr_debug("%s: leave\n", __func__);
+}
+
+static irqreturn_t sitar_mechanical_plug_detect_irq(int irq, void *data)
+{
+ int r = IRQ_HANDLED;
+ struct snd_soc_codec *codec = data;
+
+ if (unlikely(wcd9xxx_lock_sleep(codec->control_data) == false)) {
+ pr_warn("%s(): Failed to hold suspend\n", __func__);
+ r = IRQ_NONE;
+ } else {
+ sitar_hs_gpio_handler(codec);
+ wcd9xxx_unlock_sleep(codec->control_data);
+ }
+ return r;
+}
+
+static int sitar_mbhc_init_and_calibrate(struct snd_soc_codec *codec)
+{
+ int rc = 0;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ sitar->mbhc_cfg.mclk_cb_fn(codec, 1, false);
+ sitar_mbhc_init(codec);
+ sitar_mbhc_cal(codec);
+ sitar_mbhc_calc_thres(codec);
+ sitar->mbhc_cfg.mclk_cb_fn(codec, 0, false);
+ sitar_codec_calibrate_hs_polling(codec);
+
+ /* Enable Mic Bias pull down and HPH Switch to GND */
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.ctl_reg,
+ 0x01, 0x01);
+ snd_soc_update_bits(codec, SITAR_A_MBHC_HPH,
+ 0x01, 0x01);
+
+ rc = request_threaded_irq(sitar->mbhc_cfg.gpio_irq,
+ NULL,
+ sitar_mechanical_plug_detect_irq,
+ (IRQF_TRIGGER_RISING |
+ IRQF_TRIGGER_FALLING),
+ "sitar-hs-gpio", codec);
+
+ if (!IS_ERR_VALUE(rc)) {
+ rc = enable_irq_wake(sitar->mbhc_cfg.gpio_irq);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL,
+ 0x10, 0x10);
+ wcd9xxx_enable_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPL_FAULT);
+ wcd9xxx_enable_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPR_FAULT);
+ /* Bootup time detection */
+ sitar_hs_gpio_handler(codec);
+ }
+
+ return rc;
+}
+
+static void mbhc_fw_read(struct work_struct *work)
+{
+ struct delayed_work *dwork;
+ struct sitar_priv *sitar;
+ struct snd_soc_codec *codec;
+ const struct firmware *fw;
+ int ret = -1, retry = 0;
+
+ dwork = to_delayed_work(work);
+ sitar = container_of(dwork, struct sitar_priv,
+ mbhc_firmware_dwork);
+ codec = sitar->codec;
+
+ while (retry < MBHC_FW_READ_ATTEMPTS) {
+ retry++;
+ pr_info("%s:Attempt %d to request MBHC firmware\n",
+ __func__, retry);
+ ret = request_firmware(&fw, "wcd9310/wcd9310_mbhc.bin",
+ codec->dev);
+
+ if (ret != 0) {
+ usleep_range(MBHC_FW_READ_TIMEOUT,
+ MBHC_FW_READ_TIMEOUT);
+ } else {
+ pr_info("%s: MBHC Firmware read succesful\n", __func__);
+ break;
+ }
+ }
+
+ if (ret != 0) {
+ pr_err("%s: Cannot load MBHC firmware use default cal\n",
+ __func__);
+ } else if (sitar_mbhc_fw_validate(fw) == false) {
+ pr_err("%s: Invalid MBHC cal data size use default cal\n",
+ __func__);
+ release_firmware(fw);
+ } else {
+ sitar->calibration = (void *)fw->data;
+ sitar->mbhc_fw = fw;
+ }
+
+ sitar_mbhc_init_and_calibrate(codec);
+}
+
+int sitar_hs_detect(struct snd_soc_codec *codec,
+ const struct sitar_mbhc_config *cfg)
+{
+ struct sitar_priv *sitar;
+ int rc = 0;
+
+ if (!codec || !cfg->calibration) {
+ pr_err("Error: no codec or calibration\n");
+ return -EINVAL;
+ }
+
+ if (cfg->mclk_rate != SITAR_MCLK_RATE_12288KHZ) {
+ if (cfg->mclk_rate == SITAR_MCLK_RATE_9600KHZ)
+ pr_err("Error: clock rate %dHz is not yet supported\n",
+ cfg->mclk_rate);
+ else
+ pr_err("Error: unsupported clock rate %d\n",
+ cfg->mclk_rate);
+ return -EINVAL;
+ }
+
+ sitar = snd_soc_codec_get_drvdata(codec);
+ sitar->mbhc_cfg = *cfg;
+ sitar->in_gpio_handler = false;
+ sitar->current_plug = PLUG_TYPE_NONE;
+ sitar->lpi_enabled = false;
+ sitar_get_mbhc_micbias_regs(codec, &sitar->mbhc_bias_regs);
+
+ /* Put CFILT in fast mode by default */
+ snd_soc_update_bits(codec, sitar->mbhc_bias_regs.cfilt_ctl,
+ 0x40, SITAR_CFILT_FAST_MODE);
+
+ INIT_DELAYED_WORK(&sitar->mbhc_firmware_dwork, mbhc_fw_read);
+ INIT_DELAYED_WORK(&sitar->mbhc_btn_dwork, btn_lpress_fn);
+ INIT_WORK(&sitar->hphlocp_work, hphlocp_off_report);
+ INIT_WORK(&sitar->hphrocp_work, hphrocp_off_report);
+ INIT_WORK(&sitar->hs_correct_plug_work,
+ sitar_hs_correct_gpio_plug);
+
+ if (!sitar->mbhc_cfg.read_fw_bin) {
+ rc = sitar_mbhc_init_and_calibrate(codec);
+ } else {
+ schedule_delayed_work(&sitar->mbhc_firmware_dwork,
+ usecs_to_jiffies(MBHC_FW_READ_TIMEOUT));
+ }
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(sitar_hs_detect);
+
+static int sitar_determine_button(const struct sitar_priv *priv,
+ const s32 bias_mv)
+{
+ s16 *v_btn_low, *v_btn_high;
+ struct sitar_mbhc_btn_detect_cfg *btn_det;
+ int i, btn = -1;
+
+ btn_det = SITAR_MBHC_CAL_BTN_DET_PTR(priv->mbhc_cfg.calibration);
+ v_btn_low = sitar_mbhc_cal_btn_det_mp(btn_det, SITAR_BTN_DET_V_BTN_LOW);
+ v_btn_high = sitar_mbhc_cal_btn_det_mp(btn_det,
+ SITAR_BTN_DET_V_BTN_HIGH);
+ for (i = 0; i < btn_det->num_btn; i++) {
+ if ((v_btn_low[i] <= bias_mv) && (v_btn_high[i] >= bias_mv)) {
+ btn = i;
+ break;
+ }
+ }
+
+ if (btn == -1)
+ pr_debug("%s: couldn't find button number for mic mv %d\n",
+ __func__, bias_mv);
+
+ return btn;
+}
+
+static int sitar_get_button_mask(const int btn)
+{
+ int mask = 0;
+ switch (btn) {
+ case 0:
+ mask = SND_JACK_BTN_0;
+ break;
+ case 1:
+ mask = SND_JACK_BTN_1;
+ break;
+ case 2:
+ mask = SND_JACK_BTN_2;
+ break;
+ case 3:
+ mask = SND_JACK_BTN_3;
+ break;
+ case 4:
+ mask = SND_JACK_BTN_4;
+ break;
+ case 5:
+ mask = SND_JACK_BTN_5;
+ break;
+ case 6:
+ mask = SND_JACK_BTN_6;
+ break;
+ case 7:
+ mask = SND_JACK_BTN_7;
+ break;
+ }
+ return mask;
+}
+
+
+static irqreturn_t sitar_dce_handler(int irq, void *data)
+{
+ int i, mask;
+ short dce, sta, bias_value_dce;
+ s32 mv, stamv, bias_mv_dce;
+ int btn = -1, meas = 0;
+ struct sitar_priv *priv = data;
+ const struct sitar_mbhc_btn_detect_cfg *d =
+ SITAR_MBHC_CAL_BTN_DET_PTR(priv->mbhc_cfg.calibration);
+ short btnmeas[d->n_btn_meas + 1];
+ struct snd_soc_codec *codec = priv->codec;
+ struct wcd9xxx *core = dev_get_drvdata(priv->codec->dev->parent);
+ int n_btn_meas = d->n_btn_meas;
+ u8 mbhc_status = snd_soc_read(codec, SITAR_A_CDC_MBHC_B1_STATUS) & 0x3E;
+
+ pr_debug("%s: enter\n", __func__);
+
+ SITAR_ACQUIRE_LOCK(priv->codec_resource_lock);
+ if (priv->mbhc_state == MBHC_STATE_POTENTIAL_RECOVERY) {
+ pr_debug("%s: mbhc is being recovered, skip button press\n",
+ __func__);
+ goto done;
+ }
+
+ priv->mbhc_state = MBHC_STATE_POTENTIAL;
+
+ if (!priv->mbhc_polling_active) {
+ pr_warn("%s: mbhc polling is not active, skip button press\n",
+ __func__);
+ goto done;
+ }
+
+ dce = sitar_codec_read_dce_result(codec);
+ mv = sitar_codec_sta_dce_v(codec, 1, dce);
+
+ /* If GPIO interrupt already kicked in, ignore button press */
+ if (priv->in_gpio_handler) {
+ pr_debug("%s: GPIO State Changed, ignore button press\n",
+ __func__);
+ btn = -1;
+ goto done;
+ }
+
+ if (mbhc_status != SITAR_MBHC_STATUS_REL_DETECTION) {
+ if (priv->mbhc_last_resume &&
+ !time_after(jiffies, priv->mbhc_last_resume + HZ)) {
+ pr_debug("%s: Button is already released shortly after "
+ "resume\n", __func__);
+ n_btn_meas = 0;
+ } else {
+ pr_debug("%s: Button is already released without "
+ "resume", __func__);
+ sta = sitar_codec_read_sta_result(codec);
+ stamv = sitar_codec_sta_dce_v(codec, 0, sta);
+ btn = sitar_determine_button(priv, mv);
+ if (btn != sitar_determine_button(priv, stamv))
+ btn = -1;
+ goto done;
+ }
+ }
+
+ /* determine pressed button */
+ btnmeas[meas++] = sitar_determine_button(priv, mv);
+ pr_debug("%s: meas %d - DCE %d,%d, button %d\n", __func__,
+ meas - 1, dce, mv, btnmeas[meas - 1]);
+ if (n_btn_meas == 0)
+ btn = btnmeas[0];
+ for (; ((d->n_btn_meas) && (meas < (d->n_btn_meas + 1))); meas++) {
+ bias_value_dce = sitar_codec_sta_dce(codec, 1, false);
+ bias_mv_dce = sitar_codec_sta_dce_v(codec, 1, bias_value_dce);
+ btnmeas[meas] = sitar_determine_button(priv, bias_mv_dce);
+ pr_debug("%s: meas %d - DCE %d,%d, button %d\n",
+ __func__, meas, bias_value_dce, bias_mv_dce,
+ btnmeas[meas]);
+ /* if large enough measurements are collected,
+ * start to check if last all n_btn_con measurements were
+ * in same button low/high range */
+ if (meas + 1 >= d->n_btn_con) {
+ for (i = 0; i < d->n_btn_con; i++)
+ if ((btnmeas[meas] < 0) ||
+ (btnmeas[meas] != btnmeas[meas - i]))
+ break;
+ if (i == d->n_btn_con) {
+ /* button pressed */
+ btn = btnmeas[meas];
+ break;
+ } else if ((n_btn_meas - meas) < (d->n_btn_con - 1)) {
+ /* if left measurements are less than n_btn_con,
+ * it's impossible to find button number */
+ break;
+ }
+ }
+ }
+
+ if (btn >= 0) {
+ if (priv->in_gpio_handler) {
+ pr_debug("%s: GPIO already triggered, ignore button "
+ "press\n", __func__);
+ goto done;
+ }
+ mask = sitar_get_button_mask(btn);
+ priv->buttons_pressed |= mask;
+ wcd9xxx_lock_sleep(core);
+ if (schedule_delayed_work(&priv->mbhc_btn_dwork,
+ msecs_to_jiffies(400)) == 0) {
+ WARN(1, "Button pressed twice without release"
+ "event\n");
+ wcd9xxx_unlock_sleep(core);
+ }
+ } else {
+ pr_debug("%s: bogus button press, too short press?\n",
+ __func__);
+ }
+
+ done:
+ pr_debug("%s: leave\n", __func__);
+ SITAR_RELEASE_LOCK(priv->codec_resource_lock);
+ return IRQ_HANDLED;
+}
+
+static int sitar_is_fake_press(struct sitar_priv *priv)
+{
+ int i;
+ int r = 0;
+ struct snd_soc_codec *codec = priv->codec;
+ const int dces = MBHC_NUM_DCE_PLUG_DETECT;
+ short mb_v;
+
+ for (i = 0; i < dces; i++) {
+ usleep_range(10000, 10000);
+ if (i == 0) {
+ mb_v = sitar_codec_sta_dce(codec, 0, true);
+ pr_debug("%s: STA[0]: %d,%d\n", __func__, mb_v,
+ sitar_codec_sta_dce_v(codec, 0, mb_v));
+ if (mb_v < (short)priv->mbhc_data.v_b1_hu ||
+ mb_v > (short)priv->mbhc_data.v_ins_hu) {
+ r = 1;
+ break;
+ }
+ } else {
+ mb_v = sitar_codec_sta_dce(codec, 1, true);
+ pr_debug("%s: DCE[%d]: %d,%d\n", __func__, i, mb_v,
+ sitar_codec_sta_dce_v(codec, 1, mb_v));
+ if (mb_v < (short)priv->mbhc_data.v_b1_h ||
+ mb_v > (short)priv->mbhc_data.v_ins_h) {
+ r = 1;
+ break;
+ }
+ }
+ }
+
+ return r;
+}
+
+static irqreturn_t sitar_release_handler(int irq, void *data)
+{
+ int ret;
+ struct sitar_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+
+ pr_debug("%s: enter\n", __func__);
+
+ SITAR_ACQUIRE_LOCK(priv->codec_resource_lock);
+ priv->mbhc_state = MBHC_STATE_RELEASE;
+
+ if (priv->buttons_pressed & SITAR_JACK_BUTTON_MASK) {
+ ret = sitar_cancel_btn_work(priv);
+ if (ret == 0) {
+ pr_debug("%s: Reporting long button release event\n",
+ __func__);
+ if (priv->mbhc_cfg.button_jack)
+ sitar_snd_soc_jack_report(priv,
+ priv->mbhc_cfg.button_jack, 0,
+ priv->buttons_pressed);
+ } else {
+ if (sitar_is_fake_press(priv)) {
+ pr_debug("%s: Fake button press interrupt\n",
+ __func__);
+ } else if (priv->mbhc_cfg.button_jack) {
+ if (priv->in_gpio_handler) {
+ pr_debug("%s: GPIO kicked in, ignore\n",
+ __func__);
+ } else {
+ pr_debug("%s: Reporting short button 0 "
+ "press and release\n",
+ __func__);
+ sitar_snd_soc_jack_report(priv,
+ priv->mbhc_cfg.button_jack,
+ priv->buttons_pressed,
+ priv->buttons_pressed);
+ sitar_snd_soc_jack_report(priv,
+ priv->mbhc_cfg.button_jack, 0,
+ priv->buttons_pressed);
+ }
+ }
+ }
+
+ priv->buttons_pressed &= ~SITAR_JACK_BUTTON_MASK;
+ }
+
+ sitar_codec_calibrate_hs_polling(codec);
+
+ if (priv->mbhc_cfg.gpio)
+ msleep(SITAR_MBHC_GPIO_REL_DEBOUNCE_TIME_MS);
+
+ sitar_codec_start_hs_polling(codec);
+
+ pr_debug("%s: leave\n", __func__);
+ SITAR_RELEASE_LOCK(priv->codec_resource_lock);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t sitar_hphl_ocp_irq(int irq, void *data)
+{
+ struct sitar_priv *sitar = data;
+ struct snd_soc_codec *codec;
+
+ pr_info("%s: received HPHL OCP irq\n", __func__);
+
+ if (sitar) {
+ codec = sitar->codec;
+ if (sitar->hphlocp_cnt++ < SITAR_OCP_ATTEMPT) {
+ pr_info("%s: retry\n", __func__);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL, 0x10,
+ 0x00);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL, 0x10,
+ 0x10);
+ } else {
+ wcd9xxx_disable_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPL_FAULT);
+ sitar->hphlocp_cnt = 0;
+ sitar->hph_status |= SND_JACK_OC_HPHL;
+ if (sitar->mbhc_cfg.headset_jack)
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.headset_jack,
+ sitar->hph_status,
+ SITAR_JACK_MASK);
+ }
+ } else {
+ pr_err("%s: Bad sitar private data\n", __func__);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t sitar_hphr_ocp_irq(int irq, void *data)
+{
+ struct sitar_priv *sitar = data;
+ struct snd_soc_codec *codec;
+
+ pr_info("%s: received HPHR OCP irq\n", __func__);
+
+ if (sitar) {
+ codec = sitar->codec;
+ if (sitar->hphrocp_cnt++ < SITAR_OCP_ATTEMPT) {
+ pr_info("%s: retry\n", __func__);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL, 0x10,
+ 0x00);
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL, 0x10,
+ 0x10);
+ } else {
+ wcd9xxx_disable_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPR_FAULT);
+ sitar->hphrocp_cnt = 0;
+ sitar->hph_status |= SND_JACK_OC_HPHR;
+ if (sitar->mbhc_cfg.headset_jack)
+ sitar_snd_soc_jack_report(sitar,
+ sitar->mbhc_cfg.headset_jack,
+ sitar->hph_status,
+ SITAR_JACK_MASK);
+ }
+ } else {
+ pr_err("%s: Bad sitar private data\n", __func__);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t sitar_hs_insert_irq(int irq, void *data)
+{
+ struct sitar_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+
+ pr_debug("%s: enter\n", __func__);
+ SITAR_ACQUIRE_LOCK(priv->codec_resource_lock);
+ wcd9xxx_disable_irq(codec->control_data, SITAR_IRQ_MBHC_INSERTION);
+
+ snd_soc_update_bits(codec, SITAR_A_CDC_MBHC_INT_CTL, 0x03, 0x00);
+
+ /* Turn off both HPH and MIC line schmitt triggers */
+ snd_soc_update_bits(codec, priv->mbhc_bias_regs.mbhc_reg, 0x90, 0x00);
+ snd_soc_update_bits(codec, SITAR_A_MBHC_HPH, 0x13, 0x00);
+ snd_soc_update_bits(codec, priv->mbhc_bias_regs.ctl_reg, 0x01, 0x00);
+
+ pr_debug("%s: MIC trigger insertion interrupt\n", __func__);
+
+ rmb();
+ if (priv->lpi_enabled)
+ msleep(100);
+
+ rmb();
+ if (!priv->lpi_enabled) {
+ pr_debug("%s: lpi is disabled\n", __func__);
+ } else if (gpio_get_value_cansleep(priv->mbhc_cfg.gpio) ==
+ priv->mbhc_cfg.gpio_level_insert) {
+ pr_debug("%s: Valid insertion, "
+ "detect plug type\n", __func__);
+ sitar_codec_decide_gpio_plug(codec);
+ } else {
+ pr_debug("%s: Invalid insertion, "
+ "stop plug detection\n", __func__);
+ }
+ SITAR_RELEASE_LOCK(priv->codec_resource_lock);
+ return IRQ_HANDLED;
+}
+
+static bool is_valid_mic_voltage(struct snd_soc_codec *codec, s32 mic_mv)
+{
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ struct sitar_mbhc_plug_type_cfg *plug_type =
+ SITAR_MBHC_CAL_PLUG_TYPE_PTR(sitar->mbhc_cfg.calibration);
+
+ return (!(mic_mv > SITAR_MBHC_FAKE_INSERT_LOW
+ && mic_mv < SITAR_MBHC_FAKE_INSERT_HIGH)
+ && (mic_mv > plug_type->v_no_mic)
+ && (mic_mv < plug_type->v_hs_max)) ? true : false;
+}
+
+/* called under codec_resource_lock acquisition
+ * returns true if mic voltage range is back to normal insertion
+ * returns false either if timedout or removed */
+static bool sitar_hs_remove_settle(struct snd_soc_codec *codec)
+{
+ int i;
+ bool timedout, settled = false;
+ s32 mic_mv[MBHC_NUM_DCE_PLUG_DETECT];
+ short mb_v[MBHC_NUM_DCE_PLUG_DETECT];
+ unsigned long retry = 0, timeout;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+
+ timeout = jiffies + msecs_to_jiffies(SITAR_HS_DETECT_PLUG_TIME_MS);
+ while (!(timedout = time_after(jiffies, timeout))) {
+ retry++;
+ if (sitar_hs_gpio_level_remove(sitar)) {
+ pr_debug("%s: GPIO indicates removal\n", __func__);
+ break;
+ }
+
+ if (retry > 1)
+ msleep(250);
+ else
+ msleep(50);
+
+ if (sitar_hs_gpio_level_remove(sitar)) {
+ pr_debug("%s: GPIO indicates removal\n", __func__);
+ break;
+ }
+
+ sitar_turn_onoff_override(codec, true);
+ for (i = 0; i < MBHC_NUM_DCE_PLUG_DETECT; i++) {
+ mb_v[i] = __sitar_codec_sta_dce(codec, 1, true, true);
+ mic_mv[i] = sitar_codec_sta_dce_v(codec, 1 , mb_v[i]);
+ pr_debug("%s : DCE run %lu, mic_mv = %d(%x)\n",
+ __func__, retry, mic_mv[i], mb_v[i]);
+ }
+ sitar_turn_onoff_override(codec, false);
+
+ if (sitar_hs_gpio_level_remove(sitar)) {
+ pr_debug("%s: GPIO indicates removal\n", __func__);
+ break;
+ }
+
+ for (i = 0; i < MBHC_NUM_DCE_PLUG_DETECT; i++)
+ if (!is_valid_mic_voltage(codec, mic_mv[i]))
+ break;
+
+ if (i == MBHC_NUM_DCE_PLUG_DETECT) {
+ pr_debug("%s: MIC voltage settled\n", __func__);
+ settled = true;
+ msleep(200);
+ break;
+ }
+ }
+
+ if (timedout)
+ pr_debug("%s: Microphone did not settle in %d seconds\n",
+ __func__, SITAR_HS_DETECT_PLUG_TIME_MS);
+ return settled;
+}
+
+static irqreturn_t sitar_hs_remove_irq(int irq, void *data)
+{
+ struct sitar_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+
+ pr_debug("%s: enter, removal interrupt\n", __func__);
+
+ SITAR_ACQUIRE_LOCK(priv->codec_resource_lock);
+ if (sitar_hs_remove_settle(codec))
+ sitar_codec_start_hs_polling(codec);
+ pr_debug("%s: remove settle done\n", __func__);
+
+ SITAR_RELEASE_LOCK(priv->codec_resource_lock);
+ return IRQ_HANDLED;
+}
+
+
+static unsigned long slimbus_value;
+
+static irqreturn_t sitar_slimbus_irq(int irq, void *data)
+{
+ struct sitar_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+ int i, j;
+ u8 val;
+
+
+ for (i = 0; i < WCD9XXX_SLIM_NUM_PORT_REG; i++) {
+ slimbus_value = wcd9xxx_interface_reg_read(codec->control_data,
+ SITAR_SLIM_PGD_PORT_INT_STATUS0 + i);
+ for_each_set_bit(j, &slimbus_value, BITS_PER_BYTE) {
+ val = wcd9xxx_interface_reg_read(codec->control_data,
+ SITAR_SLIM_PGD_PORT_INT_SOURCE0 + i*8 + j);
+ if (val & 0x1)
+ pr_err_ratelimited("overflow error on port %x,"
+ " value %x\n", i*8 + j, val);
+ if (val & 0x2)
+ pr_err_ratelimited("underflow error on port %x,"
+ " value %x\n", i*8 + j, val);
+ }
+ wcd9xxx_interface_reg_write(codec->control_data,
+ SITAR_SLIM_PGD_PORT_INT_CLR0 + i, 0xFF);
+ }
+
+ return IRQ_HANDLED;
+}
+
+
+static int sitar_handle_pdata(struct sitar_priv *sitar)
+{
+ struct snd_soc_codec *codec = sitar->codec;
+ struct wcd9xxx_pdata *pdata = sitar->pdata;
+ int k1, k2, rc = 0;
+ u8 leg_mode = pdata->amic_settings.legacy_mode;
+ u8 txfe_bypass = pdata->amic_settings.txfe_enable;
+ u8 txfe_buff = pdata->amic_settings.txfe_buff;
+ u8 flag = pdata->amic_settings.use_pdata;
+ u8 i = 0, j = 0;
+ u8 val_txfe = 0, value = 0;
+
+ if (!pdata) {
+ rc = -ENODEV;
+ goto done;
+ }
+
+ /* Make sure settings are correct */
+ if ((pdata->micbias.ldoh_v > SITAR_LDOH_2P85_V) ||
+ (pdata->micbias.bias1_cfilt_sel > SITAR_CFILT2_SEL) ||
+ (pdata->micbias.bias2_cfilt_sel > SITAR_CFILT2_SEL)) {
+ rc = -EINVAL;
+ goto done;
+ }
+
+ /* figure out k value */
+ k1 = sitar_find_k_value(pdata->micbias.ldoh_v,
+ pdata->micbias.cfilt1_mv);
+ k2 = sitar_find_k_value(pdata->micbias.ldoh_v,
+ pdata->micbias.cfilt2_mv);
+
+ if (IS_ERR_VALUE(k1) || IS_ERR_VALUE(k2)) {
+ rc = -EINVAL;
+ goto done;
+ }
+
+ /* Set voltage level and always use LDO */
+ snd_soc_update_bits(codec, SITAR_A_LDO_H_MODE_1, 0x0C,
+ (pdata->micbias.ldoh_v << 2));
+
+ snd_soc_update_bits(codec, SITAR_A_MICB_CFILT_1_VAL, 0xFC,
+ (k1 << 2));
+ snd_soc_update_bits(codec, SITAR_A_MICB_CFILT_2_VAL, 0xFC,
+ (k2 << 2));
+
+ snd_soc_update_bits(codec, SITAR_A_MICB_1_CTL, 0x60,
+ (pdata->micbias.bias1_cfilt_sel << 5));
+ snd_soc_update_bits(codec, SITAR_A_MICB_2_CTL, 0x60,
+ (pdata->micbias.bias2_cfilt_sel << 5));
+
+ for (i = 0; i < 6; j++, i += 2) {
+ if (flag & (0x01 << i)) {
+ value = (leg_mode & (0x01 << i)) ? 0x10 : 0x00;
+ val_txfe = (txfe_bypass & (0x01 << i)) ? 0x20 : 0x00;
+ val_txfe = val_txfe |
+ ((txfe_buff & (0x01 << i)) ? 0x10 : 0x00);
+ snd_soc_update_bits(codec, SITAR_A_TX_1_2_EN + j * 10,
+ 0x10, value);
+ snd_soc_update_bits(codec,
+ SITAR_A_TX_1_2_TEST_EN + j * 10,
+ 0x30, val_txfe);
+ }
+ if (flag & (0x01 << (i + 1))) {
+ value = (leg_mode & (0x01 << (i + 1))) ? 0x01 : 0x00;
+ val_txfe = (txfe_bypass &
+ (0x01 << (i + 1))) ? 0x02 : 0x00;
+ val_txfe |= (txfe_buff &
+ (0x01 << (i + 1))) ? 0x01 : 0x00;
+ snd_soc_update_bits(codec, SITAR_A_TX_1_2_EN + j * 10,
+ 0x01, value);
+ snd_soc_update_bits(codec,
+ SITAR_A_TX_1_2_TEST_EN + j * 10,
+ 0x03, val_txfe);
+ }
+ }
+ if (flag & 0x40) {
+ value = (leg_mode & 0x40) ? 0x10 : 0x00;
+ value = value | ((txfe_bypass & 0x40) ? 0x02 : 0x00);
+ value = value | ((txfe_buff & 0x40) ? 0x01 : 0x00);
+ snd_soc_update_bits(codec, SITAR_A_TX_4_MBHC_EN,
+ 0x13, value);
+ }
+
+
+ if (pdata->ocp.use_pdata) {
+ /* not defined in CODEC specification */
+ if (pdata->ocp.hph_ocp_limit == 1 ||
+ pdata->ocp.hph_ocp_limit == 5) {
+ rc = -EINVAL;
+ goto done;
+ }
+ snd_soc_update_bits(codec, SITAR_A_RX_COM_OCP_CTL,
+ 0x0F, pdata->ocp.num_attempts);
+ snd_soc_write(codec, SITAR_A_RX_COM_OCP_COUNT,
+ ((pdata->ocp.run_time << 4) | pdata->ocp.wait_time));
+ snd_soc_update_bits(codec, SITAR_A_RX_HPH_OCP_CTL,
+ 0xE0, (pdata->ocp.hph_ocp_limit << 5));
+ }
+done:
+ return rc;
+}
+
+static const struct sitar_reg_mask_val sitar_1_1_reg_defaults[] = {
+
+ SITAR_REG_VAL(SITAR_A_MICB_1_INT_RBIAS, 0x24),
+ SITAR_REG_VAL(SITAR_A_MICB_2_INT_RBIAS, 0x24),
+
+ SITAR_REG_VAL(SITAR_A_RX_HPH_BIAS_PA, 0x57),
+ SITAR_REG_VAL(SITAR_A_RX_HPH_BIAS_LDO, 0x56),
+
+ SITAR_REG_VAL(SITAR_A_RX_EAR_BIAS_PA, 0xA6),
+ SITAR_REG_VAL(SITAR_A_RX_EAR_GAIN, 0x02),
+ SITAR_REG_VAL(SITAR_A_RX_EAR_VCM, 0x03),
+
+ SITAR_REG_VAL(SITAR_A_CDC_RX1_B5_CTL, 0x78),
+ SITAR_REG_VAL(SITAR_A_CDC_RX2_B5_CTL, 0x78),
+ SITAR_REG_VAL(SITAR_A_CDC_RX3_B5_CTL, 0x78),
+
+ SITAR_REG_VAL(SITAR_A_CDC_RX1_B6_CTL, 0x80),
+
+ SITAR_REG_VAL(SITAR_A_CDC_CLSG_FREQ_THRESH_B3_CTL, 0x1B),
+
+};
+
+static void sitar_update_reg_defaults(struct snd_soc_codec *codec)
+{
+ u32 i;
+ for (i = 0; i < ARRAY_SIZE(sitar_1_1_reg_defaults); i++)
+ snd_soc_write(codec, sitar_1_1_reg_defaults[i].reg,
+ sitar_1_1_reg_defaults[i].val);
+
+}
+static const struct sitar_reg_mask_val sitar_codec_reg_init_val[] = {
+ /* Initialize current threshold to 350MA
+ * number of wait and run cycles to 4096
+ */
+ {SITAR_A_RX_HPH_OCP_CTL, 0xF8, 0x60},
+ {SITAR_A_RX_COM_OCP_COUNT, 0xFF, 0xFF},
+
+ {SITAR_A_QFUSE_CTL, 0xFF, 0x03},
+
+ /* Initialize gain registers to use register gain */
+ {SITAR_A_RX_HPH_L_GAIN, 0x10, 0x10},
+ {SITAR_A_RX_HPH_R_GAIN, 0x10, 0x10},
+ {SITAR_A_RX_LINE_1_GAIN, 0x10, 0x10},
+ {SITAR_A_RX_LINE_2_GAIN, 0x10, 0x10},
+
+ /* Initialize mic biases to differential mode */
+ {SITAR_A_MICB_1_INT_RBIAS, 0x24, 0x24},
+ {SITAR_A_MICB_2_INT_RBIAS, 0x24, 0x24},
+
+ {SITAR_A_CDC_CONN_CLSG_CTL, 0x3C, 0x14},
+
+ /* Use 16 bit sample size for TX1 to TX6 */
+ {SITAR_A_CDC_CONN_TX_SB_B1_CTL, 0x30, 0x20},
+ {SITAR_A_CDC_CONN_TX_SB_B2_CTL, 0x30, 0x20},
+ {SITAR_A_CDC_CONN_TX_SB_B3_CTL, 0x30, 0x20},
+ {SITAR_A_CDC_CONN_TX_SB_B4_CTL, 0x30, 0x20},
+ {SITAR_A_CDC_CONN_TX_SB_B5_CTL, 0x30, 0x20},
+ {SITAR_A_CDC_CLK_TX_CLK_EN_B1_CTL, 0x1, 0x1},
+
+ /* Use 16 bit sample size for RX */
+ {SITAR_A_CDC_CONN_RX_SB_B1_CTL, 0xFF, 0xAA},
+ {SITAR_A_CDC_CONN_RX_SB_B2_CTL, 0x02, 0x02},
+
+ /*enable HPF filter for TX paths */
+ {SITAR_A_CDC_TX1_MUX_CTL, 0x8, 0x0},
+ {SITAR_A_CDC_TX2_MUX_CTL, 0x8, 0x0},
+
+ /*enable External clock select*/
+ {SITAR_A_CDC_CLK_MCLK_CTL, 0x01, 0x01},
+};
+
+static void sitar_codec_init_reg(struct snd_soc_codec *codec)
+{
+ u32 i;
+ for (i = 0; i < ARRAY_SIZE(sitar_codec_reg_init_val); i++)
+ snd_soc_update_bits(codec, sitar_codec_reg_init_val[i].reg,
+ sitar_codec_reg_init_val[i].mask,
+ sitar_codec_reg_init_val[i].val);
+}
+
+static int sitar_codec_probe(struct snd_soc_codec *codec)
+{
+ struct sitar *control;
+ struct sitar_priv *sitar;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret = 0;
+ int i;
+ u8 sitar_version;
+ int ch_cnt;
+
+ codec->control_data = dev_get_drvdata(codec->dev->parent);
+ control = codec->control_data;
+
+ sitar = kzalloc(sizeof(struct sitar_priv), GFP_KERNEL);
+ if (!sitar) {
+ dev_err(codec->dev, "Failed to allocate private data\n");
+ return -ENOMEM;
+ }
+
+ /* Make sure mbhc micbias register addresses are zeroed out */
+ memset(&sitar->mbhc_bias_regs, 0,
+ sizeof(struct mbhc_micbias_regs));
+ sitar->cfilt_k_value = 0;
+ sitar->mbhc_micbias_switched = false;
+
+ /* Make sure mbhc intenal calibration data is zeroed out */
+ memset(&sitar->mbhc_data, 0,
+ sizeof(struct mbhc_internal_cal_data));
+ sitar->mbhc_data.t_sta_dce = DEFAULT_DCE_STA_WAIT;
+ sitar->mbhc_data.t_dce = DEFAULT_DCE_WAIT;
+ sitar->mbhc_data.t_sta = DEFAULT_STA_WAIT;
+ snd_soc_codec_set_drvdata(codec, sitar);
+
+ sitar->mclk_enabled = false;
+ sitar->bandgap_type = SITAR_BANDGAP_OFF;
+ sitar->clock_active = false;
+ sitar->config_mode_active = false;
+ sitar->mbhc_polling_active = false;
+ sitar->no_mic_headset_override = false;
+ mutex_init(&sitar->codec_resource_lock);
+ sitar->codec = codec;
+ sitar->mbhc_state = MBHC_STATE_NONE;
+ sitar->mbhc_last_resume = 0;
+ sitar->pdata = dev_get_platdata(codec->dev->parent);
+ sitar_update_reg_defaults(codec);
+ sitar_codec_init_reg(codec);
+
+ ret = sitar_handle_pdata(sitar);
+ if (IS_ERR_VALUE(ret)) {
+ pr_err("%s: bad pdata\n", __func__);
+ goto err_pdata;
+ }
+
+ snd_soc_add_codec_controls(codec, sitar_snd_controls,
+ ARRAY_SIZE(sitar_snd_controls));
+ snd_soc_dapm_new_controls(dapm, sitar_dapm_widgets,
+ ARRAY_SIZE(sitar_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+ sitar_version = snd_soc_read(codec, WCD9XXX_A_CHIP_VERSION);
+ pr_info("%s : Sitar version reg 0x%2x\n", __func__, (u32)sitar_version);
+
+ sitar_version &= 0x1F;
+ pr_info("%s : Sitar version %u\n", __func__, (u32)sitar_version);
+
+ snd_soc_dapm_sync(dapm);
+
+
+ ret = wcd9xxx_request_irq(codec->control_data, SITAR_IRQ_MBHC_INSERTION,
+ sitar_hs_insert_irq, "Headset insert detect", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_MBHC_INSERTION);
+ goto err_insert_irq;
+ }
+ wcd9xxx_disable_irq(codec->control_data, SITAR_IRQ_MBHC_INSERTION);
+
+ ret = wcd9xxx_request_irq(codec->control_data, SITAR_IRQ_MBHC_REMOVAL,
+ sitar_hs_remove_irq, "Headset remove detect", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_MBHC_REMOVAL);
+ goto err_remove_irq;
+ }
+
+ ret = wcd9xxx_request_irq(codec->control_data, SITAR_IRQ_MBHC_POTENTIAL,
+ sitar_dce_handler, "DC Estimation detect", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_MBHC_POTENTIAL);
+ goto err_potential_irq;
+ }
+
+ ret = wcd9xxx_request_irq(codec->control_data, SITAR_IRQ_MBHC_RELEASE,
+ sitar_release_handler, "Button Release detect", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_MBHC_RELEASE);
+ goto err_release_irq;
+ }
+
+ ret = wcd9xxx_request_irq(codec->control_data, SITAR_IRQ_SLIMBUS,
+ sitar_slimbus_irq, "SLIMBUS Slave", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_SLIMBUS);
+ goto err_slimbus_irq;
+ }
+
+ for (i = 0; i < WCD9XXX_SLIM_NUM_PORT_REG; i++)
+ wcd9xxx_interface_reg_write(codec->control_data,
+ SITAR_SLIM_PGD_PORT_INT_EN0 + i, 0xFF);
+
+
+ ret = wcd9xxx_request_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPL_FAULT, sitar_hphl_ocp_irq,
+ "HPH_L OCP detect", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_HPH_PA_OCPL_FAULT);
+ goto err_hphl_ocp_irq;
+ }
+ wcd9xxx_disable_irq(codec->control_data, SITAR_IRQ_HPH_PA_OCPL_FAULT);
+
+ ret = wcd9xxx_request_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPR_FAULT, sitar_hphr_ocp_irq,
+ "HPH_R OCP detect", sitar);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ SITAR_IRQ_HPH_PA_OCPR_FAULT);
+ goto err_hphr_ocp_irq;
+ }
+ wcd9xxx_disable_irq(codec->control_data, SITAR_IRQ_HPH_PA_OCPR_FAULT);
+
+ for (i = 0; i < ARRAY_SIZE(sitar_dai); i++) {
+ switch (sitar_dai[i].id) {
+ case AIF1_PB:
+ ch_cnt = sitar_dai[i].playback.channels_max;
+ break;
+ case AIF1_CAP:
+ ch_cnt = sitar_dai[i].capture.channels_max;
+ break;
+ default:
+ continue;
+ }
+ sitar->dai[i].ch_num = kzalloc((sizeof(unsigned int)*
+ ch_cnt), GFP_KERNEL);
+ }
+
+ codec->ignore_pmdown_time = 1;
+
+#ifdef CONFIG_DEBUG_FS
+ debug_sitar_priv = sitar;
+#endif
+
+ return ret;
+
+err_hphr_ocp_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ SITAR_IRQ_HPH_PA_OCPL_FAULT, sitar);
+err_hphl_ocp_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ SITAR_IRQ_SLIMBUS, sitar);
+err_slimbus_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ SITAR_IRQ_MBHC_RELEASE, sitar);
+err_release_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ SITAR_IRQ_MBHC_POTENTIAL, sitar);
+err_potential_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ SITAR_IRQ_MBHC_REMOVAL, sitar);
+err_remove_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ SITAR_IRQ_MBHC_INSERTION, sitar);
+err_insert_irq:
+err_pdata:
+ mutex_destroy(&sitar->codec_resource_lock);
+ kfree(sitar);
+ return ret;
+}
+static int sitar_codec_remove(struct snd_soc_codec *codec)
+{
+ int i;
+ struct sitar_priv *sitar = snd_soc_codec_get_drvdata(codec);
+ wcd9xxx_free_irq(codec->control_data, SITAR_IRQ_SLIMBUS, sitar);
+ wcd9xxx_free_irq(codec->control_data, SITAR_IRQ_MBHC_RELEASE, sitar);
+ wcd9xxx_free_irq(codec->control_data, SITAR_IRQ_MBHC_POTENTIAL, sitar);
+ wcd9xxx_free_irq(codec->control_data, SITAR_IRQ_MBHC_REMOVAL, sitar);
+ wcd9xxx_free_irq(codec->control_data, SITAR_IRQ_MBHC_INSERTION, sitar);
+ SITAR_ACQUIRE_LOCK(sitar->codec_resource_lock);
+ sitar_codec_disable_clock_block(codec);
+ SITAR_RELEASE_LOCK(sitar->codec_resource_lock);
+ sitar_codec_enable_bandgap(codec, SITAR_BANDGAP_OFF);
+ if (sitar->mbhc_fw)
+ release_firmware(sitar->mbhc_fw);
+ for (i = 0; i < ARRAY_SIZE(sitar_dai); i++)
+ kfree(sitar->dai[i].ch_num);
+ mutex_destroy(&sitar->codec_resource_lock);
+ kfree(sitar);
+ return 0;
+}
+static struct snd_soc_codec_driver soc_codec_dev_sitar = {
+ .probe = sitar_codec_probe,
+ .remove = sitar_codec_remove,
+ .read = sitar_read,
+ .write = sitar_write,
+
+ .readable_register = sitar_readable,
+ .volatile_register = sitar_volatile,
+
+ .reg_cache_size = SITAR_CACHE_SIZE,
+ .reg_cache_default = sitar_reg_defaults,
+ .reg_word_size = 1,
+};
+
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_poke;
+
+static int codec_debug_open(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
+static ssize_t codec_debug_write(struct file *filp,
+ const char __user *ubuf, size_t cnt, loff_t *ppos)
+{
+ char lbuf[32];
+ char *buf;
+ int rc;
+
+ if (cnt > sizeof(lbuf) - 1)
+ return -EINVAL;
+
+ rc = copy_from_user(lbuf, ubuf, cnt);
+ if (rc)
+ return -EFAULT;
+
+ lbuf[cnt] = '\0';
+ buf = (char *)lbuf;
+ debug_sitar_priv->no_mic_headset_override = (*strsep(&buf, " ") == '0')
+ ? false : true;
+
+ return rc;
+}
+
+static const struct file_operations codec_debug_ops = {
+ .open = codec_debug_open,
+ .write = codec_debug_write,
+};
+#endif
+
+#ifdef CONFIG_PM
+static int sitar_suspend(struct device *dev)
+{
+ dev_dbg(dev, "%s: system suspend\n", __func__);
+ return 0;
+}
+
+static int sitar_resume(struct device *dev)
+{
+ struct platform_device *pdev = to_platform_device(dev);
+ struct sitar_priv *sitar = platform_get_drvdata(pdev);
+ dev_dbg(dev, "%s: system resume\n", __func__);
+ sitar->mbhc_last_resume = jiffies;
+ return 0;
+}
+
+static const struct dev_pm_ops sitar_pm_ops = {
+ .suspend = sitar_suspend,
+ .resume = sitar_resume,
+};
+#endif
+
+static int __devinit sitar_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+ pr_err("%s\n", __func__);
+#ifdef CONFIG_DEBUG_FS
+ debugfs_poke = debugfs_create_file("TRRS",
+ S_IFREG | S_IRUGO, NULL, (void *) "TRRS", &codec_debug_ops);
+
+#endif
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_sitar,
+ sitar_dai, ARRAY_SIZE(sitar_dai));
+ return ret;
+}
+static int __devexit sitar_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+
+#ifdef CONFIG_DEBUG_FS
+ debugfs_remove(debugfs_poke);
+#endif
+ return 0;
+}
+static struct platform_driver sitar_codec_driver = {
+ .probe = sitar_probe,
+ .remove = sitar_remove,
+ .driver = {
+ .name = "sitar_codec",
+ .owner = THIS_MODULE,
+#ifdef CONFIG_PM
+ .pm = &sitar_pm_ops,
+#endif
+ },
+};
+
+static int __init sitar_codec_init(void)
+{
+ return platform_driver_register(&sitar_codec_driver);
+}
+
+static void __exit sitar_codec_exit(void)
+{
+ platform_driver_unregister(&sitar_codec_driver);
+}
+
+module_init(sitar_codec_init);
+module_exit(sitar_codec_exit);
+
+MODULE_DESCRIPTION("Sitar codec driver");
+MODULE_VERSION("1.0");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wcd9304.h b/sound/soc/codecs/wcd9304.h
new file mode 100644
index 0000000..70b3f0b
--- /dev/null
+++ b/sound/soc/codecs/wcd9304.h
@@ -0,0 +1,251 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <sound/soc.h>
+#include <linux/mfd/wcd9xxx/wcd9xxx-slimslave.h>
+
+#define SITAR_NUM_REGISTERS 0x400
+#define SITAR_MAX_REGISTER (SITAR_NUM_REGISTERS-1)
+#define SITAR_CACHE_SIZE SITAR_NUM_REGISTERS
+#define SITAR_1_X_ONLY_REGISTERS 3
+#define SITAR_2_HIGHER_ONLY_REGISTERS 3
+
+#define SITAR_REG_VAL(reg, val) {reg, 0, val}
+
+#define DEFAULT_DCE_STA_WAIT 55
+#define DEFAULT_DCE_WAIT 60000
+#define DEFAULT_STA_WAIT 5000
+
+#define STA 0
+#define DCE 1
+
+#define SITAR_JACK_BUTTON_MASK (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 | \
+ SND_JACK_BTN_4 | SND_JACK_BTN_5 | \
+ SND_JACK_BTN_6 | SND_JACK_BTN_7)
+
+extern const u8 sitar_reg_readable[SITAR_CACHE_SIZE];
+extern const u32 sitar_1_reg_readable[SITAR_1_X_ONLY_REGISTERS];
+extern const u32 sitar_2_reg_readable[SITAR_2_HIGHER_ONLY_REGISTERS];
+extern const u8 sitar_reg_defaults[SITAR_CACHE_SIZE];
+
+enum sitar_micbias_num {
+ SITAR_MICBIAS1,
+ SITAR_MICBIAS2,
+ SITAR_MICBIAS3,
+ SITAR_MICBIAS4,
+};
+
+enum sitar_pid_current {
+ SITAR_PID_MIC_2P5_UA,
+ SITAR_PID_MIC_5_UA,
+ SITAR_PID_MIC_10_UA,
+ SITAR_PID_MIC_20_UA,
+};
+
+struct sitar_reg_mask_val {
+ u16 reg;
+ u8 mask;
+ u8 val;
+};
+
+enum sitar_mbhc_clk_freq {
+ SITAR_MCLK_12P2MHZ = 0,
+ SITAR_MCLK_9P6MHZ,
+ SITAR_NUM_CLK_FREQS,
+};
+
+enum sitar_mbhc_analog_pwr_cfg {
+ SITAR_ANALOG_PWR_COLLAPSED = 0,
+ SITAR_ANALOG_PWR_ON,
+ SITAR_NUM_ANALOG_PWR_CONFIGS,
+};
+
+enum sitar_mbhc_btn_det_mem {
+ SITAR_BTN_DET_V_BTN_LOW,
+ SITAR_BTN_DET_V_BTN_HIGH,
+ SITAR_BTN_DET_N_READY,
+ SITAR_BTN_DET_N_CIC,
+ SITAR_BTN_DET_GAIN
+};
+
+struct sitar_mbhc_general_cfg {
+ u8 t_ldoh;
+ u8 t_bg_fast_settle;
+ u8 t_shutdown_plug_rem;
+ u8 mbhc_nsa;
+ u8 mbhc_navg;
+ u8 v_micbias_l;
+ u8 v_micbias;
+ u8 mbhc_reserved;
+ u16 settle_wait;
+ u16 t_micbias_rampup;
+ u16 t_micbias_rampdown;
+ u16 t_supply_bringup;
+} __packed;
+
+struct sitar_mbhc_plug_detect_cfg {
+ u32 mic_current;
+ u32 hph_current;
+ u16 t_mic_pid;
+ u16 t_ins_complete;
+ u16 t_ins_retry;
+ u16 v_removal_delta;
+ u8 micbias_slow_ramp;
+ u8 reserved0;
+ u8 reserved1;
+ u8 reserved2;
+} __packed;
+
+struct sitar_mbhc_plug_type_cfg {
+ u8 av_detect;
+ u8 mono_detect;
+ u8 num_ins_tries;
+ u8 reserved0;
+ s16 v_no_mic;
+ s16 v_av_min;
+ s16 v_av_max;
+ s16 v_hs_min;
+ s16 v_hs_max;
+ u16 reserved1;
+} __packed;
+
+
+struct sitar_mbhc_btn_detect_cfg {
+ s8 c[8];
+ u8 nc;
+ u8 n_meas;
+ u8 mbhc_nsc;
+ u8 n_btn_meas;
+ u8 n_btn_con;
+ u8 num_btn;
+ u8 reserved0;
+ u8 reserved1;
+ u16 t_poll;
+ u16 t_bounce_wait;
+ u16 t_rel_timeout;
+ s16 v_btn_press_delta_sta;
+ s16 v_btn_press_delta_cic;
+ u16 t_btn0_timeout;
+ s16 _v_btn_low[0]; /* v_btn_low[num_btn] */
+ s16 _v_btn_high[0]; /* v_btn_high[num_btn] */
+ u8 _n_ready[SITAR_NUM_CLK_FREQS];
+ u8 _n_cic[SITAR_NUM_CLK_FREQS];
+ u8 _gain[SITAR_NUM_CLK_FREQS];
+} __packed;
+
+struct sitar_mbhc_imped_detect_cfg {
+ u8 _hs_imped_detect;
+ u8 _n_rload;
+ u8 _hph_keep_on;
+ u8 _repeat_rload_calc;
+ u16 _t_dac_ramp_time;
+ u16 _rhph_high;
+ u16 _rhph_low;
+ u16 _rload[0]; /* rload[n_rload] */
+ u16 _alpha[0]; /* alpha[n_rload] */
+ u16 _beta[3];
+} __packed;
+
+struct sitar_mbhc_config {
+ struct snd_soc_jack *headset_jack;
+ struct snd_soc_jack *button_jack;
+ bool read_fw_bin;
+ /* void* calibration contains:
+ * struct tabla_mbhc_general_cfg generic;
+ * struct tabla_mbhc_plug_detect_cfg plug_det;
+ * struct tabla_mbhc_plug_type_cfg plug_type;
+ * struct tabla_mbhc_btn_detect_cfg btn_det;
+ * struct tabla_mbhc_imped_detect_cfg imped_det;
+ * Note: various size depends on btn_det->num_btn
+ */
+ void *calibration;
+ enum sitar_micbias_num micbias;
+ int (*mclk_cb_fn) (struct snd_soc_codec*, int, bool);
+ unsigned int mclk_rate;
+ unsigned int gpio;
+ unsigned int gpio_irq;
+ int gpio_level_insert;
+};
+
+extern int sitar_hs_detect(struct snd_soc_codec *codec,
+ const struct sitar_mbhc_config *cfg);
+
+#ifndef anc_header_dec
+struct anc_header {
+ u32 reserved[3];
+ u32 num_anc_slots;
+};
+#define anc_header_dec
+#endif
+
+extern int sitar_mclk_enable(struct snd_soc_codec *codec, int mclk_enable,
+ bool dapm);
+
+extern void *sitar_mbhc_cal_btn_det_mp(const struct sitar_mbhc_btn_detect_cfg
+ *btn_det,
+ const enum sitar_mbhc_btn_det_mem mem);
+
+#define SITAR_MBHC_CAL_SIZE(buttons, rload) ( \
+ sizeof(enum sitar_micbias_num) + \
+ sizeof(struct sitar_mbhc_general_cfg) + \
+ sizeof(struct sitar_mbhc_plug_detect_cfg) + \
+ ((sizeof(s16) + sizeof(s16)) * buttons) + \
+ sizeof(struct sitar_mbhc_plug_type_cfg) + \
+ sizeof(struct sitar_mbhc_btn_detect_cfg) + \
+ sizeof(struct sitar_mbhc_imped_detect_cfg) + \
+ ((sizeof(u16) + sizeof(u16)) * rload) \
+ )
+
+#define SITAR_MBHC_CAL_GENERAL_PTR(cali) ( \
+ (struct sitar_mbhc_general_cfg *) cali)
+#define SITAR_MBHC_CAL_PLUG_DET_PTR(cali) ( \
+ (struct sitar_mbhc_plug_detect_cfg *) \
+ &(SITAR_MBHC_CAL_GENERAL_PTR(cali)[1]))
+#define SITAR_MBHC_CAL_PLUG_TYPE_PTR(cali) ( \
+ (struct sitar_mbhc_plug_type_cfg *) \
+ &(SITAR_MBHC_CAL_PLUG_DET_PTR(cali)[1]))
+#define SITAR_MBHC_CAL_BTN_DET_PTR(cali) ( \
+ (struct sitar_mbhc_btn_detect_cfg *) \
+ &(SITAR_MBHC_CAL_PLUG_TYPE_PTR(cali)[1]))
+#define SITAR_MBHC_CAL_IMPED_DET_PTR(cali) ( \
+ (struct sitar_mbhc_imped_detect_cfg *) \
+ (((void *)&SITAR_MBHC_CAL_BTN_DET_PTR(cali)[1]) + \
+ (SITAR_MBHC_CAL_BTN_DET_PTR(cali)->num_btn * \
+ (sizeof(SITAR_MBHC_CAL_BTN_DET_PTR(cali)->_v_btn_low[0]) + \
+ sizeof(SITAR_MBHC_CAL_BTN_DET_PTR(cali)->_v_btn_high[0])))) \
+ )
+
+/* minimum size of calibration data assuming there is only one button and
+ * one rload.
+ */
+#define SITAR_MBHC_CAL_MIN_SIZE ( \
+ sizeof(struct sitar_mbhc_general_cfg) + \
+ sizeof(struct sitar_mbhc_plug_detect_cfg) + \
+ sizeof(struct sitar_mbhc_plug_type_cfg) + \
+ sizeof(struct sitar_mbhc_btn_detect_cfg) + \
+ sizeof(struct sitar_mbhc_imped_detect_cfg) + \
+ (sizeof(u16) * 2))
+
+#define SITAR_MBHC_CAL_BTN_SZ(cfg_ptr) ( \
+ sizeof(struct sitar_mbhc_btn_detect_cfg) + \
+ (cfg_ptr->num_btn * (sizeof(cfg_ptr->_v_btn_low[0]) + \
+ sizeof(cfg_ptr->_v_btn_high[0]))))
+
+#define SITAR_MBHC_CAL_IMPED_MIN_SZ ( \
+ sizeof(struct sitar_mbhc_imped_detect_cfg) + \
+ sizeof(u16) * 2)
+
+#define SITAR_MBHC_CAL_IMPED_SZ(cfg_ptr) ( \
+ sizeof(struct sitar_mbhc_imped_detect_cfg) + \
+ (cfg_ptr->_n_rload * (sizeof(cfg_ptr->_rload[0]) + \
+ sizeof(cfg_ptr->_alpha[0]))))
diff --git a/sound/soc/codecs/wcd9310-tables.c b/sound/soc/codecs/wcd9310-tables.c
new file mode 100644
index 0000000..2cba59d
--- /dev/null
+++ b/sound/soc/codecs/wcd9310-tables.c
@@ -0,0 +1,1114 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/mfd/wcd9xxx/wcd9310_registers.h>
+#include "wcd9310.h"
+
+const u8 tabla_reg_readable[TABLA_CACHE_SIZE] = {
+ [TABLA_A_CHIP_CTL] = 1,
+ [TABLA_A_CHIP_STATUS] = 1,
+ [TABLA_A_CHIP_ID_BYTE_0] = 1,
+ [TABLA_A_CHIP_ID_BYTE_1] = 1,
+ [TABLA_A_CHIP_ID_BYTE_2] = 1,
+ [TABLA_A_CHIP_ID_BYTE_3] = 1,
+ [TABLA_A_CHIP_VERSION] = 1,
+ [TABLA_A_SB_VERSION] = 1,
+ [TABLA_A_SLAVE_ID_1] = 1,
+ [TABLA_A_SLAVE_ID_2] = 1,
+ [TABLA_A_SLAVE_ID_3] = 1,
+ [TABLA_A_PIN_CTL_OE0] = 1,
+ [TABLA_A_PIN_CTL_OE1] = 1,
+ [TABLA_A_PIN_CTL_DATA0] = 1,
+ [TABLA_A_PIN_CTL_DATA1] = 1,
+ [TABLA_A_HDRIVE_GENERIC] = 1,
+ [TABLA_A_HDRIVE_OVERRIDE] = 1,
+ [TABLA_A_ANA_CSR_WAIT_STATE] = 1,
+ [TABLA_A_PROCESS_MONITOR_CTL0] = 1,
+ [TABLA_A_PROCESS_MONITOR_CTL1] = 1,
+ [TABLA_A_PROCESS_MONITOR_CTL2] = 1,
+ [TABLA_A_PROCESS_MONITOR_CTL3] = 1,
+ [TABLA_A_QFUSE_CTL] = 1,
+ [TABLA_A_QFUSE_STATUS] = 1,
+ [TABLA_A_QFUSE_DATA_OUT0] = 1,
+ [TABLA_A_QFUSE_DATA_OUT1] = 1,
+ [TABLA_A_QFUSE_DATA_OUT2] = 1,
+ [TABLA_A_QFUSE_DATA_OUT3] = 1,
+ [TABLA_A_CDC_CTL] = 1,
+ [TABLA_A_LEAKAGE_CTL] = 1,
+ [TABLA_A_INTR_MODE] = 1,
+ [TABLA_A_INTR_MASK0] = 1,
+ [TABLA_A_INTR_MASK1] = 1,
+ [TABLA_A_INTR_MASK2] = 1,
+ [TABLA_A_INTR_STATUS0] = 1,
+ [TABLA_A_INTR_STATUS1] = 1,
+ [TABLA_A_INTR_STATUS2] = 1,
+ [TABLA_A_INTR_CLEAR0] = 0,
+ [TABLA_A_INTR_CLEAR1] = 0,
+ [TABLA_A_INTR_CLEAR2] = 0,
+ [TABLA_A_INTR_LEVEL0] = 1,
+ [TABLA_A_INTR_LEVEL1] = 1,
+ [TABLA_A_INTR_LEVEL2] = 1,
+ [TABLA_A_INTR_TEST0] = 1,
+ [TABLA_A_INTR_TEST1] = 1,
+ [TABLA_A_INTR_TEST2] = 1,
+ [TABLA_A_INTR_SET0] = 1,
+ [TABLA_A_INTR_SET1] = 1,
+ [TABLA_A_INTR_SET2] = 1,
+ [TABLA_A_CDC_TX_I2S_SCK_MODE] = 1,
+ [TABLA_A_CDC_TX_I2S_WS_MODE] = 1,
+ [TABLA_A_CDC_DMIC_DATA0_MODE] = 1,
+ [TABLA_A_CDC_DMIC_CLK0_MODE] = 1,
+ [TABLA_A_CDC_DMIC_DATA1_MODE] = 1,
+ [TABLA_A_CDC_DMIC_CLK1_MODE] = 1,
+ [TABLA_A_CDC_RX_I2S_SCK_MODE] = 1,
+ [TABLA_A_CDC_RX_I2S_WS_MODE] = 1,
+ [TABLA_A_CDC_DMIC_DATA2_MODE] = 1,
+ [TABLA_A_CDC_DMIC_CLK2_MODE] = 1,
+ [TABLA_A_CDC_INTR_MODE] = 1,
+ [TABLA_A_BIAS_REF_CTL] = 1,
+ [TABLA_A_BIAS_CENTRAL_BG_CTL] = 1,
+ [TABLA_A_BIAS_PRECHRG_CTL] = 1,
+ [TABLA_A_BIAS_CURR_CTL_1] = 1,
+ [TABLA_A_BIAS_CURR_CTL_2] = 1,
+ [TABLA_A_BIAS_CONFIG_MODE_BG_CTL] = 1,
+ [TABLA_A_BIAS_BG_STATUS] = 1,
+ [TABLA_A_CLK_BUFF_EN1] = 1,
+ [TABLA_A_CLK_BUFF_EN2] = 1,
+ [TABLA_A_LDO_H_MODE_1] = 1,
+ [TABLA_A_LDO_H_MODE_2] = 1,
+ [TABLA_A_LDO_H_LOOP_CTL] = 1,
+ [TABLA_A_LDO_H_COMP_1] = 1,
+ [TABLA_A_LDO_H_COMP_2] = 1,
+ [TABLA_A_LDO_H_BIAS_1] = 1,
+ [TABLA_A_LDO_H_BIAS_2] = 1,
+ [TABLA_A_LDO_H_BIAS_3] = 1,
+ [TABLA_A_LDO_L_MODE_1] = 1,
+ [TABLA_A_LDO_L_MODE_2] = 1,
+ [TABLA_A_LDO_L_LOOP_CTL] = 1,
+ [TABLA_A_LDO_L_COMP_1] = 1,
+ [TABLA_A_LDO_L_COMP_2] = 1,
+ [TABLA_A_LDO_L_BIAS_1] = 1,
+ [TABLA_A_LDO_L_BIAS_2] = 1,
+ [TABLA_A_LDO_L_BIAS_3] = 1,
+ [TABLA_A_MICB_CFILT_1_CTL] = 1,
+ [TABLA_A_MICB_CFILT_1_VAL] = 1,
+ [TABLA_A_MICB_CFILT_1_PRECHRG] = 1,
+ [TABLA_A_MICB_1_CTL] = 1,
+ [TABLA_A_MICB_1_INT_RBIAS] = 1,
+ [TABLA_A_MICB_1_MBHC] = 1,
+ [TABLA_A_MICB_CFILT_2_CTL] = 1,
+ [TABLA_A_MICB_CFILT_2_VAL] = 1,
+ [TABLA_A_MICB_CFILT_2_PRECHRG] = 1,
+ [TABLA_A_MICB_2_CTL] = 1,
+ [TABLA_A_MICB_2_INT_RBIAS] = 1,
+ [TABLA_A_MICB_2_MBHC] = 1,
+ [TABLA_A_MICB_CFILT_3_CTL] = 1,
+ [TABLA_A_MICB_CFILT_3_VAL] = 1,
+ [TABLA_A_MICB_CFILT_3_PRECHRG] = 1,
+ [TABLA_A_MICB_3_CTL] = 1,
+ [TABLA_A_MICB_3_INT_RBIAS] = 1,
+ [TABLA_A_MICB_3_MBHC] = 1,
+ [TABLA_A_TX_COM_BIAS] = 1,
+ [TABLA_A_MBHC_SCALING_MUX_1] = 1,
+ [TABLA_A_MBHC_SCALING_MUX_2] = 1,
+ [TABLA_A_TX_SUP_SWITCH_CTRL_1] = 1,
+ [TABLA_A_TX_SUP_SWITCH_CTRL_2] = 1,
+ [TABLA_A_TX_1_2_EN] = 1,
+ [TABLA_A_TX_1_2_TEST_EN] = 1,
+ [TABLA_A_TX_1_2_ADC_CH1] = 1,
+ [TABLA_A_TX_1_2_ADC_CH2] = 1,
+ [TABLA_A_TX_1_2_ATEST_REFCTRL] = 1,
+ [TABLA_A_TX_1_2_TEST_CTL] = 1,
+ [TABLA_A_TX_1_2_TEST_BLOCK_EN] = 1,
+ [TABLA_A_TX_1_2_TXFE_CLKDIV] = 1,
+ [TABLA_A_TX_1_2_SAR_ERR_CH1] = 1,
+ [TABLA_A_TX_1_2_SAR_ERR_CH2] = 1,
+ [TABLA_A_TX_3_4_EN] = 1,
+ [TABLA_A_TX_3_4_TEST_EN] = 1,
+ [TABLA_A_TX_3_4_ADC_CH3] = 1,
+ [TABLA_A_TX_3_4_ADC_CH4] = 1,
+ [TABLA_A_TX_3_4_ATEST_REFCTRL] = 1,
+ [TABLA_A_TX_3_4_TEST_CTL] = 1,
+ [TABLA_A_TX_3_4_TEST_BLOCK_EN] = 1,
+ [TABLA_A_TX_3_4_TXFE_CKDIV] = 1,
+ [TABLA_A_TX_3_4_SAR_ERR_CH3] = 1,
+ [TABLA_A_TX_3_4_SAR_ERR_CH4] = 1,
+ [TABLA_A_TX_5_6_EN] = 1,
+ [TABLA_A_TX_5_6_TEST_EN] = 1,
+ [TABLA_A_TX_5_6_ADC_CH5] = 1,
+ [TABLA_A_TX_5_6_ADC_CH6] = 1,
+ [TABLA_A_TX_5_6_ATEST_REFCTRL] = 1,
+ [TABLA_A_TX_5_6_TEST_CTL] = 1,
+ [TABLA_A_TX_5_6_TEST_BLOCK_EN] = 1,
+ [TABLA_A_TX_5_6_TXFE_CKDIV] = 1,
+ [TABLA_A_TX_5_6_SAR_ERR_CH5] = 1,
+ [TABLA_A_TX_5_6_SAR_ERR_CH6] = 1,
+ [TABLA_A_TX_7_MBHC_EN] = 1,
+ [TABLA_A_TX_7_MBHC_ATEST_REFCTRL] = 1,
+ [TABLA_A_TX_7_MBHC_ADC] = 1,
+ [TABLA_A_TX_7_MBHC_TEST_CTL] = 1,
+ [TABLA_A_TX_7_MBHC_SAR_ERR] = 1,
+ [TABLA_A_TX_7_TXFE_CLKDIV] = 1,
+ [TABLA_A_AUX_COM_CTL] = 1,
+ [TABLA_A_AUX_COM_ATEST] = 1,
+ [TABLA_A_AUX_L_EN] = 1,
+ [TABLA_A_AUX_L_GAIN] = 1,
+ [TABLA_A_AUX_L_PA_CONN] = 1,
+ [TABLA_A_AUX_L_PA_CONN_INV] = 1,
+ [TABLA_A_AUX_R_EN] = 1,
+ [TABLA_A_AUX_R_GAIN] = 1,
+ [TABLA_A_AUX_R_PA_CONN] = 1,
+ [TABLA_A_AUX_R_PA_CONN_INV] = 1,
+ [TABLA_A_CP_EN] = 1,
+ [TABLA_A_CP_CLK] = 1,
+ [TABLA_A_CP_STATIC] = 1,
+ [TABLA_A_CP_DCC1] = 1,
+ [TABLA_A_CP_DCC3] = 1,
+ [TABLA_A_CP_ATEST] = 1,
+ [TABLA_A_CP_DTEST] = 1,
+ [TABLA_A_RX_COM_TIMER_DIV] = 1,
+ [TABLA_A_RX_COM_OCP_CTL] = 1,
+ [TABLA_A_RX_COM_OCP_COUNT] = 1,
+ [TABLA_A_RX_COM_DAC_CTL] = 1,
+ [TABLA_A_RX_COM_BIAS] = 1,
+ [TABLA_A_RX_HPH_BIAS_PA] = 1,
+ [TABLA_A_RX_HPH_BIAS_LDO] = 1,
+ [TABLA_A_RX_HPH_BIAS_CNP] = 1,
+ [TABLA_A_RX_HPH_BIAS_WG] = 1,
+ [TABLA_A_RX_HPH_OCP_CTL] = 1,
+ [TABLA_A_RX_HPH_CNP_EN] = 1,
+ [TABLA_A_RX_HPH_CNP_WG_CTL] = 1,
+ [TABLA_A_RX_HPH_CNP_WG_TIME] = 1,
+ [TABLA_A_RX_HPH_L_GAIN] = 1,
+ [TABLA_A_RX_HPH_L_TEST] = 1,
+ [TABLA_A_RX_HPH_L_PA_CTL] = 1,
+ [TABLA_A_RX_HPH_L_DAC_CTL] = 1,
+ [TABLA_A_RX_HPH_L_ATEST] = 1,
+ [TABLA_A_RX_HPH_L_STATUS] = 1,
+ [TABLA_A_RX_HPH_R_GAIN] = 1,
+ [TABLA_A_RX_HPH_R_TEST] = 1,
+ [TABLA_A_RX_HPH_R_PA_CTL] = 1,
+ [TABLA_A_RX_HPH_R_DAC_CTL] = 1,
+ [TABLA_A_RX_HPH_R_ATEST] = 1,
+ [TABLA_A_RX_HPH_R_STATUS] = 1,
+ [TABLA_A_RX_EAR_BIAS_PA] = 1,
+ [TABLA_A_RX_EAR_BIAS_CMBUFF] = 1,
+ [TABLA_A_RX_EAR_EN] = 1,
+ [TABLA_A_RX_EAR_GAIN] = 1,
+ [TABLA_A_RX_EAR_CMBUFF] = 1,
+ [TABLA_A_RX_EAR_ICTL] = 1,
+ [TABLA_A_RX_EAR_CCOMP] = 1,
+ [TABLA_A_RX_EAR_VCM] = 1,
+ [TABLA_A_RX_EAR_CNP] = 1,
+ [TABLA_A_RX_EAR_ATEST] = 1,
+ [TABLA_A_RX_EAR_STATUS] = 1,
+ [TABLA_A_RX_LINE_BIAS_PA] = 1,
+ [TABLA_A_RX_LINE_BIAS_DAC] = 1,
+ [TABLA_A_RX_LINE_BIAS_CNP] = 1,
+ [TABLA_A_RX_LINE_COM] = 1,
+ [TABLA_A_RX_LINE_CNP_EN] = 1,
+ [TABLA_A_RX_LINE_CNP_WG_CTL] = 1,
+ [TABLA_A_RX_LINE_CNP_WG_TIME] = 1,
+ [TABLA_A_RX_LINE_1_GAIN] = 1,
+ [TABLA_A_RX_LINE_1_TEST] = 1,
+ [TABLA_A_RX_LINE_1_DAC_CTL] = 1,
+ [TABLA_A_RX_LINE_1_STATUS] = 1,
+ [TABLA_A_RX_LINE_2_GAIN] = 1,
+ [TABLA_A_RX_LINE_2_TEST] = 1,
+ [TABLA_A_RX_LINE_2_DAC_CTL] = 1,
+ [TABLA_A_RX_LINE_2_STATUS] = 1,
+ [TABLA_A_RX_LINE_3_GAIN] = 1,
+ [TABLA_A_RX_LINE_3_TEST] = 1,
+ [TABLA_A_RX_LINE_3_DAC_CTL] = 1,
+ [TABLA_A_RX_LINE_3_STATUS] = 1,
+ [TABLA_A_RX_LINE_4_GAIN] = 1,
+ [TABLA_A_RX_LINE_4_TEST] = 1,
+ [TABLA_A_RX_LINE_4_DAC_CTL] = 1,
+ [TABLA_A_RX_LINE_4_STATUS] = 1,
+ [TABLA_A_RX_LINE_5_GAIN] = 1,
+ [TABLA_A_RX_LINE_5_TEST] = 1,
+ [TABLA_A_RX_LINE_5_DAC_CTL] = 1,
+ [TABLA_A_RX_LINE_5_STATUS] = 1,
+ [TABLA_A_RX_LINE_CNP_DBG] = 1,
+ [TABLA_A_MBHC_HPH] = 1,
+ [TABLA_A_CONFIG_MODE_FREQ] = 1,
+ [TABLA_A_CONFIG_MODE_TEST] = 1,
+ [TABLA_A_CONFIG_MODE_STATUS] = 1,
+ [TABLA_A_CONFIG_MODE_TUNER] = 1,
+ [TABLA_A_CDC_ANC1_CTL] = 1,
+ [TABLA_A_CDC_ANC1_SHIFT] = 1,
+ [TABLA_A_CDC_ANC1_FILT1_B1_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT1_B2_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT1_B3_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT1_B4_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT2_B1_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT2_B2_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT2_B3_CTL] = 1,
+ [TABLA_A_CDC_ANC1_SPARE] = 1,
+ [TABLA_A_CDC_ANC1_FILT3_CTL] = 1,
+ [TABLA_A_CDC_ANC1_FILT4_CTL] = 1,
+ [TABLA_A_CDC_TX1_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX2_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX3_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX4_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX5_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX6_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX7_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX8_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX9_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX10_VOL_CTL_TIMER] = 1,
+ [TABLA_A_CDC_TX1_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX2_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX3_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX4_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX5_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX6_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX7_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX8_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX9_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX10_VOL_CTL_GAIN] = 1,
+ [TABLA_A_CDC_TX1_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX2_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX3_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX4_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX5_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX6_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX7_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX8_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX9_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX10_VOL_CTL_CFG] = 1,
+ [TABLA_A_CDC_TX1_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX2_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX3_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX4_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX5_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX6_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX7_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX8_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX9_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX10_MUX_CTL] = 1,
+ [TABLA_A_CDC_TX1_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX2_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX3_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX4_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX5_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX6_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX7_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX8_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX9_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX10_CLK_FS_CTL] = 1,
+ [TABLA_A_CDC_TX1_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX2_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX3_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX4_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX5_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX6_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX7_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX8_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX9_DMIC_CTL] = 1,
+ [TABLA_A_CDC_TX10_DMIC_CTL] = 1,
+ [TABLA_A_CDC_ANC2_CTL] = 1,
+ [TABLA_A_CDC_ANC2_SHIFT] = 1,
+ [TABLA_A_CDC_ANC2_FILT1_B1_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT1_B2_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT1_B3_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT1_B4_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT2_B1_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT2_B2_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT2_B3_CTL] = 1,
+ [TABLA_A_CDC_ANC2_SPARE] = 1,
+ [TABLA_A_CDC_ANC2_FILT3_CTL] = 1,
+ [TABLA_A_CDC_ANC2_FILT4_CTL] = 1,
+ [TABLA_A_CDC_SRC1_PDA_CFG] = 1,
+ [TABLA_A_CDC_SRC2_PDA_CFG] = 1,
+ [TABLA_A_CDC_SRC1_FS_CTL] = 1,
+ [TABLA_A_CDC_SRC2_FS_CTL] = 1,
+ [TABLA_A_CDC_RX1_B1_CTL] = 1,
+ [TABLA_A_CDC_RX2_B1_CTL] = 1,
+ [TABLA_A_CDC_RX3_B1_CTL] = 1,
+ [TABLA_A_CDC_RX4_B1_CTL] = 1,
+ [TABLA_A_CDC_RX5_B1_CTL] = 1,
+ [TABLA_A_CDC_RX6_B1_CTL] = 1,
+ [TABLA_A_CDC_RX7_B1_CTL] = 1,
+ [TABLA_A_CDC_RX1_B2_CTL] = 1,
+ [TABLA_A_CDC_RX2_B2_CTL] = 1,
+ [TABLA_A_CDC_RX3_B2_CTL] = 1,
+ [TABLA_A_CDC_RX4_B2_CTL] = 1,
+ [TABLA_A_CDC_RX5_B2_CTL] = 1,
+ [TABLA_A_CDC_RX6_B2_CTL] = 1,
+ [TABLA_A_CDC_RX7_B2_CTL] = 1,
+ [TABLA_A_CDC_RX1_B3_CTL] = 1,
+ [TABLA_A_CDC_RX2_B3_CTL] = 1,
+ [TABLA_A_CDC_RX3_B3_CTL] = 1,
+ [TABLA_A_CDC_RX4_B3_CTL] = 1,
+ [TABLA_A_CDC_RX5_B3_CTL] = 1,
+ [TABLA_A_CDC_RX6_B3_CTL] = 1,
+ [TABLA_A_CDC_RX7_B3_CTL] = 1,
+ [TABLA_A_CDC_RX1_B4_CTL] = 1,
+ [TABLA_A_CDC_RX2_B4_CTL] = 1,
+ [TABLA_A_CDC_RX3_B4_CTL] = 1,
+ [TABLA_A_CDC_RX4_B4_CTL] = 1,
+ [TABLA_A_CDC_RX5_B4_CTL] = 1,
+ [TABLA_A_CDC_RX6_B4_CTL] = 1,
+ [TABLA_A_CDC_RX7_B4_CTL] = 1,
+ [TABLA_A_CDC_RX1_B5_CTL] = 1,
+ [TABLA_A_CDC_RX2_B5_CTL] = 1,
+ [TABLA_A_CDC_RX3_B5_CTL] = 1,
+ [TABLA_A_CDC_RX4_B5_CTL] = 1,
+ [TABLA_A_CDC_RX5_B5_CTL] = 1,
+ [TABLA_A_CDC_RX6_B5_CTL] = 1,
+ [TABLA_A_CDC_RX7_B5_CTL] = 1,
+ [TABLA_A_CDC_RX1_B6_CTL] = 1,
+ [TABLA_A_CDC_RX2_B6_CTL] = 1,
+ [TABLA_A_CDC_RX3_B6_CTL] = 1,
+ [TABLA_A_CDC_RX4_B6_CTL] = 1,
+ [TABLA_A_CDC_RX5_B6_CTL] = 1,
+ [TABLA_A_CDC_RX6_B6_CTL] = 1,
+ [TABLA_A_CDC_RX7_B6_CTL] = 1,
+ [TABLA_A_CDC_RX1_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX2_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX3_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX4_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX5_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX6_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX7_VOL_CTL_B1_CTL] = 1,
+ [TABLA_A_CDC_RX1_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_RX2_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_RX3_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_RX4_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_RX5_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_RX6_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_RX7_VOL_CTL_B2_CTL] = 1,
+ [TABLA_A_CDC_CLK_ANC_RESET_CTL] = 1,
+ [TABLA_A_CDC_CLK_RX_RESET_CTL] = 1,
+ [TABLA_A_CDC_CLK_TX_RESET_B1_CTL] = 1,
+ [TABLA_A_CDC_CLK_TX_RESET_B2_CTL] = 1,
+ [TABLA_A_CDC_CLK_DMIC_CTL] = 1,
+ [TABLA_A_CDC_CLK_RX_I2S_CTL] = 1,
+ [TABLA_A_CDC_CLK_TX_I2S_CTL] = 1,
+ [TABLA_A_CDC_CLK_OTHR_RESET_CTL] = 1,
+ [TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL] = 1,
+ [TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL] = 1,
+ [TABLA_A_CDC_CLK_OTHR_CTL] = 1,
+ [TABLA_A_CDC_CLK_RDAC_CLK_EN_CTL] = 1,
+ [TABLA_A_CDC_CLK_ANC_CLK_EN_CTL] = 1,
+ [TABLA_A_CDC_CLK_RX_B1_CTL] = 1,
+ [TABLA_A_CDC_CLK_RX_B2_CTL] = 1,
+ [TABLA_A_CDC_CLK_MCLK_CTL] = 1,
+ [TABLA_A_CDC_CLK_PDM_CTL] = 1,
+ [TABLA_A_CDC_CLK_SD_CTL] = 1,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B1_CTL] = 1,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B2_CTL] = 1,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B3_CTL] = 1,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B4_CTL] = 1,
+ [TABLA_A_CDC_CLSG_GAIN_THRESH_CTL] = 1,
+ [TABLA_A_CDC_CLSG_TIMER_B1_CFG] = 1,
+ [TABLA_A_CDC_CLSG_TIMER_B2_CFG] = 1,
+ [TABLA_A_CDC_CLSG_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B1_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B1_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B2_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B2_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B3_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B3_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B4_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B4_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B5_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B5_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B6_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B6_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B7_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B7_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_B8_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_B8_CTL] = 1,
+ [TABLA_A_CDC_IIR1_CTL] = 1,
+ [TABLA_A_CDC_IIR2_CTL] = 1,
+ [TABLA_A_CDC_IIR1_GAIN_TIMER_CTL] = 1,
+ [TABLA_A_CDC_IIR2_GAIN_TIMER_CTL] = 1,
+ [TABLA_A_CDC_IIR1_COEF_B1_CTL] = 1,
+ [TABLA_A_CDC_IIR2_COEF_B1_CTL] = 1,
+ [TABLA_A_CDC_IIR1_COEF_B2_CTL] = 1,
+ [TABLA_A_CDC_IIR2_COEF_B2_CTL] = 1,
+ [TABLA_A_CDC_IIR1_COEF_B3_CTL] = 1,
+ [TABLA_A_CDC_IIR2_COEF_B3_CTL] = 1,
+ [TABLA_A_CDC_IIR1_COEF_B4_CTL] = 1,
+ [TABLA_A_CDC_IIR2_COEF_B4_CTL] = 1,
+ [TABLA_A_CDC_IIR1_COEF_B5_CTL] = 1,
+ [TABLA_A_CDC_IIR2_COEF_B5_CTL] = 1,
+ [TABLA_A_CDC_TOP_GAIN_UPDATE] = 1,
+ [TABLA_A_CDC_DEBUG_B1_CTL] = 1,
+ [TABLA_A_CDC_DEBUG_B2_CTL] = 1,
+ [TABLA_A_CDC_DEBUG_B3_CTL] = 1,
+ [TABLA_A_CDC_DEBUG_B4_CTL] = 1,
+ [TABLA_A_CDC_DEBUG_B5_CTL] = 1,
+ [TABLA_A_CDC_DEBUG_B6_CTL] = 1,
+ [TABLA_A_CDC_COMP1_B1_CTL] = 1,
+ [TABLA_A_CDC_COMP1_B2_CTL] = 1,
+ [TABLA_A_CDC_COMP1_B3_CTL] = 1,
+ [TABLA_A_CDC_COMP1_B4_CTL] = 1,
+ [TABLA_A_CDC_COMP1_B5_CTL] = 1,
+ [TABLA_A_CDC_COMP1_B6_CTL] = 1,
+ [TABLA_A_CDC_COMP1_SHUT_DOWN_STATUS] = 1,
+ [TABLA_A_CDC_COMP1_FS_CFG] = 1,
+ [TABLA_A_CDC_COMP2_B1_CTL] = 1,
+ [TABLA_A_CDC_COMP2_B2_CTL] = 1,
+ [TABLA_A_CDC_COMP2_B3_CTL] = 1,
+ [TABLA_A_CDC_COMP2_B4_CTL] = 1,
+ [TABLA_A_CDC_COMP2_B5_CTL] = 1,
+ [TABLA_A_CDC_COMP2_B6_CTL] = 1,
+ [TABLA_A_CDC_COMP2_SHUT_DOWN_STATUS] = 1,
+ [TABLA_A_CDC_COMP2_FS_CFG] = 1,
+ [TABLA_A_CDC_CONN_RX1_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX1_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX1_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX2_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX2_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX2_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX3_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX3_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX3_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX4_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX4_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX5_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX5_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX6_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX6_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX7_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX7_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_ANC_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_ANC_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_B4_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ1_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ1_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ1_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ1_B4_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ2_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ2_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ2_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_EQ2_B4_CTL] = 1,
+ [TABLA_A_CDC_CONN_SRC1_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_SRC1_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_SRC2_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_SRC2_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B3_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B4_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B5_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B6_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B7_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B8_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B9_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B10_CTL] = 1,
+ [TABLA_A_CDC_CONN_TX_SB_B11_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX_SB_B1_CTL] = 1,
+ [TABLA_A_CDC_CONN_RX_SB_B2_CTL] = 1,
+ [TABLA_A_CDC_CONN_CLSG_CTL] = 1,
+ [TABLA_A_CDC_CONN_SPARE] = 1,
+ [TABLA_A_CDC_MBHC_EN_CTL] = 1,
+ [TABLA_A_CDC_MBHC_FEATURE_B1_CFG] = 1,
+ [TABLA_A_CDC_MBHC_FEATURE_B2_CFG] = 1,
+ [TABLA_A_CDC_MBHC_TIMER_B1_CTL] = 1,
+ [TABLA_A_CDC_MBHC_TIMER_B2_CTL] = 1,
+ [TABLA_A_CDC_MBHC_TIMER_B3_CTL] = 1,
+ [TABLA_A_CDC_MBHC_TIMER_B4_CTL] = 1,
+ [TABLA_A_CDC_MBHC_TIMER_B5_CTL] = 1,
+ [TABLA_A_CDC_MBHC_TIMER_B6_CTL] = 1,
+ [TABLA_A_CDC_MBHC_B1_STATUS] = 1,
+ [TABLA_A_CDC_MBHC_B2_STATUS] = 1,
+ [TABLA_A_CDC_MBHC_B3_STATUS] = 1,
+ [TABLA_A_CDC_MBHC_B4_STATUS] = 1,
+ [TABLA_A_CDC_MBHC_B5_STATUS] = 1,
+ [TABLA_A_CDC_MBHC_B1_CTL] = 1,
+ [TABLA_A_CDC_MBHC_B2_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B1_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B2_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B3_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B4_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B5_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B6_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B7_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B8_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B9_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B10_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B11_CTL] = 1,
+ [TABLA_A_CDC_MBHC_VOLT_B12_CTL] = 1,
+ [TABLA_A_CDC_MBHC_CLK_CTL] = 1,
+ [TABLA_A_CDC_MBHC_INT_CTL] = 1,
+ [TABLA_A_CDC_MBHC_DEBUG_CTL] = 1,
+ [TABLA_A_CDC_MBHC_SPARE] = 1,
+};
+
+const unsigned int tabla_1_reg_readable[TABLA_1_X_ONLY_REGISTERS] = {
+ TABLA_1_A_MICB_4_CTL,
+ TABLA_1_A_MICB_4_INT_RBIAS,
+ TABLA_1_A_MICB_4_MBHC,
+};
+
+const unsigned int tabla_2_reg_readable[TABLA_2_HIGHER_ONLY_REGISTERS] = {
+ TABLA_2_A_MICB_4_CTL,
+ TABLA_2_A_MICB_4_INT_RBIAS,
+ TABLA_2_A_MICB_4_MBHC,
+};
+
+const u8 tabla_reg_defaults[TABLA_CACHE_SIZE] = {
+ [TABLA_A_CHIP_CTL] = TABLA_A_CHIP_CTL__POR,
+ [TABLA_A_CHIP_STATUS] = TABLA_A_CHIP_STATUS__POR,
+ [TABLA_A_CHIP_ID_BYTE_0] = TABLA_A_CHIP_ID_BYTE_0__POR,
+ [TABLA_A_CHIP_ID_BYTE_1] = TABLA_A_CHIP_ID_BYTE_1__POR,
+ [TABLA_A_CHIP_ID_BYTE_2] = TABLA_A_CHIP_ID_BYTE_2__POR,
+ [TABLA_A_CHIP_ID_BYTE_3] = TABLA_A_CHIP_ID_BYTE_3__POR,
+ [TABLA_A_CHIP_VERSION] = TABLA_A_CHIP_VERSION__POR,
+ [TABLA_A_SB_VERSION] = TABLA_A_SB_VERSION__POR,
+ [TABLA_A_SLAVE_ID_1] = TABLA_A_SLAVE_ID_1__POR,
+ [TABLA_A_SLAVE_ID_2] = TABLA_A_SLAVE_ID_2__POR,
+ [TABLA_A_SLAVE_ID_3] = TABLA_A_SLAVE_ID_3__POR,
+ [TABLA_A_PIN_CTL_OE0] = TABLA_A_PIN_CTL_OE0__POR,
+ [TABLA_A_PIN_CTL_OE1] = TABLA_A_PIN_CTL_OE1__POR,
+ [TABLA_A_PIN_CTL_DATA0] = TABLA_A_PIN_CTL_DATA0__POR,
+ [TABLA_A_PIN_CTL_DATA1] = TABLA_A_PIN_CTL_DATA1__POR,
+ [TABLA_A_HDRIVE_GENERIC] = TABLA_A_HDRIVE_GENERIC__POR,
+ [TABLA_A_HDRIVE_OVERRIDE] = TABLA_A_HDRIVE_OVERRIDE__POR,
+ [TABLA_A_ANA_CSR_WAIT_STATE] = TABLA_A_ANA_CSR_WAIT_STATE__POR,
+ [TABLA_A_PROCESS_MONITOR_CTL0] = TABLA_A_PROCESS_MONITOR_CTL0__POR,
+ [TABLA_A_PROCESS_MONITOR_CTL1] = TABLA_A_PROCESS_MONITOR_CTL1__POR,
+ [TABLA_A_PROCESS_MONITOR_CTL2] = TABLA_A_PROCESS_MONITOR_CTL2__POR,
+ [TABLA_A_PROCESS_MONITOR_CTL3] = TABLA_A_PROCESS_MONITOR_CTL3__POR,
+ [TABLA_A_QFUSE_CTL] = TABLA_A_QFUSE_CTL__POR,
+ [TABLA_A_QFUSE_STATUS] = TABLA_A_QFUSE_STATUS__POR,
+ [TABLA_A_QFUSE_DATA_OUT0] = TABLA_A_QFUSE_DATA_OUT0__POR,
+ [TABLA_A_QFUSE_DATA_OUT1] = TABLA_A_QFUSE_DATA_OUT1__POR,
+ [TABLA_A_QFUSE_DATA_OUT2] = TABLA_A_QFUSE_DATA_OUT2__POR,
+ [TABLA_A_QFUSE_DATA_OUT3] = TABLA_A_QFUSE_DATA_OUT3__POR,
+ [TABLA_A_CDC_CTL] = TABLA_A_CDC_CTL__POR,
+ [TABLA_A_LEAKAGE_CTL] = TABLA_A_LEAKAGE_CTL__POR,
+ [TABLA_A_INTR_MODE] = TABLA_A_INTR_MODE__POR,
+ [TABLA_A_INTR_MASK0] = TABLA_A_INTR_MASK0__POR,
+ [TABLA_A_INTR_MASK1] = TABLA_A_INTR_MASK1__POR,
+ [TABLA_A_INTR_MASK2] = TABLA_A_INTR_MASK2__POR,
+ [TABLA_A_INTR_STATUS0] = TABLA_A_INTR_STATUS0__POR,
+ [TABLA_A_INTR_STATUS1] = TABLA_A_INTR_STATUS1__POR,
+ [TABLA_A_INTR_STATUS2] = TABLA_A_INTR_STATUS2__POR,
+ [TABLA_A_INTR_CLEAR0] = TABLA_A_INTR_CLEAR0__POR,
+ [TABLA_A_INTR_CLEAR1] = TABLA_A_INTR_CLEAR1__POR,
+ [TABLA_A_INTR_CLEAR2] = TABLA_A_INTR_CLEAR2__POR,
+ [TABLA_A_INTR_LEVEL0] = TABLA_A_INTR_LEVEL0__POR,
+ [TABLA_A_INTR_LEVEL1] = TABLA_A_INTR_LEVEL1__POR,
+ [TABLA_A_INTR_LEVEL2] = TABLA_A_INTR_LEVEL2__POR,
+ [TABLA_A_INTR_TEST0] = TABLA_A_INTR_TEST0__POR,
+ [TABLA_A_INTR_TEST1] = TABLA_A_INTR_TEST1__POR,
+ [TABLA_A_INTR_TEST2] = TABLA_A_INTR_TEST2__POR,
+ [TABLA_A_INTR_SET0] = TABLA_A_INTR_SET0__POR,
+ [TABLA_A_INTR_SET1] = TABLA_A_INTR_SET1__POR,
+ [TABLA_A_INTR_SET2] = TABLA_A_INTR_SET2__POR,
+ [TABLA_A_CDC_TX_I2S_SCK_MODE] = TABLA_A_CDC_TX_I2S_SCK_MODE__POR,
+ [TABLA_A_CDC_TX_I2S_WS_MODE] = TABLA_A_CDC_TX_I2S_WS_MODE__POR,
+ [TABLA_A_CDC_DMIC_DATA0_MODE] = TABLA_A_CDC_DMIC_DATA0_MODE__POR,
+ [TABLA_A_CDC_DMIC_CLK0_MODE] = TABLA_A_CDC_DMIC_CLK0_MODE__POR,
+ [TABLA_A_CDC_DMIC_DATA1_MODE] = TABLA_A_CDC_DMIC_DATA1_MODE__POR,
+ [TABLA_A_CDC_DMIC_CLK1_MODE] = TABLA_A_CDC_DMIC_CLK1_MODE__POR,
+ [TABLA_A_CDC_RX_I2S_SCK_MODE] = TABLA_A_CDC_RX_I2S_SCK_MODE__POR,
+ [TABLA_A_CDC_RX_I2S_WS_MODE] = TABLA_A_CDC_RX_I2S_WS_MODE__POR,
+ [TABLA_A_CDC_DMIC_DATA2_MODE] = TABLA_A_CDC_DMIC_DATA2_MODE__POR,
+ [TABLA_A_CDC_DMIC_CLK2_MODE] = TABLA_A_CDC_DMIC_CLK2_MODE__POR,
+ [TABLA_A_CDC_INTR_MODE] = TABLA_A_CDC_INTR_MODE__POR,
+ [TABLA_A_BIAS_REF_CTL] = TABLA_A_BIAS_REF_CTL__POR,
+ [TABLA_A_BIAS_CENTRAL_BG_CTL] = TABLA_A_BIAS_CENTRAL_BG_CTL__POR,
+ [TABLA_A_BIAS_PRECHRG_CTL] = TABLA_A_BIAS_PRECHRG_CTL__POR,
+ [TABLA_A_BIAS_CURR_CTL_1] = TABLA_A_BIAS_CURR_CTL_1__POR,
+ [TABLA_A_BIAS_CURR_CTL_2] = TABLA_A_BIAS_CURR_CTL_2__POR,
+ [TABLA_A_BIAS_CONFIG_MODE_BG_CTL] =
+ TABLA_A_BIAS_CONFIG_MODE_BG_CTL__POR,
+ [TABLA_A_BIAS_BG_STATUS] = TABLA_A_BIAS_BG_STATUS__POR,
+ [TABLA_A_CLK_BUFF_EN1] = TABLA_A_CLK_BUFF_EN1__POR,
+ [TABLA_A_CLK_BUFF_EN2] = TABLA_A_CLK_BUFF_EN2__POR,
+ [TABLA_A_LDO_H_MODE_1] = TABLA_A_LDO_H_MODE_1__POR,
+ [TABLA_A_LDO_H_MODE_2] = TABLA_A_LDO_H_MODE_2__POR,
+ [TABLA_A_LDO_H_LOOP_CTL] = TABLA_A_LDO_H_LOOP_CTL__POR,
+ [TABLA_A_LDO_H_COMP_1] = TABLA_A_LDO_H_COMP_1__POR,
+ [TABLA_A_LDO_H_COMP_2] = TABLA_A_LDO_H_COMP_2__POR,
+ [TABLA_A_LDO_H_BIAS_1] = TABLA_A_LDO_H_BIAS_1__POR,
+ [TABLA_A_LDO_H_BIAS_2] = TABLA_A_LDO_H_BIAS_2__POR,
+ [TABLA_A_LDO_H_BIAS_3] = TABLA_A_LDO_H_BIAS_3__POR,
+ [TABLA_A_LDO_L_MODE_1] = TABLA_A_LDO_L_MODE_1__POR,
+ [TABLA_A_LDO_L_MODE_2] = TABLA_A_LDO_L_MODE_2__POR,
+ [TABLA_A_LDO_L_LOOP_CTL] = TABLA_A_LDO_L_LOOP_CTL__POR,
+ [TABLA_A_LDO_L_COMP_1] = TABLA_A_LDO_L_COMP_1__POR,
+ [TABLA_A_LDO_L_COMP_2] = TABLA_A_LDO_L_COMP_2__POR,
+ [TABLA_A_LDO_L_BIAS_1] = TABLA_A_LDO_L_BIAS_1__POR,
+ [TABLA_A_LDO_L_BIAS_2] = TABLA_A_LDO_L_BIAS_2__POR,
+ [TABLA_A_LDO_L_BIAS_3] = TABLA_A_LDO_L_BIAS_3__POR,
+ [TABLA_A_MICB_CFILT_1_CTL] = TABLA_A_MICB_CFILT_1_CTL__POR,
+ [TABLA_A_MICB_CFILT_1_VAL] = TABLA_A_MICB_CFILT_1_VAL__POR,
+ [TABLA_A_MICB_CFILT_1_PRECHRG] = TABLA_A_MICB_CFILT_1_PRECHRG__POR,
+ [TABLA_A_MICB_1_CTL] = TABLA_A_MICB_1_CTL__POR,
+ [TABLA_A_MICB_1_INT_RBIAS] = TABLA_A_MICB_1_INT_RBIAS__POR,
+ [TABLA_A_MICB_1_MBHC] = TABLA_A_MICB_1_MBHC__POR,
+ [TABLA_A_MICB_CFILT_2_CTL] = TABLA_A_MICB_CFILT_2_CTL__POR,
+ [TABLA_A_MICB_CFILT_2_VAL] = TABLA_A_MICB_CFILT_2_VAL__POR,
+ [TABLA_A_MICB_CFILT_2_PRECHRG] = TABLA_A_MICB_CFILT_2_PRECHRG__POR,
+ [TABLA_A_MICB_2_CTL] = TABLA_A_MICB_2_CTL__POR,
+ [TABLA_A_MICB_2_INT_RBIAS] = TABLA_A_MICB_2_INT_RBIAS__POR,
+ [TABLA_A_MICB_2_MBHC] = TABLA_A_MICB_2_MBHC__POR,
+ [TABLA_A_MICB_CFILT_3_CTL] = TABLA_A_MICB_CFILT_3_CTL__POR,
+ [TABLA_A_MICB_CFILT_3_VAL] = TABLA_A_MICB_CFILT_3_VAL__POR,
+ [TABLA_A_MICB_CFILT_3_PRECHRG] = TABLA_A_MICB_CFILT_3_PRECHRG__POR,
+ [TABLA_A_MICB_3_CTL] = TABLA_A_MICB_3_CTL__POR,
+ [TABLA_A_MICB_3_INT_RBIAS] = TABLA_A_MICB_3_INT_RBIAS__POR,
+ [TABLA_A_MICB_3_MBHC] = TABLA_A_MICB_3_MBHC__POR,
+ [TABLA_1_A_MICB_4_CTL] = TABLA_A_MICB_4_CTL__POR,
+ [TABLA_1_A_MICB_4_INT_RBIAS] = TABLA_A_MICB_4_INT_RBIAS__POR,
+ [TABLA_1_A_MICB_4_MBHC] = TABLA_A_MICB_4_MBHC__POR,
+ [TABLA_2_A_MICB_4_CTL] = TABLA_A_MICB_4_CTL__POR,
+ [TABLA_2_A_MICB_4_INT_RBIAS] = TABLA_A_MICB_4_INT_RBIAS__POR,
+ [TABLA_2_A_MICB_4_MBHC] = TABLA_A_MICB_4_MBHC__POR,
+ [TABLA_A_TX_COM_BIAS] = TABLA_A_TX_COM_BIAS__POR,
+ [TABLA_A_MBHC_SCALING_MUX_1] = TABLA_A_MBHC_SCALING_MUX_1__POR,
+ [TABLA_A_MBHC_SCALING_MUX_2] = TABLA_A_MBHC_SCALING_MUX_2__POR,
+ [TABLA_A_TX_SUP_SWITCH_CTRL_1] = TABLA_A_TX_SUP_SWITCH_CTRL_1__POR,
+ [TABLA_A_TX_SUP_SWITCH_CTRL_2] = TABLA_A_TX_SUP_SWITCH_CTRL_2__POR,
+ [TABLA_A_TX_1_2_EN] = TABLA_A_TX_1_2_EN__POR,
+ [TABLA_A_TX_1_2_TEST_EN] = TABLA_A_TX_1_2_TEST_EN__POR,
+ [TABLA_A_TX_1_2_ADC_CH1] = TABLA_A_TX_1_2_ADC_CH1__POR,
+ [TABLA_A_TX_1_2_ADC_CH2] = TABLA_A_TX_1_2_ADC_CH2__POR,
+ [TABLA_A_TX_1_2_ATEST_REFCTRL] = TABLA_A_TX_1_2_ATEST_REFCTRL__POR,
+ [TABLA_A_TX_1_2_TEST_CTL] = TABLA_A_TX_1_2_TEST_CTL__POR,
+ [TABLA_A_TX_1_2_TEST_BLOCK_EN] = TABLA_A_TX_1_2_TEST_BLOCK_EN__POR,
+ [TABLA_A_TX_1_2_TXFE_CLKDIV] = TABLA_A_TX_1_2_TXFE_CLKDIV__POR,
+ [TABLA_A_TX_1_2_SAR_ERR_CH1] = TABLA_A_TX_1_2_SAR_ERR_CH1__POR,
+ [TABLA_A_TX_1_2_SAR_ERR_CH2] = TABLA_A_TX_1_2_SAR_ERR_CH2__POR,
+ [TABLA_A_TX_3_4_EN] = TABLA_A_TX_3_4_EN__POR,
+ [TABLA_A_TX_3_4_TEST_EN] = TABLA_A_TX_3_4_TEST_EN__POR,
+ [TABLA_A_TX_3_4_ADC_CH3] = TABLA_A_TX_3_4_ADC_CH3__POR,
+ [TABLA_A_TX_3_4_ADC_CH4] = TABLA_A_TX_3_4_ADC_CH4__POR,
+ [TABLA_A_TX_3_4_ATEST_REFCTRL] = TABLA_A_TX_3_4_ATEST_REFCTRL__POR,
+ [TABLA_A_TX_3_4_TEST_CTL] = TABLA_A_TX_3_4_TEST_CTL__POR,
+ [TABLA_A_TX_3_4_TEST_BLOCK_EN] = TABLA_A_TX_3_4_TEST_BLOCK_EN__POR,
+ [TABLA_A_TX_3_4_TXFE_CKDIV] = TABLA_A_TX_3_4_TXFE_CKDIV__POR,
+ [TABLA_A_TX_3_4_SAR_ERR_CH3] = TABLA_A_TX_3_4_SAR_ERR_CH3__POR,
+ [TABLA_A_TX_3_4_SAR_ERR_CH4] = TABLA_A_TX_3_4_SAR_ERR_CH4__POR,
+ [TABLA_A_TX_5_6_EN] = TABLA_A_TX_5_6_EN__POR,
+ [TABLA_A_TX_5_6_TEST_EN] = TABLA_A_TX_5_6_TEST_EN__POR,
+ [TABLA_A_TX_5_6_ADC_CH5] = TABLA_A_TX_5_6_ADC_CH5__POR,
+ [TABLA_A_TX_5_6_ADC_CH6] = TABLA_A_TX_5_6_ADC_CH6__POR,
+ [TABLA_A_TX_5_6_ATEST_REFCTRL] = TABLA_A_TX_5_6_ATEST_REFCTRL__POR,
+ [TABLA_A_TX_5_6_TEST_CTL] = TABLA_A_TX_5_6_TEST_CTL__POR,
+ [TABLA_A_TX_5_6_TEST_BLOCK_EN] = TABLA_A_TX_5_6_TEST_BLOCK_EN__POR,
+ [TABLA_A_TX_5_6_TXFE_CKDIV] = TABLA_A_TX_5_6_TXFE_CKDIV__POR,
+ [TABLA_A_TX_5_6_SAR_ERR_CH5] = TABLA_A_TX_5_6_SAR_ERR_CH5__POR,
+ [TABLA_A_TX_5_6_SAR_ERR_CH6] = TABLA_A_TX_5_6_SAR_ERR_CH6__POR,
+ [TABLA_A_TX_7_MBHC_EN] = TABLA_A_TX_7_MBHC_EN__POR,
+ [TABLA_A_TX_7_MBHC_ATEST_REFCTRL] =
+ TABLA_A_TX_7_MBHC_ATEST_REFCTRL__POR,
+ [TABLA_A_TX_7_MBHC_ADC] = TABLA_A_TX_7_MBHC_ADC__POR,
+ [TABLA_A_TX_7_MBHC_TEST_CTL] = TABLA_A_TX_7_MBHC_TEST_CTL__POR,
+ [TABLA_A_TX_7_MBHC_SAR_ERR] = TABLA_A_TX_7_MBHC_SAR_ERR__POR,
+ [TABLA_A_TX_7_TXFE_CLKDIV] = TABLA_A_TX_7_TXFE_CLKDIV__POR,
+ [TABLA_A_AUX_COM_CTL] = TABLA_A_AUX_COM_CTL__POR,
+ [TABLA_A_AUX_COM_ATEST] = TABLA_A_AUX_COM_ATEST__POR,
+ [TABLA_A_AUX_L_EN] = TABLA_A_AUX_L_EN__POR,
+ [TABLA_A_AUX_L_GAIN] = TABLA_A_AUX_L_GAIN__POR,
+ [TABLA_A_AUX_L_PA_CONN] = TABLA_A_AUX_L_PA_CONN__POR,
+ [TABLA_A_AUX_L_PA_CONN_INV] = TABLA_A_AUX_L_PA_CONN_INV__POR,
+ [TABLA_A_AUX_R_EN] = TABLA_A_AUX_R_EN__POR,
+ [TABLA_A_AUX_R_GAIN] = TABLA_A_AUX_R_GAIN__POR,
+ [TABLA_A_AUX_R_PA_CONN] = TABLA_A_AUX_R_PA_CONN__POR,
+ [TABLA_A_AUX_R_PA_CONN_INV] = TABLA_A_AUX_R_PA_CONN_INV__POR,
+ [TABLA_A_CP_EN] = TABLA_A_CP_EN__POR,
+ [TABLA_A_CP_CLK] = TABLA_A_CP_CLK__POR,
+ [TABLA_A_CP_STATIC] = TABLA_A_CP_STATIC__POR,
+ [TABLA_A_CP_DCC1] = TABLA_A_CP_DCC1__POR,
+ [TABLA_A_CP_DCC3] = TABLA_A_CP_DCC3__POR,
+ [TABLA_A_CP_ATEST] = TABLA_A_CP_ATEST__POR,
+ [TABLA_A_CP_DTEST] = TABLA_A_CP_DTEST__POR,
+ [TABLA_A_RX_COM_TIMER_DIV] = TABLA_A_RX_COM_TIMER_DIV__POR,
+ [TABLA_A_RX_COM_OCP_CTL] = TABLA_A_RX_COM_OCP_CTL__POR,
+ [TABLA_A_RX_COM_OCP_COUNT] = TABLA_A_RX_COM_OCP_COUNT__POR,
+ [TABLA_A_RX_COM_DAC_CTL] = TABLA_A_RX_COM_DAC_CTL__POR,
+ [TABLA_A_RX_COM_BIAS] = TABLA_A_RX_COM_BIAS__POR,
+ [TABLA_A_RX_HPH_BIAS_PA] = TABLA_A_RX_HPH_BIAS_PA__POR,
+ [TABLA_A_RX_HPH_BIAS_LDO] = TABLA_A_RX_HPH_BIAS_LDO__POR,
+ [TABLA_A_RX_HPH_BIAS_CNP] = TABLA_A_RX_HPH_BIAS_CNP__POR,
+ [TABLA_A_RX_HPH_BIAS_WG] = TABLA_A_RX_HPH_BIAS_WG__POR,
+ [TABLA_A_RX_HPH_OCP_CTL] = TABLA_A_RX_HPH_OCP_CTL__POR,
+ [TABLA_A_RX_HPH_CNP_EN] = TABLA_A_RX_HPH_CNP_EN__POR,
+ [TABLA_A_RX_HPH_CNP_WG_CTL] = TABLA_A_RX_HPH_CNP_WG_CTL__POR,
+ [TABLA_A_RX_HPH_CNP_WG_TIME] = TABLA_A_RX_HPH_CNP_WG_TIME__POR,
+ [TABLA_A_RX_HPH_L_GAIN] = TABLA_A_RX_HPH_L_GAIN__POR,
+ [TABLA_A_RX_HPH_L_TEST] = TABLA_A_RX_HPH_L_TEST__POR,
+ [TABLA_A_RX_HPH_L_PA_CTL] = TABLA_A_RX_HPH_L_PA_CTL__POR,
+ [TABLA_A_RX_HPH_L_DAC_CTL] = TABLA_A_RX_HPH_L_DAC_CTL__POR,
+ [TABLA_A_RX_HPH_L_ATEST] = TABLA_A_RX_HPH_L_ATEST__POR,
+ [TABLA_A_RX_HPH_L_STATUS] = TABLA_A_RX_HPH_L_STATUS__POR,
+ [TABLA_A_RX_HPH_R_GAIN] = TABLA_A_RX_HPH_R_GAIN__POR,
+ [TABLA_A_RX_HPH_R_TEST] = TABLA_A_RX_HPH_R_TEST__POR,
+ [TABLA_A_RX_HPH_R_PA_CTL] = TABLA_A_RX_HPH_R_PA_CTL__POR,
+ [TABLA_A_RX_HPH_R_DAC_CTL] = TABLA_A_RX_HPH_R_DAC_CTL__POR,
+ [TABLA_A_RX_HPH_R_ATEST] = TABLA_A_RX_HPH_R_ATEST__POR,
+ [TABLA_A_RX_HPH_R_STATUS] = TABLA_A_RX_HPH_R_STATUS__POR,
+ [TABLA_A_RX_EAR_BIAS_PA] = TABLA_A_RX_EAR_BIAS_PA__POR,
+ [TABLA_A_RX_EAR_BIAS_CMBUFF] = TABLA_A_RX_EAR_BIAS_CMBUFF__POR,
+ [TABLA_A_RX_EAR_EN] = TABLA_A_RX_EAR_EN__POR,
+ [TABLA_A_RX_EAR_GAIN] = TABLA_A_RX_EAR_GAIN__POR,
+ [TABLA_A_RX_EAR_CMBUFF] = TABLA_A_RX_EAR_CMBUFF__POR,
+ [TABLA_A_RX_EAR_ICTL] = TABLA_A_RX_EAR_ICTL__POR,
+ [TABLA_A_RX_EAR_CCOMP] = TABLA_A_RX_EAR_CCOMP__POR,
+ [TABLA_A_RX_EAR_VCM] = TABLA_A_RX_EAR_VCM__POR,
+ [TABLA_A_RX_EAR_CNP] = TABLA_A_RX_EAR_CNP__POR,
+ [TABLA_A_RX_EAR_ATEST] = TABLA_A_RX_EAR_ATEST__POR,
+ [TABLA_A_RX_EAR_STATUS] = TABLA_A_RX_EAR_STATUS__POR,
+ [TABLA_A_RX_LINE_BIAS_PA] = TABLA_A_RX_LINE_BIAS_PA__POR,
+ [TABLA_A_RX_LINE_BIAS_DAC] = TABLA_A_RX_LINE_BIAS_DAC__POR,
+ [TABLA_A_RX_LINE_BIAS_CNP] = TABLA_A_RX_LINE_BIAS_CNP__POR,
+ [TABLA_A_RX_LINE_COM] = TABLA_A_RX_LINE_COM__POR,
+ [TABLA_A_RX_LINE_CNP_EN] = TABLA_A_RX_LINE_CNP_EN__POR,
+ [TABLA_A_RX_LINE_CNP_WG_CTL] = TABLA_A_RX_LINE_CNP_WG_CTL__POR,
+ [TABLA_A_RX_LINE_CNP_WG_TIME] = TABLA_A_RX_LINE_CNP_WG_TIME__POR,
+ [TABLA_A_RX_LINE_1_GAIN] = TABLA_A_RX_LINE_1_GAIN__POR,
+ [TABLA_A_RX_LINE_1_TEST] = TABLA_A_RX_LINE_1_TEST__POR,
+ [TABLA_A_RX_LINE_1_DAC_CTL] = TABLA_A_RX_LINE_1_DAC_CTL__POR,
+ [TABLA_A_RX_LINE_1_STATUS] = TABLA_A_RX_LINE_1_STATUS__POR,
+ [TABLA_A_RX_LINE_2_GAIN] = TABLA_A_RX_LINE_2_GAIN__POR,
+ [TABLA_A_RX_LINE_2_TEST] = TABLA_A_RX_LINE_2_TEST__POR,
+ [TABLA_A_RX_LINE_2_DAC_CTL] = TABLA_A_RX_LINE_2_DAC_CTL__POR,
+ [TABLA_A_RX_LINE_2_STATUS] = TABLA_A_RX_LINE_2_STATUS__POR,
+ [TABLA_A_RX_LINE_3_GAIN] = TABLA_A_RX_LINE_3_GAIN__POR,
+ [TABLA_A_RX_LINE_3_TEST] = TABLA_A_RX_LINE_3_TEST__POR,
+ [TABLA_A_RX_LINE_3_DAC_CTL] = TABLA_A_RX_LINE_3_DAC_CTL__POR,
+ [TABLA_A_RX_LINE_3_STATUS] = TABLA_A_RX_LINE_3_STATUS__POR,
+ [TABLA_A_RX_LINE_4_GAIN] = TABLA_A_RX_LINE_4_GAIN__POR,
+ [TABLA_A_RX_LINE_4_TEST] = TABLA_A_RX_LINE_4_TEST__POR,
+ [TABLA_A_RX_LINE_4_DAC_CTL] = TABLA_A_RX_LINE_4_DAC_CTL__POR,
+ [TABLA_A_RX_LINE_4_STATUS] = TABLA_A_RX_LINE_4_STATUS__POR,
+ [TABLA_A_RX_LINE_5_GAIN] = TABLA_A_RX_LINE_5_GAIN__POR,
+ [TABLA_A_RX_LINE_5_TEST] = TABLA_A_RX_LINE_5_TEST__POR,
+ [TABLA_A_RX_LINE_5_DAC_CTL] = TABLA_A_RX_LINE_5_DAC_CTL__POR,
+ [TABLA_A_RX_LINE_5_STATUS] = TABLA_A_RX_LINE_5_STATUS__POR,
+ [TABLA_A_RX_LINE_CNP_DBG] = TABLA_A_RX_LINE_CNP_DBG__POR,
+ [TABLA_A_MBHC_HPH] = TABLA_A_MBHC_HPH__POR,
+ [TABLA_A_CONFIG_MODE_FREQ] = TABLA_A_CONFIG_MODE_FREQ__POR,
+ [TABLA_A_CONFIG_MODE_TEST] = TABLA_A_CONFIG_MODE_TEST__POR,
+ [TABLA_A_CONFIG_MODE_STATUS] = TABLA_A_CONFIG_MODE_STATUS__POR,
+ [TABLA_A_CONFIG_MODE_TUNER] = TABLA_A_CONFIG_MODE_TUNER__POR,
+ [TABLA_A_CDC_ANC1_CTL] = TABLA_A_CDC_ANC1_CTL__POR,
+ [TABLA_A_CDC_ANC1_SHIFT] = TABLA_A_CDC_ANC1_SHIFT__POR,
+ [TABLA_A_CDC_ANC1_FILT1_B1_CTL] = TABLA_A_CDC_ANC1_FILT1_B1_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT1_B2_CTL] = TABLA_A_CDC_ANC1_FILT1_B2_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT1_B3_CTL] = TABLA_A_CDC_ANC1_FILT1_B3_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT1_B4_CTL] = TABLA_A_CDC_ANC1_FILT1_B4_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT2_B1_CTL] = TABLA_A_CDC_ANC1_FILT2_B1_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT2_B2_CTL] = TABLA_A_CDC_ANC1_FILT2_B2_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT2_B3_CTL] = TABLA_A_CDC_ANC1_FILT2_B3_CTL__POR,
+ [TABLA_A_CDC_ANC1_SPARE] = TABLA_A_CDC_ANC1_SPARE__POR,
+ [TABLA_A_CDC_ANC1_FILT3_CTL] = TABLA_A_CDC_ANC1_FILT3_CTL__POR,
+ [TABLA_A_CDC_ANC1_FILT4_CTL] = TABLA_A_CDC_ANC1_FILT4_CTL__POR,
+ [TABLA_A_CDC_TX1_VOL_CTL_TIMER] = TABLA_A_CDC_TX1_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX2_VOL_CTL_TIMER] = TABLA_A_CDC_TX2_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX3_VOL_CTL_TIMER] = TABLA_A_CDC_TX3_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX4_VOL_CTL_TIMER] = TABLA_A_CDC_TX4_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX5_VOL_CTL_TIMER] = TABLA_A_CDC_TX5_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX6_VOL_CTL_TIMER] = TABLA_A_CDC_TX6_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX7_VOL_CTL_TIMER] = TABLA_A_CDC_TX7_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX8_VOL_CTL_TIMER] = TABLA_A_CDC_TX8_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX9_VOL_CTL_TIMER] = TABLA_A_CDC_TX9_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX10_VOL_CTL_TIMER] = TABLA_A_CDC_TX10_VOL_CTL_TIMER__POR,
+ [TABLA_A_CDC_TX1_VOL_CTL_GAIN] = TABLA_A_CDC_TX1_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX2_VOL_CTL_GAIN] = TABLA_A_CDC_TX2_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX3_VOL_CTL_GAIN] = TABLA_A_CDC_TX3_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX4_VOL_CTL_GAIN] = TABLA_A_CDC_TX4_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX5_VOL_CTL_GAIN] = TABLA_A_CDC_TX5_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX6_VOL_CTL_GAIN] = TABLA_A_CDC_TX6_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX7_VOL_CTL_GAIN] = TABLA_A_CDC_TX7_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX8_VOL_CTL_GAIN] = TABLA_A_CDC_TX8_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX9_VOL_CTL_GAIN] = TABLA_A_CDC_TX9_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX10_VOL_CTL_GAIN] = TABLA_A_CDC_TX10_VOL_CTL_GAIN__POR,
+ [TABLA_A_CDC_TX1_VOL_CTL_CFG] = TABLA_A_CDC_TX1_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX2_VOL_CTL_CFG] = TABLA_A_CDC_TX2_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX3_VOL_CTL_CFG] = TABLA_A_CDC_TX3_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX4_VOL_CTL_CFG] = TABLA_A_CDC_TX4_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX5_VOL_CTL_CFG] = TABLA_A_CDC_TX5_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX6_VOL_CTL_CFG] = TABLA_A_CDC_TX6_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX7_VOL_CTL_CFG] = TABLA_A_CDC_TX7_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX8_VOL_CTL_CFG] = TABLA_A_CDC_TX8_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX9_VOL_CTL_CFG] = TABLA_A_CDC_TX9_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX10_VOL_CTL_CFG] = TABLA_A_CDC_TX10_VOL_CTL_CFG__POR,
+ [TABLA_A_CDC_TX1_MUX_CTL] = TABLA_A_CDC_TX1_MUX_CTL__POR,
+ [TABLA_A_CDC_TX2_MUX_CTL] = TABLA_A_CDC_TX2_MUX_CTL__POR,
+ [TABLA_A_CDC_TX3_MUX_CTL] = TABLA_A_CDC_TX3_MUX_CTL__POR,
+ [TABLA_A_CDC_TX4_MUX_CTL] = TABLA_A_CDC_TX4_MUX_CTL__POR,
+ [TABLA_A_CDC_TX5_MUX_CTL] = TABLA_A_CDC_TX5_MUX_CTL__POR,
+ [TABLA_A_CDC_TX6_MUX_CTL] = TABLA_A_CDC_TX6_MUX_CTL__POR,
+ [TABLA_A_CDC_TX7_MUX_CTL] = TABLA_A_CDC_TX7_MUX_CTL__POR,
+ [TABLA_A_CDC_TX8_MUX_CTL] = TABLA_A_CDC_TX8_MUX_CTL__POR,
+ [TABLA_A_CDC_TX9_MUX_CTL] = TABLA_A_CDC_TX9_MUX_CTL__POR,
+ [TABLA_A_CDC_TX10_MUX_CTL] = TABLA_A_CDC_TX10_MUX_CTL__POR,
+ [TABLA_A_CDC_TX1_CLK_FS_CTL] = TABLA_A_CDC_TX1_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX2_CLK_FS_CTL] = TABLA_A_CDC_TX2_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX3_CLK_FS_CTL] = TABLA_A_CDC_TX3_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX4_CLK_FS_CTL] = TABLA_A_CDC_TX4_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX5_CLK_FS_CTL] = TABLA_A_CDC_TX5_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX6_CLK_FS_CTL] = TABLA_A_CDC_TX6_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX7_CLK_FS_CTL] = TABLA_A_CDC_TX7_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX8_CLK_FS_CTL] = TABLA_A_CDC_TX8_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX9_CLK_FS_CTL] = TABLA_A_CDC_TX9_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX10_CLK_FS_CTL] = TABLA_A_CDC_TX10_CLK_FS_CTL__POR,
+ [TABLA_A_CDC_TX1_DMIC_CTL] = TABLA_A_CDC_TX1_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX2_DMIC_CTL] = TABLA_A_CDC_TX2_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX3_DMIC_CTL] = TABLA_A_CDC_TX3_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX4_DMIC_CTL] = TABLA_A_CDC_TX4_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX5_DMIC_CTL] = TABLA_A_CDC_TX5_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX6_DMIC_CTL] = TABLA_A_CDC_TX6_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX7_DMIC_CTL] = TABLA_A_CDC_TX7_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX8_DMIC_CTL] = TABLA_A_CDC_TX8_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX9_DMIC_CTL] = TABLA_A_CDC_TX9_DMIC_CTL__POR,
+ [TABLA_A_CDC_TX10_DMIC_CTL] = TABLA_A_CDC_TX10_DMIC_CTL__POR,
+ [TABLA_A_CDC_SRC1_PDA_CFG] = TABLA_A_CDC_SRC1_PDA_CFG__POR,
+ [TABLA_A_CDC_SRC2_PDA_CFG] = TABLA_A_CDC_SRC2_PDA_CFG__POR,
+ [TABLA_A_CDC_SRC1_FS_CTL] = TABLA_A_CDC_SRC1_FS_CTL__POR,
+ [TABLA_A_CDC_SRC2_FS_CTL] = TABLA_A_CDC_SRC2_FS_CTL__POR,
+ [TABLA_A_CDC_RX1_B1_CTL] = TABLA_A_CDC_RX1_B1_CTL__POR,
+ [TABLA_A_CDC_RX2_B1_CTL] = TABLA_A_CDC_RX2_B1_CTL__POR,
+ [TABLA_A_CDC_RX3_B1_CTL] = TABLA_A_CDC_RX3_B1_CTL__POR,
+ [TABLA_A_CDC_RX4_B1_CTL] = TABLA_A_CDC_RX4_B1_CTL__POR,
+ [TABLA_A_CDC_RX5_B1_CTL] = TABLA_A_CDC_RX5_B1_CTL__POR,
+ [TABLA_A_CDC_RX6_B1_CTL] = TABLA_A_CDC_RX6_B1_CTL__POR,
+ [TABLA_A_CDC_RX7_B1_CTL] = TABLA_A_CDC_RX7_B1_CTL__POR,
+ [TABLA_A_CDC_RX1_B2_CTL] = TABLA_A_CDC_RX1_B2_CTL__POR,
+ [TABLA_A_CDC_RX2_B2_CTL] = TABLA_A_CDC_RX2_B2_CTL__POR,
+ [TABLA_A_CDC_RX3_B2_CTL] = TABLA_A_CDC_RX3_B2_CTL__POR,
+ [TABLA_A_CDC_RX4_B2_CTL] = TABLA_A_CDC_RX4_B2_CTL__POR,
+ [TABLA_A_CDC_RX5_B2_CTL] = TABLA_A_CDC_RX5_B2_CTL__POR,
+ [TABLA_A_CDC_RX6_B2_CTL] = TABLA_A_CDC_RX6_B2_CTL__POR,
+ [TABLA_A_CDC_RX7_B2_CTL] = TABLA_A_CDC_RX7_B2_CTL__POR,
+ [TABLA_A_CDC_RX1_B3_CTL] = TABLA_A_CDC_RX1_B3_CTL__POR,
+ [TABLA_A_CDC_RX2_B3_CTL] = TABLA_A_CDC_RX2_B3_CTL__POR,
+ [TABLA_A_CDC_RX3_B3_CTL] = TABLA_A_CDC_RX3_B3_CTL__POR,
+ [TABLA_A_CDC_RX4_B3_CTL] = TABLA_A_CDC_RX4_B3_CTL__POR,
+ [TABLA_A_CDC_RX5_B3_CTL] = TABLA_A_CDC_RX5_B3_CTL__POR,
+ [TABLA_A_CDC_RX6_B3_CTL] = TABLA_A_CDC_RX6_B3_CTL__POR,
+ [TABLA_A_CDC_RX7_B3_CTL] = TABLA_A_CDC_RX7_B3_CTL__POR,
+ [TABLA_A_CDC_RX1_B4_CTL] = TABLA_A_CDC_RX1_B4_CTL__POR,
+ [TABLA_A_CDC_RX2_B4_CTL] = TABLA_A_CDC_RX2_B4_CTL__POR,
+ [TABLA_A_CDC_RX3_B4_CTL] = TABLA_A_CDC_RX3_B4_CTL__POR,
+ [TABLA_A_CDC_RX4_B4_CTL] = TABLA_A_CDC_RX4_B4_CTL__POR,
+ [TABLA_A_CDC_RX5_B4_CTL] = TABLA_A_CDC_RX5_B4_CTL__POR,
+ [TABLA_A_CDC_RX6_B4_CTL] = TABLA_A_CDC_RX6_B4_CTL__POR,
+ [TABLA_A_CDC_RX7_B4_CTL] = TABLA_A_CDC_RX7_B4_CTL__POR,
+ [TABLA_A_CDC_RX1_B5_CTL] = TABLA_A_CDC_RX1_B5_CTL__POR,
+ [TABLA_A_CDC_RX2_B5_CTL] = TABLA_A_CDC_RX2_B5_CTL__POR,
+ [TABLA_A_CDC_RX3_B5_CTL] = TABLA_A_CDC_RX3_B5_CTL__POR,
+ [TABLA_A_CDC_RX4_B5_CTL] = TABLA_A_CDC_RX4_B5_CTL__POR,
+ [TABLA_A_CDC_RX5_B5_CTL] = TABLA_A_CDC_RX5_B5_CTL__POR,
+ [TABLA_A_CDC_RX6_B5_CTL] = TABLA_A_CDC_RX6_B5_CTL__POR,
+ [TABLA_A_CDC_RX7_B5_CTL] = TABLA_A_CDC_RX7_B5_CTL__POR,
+ [TABLA_A_CDC_RX1_B6_CTL] = TABLA_A_CDC_RX1_B6_CTL__POR,
+ [TABLA_A_CDC_RX2_B6_CTL] = TABLA_A_CDC_RX2_B6_CTL__POR,
+ [TABLA_A_CDC_RX3_B6_CTL] = TABLA_A_CDC_RX3_B6_CTL__POR,
+ [TABLA_A_CDC_RX4_B6_CTL] = TABLA_A_CDC_RX4_B6_CTL__POR,
+ [TABLA_A_CDC_RX5_B6_CTL] = TABLA_A_CDC_RX5_B6_CTL__POR,
+ [TABLA_A_CDC_RX6_B6_CTL] = TABLA_A_CDC_RX6_B6_CTL__POR,
+ [TABLA_A_CDC_RX7_B6_CTL] = TABLA_A_CDC_RX7_B6_CTL__POR,
+ [TABLA_A_CDC_RX1_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX1_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX2_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX2_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX3_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX3_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX4_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX4_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX5_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX5_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX6_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX6_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX7_VOL_CTL_B1_CTL] = TABLA_A_CDC_RX7_VOL_CTL_B1_CTL__POR,
+ [TABLA_A_CDC_RX1_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX1_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_RX2_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX2_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_RX3_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX3_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_RX4_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX4_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_RX5_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX5_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_RX6_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX6_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_RX7_VOL_CTL_B2_CTL] = TABLA_A_CDC_RX7_VOL_CTL_B2_CTL__POR,
+ [TABLA_A_CDC_CLK_RX_RESET_CTL] = TABLA_A_CDC_CLK_RX_RESET_CTL__POR,
+ [TABLA_A_CDC_CLK_ANC_RESET_CTL] = TABLA_A_CDC_CLK_ANC_RESET_CTL__POR,
+ [TABLA_A_CDC_CLK_TX_RESET_B1_CTL] =
+ TABLA_A_CDC_CLK_TX_RESET_B1_CTL__POR,
+ [TABLA_A_CDC_CLK_TX_RESET_B2_CTL] =
+ TABLA_A_CDC_CLK_TX_RESET_B2_CTL__POR,
+ [TABLA_A_CDC_CLK_DMIC_CTL] = TABLA_A_CDC_CLK_DMIC_CTL__POR,
+ [TABLA_A_CDC_CLK_RX_I2S_CTL] = TABLA_A_CDC_CLK_RX_I2S_CTL__POR,
+ [TABLA_A_CDC_CLK_TX_I2S_CTL] = TABLA_A_CDC_CLK_TX_I2S_CTL__POR,
+ [TABLA_A_CDC_CLK_OTHR_RESET_CTL] = TABLA_A_CDC_CLK_OTHR_RESET_CTL__POR,
+ [TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL] =
+ TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL__POR,
+ [TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL] =
+ TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL__POR,
+ [TABLA_A_CDC_CLK_OTHR_CTL] = TABLA_A_CDC_CLK_OTHR_CTL__POR,
+ [TABLA_A_CDC_CLK_RDAC_CLK_EN_CTL] =
+ TABLA_A_CDC_CLK_RDAC_CLK_EN_CTL__POR,
+ [TABLA_A_CDC_CLK_ANC_CLK_EN_CTL] = TABLA_A_CDC_CLK_ANC_CLK_EN_CTL__POR,
+ [TABLA_A_CDC_CLK_RX_B1_CTL] = TABLA_A_CDC_CLK_RX_B1_CTL__POR,
+ [TABLA_A_CDC_CLK_RX_B2_CTL] = TABLA_A_CDC_CLK_RX_B2_CTL__POR,
+ [TABLA_A_CDC_CLK_MCLK_CTL] = TABLA_A_CDC_CLK_MCLK_CTL__POR,
+ [TABLA_A_CDC_CLK_PDM_CTL] = TABLA_A_CDC_CLK_PDM_CTL__POR,
+ [TABLA_A_CDC_CLK_SD_CTL] = TABLA_A_CDC_CLK_SD_CTL__POR,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B1_CTL] =
+ TABLA_A_CDC_CLSG_FREQ_THRESH_B1_CTL__POR,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B2_CTL] =
+ TABLA_A_CDC_CLSG_FREQ_THRESH_B2_CTL__POR,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B3_CTL] =
+ TABLA_A_CDC_CLSG_FREQ_THRESH_B3_CTL__POR,
+ [TABLA_A_CDC_CLSG_FREQ_THRESH_B4_CTL] =
+ TABLA_A_CDC_CLSG_FREQ_THRESH_B4_CTL__POR,
+ [TABLA_A_CDC_CLSG_GAIN_THRESH_CTL] =
+ TABLA_A_CDC_CLSG_GAIN_THRESH_CTL__POR,
+ [TABLA_A_CDC_CLSG_TIMER_B1_CFG] = TABLA_A_CDC_CLSG_TIMER_B1_CFG__POR,
+ [TABLA_A_CDC_CLSG_TIMER_B2_CFG] = TABLA_A_CDC_CLSG_TIMER_B2_CFG__POR,
+ [TABLA_A_CDC_CLSG_CTL] = TABLA_A_CDC_CLSG_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B1_CTL] = TABLA_A_CDC_IIR1_GAIN_B1_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B1_CTL] = TABLA_A_CDC_IIR2_GAIN_B1_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B2_CTL] = TABLA_A_CDC_IIR1_GAIN_B2_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B2_CTL] = TABLA_A_CDC_IIR2_GAIN_B2_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B3_CTL] = TABLA_A_CDC_IIR1_GAIN_B3_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B3_CTL] = TABLA_A_CDC_IIR2_GAIN_B3_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B4_CTL] = TABLA_A_CDC_IIR1_GAIN_B4_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B4_CTL] = TABLA_A_CDC_IIR2_GAIN_B4_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B5_CTL] = TABLA_A_CDC_IIR1_GAIN_B5_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B5_CTL] = TABLA_A_CDC_IIR2_GAIN_B5_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B6_CTL] = TABLA_A_CDC_IIR1_GAIN_B6_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B6_CTL] = TABLA_A_CDC_IIR2_GAIN_B6_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B7_CTL] = TABLA_A_CDC_IIR1_GAIN_B7_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B7_CTL] = TABLA_A_CDC_IIR2_GAIN_B7_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_B8_CTL] = TABLA_A_CDC_IIR1_GAIN_B8_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_B8_CTL] = TABLA_A_CDC_IIR2_GAIN_B8_CTL__POR,
+ [TABLA_A_CDC_IIR1_CTL] = TABLA_A_CDC_IIR1_CTL__POR,
+ [TABLA_A_CDC_IIR2_CTL] = TABLA_A_CDC_IIR2_CTL__POR,
+ [TABLA_A_CDC_IIR1_GAIN_TIMER_CTL] =
+ TABLA_A_CDC_IIR1_GAIN_TIMER_CTL__POR,
+ [TABLA_A_CDC_IIR2_GAIN_TIMER_CTL] =
+ TABLA_A_CDC_IIR2_GAIN_TIMER_CTL__POR,
+ [TABLA_A_CDC_IIR1_COEF_B1_CTL] = TABLA_A_CDC_IIR1_COEF_B1_CTL__POR,
+ [TABLA_A_CDC_IIR2_COEF_B1_CTL] = TABLA_A_CDC_IIR2_COEF_B1_CTL__POR,
+ [TABLA_A_CDC_IIR1_COEF_B2_CTL] = TABLA_A_CDC_IIR1_COEF_B2_CTL__POR,
+ [TABLA_A_CDC_IIR2_COEF_B2_CTL] = TABLA_A_CDC_IIR2_COEF_B2_CTL__POR,
+ [TABLA_A_CDC_IIR1_COEF_B3_CTL] = TABLA_A_CDC_IIR1_COEF_B3_CTL__POR,
+ [TABLA_A_CDC_IIR2_COEF_B3_CTL] = TABLA_A_CDC_IIR2_COEF_B3_CTL__POR,
+ [TABLA_A_CDC_IIR1_COEF_B4_CTL] = TABLA_A_CDC_IIR1_COEF_B4_CTL__POR,
+ [TABLA_A_CDC_IIR2_COEF_B4_CTL] = TABLA_A_CDC_IIR2_COEF_B4_CTL__POR,
+ [TABLA_A_CDC_IIR1_COEF_B5_CTL] = TABLA_A_CDC_IIR1_COEF_B5_CTL__POR,
+ [TABLA_A_CDC_IIR2_COEF_B5_CTL] = TABLA_A_CDC_IIR2_COEF_B5_CTL__POR,
+ [TABLA_A_CDC_TOP_GAIN_UPDATE] = TABLA_A_CDC_TOP_GAIN_UPDATE__POR,
+ [TABLA_A_CDC_DEBUG_B1_CTL] = TABLA_A_CDC_DEBUG_B1_CTL__POR,
+ [TABLA_A_CDC_DEBUG_B2_CTL] = TABLA_A_CDC_DEBUG_B2_CTL__POR,
+ [TABLA_A_CDC_DEBUG_B3_CTL] = TABLA_A_CDC_DEBUG_B3_CTL__POR,
+ [TABLA_A_CDC_DEBUG_B4_CTL] = TABLA_A_CDC_DEBUG_B4_CTL__POR,
+ [TABLA_A_CDC_DEBUG_B5_CTL] = TABLA_A_CDC_DEBUG_B5_CTL__POR,
+ [TABLA_A_CDC_DEBUG_B6_CTL] = TABLA_A_CDC_DEBUG_B6_CTL__POR,
+ [TABLA_A_CDC_COMP1_B1_CTL] = TABLA_A_CDC_COMP1_B1_CTL__POR,
+ [TABLA_A_CDC_COMP1_B2_CTL] = TABLA_A_CDC_COMP1_B2_CTL__POR,
+ [TABLA_A_CDC_COMP1_B3_CTL] = TABLA_A_CDC_COMP1_B3_CTL__POR,
+ [TABLA_A_CDC_COMP1_B4_CTL] = TABLA_A_CDC_COMP1_B4_CTL__POR,
+ [TABLA_A_CDC_COMP1_B5_CTL] = TABLA_A_CDC_COMP1_B5_CTL__POR,
+ [TABLA_A_CDC_COMP1_B6_CTL] = TABLA_A_CDC_COMP1_B6_CTL__POR,
+ [TABLA_A_CDC_COMP1_SHUT_DOWN_STATUS] =
+ TABLA_A_CDC_COMP1_SHUT_DOWN_STATUS__POR,
+ [TABLA_A_CDC_COMP1_FS_CFG] = TABLA_A_CDC_COMP1_FS_CFG__POR,
+ [TABLA_A_CDC_COMP2_B1_CTL] = TABLA_A_CDC_COMP2_B1_CTL__POR,
+ [TABLA_A_CDC_COMP2_B2_CTL] = TABLA_A_CDC_COMP2_B2_CTL__POR,
+ [TABLA_A_CDC_COMP2_B3_CTL] = TABLA_A_CDC_COMP2_B3_CTL__POR,
+ [TABLA_A_CDC_COMP2_B4_CTL] = TABLA_A_CDC_COMP2_B4_CTL__POR,
+ [TABLA_A_CDC_COMP2_B5_CTL] = TABLA_A_CDC_COMP2_B5_CTL__POR,
+ [TABLA_A_CDC_COMP2_B6_CTL] = TABLA_A_CDC_COMP2_B6_CTL__POR,
+ [TABLA_A_CDC_COMP1_SHUT_DOWN_STATUS] =
+ TABLA_A_CDC_COMP1_SHUT_DOWN_STATUS__POR,
+ [TABLA_A_CDC_COMP2_FS_CFG] = TABLA_A_CDC_COMP2_FS_CFG__POR,
+ [TABLA_A_CDC_CONN_RX1_B1_CTL] = TABLA_A_CDC_CONN_RX1_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX1_B2_CTL] = TABLA_A_CDC_CONN_RX1_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_RX1_B3_CTL] = TABLA_A_CDC_CONN_RX1_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_RX2_B1_CTL] = TABLA_A_CDC_CONN_RX2_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX2_B2_CTL] = TABLA_A_CDC_CONN_RX2_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_RX2_B3_CTL] = TABLA_A_CDC_CONN_RX2_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_RX3_B1_CTL] = TABLA_A_CDC_CONN_RX3_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX3_B2_CTL] = TABLA_A_CDC_CONN_RX3_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_RX3_B3_CTL] = TABLA_A_CDC_CONN_RX3_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_RX4_B1_CTL] = TABLA_A_CDC_CONN_RX4_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX4_B2_CTL] = TABLA_A_CDC_CONN_RX4_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_RX5_B1_CTL] = TABLA_A_CDC_CONN_RX5_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX5_B2_CTL] = TABLA_A_CDC_CONN_RX5_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_RX6_B1_CTL] = TABLA_A_CDC_CONN_RX6_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX6_B2_CTL] = TABLA_A_CDC_CONN_RX6_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_RX7_B1_CTL] = TABLA_A_CDC_CONN_RX7_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX7_B2_CTL] = TABLA_A_CDC_CONN_RX7_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_ANC_B1_CTL] = TABLA_A_CDC_CONN_ANC_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_ANC_B2_CTL] = TABLA_A_CDC_CONN_ANC_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_B1_CTL] = TABLA_A_CDC_CONN_TX_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_B2_CTL] = TABLA_A_CDC_CONN_TX_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_B3_CTL] = TABLA_A_CDC_CONN_TX_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_B4_CTL] = TABLA_A_CDC_CONN_TX_B4_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ1_B1_CTL] = TABLA_A_CDC_CONN_EQ1_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ1_B2_CTL] = TABLA_A_CDC_CONN_EQ1_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ1_B3_CTL] = TABLA_A_CDC_CONN_EQ1_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ1_B4_CTL] = TABLA_A_CDC_CONN_EQ1_B4_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ2_B1_CTL] = TABLA_A_CDC_CONN_EQ2_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ2_B2_CTL] = TABLA_A_CDC_CONN_EQ2_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ2_B3_CTL] = TABLA_A_CDC_CONN_EQ2_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_EQ2_B4_CTL] = TABLA_A_CDC_CONN_EQ2_B4_CTL__POR,
+ [TABLA_A_CDC_CONN_SRC1_B1_CTL] = TABLA_A_CDC_CONN_SRC1_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_SRC1_B2_CTL] = TABLA_A_CDC_CONN_SRC1_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_SRC2_B1_CTL] = TABLA_A_CDC_CONN_SRC2_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_SRC2_B2_CTL] = TABLA_A_CDC_CONN_SRC2_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B1_CTL] = TABLA_A_CDC_CONN_TX_SB_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B2_CTL] = TABLA_A_CDC_CONN_TX_SB_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B3_CTL] = TABLA_A_CDC_CONN_TX_SB_B3_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B4_CTL] = TABLA_A_CDC_CONN_TX_SB_B4_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B5_CTL] = TABLA_A_CDC_CONN_TX_SB_B5_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B6_CTL] = TABLA_A_CDC_CONN_TX_SB_B6_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B7_CTL] = TABLA_A_CDC_CONN_TX_SB_B7_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B8_CTL] = TABLA_A_CDC_CONN_TX_SB_B8_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B9_CTL] = TABLA_A_CDC_CONN_TX_SB_B9_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B10_CTL] = TABLA_A_CDC_CONN_TX_SB_B10_CTL__POR,
+ [TABLA_A_CDC_CONN_TX_SB_B11_CTL] = TABLA_A_CDC_CONN_TX_SB_B11_CTL__POR,
+ [TABLA_A_CDC_CONN_RX_SB_B1_CTL] = TABLA_A_CDC_CONN_RX_SB_B1_CTL__POR,
+ [TABLA_A_CDC_CONN_RX_SB_B2_CTL] = TABLA_A_CDC_CONN_RX_SB_B2_CTL__POR,
+ [TABLA_A_CDC_CONN_CLSG_CTL] = TABLA_A_CDC_CONN_CLSG_CTL__POR,
+ [TABLA_A_CDC_CONN_SPARE] = TABLA_A_CDC_CONN_SPARE__POR,
+ [TABLA_A_CDC_MBHC_EN_CTL] = TABLA_A_CDC_MBHC_EN_CTL__POR,
+ [TABLA_A_CDC_MBHC_FEATURE_B1_CFG] =
+ TABLA_A_CDC_MBHC_FEATURE_B1_CFG__POR,
+ [TABLA_A_CDC_MBHC_FEATURE_B2_CFG] =
+ TABLA_A_CDC_MBHC_FEATURE_B2_CFG__POR,
+ [TABLA_A_CDC_MBHC_TIMER_B1_CTL] = TABLA_A_CDC_MBHC_TIMER_B1_CTL__POR,
+ [TABLA_A_CDC_MBHC_TIMER_B2_CTL] = TABLA_A_CDC_MBHC_TIMER_B2_CTL__POR,
+ [TABLA_A_CDC_MBHC_TIMER_B3_CTL] = TABLA_A_CDC_MBHC_TIMER_B3_CTL__POR,
+ [TABLA_A_CDC_MBHC_TIMER_B4_CTL] = TABLA_A_CDC_MBHC_TIMER_B4_CTL__POR,
+ [TABLA_A_CDC_MBHC_TIMER_B5_CTL] = TABLA_A_CDC_MBHC_TIMER_B5_CTL__POR,
+ [TABLA_A_CDC_MBHC_TIMER_B6_CTL] = TABLA_A_CDC_MBHC_TIMER_B6_CTL__POR,
+ [TABLA_A_CDC_MBHC_B1_STATUS] = TABLA_A_CDC_MBHC_B1_STATUS__POR,
+ [TABLA_A_CDC_MBHC_B2_STATUS] = TABLA_A_CDC_MBHC_B2_STATUS__POR,
+ [TABLA_A_CDC_MBHC_B3_STATUS] = TABLA_A_CDC_MBHC_B3_STATUS__POR,
+ [TABLA_A_CDC_MBHC_B4_STATUS] = TABLA_A_CDC_MBHC_B4_STATUS__POR,
+ [TABLA_A_CDC_MBHC_B5_STATUS] = TABLA_A_CDC_MBHC_B5_STATUS__POR,
+ [TABLA_A_CDC_MBHC_B1_CTL] = TABLA_A_CDC_MBHC_B1_CTL__POR,
+ [TABLA_A_CDC_MBHC_B2_CTL] = TABLA_A_CDC_MBHC_B2_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B1_CTL] = TABLA_A_CDC_MBHC_VOLT_B1_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B2_CTL] = TABLA_A_CDC_MBHC_VOLT_B2_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B3_CTL] = TABLA_A_CDC_MBHC_VOLT_B3_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B4_CTL] = TABLA_A_CDC_MBHC_VOLT_B4_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B5_CTL] = TABLA_A_CDC_MBHC_VOLT_B5_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B6_CTL] = TABLA_A_CDC_MBHC_VOLT_B6_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B7_CTL] = TABLA_A_CDC_MBHC_VOLT_B7_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B8_CTL] = TABLA_A_CDC_MBHC_VOLT_B8_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B9_CTL] = TABLA_A_CDC_MBHC_VOLT_B9_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B10_CTL] = TABLA_A_CDC_MBHC_VOLT_B10_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B11_CTL] = TABLA_A_CDC_MBHC_VOLT_B11_CTL__POR,
+ [TABLA_A_CDC_MBHC_VOLT_B12_CTL] = TABLA_A_CDC_MBHC_VOLT_B12_CTL__POR,
+ [TABLA_A_CDC_MBHC_CLK_CTL] = TABLA_A_CDC_MBHC_CLK_CTL__POR,
+ [TABLA_A_CDC_MBHC_INT_CTL] = TABLA_A_CDC_MBHC_INT_CTL__POR,
+ [TABLA_A_CDC_MBHC_DEBUG_CTL] = TABLA_A_CDC_MBHC_DEBUG_CTL__POR,
+ [TABLA_A_CDC_MBHC_SPARE] = TABLA_A_CDC_MBHC_SPARE__POR,
+};
diff --git a/sound/soc/codecs/wcd9310.c b/sound/soc/codecs/wcd9310.c
new file mode 100644
index 0000000..4b4dae1
--- /dev/null
+++ b/sound/soc/codecs/wcd9310.c
@@ -0,0 +1,7792 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/firmware.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/printk.h>
+#include <linux/ratelimit.h>
+#include <linux/debugfs.h>
+#include <linux/mfd/wcd9xxx/core.h>
+#include <linux/mfd/wcd9xxx/wcd9xxx_registers.h>
+#include <linux/mfd/wcd9xxx/wcd9310_registers.h>
+#include <linux/mfd/wcd9xxx/pdata.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <linux/bitops.h>
+#include <linux/delay.h>
+#include <linux/pm_runtime.h>
+#include <linux/kernel.h>
+#include <linux/gpio.h>
+#include "wcd9310.h"
+
+#define WCD9310_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
+
+
+#define NUM_DECIMATORS 10
+#define NUM_INTERPOLATORS 7
+#define BITS_PER_REG 8
+#define TABLA_CFILT_FAST_MODE 0x00
+#define TABLA_CFILT_SLOW_MODE 0x40
+#define MBHC_FW_READ_ATTEMPTS 15
+#define MBHC_FW_READ_TIMEOUT 2000000
+
+enum {
+ MBHC_USE_HPHL_TRIGGER = 1,
+ MBHC_USE_MB_TRIGGER = 2
+};
+
+#define MBHC_NUM_DCE_PLUG_DETECT 3
+#define NUM_ATTEMPTS_INSERT_DETECT 25
+#define NUM_ATTEMPTS_TO_REPORT 5
+
+#define TABLA_JACK_MASK (SND_JACK_HEADSET | SND_JACK_OC_HPHL | \
+ SND_JACK_OC_HPHR | SND_JACK_UNSUPPORTED)
+
+#define TABLA_I2S_MASTER_MODE_MASK 0x08
+
+#define TABLA_OCP_ATTEMPT 1
+
+#define AIF1_PB 1
+#define AIF1_CAP 2
+#define AIF2_PB 3
+#define AIF2_CAP 4
+#define AIF3_CAP 5
+#define AIF3_PB 6
+
+#define NUM_CODEC_DAIS 6
+#define TABLA_COMP_DIGITAL_GAIN_OFFSET 3
+
+struct tabla_codec_dai_data {
+ u32 rate;
+ u32 *ch_num;
+ u32 ch_act;
+ u32 ch_tot;
+};
+
+#define TABLA_MCLK_RATE_12288KHZ 12288000
+#define TABLA_MCLK_RATE_9600KHZ 9600000
+
+#define TABLA_FAKE_INS_THRESHOLD_MS 2500
+#define TABLA_FAKE_REMOVAL_MIN_PERIOD_MS 50
+
+#define TABLA_MBHC_BUTTON_MIN 0x8000
+
+#define TABLA_MBHC_FAKE_INSERT_LOW 10
+#define TABLA_MBHC_FAKE_INSERT_HIGH 80
+#define TABLA_MBHC_FAKE_INS_HIGH_NO_GPIO 150
+
+#define TABLA_MBHC_STATUS_REL_DETECTION 0x0C
+
+#define TABLA_MBHC_GPIO_REL_DEBOUNCE_TIME_MS 200
+
+#define TABLA_MBHC_FAKE_INS_DELTA_MV 200
+#define TABLA_MBHC_FAKE_INS_DELTA_SCALED_MV 300
+
+#define TABLA_HS_DETECT_PLUG_TIME_MS (5 * 1000)
+#define TABLA_HS_DETECT_PLUG_INERVAL_MS 100
+
+#define TABLA_GPIO_IRQ_DEBOUNCE_TIME_US 5000
+
+#define TABLA_MBHC_GND_MIC_SWAP_THRESHOLD 2
+
+#define TABLA_ACQUIRE_LOCK(x) do { mutex_lock(&x); } while (0)
+#define TABLA_RELEASE_LOCK(x) do { mutex_unlock(&x); } while (0)
+
+static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
+static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
+static struct snd_soc_dai_driver tabla_dai[];
+static const DECLARE_TLV_DB_SCALE(aux_pga_gain, 0, 2, 0);
+
+enum tabla_bandgap_type {
+ TABLA_BANDGAP_OFF = 0,
+ TABLA_BANDGAP_AUDIO_MODE,
+ TABLA_BANDGAP_MBHC_MODE,
+};
+
+struct mbhc_micbias_regs {
+ u16 cfilt_val;
+ u16 cfilt_ctl;
+ u16 mbhc_reg;
+ u16 int_rbias;
+ u16 ctl_reg;
+ u8 cfilt_sel;
+};
+
+/* Codec supports 2 IIR filters */
+enum {
+ IIR1 = 0,
+ IIR2,
+ IIR_MAX,
+};
+/* Codec supports 5 bands */
+enum {
+ BAND1 = 0,
+ BAND2,
+ BAND3,
+ BAND4,
+ BAND5,
+ BAND_MAX,
+};
+
+enum {
+ COMPANDER_1 = 0,
+ COMPANDER_2,
+ COMPANDER_MAX,
+};
+
+enum {
+ COMPANDER_FS_8KHZ = 0,
+ COMPANDER_FS_16KHZ,
+ COMPANDER_FS_32KHZ,
+ COMPANDER_FS_48KHZ,
+ COMPANDER_FS_96KHZ,
+ COMPANDER_FS_192KHZ,
+ COMPANDER_FS_MAX,
+};
+
+/* Flags to track of PA and DAC state.
+ * PA and DAC should be tracked separately as AUXPGA loopback requires
+ * only PA to be turned on without DAC being on. */
+enum tabla_priv_ack_flags {
+ TABLA_HPHL_PA_OFF_ACK = 0,
+ TABLA_HPHR_PA_OFF_ACK,
+ TABLA_HPHL_DAC_OFF_ACK,
+ TABLA_HPHR_DAC_OFF_ACK
+};
+
+
+struct comp_sample_dependent_params {
+ u32 peak_det_timeout;
+ u32 rms_meter_div_fact;
+ u32 rms_meter_resamp_fact;
+};
+
+/* Data used by MBHC */
+struct mbhc_internal_cal_data {
+ u16 dce_z;
+ u16 dce_mb;
+ u16 sta_z;
+ u16 sta_mb;
+ u32 t_sta_dce;
+ u32 t_dce;
+ u32 t_sta;
+ u32 micb_mv;
+ u16 v_ins_hu;
+ u16 v_ins_h;
+ u16 v_b1_hu;
+ u16 v_b1_h;
+ u16 v_b1_huc;
+ u16 v_brh;
+ u16 v_brl;
+ u16 v_no_mic;
+ u8 npoll;
+ u8 nbounce_wait;
+ s16 adj_v_hs_max;
+ u16 adj_v_ins_hu;
+ u16 adj_v_ins_h;
+ s16 v_inval_ins_low;
+ s16 v_inval_ins_high;
+};
+
+struct tabla_reg_address {
+ u16 micb_4_ctl;
+ u16 micb_4_int_rbias;
+ u16 micb_4_mbhc;
+};
+
+enum tabla_mbhc_plug_type {
+ PLUG_TYPE_INVALID = -1,
+ PLUG_TYPE_NONE,
+ PLUG_TYPE_HEADSET,
+ PLUG_TYPE_HEADPHONE,
+ PLUG_TYPE_HIGH_HPH,
+ PLUG_TYPE_GND_MIC_SWAP,
+};
+
+enum tabla_mbhc_state {
+ MBHC_STATE_NONE = -1,
+ MBHC_STATE_POTENTIAL,
+ MBHC_STATE_POTENTIAL_RECOVERY,
+ MBHC_STATE_RELEASE,
+};
+
+struct hpf_work {
+ struct tabla_priv *tabla;
+ u32 decimator;
+ u8 tx_hpf_cut_of_freq;
+ struct delayed_work dwork;
+};
+
+static struct hpf_work tx_hpf_work[NUM_DECIMATORS];
+
+struct tabla_priv {
+ struct snd_soc_codec *codec;
+ struct tabla_reg_address reg_addr;
+ u32 adc_count;
+ u32 cfilt1_cnt;
+ u32 cfilt2_cnt;
+ u32 cfilt3_cnt;
+ u32 rx_bias_count;
+ s32 dmic_1_2_clk_cnt;
+ s32 dmic_3_4_clk_cnt;
+ s32 dmic_5_6_clk_cnt;
+
+ enum tabla_bandgap_type bandgap_type;
+ bool mclk_enabled;
+ bool clock_active;
+ bool config_mode_active;
+ bool mbhc_polling_active;
+ unsigned long mbhc_fake_ins_start;
+ int buttons_pressed;
+ enum tabla_mbhc_state mbhc_state;
+ struct tabla_mbhc_config mbhc_cfg;
+ struct mbhc_internal_cal_data mbhc_data;
+
+ struct wcd9xxx_pdata *pdata;
+ u32 anc_slot;
+
+ bool no_mic_headset_override;
+ /* Delayed work to report long button press */
+ struct delayed_work mbhc_btn_dwork;
+
+ struct mbhc_micbias_regs mbhc_bias_regs;
+ bool mbhc_micbias_switched;
+
+ /* track PA/DAC state */
+ unsigned long hph_pa_dac_state;
+
+ /*track tabla interface type*/
+ u8 intf_type;
+
+ u32 hph_status; /* track headhpone status */
+ /* define separate work for left and right headphone OCP to avoid
+ * additional checking on which OCP event to report so no locking
+ * to ensure synchronization is required
+ */
+ struct work_struct hphlocp_work; /* reporting left hph ocp off */
+ struct work_struct hphrocp_work; /* reporting right hph ocp off */
+
+ u8 hphlocp_cnt; /* headphone left ocp retry */
+ u8 hphrocp_cnt; /* headphone right ocp retry */
+
+ /* Work to perform MBHC Firmware Read */
+ struct delayed_work mbhc_firmware_dwork;
+ const struct firmware *mbhc_fw;
+
+ /* num of slim ports required */
+ struct tabla_codec_dai_data dai[NUM_CODEC_DAIS];
+
+ /*compander*/
+ int comp_enabled[COMPANDER_MAX];
+ u32 comp_fs[COMPANDER_MAX];
+
+ /* Maintain the status of AUX PGA */
+ int aux_pga_cnt;
+ u8 aux_l_gain;
+ u8 aux_r_gain;
+
+ struct delayed_work mbhc_insert_dwork;
+ unsigned long mbhc_last_resume; /* in jiffies */
+
+ u8 current_plug;
+ struct work_struct hs_correct_plug_work;
+ bool hs_detect_work_stop;
+ bool hs_polling_irq_prepared;
+ bool lpi_enabled; /* low power insertion detection */
+ bool in_gpio_handler;
+ /* Currently, only used for mbhc purpose, to protect
+ * concurrent execution of mbhc threaded irq handlers and
+ * kill race between DAPM and MBHC.But can serve as a
+ * general lock to protect codec resource
+ */
+ struct mutex codec_resource_lock;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_poke;
+ struct dentry *debugfs_mbhc;
+#endif
+};
+
+static const u32 comp_shift[] = {
+ 0,
+ 2,
+};
+
+static const int comp_rx_path[] = {
+ COMPANDER_1,
+ COMPANDER_1,
+ COMPANDER_2,
+ COMPANDER_2,
+ COMPANDER_2,
+ COMPANDER_2,
+ COMPANDER_MAX,
+};
+
+static const struct comp_sample_dependent_params comp_samp_params[] = {
+ {
+ .peak_det_timeout = 0x2,
+ .rms_meter_div_fact = 0x8 << 4,
+ .rms_meter_resamp_fact = 0x21,
+ },
+ {
+ .peak_det_timeout = 0x3,
+ .rms_meter_div_fact = 0x9 << 4,
+ .rms_meter_resamp_fact = 0x28,
+ },
+
+ {
+ .peak_det_timeout = 0x5,
+ .rms_meter_div_fact = 0xB << 4,
+ .rms_meter_resamp_fact = 0x28,
+ },
+
+ {
+ .peak_det_timeout = 0x5,
+ .rms_meter_div_fact = 0xB << 4,
+ .rms_meter_resamp_fact = 0x28,
+ },
+};
+
+static unsigned short rx_digital_gain_reg[] = {
+ TABLA_A_CDC_RX1_VOL_CTL_B2_CTL,
+ TABLA_A_CDC_RX2_VOL_CTL_B2_CTL,
+ TABLA_A_CDC_RX3_VOL_CTL_B2_CTL,
+ TABLA_A_CDC_RX4_VOL_CTL_B2_CTL,
+ TABLA_A_CDC_RX5_VOL_CTL_B2_CTL,
+ TABLA_A_CDC_RX6_VOL_CTL_B2_CTL,
+ TABLA_A_CDC_RX7_VOL_CTL_B2_CTL,
+};
+
+
+static unsigned short tx_digital_gain_reg[] = {
+ TABLA_A_CDC_TX1_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX2_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX3_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX4_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX5_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX6_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX7_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX8_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX9_VOL_CTL_GAIN,
+ TABLA_A_CDC_TX10_VOL_CTL_GAIN,
+};
+
+static int tabla_codec_enable_charge_pump(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %d\n", __func__, event);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_OTHR_CTL, 0x01,
+ 0x01);
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLSG_CTL, 0x08, 0x08);
+ usleep_range(200, 200);
+ snd_soc_update_bits(codec, TABLA_A_CP_STATIC, 0x10, 0x00);
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_OTHR_RESET_CTL, 0x10,
+ 0x10);
+ usleep_range(20, 20);
+ snd_soc_update_bits(codec, TABLA_A_CP_STATIC, 0x08, 0x08);
+ snd_soc_update_bits(codec, TABLA_A_CP_STATIC, 0x10, 0x10);
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLSG_CTL, 0x08, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_OTHR_CTL, 0x01,
+ 0x00);
+ snd_soc_update_bits(codec, TABLA_A_CP_STATIC, 0x08, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static int tabla_get_anc_slot(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ ucontrol->value.integer.value[0] = tabla->anc_slot;
+ return 0;
+}
+
+static int tabla_put_anc_slot(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ tabla->anc_slot = ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int tabla_pa_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 ear_pa_gain;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ ear_pa_gain = snd_soc_read(codec, TABLA_A_RX_EAR_GAIN);
+
+ ear_pa_gain = ear_pa_gain >> 5;
+
+ if (ear_pa_gain == 0x00) {
+ ucontrol->value.integer.value[0] = 0;
+ } else if (ear_pa_gain == 0x04) {
+ ucontrol->value.integer.value[0] = 1;
+ } else {
+ pr_err("%s: ERROR: Unsupported Ear Gain = 0x%x\n",
+ __func__, ear_pa_gain);
+ return -EINVAL;
+ }
+
+ pr_debug("%s: ear_pa_gain = 0x%x\n", __func__, ear_pa_gain);
+
+ return 0;
+}
+
+static int tabla_pa_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u8 ear_pa_gain;
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s: ucontrol->value.integer.value[0] = %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ ear_pa_gain = 0x00;
+ break;
+ case 1:
+ ear_pa_gain = 0x80;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, TABLA_A_RX_EAR_GAIN, 0xE0, ear_pa_gain);
+ return 0;
+}
+
+static int tabla_get_iir_enable_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ ucontrol->value.integer.value[0] =
+ snd_soc_read(codec, (TABLA_A_CDC_IIR1_CTL + 16 * iir_idx)) &
+ (1 << band_idx);
+
+ pr_debug("%s: IIR #%d band #%d enable %d\n", __func__,
+ iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int tabla_put_iir_enable_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+ int value = ucontrol->value.integer.value[0];
+
+ /* Mask first 5 bits, 6-8 are reserved */
+ snd_soc_update_bits(codec, (TABLA_A_CDC_IIR1_CTL + 16 * iir_idx),
+ (1 << band_idx), (value << band_idx));
+
+ pr_debug("%s: IIR #%d band #%d enable %d\n", __func__,
+ iir_idx, band_idx, value);
+ return 0;
+}
+static uint32_t get_iir_band_coeff(struct snd_soc_codec *codec,
+ int iir_idx, int band_idx,
+ int coeff_idx)
+{
+ /* Address does not automatically update if reading */
+ snd_soc_write(codec,
+ (TABLA_A_CDC_IIR1_COEF_B1_CTL + 16 * iir_idx),
+ (band_idx * BAND_MAX + coeff_idx) & 0x1F);
+
+ /* Mask bits top 2 bits since they are reserved */
+ return ((snd_soc_read(codec,
+ (TABLA_A_CDC_IIR1_COEF_B2_CTL + 16 * iir_idx)) << 24) |
+ (snd_soc_read(codec,
+ (TABLA_A_CDC_IIR1_COEF_B3_CTL + 16 * iir_idx)) << 16) |
+ (snd_soc_read(codec,
+ (TABLA_A_CDC_IIR1_COEF_B4_CTL + 16 * iir_idx)) << 8) |
+ (snd_soc_read(codec,
+ (TABLA_A_CDC_IIR1_COEF_B5_CTL + 16 * iir_idx)))) &
+ 0x3FFFFFFF;
+}
+
+static int tabla_get_iir_band_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ ucontrol->value.integer.value[0] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 0);
+ ucontrol->value.integer.value[1] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 1);
+ ucontrol->value.integer.value[2] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 2);
+ ucontrol->value.integer.value[3] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 3);
+ ucontrol->value.integer.value[4] =
+ get_iir_band_coeff(codec, iir_idx, band_idx, 4);
+
+ pr_debug("%s: IIR #%d band #%d b0 = 0x%x\n"
+ "%s: IIR #%d band #%d b1 = 0x%x\n"
+ "%s: IIR #%d band #%d b2 = 0x%x\n"
+ "%s: IIR #%d band #%d a1 = 0x%x\n"
+ "%s: IIR #%d band #%d a2 = 0x%x\n",
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[0],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[1],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[2],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[3],
+ __func__, iir_idx, band_idx,
+ (uint32_t)ucontrol->value.integer.value[4]);
+ return 0;
+}
+
+static void set_iir_band_coeff(struct snd_soc_codec *codec,
+ int iir_idx, int band_idx,
+ int coeff_idx, uint32_t value)
+{
+ /* Mask top 3 bits, 6-8 are reserved */
+ /* Update address manually each time */
+ snd_soc_write(codec,
+ (TABLA_A_CDC_IIR1_COEF_B1_CTL + 16 * iir_idx),
+ (band_idx * BAND_MAX + coeff_idx) & 0x1F);
+
+ /* Mask top 2 bits, 7-8 are reserved */
+ snd_soc_write(codec,
+ (TABLA_A_CDC_IIR1_COEF_B2_CTL + 16 * iir_idx),
+ (value >> 24) & 0x3F);
+
+ /* Isolate 8bits at a time */
+ snd_soc_write(codec,
+ (TABLA_A_CDC_IIR1_COEF_B3_CTL + 16 * iir_idx),
+ (value >> 16) & 0xFF);
+
+ snd_soc_write(codec,
+ (TABLA_A_CDC_IIR1_COEF_B4_CTL + 16 * iir_idx),
+ (value >> 8) & 0xFF);
+
+ snd_soc_write(codec,
+ (TABLA_A_CDC_IIR1_COEF_B5_CTL + 16 * iir_idx),
+ value & 0xFF);
+}
+
+static int tabla_put_iir_band_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int iir_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ set_iir_band_coeff(codec, iir_idx, band_idx, 0,
+ ucontrol->value.integer.value[0]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 1,
+ ucontrol->value.integer.value[1]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 2,
+ ucontrol->value.integer.value[2]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 3,
+ ucontrol->value.integer.value[3]);
+ set_iir_band_coeff(codec, iir_idx, band_idx, 4,
+ ucontrol->value.integer.value[4]);
+
+ pr_debug("%s: IIR #%d band #%d b0 = 0x%x\n"
+ "%s: IIR #%d band #%d b1 = 0x%x\n"
+ "%s: IIR #%d band #%d b2 = 0x%x\n"
+ "%s: IIR #%d band #%d a1 = 0x%x\n"
+ "%s: IIR #%d band #%d a2 = 0x%x\n",
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 0),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 1),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 2),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 3),
+ __func__, iir_idx, band_idx,
+ get_iir_band_coeff(codec, iir_idx, band_idx, 4));
+ return 0;
+}
+
+static int tabla_compander_gain_offset(
+ struct snd_soc_codec *codec, u32 enable,
+ unsigned int reg, int mask, int event)
+{
+ int pa_mode = snd_soc_read(codec, reg) & mask;
+ int gain_offset = 0;
+ /* if PMU && enable is 1-> offset is 3
+ * if PMU && enable is 0-> offset is 0
+ * if PMD && pa_mode is PA -> offset is 0: PMU compander is off
+ * if PMD && pa_mode is comp -> offset is -3: PMU compander is on.
+ */
+
+ if (SND_SOC_DAPM_EVENT_ON(event) && (enable != 0))
+ gain_offset = TABLA_COMP_DIGITAL_GAIN_OFFSET;
+ if (SND_SOC_DAPM_EVENT_OFF(event) && (pa_mode == 0))
+ gain_offset = -TABLA_COMP_DIGITAL_GAIN_OFFSET;
+ return gain_offset;
+}
+
+
+static int tabla_config_gain_compander(
+ struct snd_soc_codec *codec,
+ u32 compander, u32 enable, int event)
+{
+ int value = 0;
+ int mask = 1 << 4;
+ int gain = 0;
+ int gain_offset;
+ if (compander >= COMPANDER_MAX) {
+ pr_err("%s: Error, invalid compander channel\n", __func__);
+ return -EINVAL;
+ }
+
+ if ((enable == 0) || SND_SOC_DAPM_EVENT_OFF(event))
+ value = 1 << 4;
+
+ if (compander == COMPANDER_1) {
+ gain_offset = tabla_compander_gain_offset(codec, enable,
+ TABLA_A_RX_HPH_L_GAIN, mask, event);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_L_GAIN, mask, value);
+ gain = snd_soc_read(codec, TABLA_A_CDC_RX1_VOL_CTL_B2_CTL);
+ snd_soc_update_bits(codec, TABLA_A_CDC_RX1_VOL_CTL_B2_CTL,
+ 0xFF, gain - gain_offset);
+ gain_offset = tabla_compander_gain_offset(codec, enable,
+ TABLA_A_RX_HPH_R_GAIN, mask, event);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_R_GAIN, mask, value);
+ gain = snd_soc_read(codec, TABLA_A_CDC_RX2_VOL_CTL_B2_CTL);
+ snd_soc_update_bits(codec, TABLA_A_CDC_RX2_VOL_CTL_B2_CTL,
+ 0xFF, gain - gain_offset);
+ } else if (compander == COMPANDER_2) {
+ gain_offset = tabla_compander_gain_offset(codec, enable,
+ TABLA_A_RX_LINE_1_GAIN, mask, event);
+ snd_soc_update_bits(codec, TABLA_A_RX_LINE_1_GAIN, mask, value);
+ gain = snd_soc_read(codec, TABLA_A_CDC_RX3_VOL_CTL_B2_CTL);
+ snd_soc_update_bits(codec, TABLA_A_CDC_RX3_VOL_CTL_B2_CTL,
+ 0xFF, gain - gain_offset);
+ gain_offset = tabla_compander_gain_offset(codec, enable,
+ TABLA_A_RX_LINE_3_GAIN, mask, event);
+ snd_soc_update_bits(codec, TABLA_A_RX_LINE_3_GAIN, mask, value);
+ gain = snd_soc_read(codec, TABLA_A_CDC_RX4_VOL_CTL_B2_CTL);
+ snd_soc_update_bits(codec, TABLA_A_CDC_RX4_VOL_CTL_B2_CTL,
+ 0xFF, gain - gain_offset);
+ gain_offset = tabla_compander_gain_offset(codec, enable,
+ TABLA_A_RX_LINE_2_GAIN, mask, event);
+ snd_soc_update_bits(codec, TABLA_A_RX_LINE_2_GAIN, mask, value);
+ gain = snd_soc_read(codec, TABLA_A_CDC_RX5_VOL_CTL_B2_CTL);
+ snd_soc_update_bits(codec, TABLA_A_CDC_RX5_VOL_CTL_B2_CTL,
+ 0xFF, gain - gain_offset);
+ gain_offset = tabla_compander_gain_offset(codec, enable,
+ TABLA_A_RX_LINE_4_GAIN, mask, event);
+ snd_soc_update_bits(codec, TABLA_A_RX_LINE_4_GAIN, mask, value);
+ gain = snd_soc_read(codec, TABLA_A_CDC_RX6_VOL_CTL_B2_CTL);
+ snd_soc_update_bits(codec, TABLA_A_CDC_RX6_VOL_CTL_B2_CTL,
+ 0xFF, gain - gain_offset);
+ }
+ return 0;
+}
+static int tabla_get_compander(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int comp = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->max;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = tabla->comp_enabled[comp];
+
+ return 0;
+}
+
+static int tabla_set_compander(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ int comp = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->max;
+ int value = ucontrol->value.integer.value[0];
+
+ if (value == tabla->comp_enabled[comp]) {
+ pr_debug("%s: compander #%d enable %d no change\n",
+ __func__, comp, value);
+ return 0;
+ }
+ tabla->comp_enabled[comp] = value;
+ return 0;
+}
+
+
+static int tabla_config_compander(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol,
+ int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u32 rate = tabla->comp_fs[w->shift];
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (tabla->comp_enabled[w->shift] != 0) {
+ /* Enable both L/R compander clocks */
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_RX_B2_CTL,
+ 0x03 << comp_shift[w->shift],
+ 0x03 << comp_shift[w->shift]);
+ /* Clar the HALT for the compander*/
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_COMP1_B1_CTL +
+ w->shift * 8, 1 << 2, 0);
+ /* Toggle compander reset bits*/
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_OTHR_RESET_CTL,
+ 0x03 << comp_shift[w->shift],
+ 0x03 << comp_shift[w->shift]);
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_OTHR_RESET_CTL,
+ 0x03 << comp_shift[w->shift], 0);
+ tabla_config_gain_compander(codec, w->shift, 1, event);
+ /* Update the RMS meter resampling*/
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_COMP1_B3_CTL +
+ w->shift * 8, 0xFF, 0x01);
+ /* Wait for 1ms*/
+ usleep_range(1000, 1000);
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ /* Set sample rate dependent paramater*/
+ if (tabla->comp_enabled[w->shift] != 0) {
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_FS_CFG +
+ w->shift * 8, 0x03, rate);
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_B2_CTL +
+ w->shift * 8, 0x0F,
+ comp_samp_params[rate].peak_det_timeout);
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_B2_CTL +
+ w->shift * 8, 0xF0,
+ comp_samp_params[rate].rms_meter_div_fact);
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_B3_CTL +
+ w->shift * 8, 0xFF,
+ comp_samp_params[rate].rms_meter_resamp_fact);
+ /* Compander enable -> 0x370/0x378*/
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_B1_CTL +
+ w->shift * 8, 0x03, 0x03);
+ }
+ break;
+ case SND_SOC_DAPM_PRE_PMD:
+ /* Halt the compander*/
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_B1_CTL +
+ w->shift * 8, 1 << 2, 1 << 2);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* Restore the gain */
+ tabla_config_gain_compander(codec, w->shift,
+ tabla->comp_enabled[w->shift], event);
+ /* Disable the compander*/
+ snd_soc_update_bits(codec, TABLA_A_CDC_COMP1_B1_CTL +
+ w->shift * 8, 0x03, 0x00);
+ /* Turn off the clock for compander in pair*/
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_RX_B2_CTL,
+ 0x03 << comp_shift[w->shift], 0);
+ break;
+ }
+ return 0;
+}
+
+static const char *tabla_ear_pa_gain_text[] = {"POS_6_DB", "POS_2_DB"};
+static const struct soc_enum tabla_ear_pa_gain_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, tabla_ear_pa_gain_text),
+};
+
+/*cut of frequency for high pass filter*/
+static const char *cf_text[] = {
+ "MIN_3DB_4Hz", "MIN_3DB_75Hz", "MIN_3DB_150Hz"
+};
+
+static const struct soc_enum cf_dec1_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX1_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec2_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX2_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec3_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX3_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec4_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX4_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec5_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX5_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec6_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX6_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec7_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX7_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec8_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX8_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec9_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX9_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_dec10_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_TX10_MUX_CTL, 4, 3, cf_text);
+
+static const struct soc_enum cf_rxmix1_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX1_B4_CTL, 1, 3, cf_text);
+
+static const struct soc_enum cf_rxmix2_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX2_B4_CTL, 1, 3, cf_text);
+
+static const struct soc_enum cf_rxmix3_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX3_B4_CTL, 1, 3, cf_text);
+
+static const struct soc_enum cf_rxmix4_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX4_B4_CTL, 1, 3, cf_text);
+
+static const struct soc_enum cf_rxmix5_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX5_B4_CTL, 1, 3, cf_text)
+;
+static const struct soc_enum cf_rxmix6_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX6_B4_CTL, 1, 3, cf_text);
+
+static const struct soc_enum cf_rxmix7_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX7_B4_CTL, 1, 3, cf_text);
+
+static const struct snd_kcontrol_new tabla_snd_controls[] = {
+
+ SOC_ENUM_EXT("EAR PA Gain", tabla_ear_pa_gain_enum[0],
+ tabla_pa_gain_get, tabla_pa_gain_put),
+
+ SOC_SINGLE_TLV("LINEOUT1 Volume", TABLA_A_RX_LINE_1_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("LINEOUT2 Volume", TABLA_A_RX_LINE_2_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("LINEOUT3 Volume", TABLA_A_RX_LINE_3_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("LINEOUT4 Volume", TABLA_A_RX_LINE_4_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("LINEOUT5 Volume", TABLA_A_RX_LINE_5_GAIN, 0, 12, 1,
+ line_gain),
+
+ SOC_SINGLE_TLV("HPHL Volume", TABLA_A_RX_HPH_L_GAIN, 0, 12, 1,
+ line_gain),
+ SOC_SINGLE_TLV("HPHR Volume", TABLA_A_RX_HPH_R_GAIN, 0, 12, 1,
+ line_gain),
+
+ SOC_SINGLE_S8_TLV("RX1 Digital Volume", TABLA_A_CDC_RX1_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX2 Digital Volume", TABLA_A_CDC_RX2_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX3 Digital Volume", TABLA_A_CDC_RX3_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX4 Digital Volume", TABLA_A_CDC_RX4_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX5 Digital Volume", TABLA_A_CDC_RX5_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX6 Digital Volume", TABLA_A_CDC_RX6_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+ SOC_SINGLE_S8_TLV("RX7 Digital Volume", TABLA_A_CDC_RX7_VOL_CTL_B2_CTL,
+ -84, 40, digital_gain),
+
+ SOC_SINGLE_S8_TLV("DEC1 Volume", TABLA_A_CDC_TX1_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC2 Volume", TABLA_A_CDC_TX2_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC3 Volume", TABLA_A_CDC_TX3_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC4 Volume", TABLA_A_CDC_TX4_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC5 Volume", TABLA_A_CDC_TX5_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC6 Volume", TABLA_A_CDC_TX6_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC7 Volume", TABLA_A_CDC_TX7_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC8 Volume", TABLA_A_CDC_TX8_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC9 Volume", TABLA_A_CDC_TX9_VOL_CTL_GAIN, -84, 40,
+ digital_gain),
+ SOC_SINGLE_S8_TLV("DEC10 Volume", TABLA_A_CDC_TX10_VOL_CTL_GAIN, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP1 Volume", TABLA_A_CDC_IIR1_GAIN_B1_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP2 Volume", TABLA_A_CDC_IIR1_GAIN_B2_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP3 Volume", TABLA_A_CDC_IIR1_GAIN_B3_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_S8_TLV("IIR1 INP4 Volume", TABLA_A_CDC_IIR1_GAIN_B4_CTL, -84,
+ 40, digital_gain),
+ SOC_SINGLE_TLV("ADC1 Volume", TABLA_A_TX_1_2_EN, 5, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC2 Volume", TABLA_A_TX_1_2_EN, 1, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC3 Volume", TABLA_A_TX_3_4_EN, 5, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC4 Volume", TABLA_A_TX_3_4_EN, 1, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC5 Volume", TABLA_A_TX_5_6_EN, 5, 3, 0, analog_gain),
+ SOC_SINGLE_TLV("ADC6 Volume", TABLA_A_TX_5_6_EN, 1, 3, 0, analog_gain),
+
+ SOC_SINGLE_TLV("AUX_PGA_LEFT Volume", TABLA_A_AUX_L_GAIN, 0, 39, 0,
+ aux_pga_gain),
+ SOC_SINGLE_TLV("AUX_PGA_RIGHT Volume", TABLA_A_AUX_R_GAIN, 0, 39, 0,
+ aux_pga_gain),
+
+ SOC_SINGLE("MICBIAS1 CAPLESS Switch", TABLA_A_MICB_1_CTL, 4, 1, 1),
+ SOC_SINGLE("MICBIAS2 CAPLESS Switch", TABLA_A_MICB_2_CTL, 4, 1, 1),
+ SOC_SINGLE("MICBIAS3 CAPLESS Switch", TABLA_A_MICB_3_CTL, 4, 1, 1),
+
+ SOC_SINGLE_EXT("ANC Slot", SND_SOC_NOPM, 0, 0, 100, tabla_get_anc_slot,
+ tabla_put_anc_slot),
+ SOC_ENUM("TX1 HPF cut off", cf_dec1_enum),
+ SOC_ENUM("TX2 HPF cut off", cf_dec2_enum),
+ SOC_ENUM("TX3 HPF cut off", cf_dec3_enum),
+ SOC_ENUM("TX4 HPF cut off", cf_dec4_enum),
+ SOC_ENUM("TX5 HPF cut off", cf_dec5_enum),
+ SOC_ENUM("TX6 HPF cut off", cf_dec6_enum),
+ SOC_ENUM("TX7 HPF cut off", cf_dec7_enum),
+ SOC_ENUM("TX8 HPF cut off", cf_dec8_enum),
+ SOC_ENUM("TX9 HPF cut off", cf_dec9_enum),
+ SOC_ENUM("TX10 HPF cut off", cf_dec10_enum),
+
+ SOC_SINGLE("TX1 HPF Switch", TABLA_A_CDC_TX1_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX2 HPF Switch", TABLA_A_CDC_TX2_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX3 HPF Switch", TABLA_A_CDC_TX3_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX4 HPF Switch", TABLA_A_CDC_TX4_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX5 HPF Switch", TABLA_A_CDC_TX5_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX6 HPF Switch", TABLA_A_CDC_TX6_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX7 HPF Switch", TABLA_A_CDC_TX7_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX8 HPF Switch", TABLA_A_CDC_TX8_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX9 HPF Switch", TABLA_A_CDC_TX9_MUX_CTL, 3, 1, 0),
+ SOC_SINGLE("TX10 HPF Switch", TABLA_A_CDC_TX10_MUX_CTL, 3, 1, 0),
+
+ SOC_SINGLE("RX1 HPF Switch", TABLA_A_CDC_RX1_B5_CTL, 2, 1, 0),
+ SOC_SINGLE("RX2 HPF Switch", TABLA_A_CDC_RX2_B5_CTL, 2, 1, 0),
+ SOC_SINGLE("RX3 HPF Switch", TABLA_A_CDC_RX3_B5_CTL, 2, 1, 0),
+ SOC_SINGLE("RX4 HPF Switch", TABLA_A_CDC_RX4_B5_CTL, 2, 1, 0),
+ SOC_SINGLE("RX5 HPF Switch", TABLA_A_CDC_RX5_B5_CTL, 2, 1, 0),
+ SOC_SINGLE("RX6 HPF Switch", TABLA_A_CDC_RX6_B5_CTL, 2, 1, 0),
+ SOC_SINGLE("RX7 HPF Switch", TABLA_A_CDC_RX7_B5_CTL, 2, 1, 0),
+
+ SOC_ENUM("RX1 HPF cut off", cf_rxmix1_enum),
+ SOC_ENUM("RX2 HPF cut off", cf_rxmix2_enum),
+ SOC_ENUM("RX3 HPF cut off", cf_rxmix3_enum),
+ SOC_ENUM("RX4 HPF cut off", cf_rxmix4_enum),
+ SOC_ENUM("RX5 HPF cut off", cf_rxmix5_enum),
+ SOC_ENUM("RX6 HPF cut off", cf_rxmix6_enum),
+ SOC_ENUM("RX7 HPF cut off", cf_rxmix7_enum),
+
+ SOC_SINGLE_EXT("IIR1 Enable Band1", IIR1, BAND1, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band2", IIR1, BAND2, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band3", IIR1, BAND3, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band4", IIR1, BAND4, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR1 Enable Band5", IIR1, BAND5, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band1", IIR2, BAND1, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band2", IIR2, BAND2, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band3", IIR2, BAND3, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band4", IIR2, BAND4, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+ SOC_SINGLE_EXT("IIR2 Enable Band5", IIR2, BAND5, 1, 0,
+ tabla_get_iir_enable_audio_mixer, tabla_put_iir_enable_audio_mixer),
+
+ SOC_SINGLE_MULTI_EXT("IIR1 Band1", IIR1, BAND1, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band2", IIR1, BAND2, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band3", IIR1, BAND3, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band4", IIR1, BAND4, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR1 Band5", IIR1, BAND5, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band1", IIR2, BAND1, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band2", IIR2, BAND2, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band3", IIR2, BAND3, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band4", IIR2, BAND4, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("IIR2 Band5", IIR2, BAND5, 255, 0, 5,
+ tabla_get_iir_band_audio_mixer, tabla_put_iir_band_audio_mixer),
+ SOC_SINGLE_EXT("COMP1 Switch", SND_SOC_NOPM, 1, COMPANDER_1, 0,
+ tabla_get_compander, tabla_set_compander),
+ SOC_SINGLE_EXT("COMP2 Switch", SND_SOC_NOPM, 0, COMPANDER_2, 0,
+ tabla_get_compander, tabla_set_compander),
+};
+
+static const struct snd_kcontrol_new tabla_1_x_snd_controls[] = {
+ SOC_SINGLE("MICBIAS4 CAPLESS Switch", TABLA_1_A_MICB_4_CTL, 4, 1, 1),
+};
+
+static const struct snd_kcontrol_new tabla_2_higher_snd_controls[] = {
+ SOC_SINGLE("MICBIAS4 CAPLESS Switch", TABLA_2_A_MICB_4_CTL, 4, 1, 1),
+};
+
+static const char *rx_mix1_text[] = {
+ "ZERO", "SRC1", "SRC2", "IIR1", "IIR2", "RX1", "RX2", "RX3", "RX4",
+ "RX5", "RX6", "RX7"
+};
+
+static const char *rx_mix2_text[] = {
+ "ZERO", "SRC1", "SRC2", "IIR1", "IIR2"
+};
+
+static const char *rx_dsm_text[] = {
+ "CIC_OUT", "DSM_INV"
+};
+
+static const char *sb_tx1_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC1"
+};
+
+static const char *sb_tx2_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC2"
+};
+
+static const char *sb_tx3_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC3"
+};
+
+static const char *sb_tx4_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC4"
+};
+
+static const char *sb_tx5_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC5"
+};
+
+static const char *sb_tx6_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC6"
+};
+
+static const char const *sb_tx7_to_tx10_mux_text[] = {
+ "ZERO", "RMIX1", "RMIX2", "RMIX3", "RMIX4", "RMIX5", "RMIX6", "RMIX7",
+ "DEC1", "DEC2", "DEC3", "DEC4", "DEC5", "DEC6", "DEC7", "DEC8",
+ "DEC9", "DEC10"
+};
+
+static const char *dec1_mux_text[] = {
+ "ZERO", "DMIC1", "ADC6",
+};
+
+static const char *dec2_mux_text[] = {
+ "ZERO", "DMIC2", "ADC5",
+};
+
+static const char *dec3_mux_text[] = {
+ "ZERO", "DMIC3", "ADC4",
+};
+
+static const char *dec4_mux_text[] = {
+ "ZERO", "DMIC4", "ADC3",
+};
+
+static const char *dec5_mux_text[] = {
+ "ZERO", "DMIC5", "ADC2",
+};
+
+static const char *dec6_mux_text[] = {
+ "ZERO", "DMIC6", "ADC1",
+};
+
+static const char const *dec7_mux_text[] = {
+ "ZERO", "DMIC1", "DMIC6", "ADC1", "ADC6", "ANC1_FB", "ANC2_FB",
+};
+
+static const char *dec8_mux_text[] = {
+ "ZERO", "DMIC2", "DMIC5", "ADC2", "ADC5",
+};
+
+static const char *dec9_mux_text[] = {
+ "ZERO", "DMIC4", "DMIC5", "ADC2", "ADC3", "ADCMB", "ANC1_FB", "ANC2_FB",
+};
+
+static const char *dec10_mux_text[] = {
+ "ZERO", "DMIC3", "DMIC6", "ADC1", "ADC4", "ADCMB", "ANC1_FB", "ANC2_FB",
+};
+
+static const char const *anc_mux_text[] = {
+ "ZERO", "ADC1", "ADC2", "ADC3", "ADC4", "ADC5", "ADC6", "ADC_MB",
+ "RSVD_1", "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5", "DMIC6"
+};
+
+static const char const *anc1_fb_mux_text[] = {
+ "ZERO", "EAR_HPH_L", "EAR_LINE_1",
+};
+
+static const char *iir1_inp1_text[] = {
+ "ZERO", "DEC1", "DEC2", "DEC3", "DEC4", "DEC5", "DEC6", "DEC7", "DEC8",
+ "DEC9", "DEC10", "RX1", "RX2", "RX3", "RX4", "RX5", "RX6", "RX7"
+};
+
+static const struct soc_enum rx_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX1_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX1_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx_mix1_inp3_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX1_B2_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx2_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX2_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx2_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX2_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx3_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX3_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx3_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX3_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx4_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX4_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx4_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX4_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx5_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX5_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx5_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX5_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx6_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX6_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx6_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX6_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx7_mix1_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX7_B1_CTL, 0, 12, rx_mix1_text);
+
+static const struct soc_enum rx7_mix1_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX7_B1_CTL, 4, 12, rx_mix1_text);
+
+static const struct soc_enum rx1_mix2_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX1_B3_CTL, 0, 5, rx_mix2_text);
+
+static const struct soc_enum rx1_mix2_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX1_B3_CTL, 3, 5, rx_mix2_text);
+
+static const struct soc_enum rx2_mix2_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX2_B3_CTL, 0, 5, rx_mix2_text);
+
+static const struct soc_enum rx2_mix2_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX2_B3_CTL, 3, 5, rx_mix2_text);
+
+static const struct soc_enum rx3_mix2_inp1_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX3_B3_CTL, 0, 5, rx_mix2_text);
+
+static const struct soc_enum rx3_mix2_inp2_chain_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_RX3_B3_CTL, 3, 5, rx_mix2_text);
+
+static const struct soc_enum rx4_dsm_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX4_B6_CTL, 4, 2, rx_dsm_text);
+
+static const struct soc_enum rx6_dsm_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_RX6_B6_CTL, 4, 2, rx_dsm_text);
+
+static const struct soc_enum sb_tx1_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B1_CTL, 0, 9, sb_tx1_mux_text);
+
+static const struct soc_enum sb_tx2_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B2_CTL, 0, 9, sb_tx2_mux_text);
+
+static const struct soc_enum sb_tx3_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B3_CTL, 0, 9, sb_tx3_mux_text);
+
+static const struct soc_enum sb_tx4_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B4_CTL, 0, 9, sb_tx4_mux_text);
+
+static const struct soc_enum sb_tx5_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B5_CTL, 0, 9, sb_tx5_mux_text);
+
+static const struct soc_enum sb_tx6_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B6_CTL, 0, 9, sb_tx6_mux_text);
+
+static const struct soc_enum sb_tx7_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B7_CTL, 0, 18,
+ sb_tx7_to_tx10_mux_text);
+
+static const struct soc_enum sb_tx8_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B8_CTL, 0, 18,
+ sb_tx7_to_tx10_mux_text);
+
+static const struct soc_enum sb_tx9_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B9_CTL, 0, 18,
+ sb_tx7_to_tx10_mux_text);
+
+static const struct soc_enum sb_tx10_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_SB_B10_CTL, 0, 18,
+ sb_tx7_to_tx10_mux_text);
+
+static const struct soc_enum dec1_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B1_CTL, 0, 3, dec1_mux_text);
+
+static const struct soc_enum dec2_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B1_CTL, 2, 3, dec2_mux_text);
+
+static const struct soc_enum dec3_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B1_CTL, 4, 3, dec3_mux_text);
+
+static const struct soc_enum dec4_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B1_CTL, 6, 3, dec4_mux_text);
+
+static const struct soc_enum dec5_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B2_CTL, 0, 3, dec5_mux_text);
+
+static const struct soc_enum dec6_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B2_CTL, 2, 3, dec6_mux_text);
+
+static const struct soc_enum dec7_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B2_CTL, 4, 7, dec7_mux_text);
+
+static const struct soc_enum dec8_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B3_CTL, 0, 7, dec8_mux_text);
+
+static const struct soc_enum dec9_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B3_CTL, 3, 8, dec9_mux_text);
+
+static const struct soc_enum dec10_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_TX_B4_CTL, 0, 8, dec10_mux_text);
+
+static const struct soc_enum anc1_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_ANC_B1_CTL, 0, 16, anc_mux_text);
+
+static const struct soc_enum anc2_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_ANC_B1_CTL, 4, 16, anc_mux_text);
+
+static const struct soc_enum anc1_fb_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_ANC_B2_CTL, 0, 3, anc1_fb_mux_text);
+
+static const struct soc_enum iir1_inp1_mux_enum =
+ SOC_ENUM_SINGLE(TABLA_A_CDC_CONN_EQ1_B1_CTL, 0, 18, iir1_inp1_text);
+
+static const struct snd_kcontrol_new rx_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX1 MIX1 INP1 Mux", rx_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX1 MIX1 INP2 Mux", rx_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx_mix1_inp3_mux =
+ SOC_DAPM_ENUM("RX1 MIX1 INP3 Mux", rx_mix1_inp3_chain_enum);
+
+static const struct snd_kcontrol_new rx2_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX2 MIX1 INP1 Mux", rx2_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx2_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX2 MIX1 INP2 Mux", rx2_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx3_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX3 MIX1 INP1 Mux", rx3_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx3_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX3 MIX1 INP2 Mux", rx3_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx4_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX4 MIX1 INP1 Mux", rx4_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx4_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX4 MIX1 INP2 Mux", rx4_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx5_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX5 MIX1 INP1 Mux", rx5_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx5_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX5 MIX1 INP2 Mux", rx5_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx6_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX6 MIX1 INP1 Mux", rx6_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx6_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX6 MIX1 INP2 Mux", rx6_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx7_mix1_inp1_mux =
+ SOC_DAPM_ENUM("RX7 MIX1 INP1 Mux", rx7_mix1_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx7_mix1_inp2_mux =
+ SOC_DAPM_ENUM("RX7 MIX1 INP2 Mux", rx7_mix1_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx1_mix2_inp1_mux =
+ SOC_DAPM_ENUM("RX1 MIX2 INP1 Mux", rx1_mix2_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx1_mix2_inp2_mux =
+ SOC_DAPM_ENUM("RX1 MIX2 INP2 Mux", rx1_mix2_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx2_mix2_inp1_mux =
+ SOC_DAPM_ENUM("RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx2_mix2_inp2_mux =
+ SOC_DAPM_ENUM("RX2 MIX2 INP2 Mux", rx2_mix2_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx3_mix2_inp1_mux =
+ SOC_DAPM_ENUM("RX3 MIX2 INP1 Mux", rx3_mix2_inp1_chain_enum);
+
+static const struct snd_kcontrol_new rx3_mix2_inp2_mux =
+ SOC_DAPM_ENUM("RX3 MIX2 INP2 Mux", rx3_mix2_inp2_chain_enum);
+
+static const struct snd_kcontrol_new rx4_dsm_mux =
+ SOC_DAPM_ENUM("RX4 DSM MUX Mux", rx4_dsm_enum);
+
+static const struct snd_kcontrol_new rx6_dsm_mux =
+ SOC_DAPM_ENUM("RX6 DSM MUX Mux", rx6_dsm_enum);
+
+static const struct snd_kcontrol_new sb_tx1_mux =
+ SOC_DAPM_ENUM("SLIM TX1 MUX Mux", sb_tx1_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx2_mux =
+ SOC_DAPM_ENUM("SLIM TX2 MUX Mux", sb_tx2_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx3_mux =
+ SOC_DAPM_ENUM("SLIM TX3 MUX Mux", sb_tx3_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx4_mux =
+ SOC_DAPM_ENUM("SLIM TX4 MUX Mux", sb_tx4_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx5_mux =
+ SOC_DAPM_ENUM("SLIM TX5 MUX Mux", sb_tx5_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx6_mux =
+ SOC_DAPM_ENUM("SLIM TX6 MUX Mux", sb_tx6_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx7_mux =
+ SOC_DAPM_ENUM("SLIM TX7 MUX Mux", sb_tx7_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx8_mux =
+ SOC_DAPM_ENUM("SLIM TX8 MUX Mux", sb_tx8_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx9_mux =
+ SOC_DAPM_ENUM("SLIM TX9 MUX Mux", sb_tx9_mux_enum);
+
+static const struct snd_kcontrol_new sb_tx10_mux =
+ SOC_DAPM_ENUM("SLIM TX10 MUX Mux", sb_tx10_mux_enum);
+
+
+static int wcd9310_put_dec_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *w = wlist->widgets[0];
+ struct snd_soc_codec *codec = w->codec;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int dec_mux, decimator;
+ char *dec_name = NULL;
+ char *widget_name = NULL;
+ char *temp;
+ u16 tx_mux_ctl_reg;
+ u8 adc_dmic_sel = 0x0;
+ int ret = 0;
+
+ if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ return -EINVAL;
+
+ dec_mux = ucontrol->value.enumerated.item[0];
+
+ widget_name = kstrndup(w->name, 15, GFP_KERNEL);
+ if (!widget_name)
+ return -ENOMEM;
+ temp = widget_name;
+
+ dec_name = strsep(&widget_name, " ");
+ widget_name = temp;
+ if (!dec_name) {
+ pr_err("%s: Invalid decimator = %s\n", __func__, w->name);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ ret = kstrtouint(strpbrk(dec_name, "123456789"), 10, &decimator);
+ if (ret < 0) {
+ pr_err("%s: Invalid decimator = %s\n", __func__, dec_name);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ dev_dbg(w->dapm->dev, "%s(): widget = %s dec_name = %s decimator = %u"
+ " dec_mux = %u\n", __func__, w->name, dec_name, decimator,
+ dec_mux);
+
+
+ switch (decimator) {
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ case 5:
+ case 6:
+ if (dec_mux == 1)
+ adc_dmic_sel = 0x1;
+ else
+ adc_dmic_sel = 0x0;
+ break;
+ case 7:
+ case 8:
+ case 9:
+ case 10:
+ if ((dec_mux == 1) || (dec_mux == 2))
+ adc_dmic_sel = 0x1;
+ else
+ adc_dmic_sel = 0x0;
+ break;
+ default:
+ pr_err("%s: Invalid Decimator = %u\n", __func__, decimator);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ tx_mux_ctl_reg = TABLA_A_CDC_TX1_MUX_CTL + 8 * (decimator - 1);
+
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x1, adc_dmic_sel);
+
+ ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol);
+
+out:
+ kfree(widget_name);
+ return ret;
+}
+
+#define WCD9310_DEC_ENUM(xname, xenum) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = snd_soc_dapm_get_enum_double, \
+ .put = wcd9310_put_dec_enum, \
+ .private_value = (unsigned long)&xenum }
+
+static const struct snd_kcontrol_new dec1_mux =
+ WCD9310_DEC_ENUM("DEC1 MUX Mux", dec1_mux_enum);
+
+static const struct snd_kcontrol_new dec2_mux =
+ WCD9310_DEC_ENUM("DEC2 MUX Mux", dec2_mux_enum);
+
+static const struct snd_kcontrol_new dec3_mux =
+ WCD9310_DEC_ENUM("DEC3 MUX Mux", dec3_mux_enum);
+
+static const struct snd_kcontrol_new dec4_mux =
+ WCD9310_DEC_ENUM("DEC4 MUX Mux", dec4_mux_enum);
+
+static const struct snd_kcontrol_new dec5_mux =
+ WCD9310_DEC_ENUM("DEC5 MUX Mux", dec5_mux_enum);
+
+static const struct snd_kcontrol_new dec6_mux =
+ WCD9310_DEC_ENUM("DEC6 MUX Mux", dec6_mux_enum);
+
+static const struct snd_kcontrol_new dec7_mux =
+ WCD9310_DEC_ENUM("DEC7 MUX Mux", dec7_mux_enum);
+
+static const struct snd_kcontrol_new dec8_mux =
+ WCD9310_DEC_ENUM("DEC8 MUX Mux", dec8_mux_enum);
+
+static const struct snd_kcontrol_new dec9_mux =
+ WCD9310_DEC_ENUM("DEC9 MUX Mux", dec9_mux_enum);
+
+static const struct snd_kcontrol_new dec10_mux =
+ WCD9310_DEC_ENUM("DEC10 MUX Mux", dec10_mux_enum);
+
+static const struct snd_kcontrol_new iir1_inp1_mux =
+ SOC_DAPM_ENUM("IIR1 INP1 Mux", iir1_inp1_mux_enum);
+
+static const struct snd_kcontrol_new anc1_mux =
+ SOC_DAPM_ENUM("ANC1 MUX Mux", anc1_mux_enum);
+
+static const struct snd_kcontrol_new anc2_mux =
+ SOC_DAPM_ENUM("ANC2 MUX Mux", anc2_mux_enum);
+
+static const struct snd_kcontrol_new anc1_fb_mux =
+ SOC_DAPM_ENUM("ANC1 FB MUX Mux", anc1_fb_mux_enum);
+
+static const struct snd_kcontrol_new dac1_switch[] = {
+ SOC_DAPM_SINGLE("Switch", TABLA_A_RX_EAR_EN, 5, 1, 0)
+};
+static const struct snd_kcontrol_new hphl_switch[] = {
+ SOC_DAPM_SINGLE("Switch", TABLA_A_RX_HPH_L_DAC_CTL, 6, 1, 0)
+};
+
+static const struct snd_kcontrol_new hphl_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 7, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 7, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 7, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new hphr_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 6, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 6, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 6, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new lineout1_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 5, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 5, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 5, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new lineout2_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 4, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 4, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 4, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new lineout3_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 3, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 3, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 3, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new lineout4_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 2, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 2, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 2, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new lineout5_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 1, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 1, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 1, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new ear_pa_mix[] = {
+ SOC_DAPM_SINGLE("AUX_PGA_L Switch", TABLA_A_AUX_L_PA_CONN,
+ 0, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R Switch", TABLA_A_AUX_R_PA_CONN,
+ 0, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_L_INV Switch",
+ TABLA_A_AUX_L_PA_CONN_INV, 0, 1, 0),
+ SOC_DAPM_SINGLE("AUX_PGA_R_INV Switch",
+ TABLA_A_AUX_R_PA_CONN_INV, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new lineout3_ground_switch =
+ SOC_DAPM_SINGLE("Switch", TABLA_A_RX_LINE_3_DAC_CTL, 6, 1, 0);
+
+static const struct snd_kcontrol_new lineout4_ground_switch =
+ SOC_DAPM_SINGLE("Switch", TABLA_A_RX_LINE_4_DAC_CTL, 6, 1, 0);
+
+static void tabla_codec_enable_adc_block(struct snd_soc_codec *codec,
+ int enable)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s %d\n", __func__, enable);
+
+ if (enable) {
+ tabla->adc_count++;
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_OTHR_CTL, 0x2, 0x2);
+ } else {
+ tabla->adc_count--;
+ if (!tabla->adc_count)
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_OTHR_CTL,
+ 0x2, 0x0);
+ }
+}
+
+static int tabla_codec_enable_adc(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 adc_reg;
+ u8 init_bit_shift;
+
+ pr_debug("%s %d\n", __func__, event);
+
+ if (w->reg == TABLA_A_TX_1_2_EN)
+ adc_reg = TABLA_A_TX_1_2_TEST_CTL;
+ else if (w->reg == TABLA_A_TX_3_4_EN)
+ adc_reg = TABLA_A_TX_3_4_TEST_CTL;
+ else if (w->reg == TABLA_A_TX_5_6_EN)
+ adc_reg = TABLA_A_TX_5_6_TEST_CTL;
+ else {
+ pr_err("%s: Error, invalid adc register\n", __func__);
+ return -EINVAL;
+ }
+
+ if (w->shift == 3)
+ init_bit_shift = 6;
+ else if (w->shift == 7)
+ init_bit_shift = 7;
+ else {
+ pr_err("%s: Error, invalid init bit postion adc register\n",
+ __func__);
+ return -EINVAL;
+ }
+
+
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ tabla_codec_enable_adc_block(codec, 1);
+ snd_soc_update_bits(codec, adc_reg, 1 << init_bit_shift,
+ 1 << init_bit_shift);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+
+ snd_soc_update_bits(codec, adc_reg, 1 << init_bit_shift, 0x00);
+
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ tabla_codec_enable_adc_block(codec, 0);
+ break;
+ }
+ return 0;
+}
+
+static void tabla_codec_enable_audio_mode_bandgap(struct snd_soc_codec *codec)
+{
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x80);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x04,
+ 0x04);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x01,
+ 0x01);
+ usleep_range(1000, 1000);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x00);
+}
+
+static void tabla_codec_enable_bandgap(struct snd_soc_codec *codec,
+ enum tabla_bandgap_type choice)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ /* TODO lock resources accessed by audio streams and threaded
+ * interrupt handlers
+ */
+
+ pr_debug("%s, choice is %d, current is %d\n", __func__, choice,
+ tabla->bandgap_type);
+
+ if (tabla->bandgap_type == choice)
+ return;
+
+ if ((tabla->bandgap_type == TABLA_BANDGAP_OFF) &&
+ (choice == TABLA_BANDGAP_AUDIO_MODE)) {
+ tabla_codec_enable_audio_mode_bandgap(codec);
+ } else if (choice == TABLA_BANDGAP_MBHC_MODE) {
+ /* bandgap mode becomes fast,
+ * mclk should be off or clk buff source souldn't be VBG
+ * Let's turn off mclk always */
+ WARN_ON(snd_soc_read(codec, TABLA_A_CLK_BUFF_EN2) & (1 << 2));
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x2,
+ 0x2);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x80);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x4,
+ 0x4);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x01,
+ 0x01);
+ usleep_range(1000, 1000);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x80,
+ 0x00);
+ } else if ((tabla->bandgap_type == TABLA_BANDGAP_MBHC_MODE) &&
+ (choice == TABLA_BANDGAP_AUDIO_MODE)) {
+ snd_soc_write(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x00);
+ usleep_range(100, 100);
+ tabla_codec_enable_audio_mode_bandgap(codec);
+ } else if (choice == TABLA_BANDGAP_OFF) {
+ snd_soc_write(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x00);
+ } else {
+ pr_err("%s: Error, Invalid bandgap settings\n", __func__);
+ }
+ tabla->bandgap_type = choice;
+}
+
+static void tabla_codec_disable_clock_block(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ pr_debug("%s\n", __func__);
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN2, 0x04, 0x00);
+ usleep_range(50, 50);
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN2, 0x02, 0x02);
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1, 0x05, 0x00);
+ usleep_range(50, 50);
+ tabla->clock_active = false;
+}
+
+static int tabla_codec_mclk_index(const struct tabla_priv *tabla)
+{
+ if (tabla->mbhc_cfg.mclk_rate == TABLA_MCLK_RATE_12288KHZ)
+ return 0;
+ else if (tabla->mbhc_cfg.mclk_rate == TABLA_MCLK_RATE_9600KHZ)
+ return 1;
+ else {
+ BUG_ON(1);
+ return -EINVAL;
+ }
+}
+
+static void tabla_enable_rx_bias(struct snd_soc_codec *codec, u32 enable)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ if (enable) {
+ tabla->rx_bias_count++;
+ if (tabla->rx_bias_count == 1)
+ snd_soc_update_bits(codec, TABLA_A_RX_COM_BIAS,
+ 0x80, 0x80);
+ } else {
+ tabla->rx_bias_count--;
+ if (!tabla->rx_bias_count)
+ snd_soc_update_bits(codec, TABLA_A_RX_COM_BIAS,
+ 0x80, 0x00);
+ }
+}
+
+static int tabla_codec_enable_config_mode(struct snd_soc_codec *codec,
+ int enable)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: enable = %d\n", __func__, enable);
+ if (enable) {
+ snd_soc_update_bits(codec, TABLA_A_CONFIG_MODE_FREQ, 0x10, 0);
+ /* bandgap mode to fast */
+ snd_soc_write(codec, TABLA_A_BIAS_CONFIG_MODE_BG_CTL, 0x17);
+ usleep_range(5, 5);
+ snd_soc_update_bits(codec, TABLA_A_CONFIG_MODE_FREQ, 0x80,
+ 0x80);
+ snd_soc_update_bits(codec, TABLA_A_CONFIG_MODE_TEST, 0x80,
+ 0x80);
+ usleep_range(10, 10);
+ snd_soc_update_bits(codec, TABLA_A_CONFIG_MODE_TEST, 0x80, 0);
+ usleep_range(10000, 10000);
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1, 0x08, 0x08);
+ } else {
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CONFIG_MODE_BG_CTL, 0x1,
+ 0);
+ snd_soc_update_bits(codec, TABLA_A_CONFIG_MODE_FREQ, 0x80, 0);
+ /* clk source to ext clk and clk buff ref to VBG */
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1, 0x0C, 0x04);
+ }
+ tabla->config_mode_active = enable ? true : false;
+
+ return 0;
+}
+
+static int tabla_codec_enable_clock_block(struct snd_soc_codec *codec,
+ int config_mode)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: config_mode = %d\n", __func__, config_mode);
+
+ /* transit to RCO requires mclk off */
+ WARN_ON(snd_soc_read(codec, TABLA_A_CLK_BUFF_EN2) & (1 << 2));
+ if (config_mode) {
+ /* enable RCO and switch to it */
+ tabla_codec_enable_config_mode(codec, 1);
+ snd_soc_write(codec, TABLA_A_CLK_BUFF_EN2, 0x02);
+ usleep_range(1000, 1000);
+ } else {
+ /* switch to MCLK */
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1, 0x08, 0x00);
+
+ if (tabla->mbhc_polling_active) {
+ snd_soc_write(codec, TABLA_A_CLK_BUFF_EN2, 0x02);
+ tabla_codec_enable_config_mode(codec, 0);
+ }
+ }
+
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1, 0x01, 0x01);
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN2, 0x02, 0x00);
+ /* on MCLK */
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN2, 0x04, 0x04);
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_MCLK_CTL, 0x01, 0x01);
+ usleep_range(50, 50);
+ tabla->clock_active = true;
+ return 0;
+}
+
+static int tabla_codec_enable_aux_pga(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_AUDIO_MODE);
+ tabla_enable_rx_bias(codec, 1);
+
+ snd_soc_update_bits(codec, TABLA_A_AUX_COM_CTL,
+ 0x08, 0x08);
+ /* Enable Zero Cross detect for AUX PGA channel
+ * and set the initial AUX PGA gain to NEG_0P0_DB
+ * to avoid glitches.
+ */
+ if (w->reg == TABLA_A_AUX_L_EN) {
+ snd_soc_update_bits(codec, TABLA_A_AUX_L_EN,
+ 0x20, 0x20);
+ tabla->aux_l_gain = snd_soc_read(codec,
+ TABLA_A_AUX_L_GAIN);
+ snd_soc_write(codec, TABLA_A_AUX_L_GAIN, 0x1F);
+ } else {
+ snd_soc_update_bits(codec, TABLA_A_AUX_R_EN,
+ 0x20, 0x20);
+ tabla->aux_r_gain = snd_soc_read(codec,
+ TABLA_A_AUX_R_GAIN);
+ snd_soc_write(codec, TABLA_A_AUX_R_GAIN, 0x1F);
+ }
+ if (tabla->aux_pga_cnt++ == 1
+ && !tabla->mclk_enabled) {
+ tabla_codec_enable_clock_block(codec, 1);
+ pr_debug("AUX PGA enabled RC osc\n");
+ }
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
+ if (w->reg == TABLA_A_AUX_L_EN)
+ snd_soc_write(codec, TABLA_A_AUX_L_GAIN,
+ tabla->aux_l_gain);
+ else
+ snd_soc_write(codec, TABLA_A_AUX_R_GAIN,
+ tabla->aux_r_gain);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ /* Mute AUX PGA channel in use before disabling AUX PGA */
+ if (w->reg == TABLA_A_AUX_L_EN) {
+ tabla->aux_l_gain = snd_soc_read(codec,
+ TABLA_A_AUX_L_GAIN);
+ snd_soc_write(codec, TABLA_A_AUX_L_GAIN, 0x1F);
+ } else {
+ tabla->aux_r_gain = snd_soc_read(codec,
+ TABLA_A_AUX_R_GAIN);
+ snd_soc_write(codec, TABLA_A_AUX_R_GAIN, 0x1F);
+ }
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ tabla_enable_rx_bias(codec, 0);
+
+ snd_soc_update_bits(codec, TABLA_A_AUX_COM_CTL,
+ 0x08, 0x00);
+ if (w->reg == TABLA_A_AUX_L_EN) {
+ snd_soc_write(codec, TABLA_A_AUX_L_GAIN,
+ tabla->aux_l_gain);
+ snd_soc_update_bits(codec, TABLA_A_AUX_L_EN,
+ 0x20, 0x00);
+ } else {
+ snd_soc_write(codec, TABLA_A_AUX_R_GAIN,
+ tabla->aux_r_gain);
+ snd_soc_update_bits(codec, TABLA_A_AUX_R_EN,
+ 0x20, 0x00);
+ }
+
+ if (tabla->aux_pga_cnt-- == 0) {
+ if (tabla->mbhc_polling_active)
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_MBHC_MODE);
+ else
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_OFF);
+
+ if (!tabla->mclk_enabled &&
+ !tabla->mbhc_polling_active) {
+ tabla_codec_enable_clock_block(codec, 0);
+ }
+ }
+ break;
+ }
+ return 0;
+}
+
+static int tabla_codec_enable_lineout(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 lineout_gain_reg;
+
+ pr_debug("%s %d %s\n", __func__, event, w->name);
+
+ switch (w->shift) {
+ case 0:
+ lineout_gain_reg = TABLA_A_RX_LINE_1_GAIN;
+ break;
+ case 1:
+ lineout_gain_reg = TABLA_A_RX_LINE_2_GAIN;
+ break;
+ case 2:
+ lineout_gain_reg = TABLA_A_RX_LINE_3_GAIN;
+ break;
+ case 3:
+ lineout_gain_reg = TABLA_A_RX_LINE_4_GAIN;
+ break;
+ case 4:
+ lineout_gain_reg = TABLA_A_RX_LINE_5_GAIN;
+ break;
+ default:
+ pr_err("%s: Error, incorrect lineout register value\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, lineout_gain_reg, 0x40, 0x40);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ pr_debug("%s: sleeping 16 ms after %s PA turn on\n",
+ __func__, w->name);
+ usleep_range(16000, 16000);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, lineout_gain_reg, 0x40, 0x00);
+ break;
+ }
+ return 0;
+}
+
+
+static int tabla_codec_enable_dmic(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u8 dmic_clk_en;
+ s32 *dmic_clk_cnt;
+ unsigned int dmic;
+ int ret;
+
+ ret = kstrtouint(strpbrk(w->name, "123456"), 10, &dmic);
+ if (ret < 0) {
+ pr_err("%s: Invalid DMIC line on the codec\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (dmic) {
+ case 1:
+ case 2:
+ dmic_clk_en = 0x01;
+ dmic_clk_cnt = &(tabla->dmic_1_2_clk_cnt);
+
+ pr_debug("%s() event %d DMIC%d dmic_1_2_clk_cnt %d\n",
+ __func__, event, dmic, *dmic_clk_cnt);
+
+ break;
+
+ case 3:
+ case 4:
+ dmic_clk_en = 0x04;
+ dmic_clk_cnt = &(tabla->dmic_3_4_clk_cnt);
+
+ pr_debug("%s() event %d DMIC%d dmic_3_4_clk_cnt %d\n",
+ __func__, event, dmic, *dmic_clk_cnt);
+ break;
+
+ case 5:
+ case 6:
+ dmic_clk_en = 0x10;
+ dmic_clk_cnt = &(tabla->dmic_5_6_clk_cnt);
+
+ pr_debug("%s() event %d DMIC%d dmic_5_6_clk_cnt %d\n",
+ __func__, event, dmic, *dmic_clk_cnt);
+
+ break;
+
+ default:
+ pr_err("%s: Invalid DMIC Selection\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+
+ (*dmic_clk_cnt)++;
+ if (*dmic_clk_cnt == 1)
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_DMIC_CTL,
+ dmic_clk_en, dmic_clk_en);
+
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+
+ (*dmic_clk_cnt)--;
+ if (*dmic_clk_cnt == 0)
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_DMIC_CTL,
+ dmic_clk_en, 0);
+ break;
+ }
+ return 0;
+}
+
+static int tabla_codec_enable_anc(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ const char *filename;
+ const struct firmware *fw;
+ int i;
+ int ret;
+ int num_anc_slots;
+ struct anc_header *anc_head;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u32 anc_writes_size = 0;
+ int anc_size_remaining;
+ u32 *anc_ptr;
+ u16 reg;
+ u8 mask, val, old_val;
+
+ pr_debug("%s %d\n", __func__, event);
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+
+ filename = "wcd9310/wcd9310_anc.bin";
+
+ ret = request_firmware(&fw, filename, codec->dev);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to acquire ANC data: %d\n",
+ ret);
+ return -ENODEV;
+ }
+
+ if (fw->size < sizeof(struct anc_header)) {
+ dev_err(codec->dev, "Not enough data\n");
+ release_firmware(fw);
+ return -ENOMEM;
+ }
+
+ /* First number is the number of register writes */
+ anc_head = (struct anc_header *)(fw->data);
+ anc_ptr = (u32 *)((u32)fw->data + sizeof(struct anc_header));
+ anc_size_remaining = fw->size - sizeof(struct anc_header);
+ num_anc_slots = anc_head->num_anc_slots;
+
+ if (tabla->anc_slot >= num_anc_slots) {
+ dev_err(codec->dev, "Invalid ANC slot selected\n");
+ release_firmware(fw);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < num_anc_slots; i++) {
+
+ if (anc_size_remaining < TABLA_PACKED_REG_SIZE) {
+ dev_err(codec->dev, "Invalid register format\n");
+ release_firmware(fw);
+ return -EINVAL;
+ }
+ anc_writes_size = (u32)(*anc_ptr);
+ anc_size_remaining -= sizeof(u32);
+ anc_ptr += 1;
+
+ if (anc_writes_size * TABLA_PACKED_REG_SIZE
+ > anc_size_remaining) {
+ dev_err(codec->dev, "Invalid register format\n");
+ release_firmware(fw);
+ return -ENOMEM;
+ }
+
+ if (tabla->anc_slot == i)
+ break;
+
+ anc_size_remaining -= (anc_writes_size *
+ TABLA_PACKED_REG_SIZE);
+ anc_ptr += anc_writes_size;
+ }
+ if (i == num_anc_slots) {
+ dev_err(codec->dev, "Selected ANC slot not present\n");
+ release_firmware(fw);
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < anc_writes_size; i++) {
+ TABLA_CODEC_UNPACK_ENTRY(anc_ptr[i], reg,
+ mask, val);
+ old_val = snd_soc_read(codec, reg);
+ snd_soc_write(codec, reg, (old_val & ~mask) |
+ (val & mask));
+ }
+ release_firmware(fw);
+
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_write(codec, TABLA_A_CDC_CLK_ANC_RESET_CTL, 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_CLK_ANC_CLK_EN_CTL, 0);
+ break;
+ }
+ return 0;
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_start_hs_polling(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ struct wcd9xxx *tabla_core = dev_get_drvdata(codec->dev->parent);
+ int mbhc_state = tabla->mbhc_state;
+
+ pr_debug("%s: enter\n", __func__);
+ if (!tabla->mbhc_polling_active) {
+ pr_debug("Polling is not active, do not start polling\n");
+ return;
+ }
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x84);
+
+ if (!tabla->no_mic_headset_override) {
+ if (mbhc_state == MBHC_STATE_POTENTIAL) {
+ pr_debug("%s recovering MBHC state macine\n", __func__);
+ tabla->mbhc_state = MBHC_STATE_POTENTIAL_RECOVERY;
+ /* set to max button press threshold */
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B2_CTL,
+ 0x7F);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B1_CTL,
+ 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B4_CTL,
+ (TABLA_IS_1_X(tabla_core->version) ?
+ 0x07 : 0x7F));
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B3_CTL,
+ 0xFF);
+ /* set to max */
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B6_CTL,
+ 0x7F);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B5_CTL,
+ 0xFF);
+ }
+ }
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x1);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x0);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x1);
+ pr_debug("%s: leave\n", __func__);
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_pause_hs_polling(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: enter\n", __func__);
+ if (!tabla->mbhc_polling_active) {
+ pr_debug("polling not active, nothing to pause\n");
+ return;
+ }
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ pr_debug("%s: leave\n", __func__);
+}
+
+static void tabla_codec_switch_cfilt_mode(struct snd_soc_codec *codec, int mode)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u8 reg_mode_val, cur_mode_val;
+ bool mbhc_was_polling = false;
+
+ if (mode)
+ reg_mode_val = TABLA_CFILT_FAST_MODE;
+ else
+ reg_mode_val = TABLA_CFILT_SLOW_MODE;
+
+ cur_mode_val = snd_soc_read(codec,
+ tabla->mbhc_bias_regs.cfilt_ctl) & 0x40;
+
+ if (cur_mode_val != reg_mode_val) {
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ if (tabla->mbhc_polling_active) {
+ tabla_codec_pause_hs_polling(codec);
+ mbhc_was_polling = true;
+ }
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.cfilt_ctl, 0x40, reg_mode_val);
+ if (mbhc_was_polling)
+ tabla_codec_start_hs_polling(codec);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ pr_debug("%s: CFILT mode change (%x to %x)\n", __func__,
+ cur_mode_val, reg_mode_val);
+ } else {
+ pr_debug("%s: CFILT Value is already %x\n",
+ __func__, cur_mode_val);
+ }
+}
+
+static void tabla_codec_update_cfilt_usage(struct snd_soc_codec *codec,
+ u8 cfilt_sel, int inc)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u32 *cfilt_cnt_ptr = NULL;
+ u16 micb_cfilt_reg;
+
+ switch (cfilt_sel) {
+ case TABLA_CFILT1_SEL:
+ cfilt_cnt_ptr = &tabla->cfilt1_cnt;
+ micb_cfilt_reg = TABLA_A_MICB_CFILT_1_CTL;
+ break;
+ case TABLA_CFILT2_SEL:
+ cfilt_cnt_ptr = &tabla->cfilt2_cnt;
+ micb_cfilt_reg = TABLA_A_MICB_CFILT_2_CTL;
+ break;
+ case TABLA_CFILT3_SEL:
+ cfilt_cnt_ptr = &tabla->cfilt3_cnt;
+ micb_cfilt_reg = TABLA_A_MICB_CFILT_3_CTL;
+ break;
+ default:
+ return; /* should not happen */
+ }
+
+ if (inc) {
+ if (!(*cfilt_cnt_ptr)++) {
+ /* Switch CFILT to slow mode if MBHC CFILT being used */
+ if (cfilt_sel == tabla->mbhc_bias_regs.cfilt_sel)
+ tabla_codec_switch_cfilt_mode(codec, 0);
+
+ snd_soc_update_bits(codec, micb_cfilt_reg, 0x80, 0x80);
+ }
+ } else {
+ /* check if count not zero, decrement
+ * then check if zero, go ahead disable cfilter
+ */
+ if ((*cfilt_cnt_ptr) && !--(*cfilt_cnt_ptr)) {
+ snd_soc_update_bits(codec, micb_cfilt_reg, 0x80, 0);
+
+ /* Switch CFILT to fast mode if MBHC CFILT being used */
+ if (cfilt_sel == tabla->mbhc_bias_regs.cfilt_sel)
+ tabla_codec_switch_cfilt_mode(codec, 1);
+ }
+ }
+}
+
+static int tabla_find_k_value(unsigned int ldoh_v, unsigned int cfilt_mv)
+{
+ int rc = -EINVAL;
+ unsigned min_mv, max_mv;
+
+ switch (ldoh_v) {
+ case TABLA_LDOH_1P95_V:
+ min_mv = 160;
+ max_mv = 1800;
+ break;
+ case TABLA_LDOH_2P35_V:
+ min_mv = 200;
+ max_mv = 2200;
+ break;
+ case TABLA_LDOH_2P75_V:
+ min_mv = 240;
+ max_mv = 2600;
+ break;
+ case TABLA_LDOH_2P85_V:
+ min_mv = 250;
+ max_mv = 2700;
+ break;
+ default:
+ goto done;
+ }
+
+ if (cfilt_mv < min_mv || cfilt_mv > max_mv)
+ goto done;
+
+ for (rc = 4; rc <= 44; rc++) {
+ min_mv = max_mv * (rc) / 44;
+ if (min_mv >= cfilt_mv) {
+ rc -= 4;
+ break;
+ }
+ }
+done:
+ return rc;
+}
+
+static bool tabla_is_hph_pa_on(struct snd_soc_codec *codec)
+{
+ u8 hph_reg_val = 0;
+ hph_reg_val = snd_soc_read(codec, TABLA_A_RX_HPH_CNP_EN);
+
+ return (hph_reg_val & 0x30) ? true : false;
+}
+
+static bool tabla_is_hph_dac_on(struct snd_soc_codec *codec, int left)
+{
+ u8 hph_reg_val = 0;
+ if (left)
+ hph_reg_val = snd_soc_read(codec,
+ TABLA_A_RX_HPH_L_DAC_CTL);
+ else
+ hph_reg_val = snd_soc_read(codec,
+ TABLA_A_RX_HPH_R_DAC_CTL);
+
+ return (hph_reg_val & 0xC0) ? true : false;
+}
+
+static void tabla_turn_onoff_override(struct snd_soc_codec *codec, bool on)
+{
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x04, on << 2);
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_drive_v_to_micbias(struct snd_soc_codec *codec,
+ int usec)
+{
+ int cfilt_k_val;
+ bool set = true;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ if (tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV &&
+ tabla->mbhc_micbias_switched) {
+ pr_debug("%s: set mic V to micbias V\n", __func__);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x2, 0x2);
+ tabla_turn_onoff_override(codec, true);
+ while (1) {
+ cfilt_k_val = tabla_find_k_value(
+ tabla->pdata->micbias.ldoh_v,
+ set ? tabla->mbhc_data.micb_mv :
+ VDDIO_MICBIAS_MV);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.cfilt_val,
+ 0xFC, (cfilt_k_val << 2));
+ if (!set)
+ break;
+ usleep_range(usec, usec);
+ set = false;
+ }
+ tabla_turn_onoff_override(codec, false);
+ }
+}
+
+/* called under codec_resource_lock acquisition */
+static void __tabla_codec_switch_micbias(struct snd_soc_codec *codec,
+ int vddio_switch, bool restartpolling,
+ bool checkpolling)
+{
+ int cfilt_k_val;
+ bool override;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ if (vddio_switch && !tabla->mbhc_micbias_switched &&
+ (!checkpolling || tabla->mbhc_polling_active)) {
+ if (restartpolling)
+ tabla_codec_pause_hs_polling(codec);
+ override = snd_soc_read(codec, TABLA_A_CDC_MBHC_B1_CTL) & 0x04;
+ if (!override)
+ tabla_turn_onoff_override(codec, true);
+ /* Adjust threshold if Mic Bias voltage changes */
+ if (tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV) {
+ cfilt_k_val = tabla_find_k_value(
+ tabla->pdata->micbias.ldoh_v,
+ VDDIO_MICBIAS_MV);
+ usleep_range(10000, 10000);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.cfilt_val,
+ 0xFC, (cfilt_k_val << 2));
+ usleep_range(10000, 10000);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B1_CTL,
+ tabla->mbhc_data.adj_v_ins_hu & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B2_CTL,
+ (tabla->mbhc_data.adj_v_ins_hu >> 8) &
+ 0xFF);
+ pr_debug("%s: Programmed MBHC thresholds to VDDIO\n",
+ __func__);
+ }
+
+ /* enable MIC BIAS Switch to VDDIO */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg,
+ 0x80, 0x80);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg,
+ 0x10, 0x00);
+ if (!override)
+ tabla_turn_onoff_override(codec, false);
+ if (restartpolling)
+ tabla_codec_start_hs_polling(codec);
+
+ tabla->mbhc_micbias_switched = true;
+ pr_debug("%s: VDDIO switch enabled\n", __func__);
+ } else if (!vddio_switch && tabla->mbhc_micbias_switched) {
+ if ((!checkpolling || tabla->mbhc_polling_active) &&
+ restartpolling)
+ tabla_codec_pause_hs_polling(codec);
+ /* Reprogram thresholds */
+ if (tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV) {
+ cfilt_k_val = tabla_find_k_value(
+ tabla->pdata->micbias.ldoh_v,
+ tabla->mbhc_data.micb_mv);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.cfilt_val,
+ 0xFC, (cfilt_k_val << 2));
+ usleep_range(10000, 10000);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B1_CTL,
+ tabla->mbhc_data.v_ins_hu & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B2_CTL,
+ (tabla->mbhc_data.v_ins_hu >> 8) & 0xFF);
+ pr_debug("%s: Programmed MBHC thresholds to MICBIAS\n",
+ __func__);
+ }
+
+ /* Disable MIC BIAS Switch to VDDIO */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg,
+ 0x80, 0x00);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg,
+ 0x10, 0x00);
+
+ if ((!checkpolling || tabla->mbhc_polling_active) &&
+ restartpolling)
+ tabla_codec_start_hs_polling(codec);
+
+ tabla->mbhc_micbias_switched = false;
+ pr_debug("%s: VDDIO switch disabled\n", __func__);
+ }
+}
+
+static void tabla_codec_switch_micbias(struct snd_soc_codec *codec,
+ int vddio_switch)
+{
+ return __tabla_codec_switch_micbias(codec, vddio_switch, true, true);
+}
+
+static int tabla_codec_enable_micbias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u16 micb_int_reg;
+ int micb_line;
+ u8 cfilt_sel_val = 0;
+ char *internal1_text = "Internal1";
+ char *internal2_text = "Internal2";
+ char *internal3_text = "Internal3";
+
+ pr_debug("%s %d\n", __func__, event);
+ switch (w->reg) {
+ case TABLA_A_MICB_1_CTL:
+ micb_int_reg = TABLA_A_MICB_1_INT_RBIAS;
+ cfilt_sel_val = tabla->pdata->micbias.bias1_cfilt_sel;
+ micb_line = TABLA_MICBIAS1;
+ break;
+ case TABLA_A_MICB_2_CTL:
+ micb_int_reg = TABLA_A_MICB_2_INT_RBIAS;
+ cfilt_sel_val = tabla->pdata->micbias.bias2_cfilt_sel;
+ micb_line = TABLA_MICBIAS2;
+ break;
+ case TABLA_A_MICB_3_CTL:
+ micb_int_reg = TABLA_A_MICB_3_INT_RBIAS;
+ cfilt_sel_val = tabla->pdata->micbias.bias3_cfilt_sel;
+ micb_line = TABLA_MICBIAS3;
+ break;
+ case TABLA_1_A_MICB_4_CTL:
+ case TABLA_2_A_MICB_4_CTL:
+ micb_int_reg = tabla->reg_addr.micb_4_int_rbias;
+ cfilt_sel_val = tabla->pdata->micbias.bias4_cfilt_sel;
+ micb_line = TABLA_MICBIAS4;
+ break;
+ default:
+ pr_err("%s: Error, invalid micbias register\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ /* Decide whether to switch the micbias for MBHC */
+ if (w->reg == tabla->mbhc_bias_regs.ctl_reg) {
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_switch_micbias(codec, 0);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ }
+
+ snd_soc_update_bits(codec, w->reg, 0x0E, 0x0A);
+ tabla_codec_update_cfilt_usage(codec, cfilt_sel_val, 1);
+
+ if (strnstr(w->name, internal1_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0xE0, 0xE0);
+ else if (strnstr(w->name, internal2_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x1C, 0x1C);
+ else if (strnstr(w->name, internal3_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x3, 0x3);
+
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+
+ usleep_range(20000, 20000);
+
+ if (tabla->mbhc_polling_active &&
+ tabla->mbhc_cfg.micbias == micb_line) {
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_pause_hs_polling(codec);
+ tabla_codec_start_hs_polling(codec);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ }
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ if ((w->reg == tabla->mbhc_bias_regs.ctl_reg) &&
+ tabla_is_hph_pa_on(codec)) {
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_switch_micbias(codec, 1);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ }
+
+ if (strnstr(w->name, internal1_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x80, 0x00);
+ else if (strnstr(w->name, internal2_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x10, 0x00);
+ else if (strnstr(w->name, internal3_text, 30))
+ snd_soc_update_bits(codec, micb_int_reg, 0x2, 0x0);
+
+ tabla_codec_update_cfilt_usage(codec, cfilt_sel_val, 0);
+ break;
+ }
+
+ return 0;
+}
+
+
+static void tx_hpf_corner_freq_callback(struct work_struct *work)
+{
+ struct delayed_work *hpf_delayed_work;
+ struct hpf_work *hpf_work;
+ struct tabla_priv *tabla;
+ struct snd_soc_codec *codec;
+ u16 tx_mux_ctl_reg;
+ u8 hpf_cut_of_freq;
+
+ hpf_delayed_work = to_delayed_work(work);
+ hpf_work = container_of(hpf_delayed_work, struct hpf_work, dwork);
+ tabla = hpf_work->tabla;
+ codec = hpf_work->tabla->codec;
+ hpf_cut_of_freq = hpf_work->tx_hpf_cut_of_freq;
+
+ tx_mux_ctl_reg = TABLA_A_CDC_TX1_MUX_CTL +
+ (hpf_work->decimator - 1) * 8;
+
+ pr_debug("%s(): decimator %u hpf_cut_of_freq 0x%x\n", __func__,
+ hpf_work->decimator, (unsigned int)hpf_cut_of_freq);
+
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x30, hpf_cut_of_freq << 4);
+}
+
+#define TX_MUX_CTL_CUT_OFF_FREQ_MASK 0x30
+#define CF_MIN_3DB_4HZ 0x0
+#define CF_MIN_3DB_75HZ 0x1
+#define CF_MIN_3DB_150HZ 0x2
+
+static int tabla_codec_enable_dec(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int decimator;
+ char *dec_name = NULL;
+ char *widget_name = NULL;
+ char *temp;
+ int ret = 0;
+ u16 dec_reset_reg, tx_vol_ctl_reg, tx_mux_ctl_reg;
+ u8 dec_hpf_cut_of_freq;
+ int offset;
+
+
+ pr_debug("%s %d\n", __func__, event);
+
+ widget_name = kstrndup(w->name, 15, GFP_KERNEL);
+ if (!widget_name)
+ return -ENOMEM;
+ temp = widget_name;
+
+ dec_name = strsep(&widget_name, " ");
+ widget_name = temp;
+ if (!dec_name) {
+ pr_err("%s: Invalid decimator = %s\n", __func__, w->name);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ ret = kstrtouint(strpbrk(dec_name, "123456789"), 10, &decimator);
+ if (ret < 0) {
+ pr_err("%s: Invalid decimator = %s\n", __func__, dec_name);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ pr_debug("%s(): widget = %s dec_name = %s decimator = %u\n", __func__,
+ w->name, dec_name, decimator);
+
+ if (w->reg == TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL) {
+ dec_reset_reg = TABLA_A_CDC_CLK_TX_RESET_B1_CTL;
+ offset = 0;
+ } else if (w->reg == TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL) {
+ dec_reset_reg = TABLA_A_CDC_CLK_TX_RESET_B2_CTL;
+ offset = 8;
+ } else {
+ pr_err("%s: Error, incorrect dec\n", __func__);
+ return -EINVAL;
+ }
+
+ tx_vol_ctl_reg = TABLA_A_CDC_TX1_VOL_CTL_CFG + 8 * (decimator -1);
+ tx_mux_ctl_reg = TABLA_A_CDC_TX1_MUX_CTL + 8 * (decimator - 1);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+
+ // Enableable TX digital mute */
+ snd_soc_update_bits(codec, tx_vol_ctl_reg, 0x01, 0x01);
+
+ snd_soc_update_bits(codec, dec_reset_reg, 1 << w->shift,
+ 1 << w->shift);
+ snd_soc_update_bits(codec, dec_reset_reg, 1 << w->shift, 0x0);
+
+ dec_hpf_cut_of_freq = snd_soc_read(codec, tx_mux_ctl_reg);
+
+ dec_hpf_cut_of_freq = (dec_hpf_cut_of_freq & 0x30) >> 4;
+
+ tx_hpf_work[decimator - 1].tx_hpf_cut_of_freq =
+ dec_hpf_cut_of_freq;
+
+ if ((dec_hpf_cut_of_freq != CF_MIN_3DB_150HZ)) {
+
+ /* set cut of freq to CF_MIN_3DB_150HZ (0x1); */
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x30,
+ CF_MIN_3DB_150HZ << 4);
+ }
+
+ /* enable HPF */
+ snd_soc_update_bits(codec, tx_mux_ctl_reg , 0x08, 0x00);
+
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
+
+ /* Disable TX digital mute */
+ snd_soc_update_bits(codec, tx_vol_ctl_reg, 0x01, 0x00);
+
+ if (tx_hpf_work[decimator - 1].tx_hpf_cut_of_freq !=
+ CF_MIN_3DB_150HZ) {
+
+ schedule_delayed_work(&tx_hpf_work[decimator - 1].dwork,
+ msecs_to_jiffies(300));
+ }
+ /* apply the digital gain after the decimator is enabled*/
+ if ((w->shift) < ARRAY_SIZE(rx_digital_gain_reg))
+ snd_soc_write(codec,
+ tx_digital_gain_reg[w->shift + offset],
+ snd_soc_read(codec,
+ tx_digital_gain_reg[w->shift + offset])
+ );
+
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+
+ snd_soc_update_bits(codec, tx_vol_ctl_reg, 0x01, 0x01);
+ cancel_delayed_work_sync(&tx_hpf_work[decimator - 1].dwork);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x08, 0x08);
+ snd_soc_update_bits(codec, tx_mux_ctl_reg, 0x30,
+ (tx_hpf_work[decimator - 1].tx_hpf_cut_of_freq) << 4);
+
+ break;
+ }
+out:
+ kfree(widget_name);
+ return ret;
+}
+
+static int tabla_codec_reset_interpolator(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %d %s\n", __func__, event, w->name);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_RX_RESET_CTL,
+ 1 << w->shift, 1 << w->shift);
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_RX_RESET_CTL,
+ 1 << w->shift, 0x0);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ /* apply the digital gain after the interpolator is enabled*/
+ if ((w->shift) < ARRAY_SIZE(rx_digital_gain_reg))
+ snd_soc_write(codec,
+ rx_digital_gain_reg[w->shift],
+ snd_soc_read(codec,
+ rx_digital_gain_reg[w->shift])
+ );
+ break;
+ }
+ return 0;
+}
+
+static int tabla_codec_enable_ldo_h(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ case SND_SOC_DAPM_POST_PMD:
+ usleep_range(1000, 1000);
+ break;
+ }
+ return 0;
+}
+
+static int tabla_codec_enable_rx_bias(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ tabla_enable_rx_bias(codec, 1);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ tabla_enable_rx_bias(codec, 0);
+ break;
+ }
+ return 0;
+}
+static int tabla_hphr_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %s %d\n", __func__, w->name, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x40);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static void tabla_snd_soc_jack_report(struct tabla_priv *tabla,
+ struct snd_soc_jack *jack, int status,
+ int mask)
+{
+ /* XXX: wake_lock_timeout()? */
+ snd_soc_jack_report_no_dapm(jack, status, mask);
+}
+
+static void hphocp_off_report(struct tabla_priv *tabla,
+ u32 jack_status, int irq)
+{
+ struct snd_soc_codec *codec;
+ if (!tabla) {
+ pr_err("%s: Bad tabla private data\n", __func__);
+ return;
+ }
+
+ pr_debug("%s: clear ocp status %x\n", __func__, jack_status);
+ codec = tabla->codec;
+ if (tabla->hph_status & jack_status) {
+ tabla->hph_status &= ~jack_status;
+ if (tabla->mbhc_cfg.headset_jack)
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.headset_jack,
+ tabla->hph_status,
+ TABLA_JACK_MASK);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10, 0x10);
+ /* reset retry counter as PA is turned off signifying
+ * start of new OCP detection session
+ */
+ if (TABLA_IRQ_HPH_PA_OCPL_FAULT)
+ tabla->hphlocp_cnt = 0;
+ else
+ tabla->hphrocp_cnt = 0;
+ wcd9xxx_enable_irq(codec->control_data, irq);
+ }
+}
+
+static void hphlocp_off_report(struct work_struct *work)
+{
+ struct tabla_priv *tabla = container_of(work, struct tabla_priv,
+ hphlocp_work);
+ hphocp_off_report(tabla, SND_JACK_OC_HPHL, TABLA_IRQ_HPH_PA_OCPL_FAULT);
+}
+
+static void hphrocp_off_report(struct work_struct *work)
+{
+ struct tabla_priv *tabla = container_of(work, struct tabla_priv,
+ hphrocp_work);
+ hphocp_off_report(tabla, SND_JACK_OC_HPHR, TABLA_IRQ_HPH_PA_OCPR_FAULT);
+}
+
+static int tabla_hph_pa_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u8 mbhc_micb_ctl_val;
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ mbhc_micb_ctl_val = snd_soc_read(codec,
+ tabla->mbhc_bias_regs.ctl_reg);
+
+ if (!(mbhc_micb_ctl_val & 0x80)) {
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_switch_micbias(codec, 1);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ }
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ /* schedule work is required because at the time HPH PA DAPM
+ * event callback is called by DAPM framework, CODEC dapm mutex
+ * would have been locked while snd_soc_jack_report also
+ * attempts to acquire same lock.
+ */
+ if (w->shift == 5) {
+ clear_bit(TABLA_HPHL_PA_OFF_ACK,
+ &tabla->hph_pa_dac_state);
+ clear_bit(TABLA_HPHL_DAC_OFF_ACK,
+ &tabla->hph_pa_dac_state);
+ if (tabla->hph_status & SND_JACK_OC_HPHL)
+ schedule_work(&tabla->hphlocp_work);
+ } else if (w->shift == 4) {
+ clear_bit(TABLA_HPHR_PA_OFF_ACK,
+ &tabla->hph_pa_dac_state);
+ clear_bit(TABLA_HPHR_DAC_OFF_ACK,
+ &tabla->hph_pa_dac_state);
+ if (tabla->hph_status & SND_JACK_OC_HPHR)
+ schedule_work(&tabla->hphrocp_work);
+ }
+
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_switch_micbias(codec, 0);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+
+ pr_debug("%s: sleep 10 ms after %s PA disable.\n", __func__,
+ w->name);
+ usleep_range(10000, 10000);
+ break;
+ }
+ return 0;
+}
+
+static void tabla_get_mbhc_micbias_regs(struct snd_soc_codec *codec,
+ struct mbhc_micbias_regs *micbias_regs)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ unsigned int cfilt;
+
+ switch (tabla->mbhc_cfg.micbias) {
+ case TABLA_MICBIAS1:
+ cfilt = tabla->pdata->micbias.bias1_cfilt_sel;
+ micbias_regs->mbhc_reg = TABLA_A_MICB_1_MBHC;
+ micbias_regs->int_rbias = TABLA_A_MICB_1_INT_RBIAS;
+ micbias_regs->ctl_reg = TABLA_A_MICB_1_CTL;
+ break;
+ case TABLA_MICBIAS2:
+ cfilt = tabla->pdata->micbias.bias2_cfilt_sel;
+ micbias_regs->mbhc_reg = TABLA_A_MICB_2_MBHC;
+ micbias_regs->int_rbias = TABLA_A_MICB_2_INT_RBIAS;
+ micbias_regs->ctl_reg = TABLA_A_MICB_2_CTL;
+ break;
+ case TABLA_MICBIAS3:
+ cfilt = tabla->pdata->micbias.bias3_cfilt_sel;
+ micbias_regs->mbhc_reg = TABLA_A_MICB_3_MBHC;
+ micbias_regs->int_rbias = TABLA_A_MICB_3_INT_RBIAS;
+ micbias_regs->ctl_reg = TABLA_A_MICB_3_CTL;
+ break;
+ case TABLA_MICBIAS4:
+ cfilt = tabla->pdata->micbias.bias4_cfilt_sel;
+ micbias_regs->mbhc_reg = tabla->reg_addr.micb_4_mbhc;
+ micbias_regs->int_rbias = tabla->reg_addr.micb_4_int_rbias;
+ micbias_regs->ctl_reg = tabla->reg_addr.micb_4_ctl;
+ break;
+ default:
+ /* Should never reach here */
+ pr_err("%s: Invalid MIC BIAS for MBHC\n", __func__);
+ return;
+ }
+
+ micbias_regs->cfilt_sel = cfilt;
+
+ switch (cfilt) {
+ case TABLA_CFILT1_SEL:
+ micbias_regs->cfilt_val = TABLA_A_MICB_CFILT_1_VAL;
+ micbias_regs->cfilt_ctl = TABLA_A_MICB_CFILT_1_CTL;
+ tabla->mbhc_data.micb_mv = tabla->pdata->micbias.cfilt1_mv;
+ break;
+ case TABLA_CFILT2_SEL:
+ micbias_regs->cfilt_val = TABLA_A_MICB_CFILT_2_VAL;
+ micbias_regs->cfilt_ctl = TABLA_A_MICB_CFILT_2_CTL;
+ tabla->mbhc_data.micb_mv = tabla->pdata->micbias.cfilt2_mv;
+ break;
+ case TABLA_CFILT3_SEL:
+ micbias_regs->cfilt_val = TABLA_A_MICB_CFILT_3_VAL;
+ micbias_regs->cfilt_ctl = TABLA_A_MICB_CFILT_3_CTL;
+ tabla->mbhc_data.micb_mv = tabla->pdata->micbias.cfilt3_mv;
+ break;
+ }
+}
+static const struct snd_soc_dapm_widget tabla_dapm_i2s_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("RX_I2S_CLK", TABLA_A_CDC_CLK_RX_I2S_CTL,
+ 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("TX_I2S_CLK", TABLA_A_CDC_CLK_TX_I2S_CTL, 4,
+ 0, NULL, 0),
+};
+
+static int tabla_lineout_dac_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+
+ pr_debug("%s %s %d\n", __func__, w->name, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x40);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_update_bits(codec, w->reg, 0x40, 0x00);
+ break;
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget tabla_1_x_dapm_widgets[] = {
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS4 External", TABLA_1_A_MICB_4_CTL, 7,
+ 0, tabla_codec_enable_micbias,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_widget tabla_2_higher_dapm_widgets[] = {
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS4 External", TABLA_2_A_MICB_4_CTL, 7,
+ 0, tabla_codec_enable_micbias,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_i2s_map[] = {
+ {"RX_I2S_CLK", NULL, "CDC_CONN"},
+ {"SLIM RX1", NULL, "RX_I2S_CLK"},
+ {"SLIM RX2", NULL, "RX_I2S_CLK"},
+ {"SLIM RX3", NULL, "RX_I2S_CLK"},
+ {"SLIM RX4", NULL, "RX_I2S_CLK"},
+
+ {"SLIM TX7", NULL, "TX_I2S_CLK"},
+ {"SLIM TX8", NULL, "TX_I2S_CLK"},
+ {"SLIM TX9", NULL, "TX_I2S_CLK"},
+ {"SLIM TX10", NULL, "TX_I2S_CLK"},
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* SLIMBUS Connections */
+
+ {"SLIM TX1", NULL, "SLIM TX1 MUX"},
+ {"SLIM TX1 MUX", "DEC1", "DEC1 MUX"},
+
+ {"SLIM TX2", NULL, "SLIM TX2 MUX"},
+ {"SLIM TX2 MUX", "DEC2", "DEC2 MUX"},
+
+ {"SLIM TX3", NULL, "SLIM TX3 MUX"},
+ {"SLIM TX3 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX3 MUX", "RMIX1", "RX1 MIX1"},
+ {"SLIM TX3 MUX", "RMIX2", "RX2 MIX1"},
+ {"SLIM TX3 MUX", "RMIX3", "RX3 MIX1"},
+ {"SLIM TX3 MUX", "RMIX4", "RX4 MIX1"},
+ {"SLIM TX3 MUX", "RMIX5", "RX5 MIX1"},
+ {"SLIM TX3 MUX", "RMIX6", "RX6 MIX1"},
+ {"SLIM TX3 MUX", "RMIX7", "RX7 MIX1"},
+
+ {"SLIM TX4", NULL, "SLIM TX4 MUX"},
+ {"SLIM TX4 MUX", "DEC4", "DEC4 MUX"},
+
+ {"SLIM TX5", NULL, "SLIM TX5 MUX"},
+ {"SLIM TX5 MUX", "DEC5", "DEC5 MUX"},
+ {"SLIM TX5 MUX", "RMIX1", "RX1 MIX1"},
+ {"SLIM TX5 MUX", "RMIX2", "RX2 MIX1"},
+ {"SLIM TX5 MUX", "RMIX3", "RX3 MIX1"},
+ {"SLIM TX5 MUX", "RMIX4", "RX4 MIX1"},
+ {"SLIM TX5 MUX", "RMIX5", "RX5 MIX1"},
+ {"SLIM TX5 MUX", "RMIX6", "RX6 MIX1"},
+ {"SLIM TX5 MUX", "RMIX7", "RX7 MIX1"},
+
+ {"SLIM TX6", NULL, "SLIM TX6 MUX"},
+ {"SLIM TX6 MUX", "DEC6", "DEC6 MUX"},
+
+ {"SLIM TX7", NULL, "SLIM TX7 MUX"},
+ {"SLIM TX7 MUX", "DEC1", "DEC1 MUX"},
+ {"SLIM TX7 MUX", "DEC2", "DEC2 MUX"},
+ {"SLIM TX7 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX7 MUX", "DEC4", "DEC4 MUX"},
+ {"SLIM TX7 MUX", "DEC5", "DEC5 MUX"},
+ {"SLIM TX7 MUX", "DEC6", "DEC6 MUX"},
+ {"SLIM TX7 MUX", "DEC7", "DEC7 MUX"},
+ {"SLIM TX7 MUX", "DEC8", "DEC8 MUX"},
+ {"SLIM TX7 MUX", "DEC9", "DEC9 MUX"},
+ {"SLIM TX7 MUX", "DEC10", "DEC10 MUX"},
+ {"SLIM TX7 MUX", "RMIX1", "RX1 MIX1"},
+ {"SLIM TX7 MUX", "RMIX2", "RX2 MIX1"},
+ {"SLIM TX7 MUX", "RMIX3", "RX3 MIX1"},
+ {"SLIM TX7 MUX", "RMIX4", "RX4 MIX1"},
+ {"SLIM TX7 MUX", "RMIX5", "RX5 MIX1"},
+ {"SLIM TX7 MUX", "RMIX6", "RX6 MIX1"},
+ {"SLIM TX7 MUX", "RMIX7", "RX7 MIX1"},
+
+ {"SLIM TX8", NULL, "SLIM TX8 MUX"},
+ {"SLIM TX8 MUX", "DEC1", "DEC1 MUX"},
+ {"SLIM TX8 MUX", "DEC2", "DEC2 MUX"},
+ {"SLIM TX8 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX8 MUX", "DEC4", "DEC4 MUX"},
+ {"SLIM TX8 MUX", "DEC5", "DEC5 MUX"},
+ {"SLIM TX8 MUX", "DEC6", "DEC6 MUX"},
+ {"SLIM TX8 MUX", "DEC7", "DEC7 MUX"},
+ {"SLIM TX8 MUX", "DEC8", "DEC8 MUX"},
+ {"SLIM TX8 MUX", "DEC9", "DEC9 MUX"},
+ {"SLIM TX8 MUX", "DEC10", "DEC10 MUX"},
+
+ {"SLIM TX9", NULL, "SLIM TX9 MUX"},
+ {"SLIM TX9 MUX", "DEC1", "DEC1 MUX"},
+ {"SLIM TX9 MUX", "DEC2", "DEC2 MUX"},
+ {"SLIM TX9 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX9 MUX", "DEC4", "DEC4 MUX"},
+ {"SLIM TX9 MUX", "DEC5", "DEC5 MUX"},
+ {"SLIM TX9 MUX", "DEC6", "DEC6 MUX"},
+ {"SLIM TX9 MUX", "DEC7", "DEC7 MUX"},
+ {"SLIM TX9 MUX", "DEC8", "DEC8 MUX"},
+ {"SLIM TX9 MUX", "DEC9", "DEC9 MUX"},
+ {"SLIM TX9 MUX", "DEC10", "DEC10 MUX"},
+
+ {"SLIM TX10", NULL, "SLIM TX10 MUX"},
+ {"SLIM TX10 MUX", "DEC1", "DEC1 MUX"},
+ {"SLIM TX10 MUX", "DEC2", "DEC2 MUX"},
+ {"SLIM TX10 MUX", "DEC3", "DEC3 MUX"},
+ {"SLIM TX10 MUX", "DEC4", "DEC4 MUX"},
+ {"SLIM TX10 MUX", "DEC5", "DEC5 MUX"},
+ {"SLIM TX10 MUX", "DEC6", "DEC6 MUX"},
+ {"SLIM TX10 MUX", "DEC7", "DEC7 MUX"},
+ {"SLIM TX10 MUX", "DEC8", "DEC8 MUX"},
+ {"SLIM TX10 MUX", "DEC9", "DEC9 MUX"},
+ {"SLIM TX10 MUX", "DEC10", "DEC10 MUX"},
+
+ /* Earpiece (RX MIX1) */
+ {"EAR", NULL, "EAR PA"},
+ {"EAR PA", NULL, "EAR_PA_MIXER"},
+ {"EAR_PA_MIXER", NULL, "DAC1"},
+ {"DAC1", NULL, "CP"},
+
+ {"ANC1 FB MUX", "EAR_HPH_L", "RX1 MIX2"},
+ {"ANC1 FB MUX", "EAR_LINE_1", "RX2 MIX2"},
+ {"ANC", NULL, "ANC1 FB MUX"},
+
+ /* Headset (RX MIX1 and RX MIX2) */
+ {"HEADPHONE", NULL, "HPHL"},
+ {"HEADPHONE", NULL, "HPHR"},
+
+ {"HPHL", NULL, "HPHL_PA_MIXER"},
+ {"HPHL_PA_MIXER", NULL, "HPHL DAC"},
+
+ {"HPHR", NULL, "HPHR_PA_MIXER"},
+ {"HPHR_PA_MIXER", NULL, "HPHR DAC"},
+
+ {"HPHL DAC", NULL, "CP"},
+ {"HPHR DAC", NULL, "CP"},
+
+ {"ANC", NULL, "ANC1 MUX"},
+ {"ANC", NULL, "ANC2 MUX"},
+ {"ANC1 MUX", "ADC1", "ADC1"},
+ {"ANC1 MUX", "ADC2", "ADC2"},
+ {"ANC1 MUX", "ADC3", "ADC3"},
+ {"ANC1 MUX", "ADC4", "ADC4"},
+ {"ANC2 MUX", "ADC1", "ADC1"},
+ {"ANC2 MUX", "ADC2", "ADC2"},
+ {"ANC2 MUX", "ADC3", "ADC3"},
+ {"ANC2 MUX", "ADC4", "ADC4"},
+
+ {"ANC", NULL, "CDC_CONN"},
+
+ {"DAC1", "Switch", "RX1 CHAIN"},
+ {"HPHL DAC", "Switch", "RX1 CHAIN"},
+ {"HPHR DAC", NULL, "RX2 CHAIN"},
+
+ {"LINEOUT1", NULL, "LINEOUT1 PA"},
+ {"LINEOUT2", NULL, "LINEOUT2 PA"},
+ {"LINEOUT3", NULL, "LINEOUT3 PA"},
+ {"LINEOUT4", NULL, "LINEOUT4 PA"},
+ {"LINEOUT5", NULL, "LINEOUT5 PA"},
+
+ {"LINEOUT1 PA", NULL, "LINEOUT1_PA_MIXER"},
+ {"LINEOUT1_PA_MIXER", NULL, "LINEOUT1 DAC"},
+ {"LINEOUT2 PA", NULL, "LINEOUT2_PA_MIXER"},
+ {"LINEOUT2_PA_MIXER", NULL, "LINEOUT2 DAC"},
+ {"LINEOUT3 PA", NULL, "LINEOUT3_PA_MIXER"},
+ {"LINEOUT3_PA_MIXER", NULL, "LINEOUT3 DAC"},
+ {"LINEOUT4 PA", NULL, "LINEOUT4_PA_MIXER"},
+ {"LINEOUT4_PA_MIXER", NULL, "LINEOUT4 DAC"},
+ {"LINEOUT5 PA", NULL, "LINEOUT5_PA_MIXER"},
+ {"LINEOUT5_PA_MIXER", NULL, "LINEOUT5 DAC"},
+
+ {"LINEOUT1 DAC", NULL, "RX3 MIX2"},
+ {"LINEOUT5 DAC", NULL, "RX7 MIX1"},
+
+ {"RX1 CHAIN", NULL, "RX1 MIX2"},
+ {"RX2 CHAIN", NULL, "RX2 MIX2"},
+ {"RX1 CHAIN", NULL, "ANC"},
+ {"RX2 CHAIN", NULL, "ANC"},
+
+ {"CP", NULL, "RX_BIAS"},
+ {"LINEOUT1 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT2 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT3 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT4 DAC", NULL, "RX_BIAS"},
+ {"LINEOUT5 DAC", NULL, "RX_BIAS"},
+
+ {"RX1 MIX1", NULL, "COMP1_CLK"},
+ {"RX2 MIX1", NULL, "COMP1_CLK"},
+ {"RX3 MIX1", NULL, "COMP2_CLK"},
+ {"RX5 MIX1", NULL, "COMP2_CLK"},
+
+
+ {"RX1 MIX1", NULL, "RX1 MIX1 INP1"},
+ {"RX1 MIX1", NULL, "RX1 MIX1 INP2"},
+ {"RX1 MIX1", NULL, "RX1 MIX1 INP3"},
+ {"RX2 MIX1", NULL, "RX2 MIX1 INP1"},
+ {"RX2 MIX1", NULL, "RX2 MIX1 INP2"},
+ {"RX3 MIX1", NULL, "RX3 MIX1 INP1"},
+ {"RX3 MIX1", NULL, "RX3 MIX1 INP2"},
+ {"RX4 MIX1", NULL, "RX4 MIX1 INP1"},
+ {"RX4 MIX1", NULL, "RX4 MIX1 INP2"},
+ {"RX5 MIX1", NULL, "RX5 MIX1 INP1"},
+ {"RX5 MIX1", NULL, "RX5 MIX1 INP2"},
+ {"RX6 MIX1", NULL, "RX6 MIX1 INP1"},
+ {"RX6 MIX1", NULL, "RX6 MIX1 INP2"},
+ {"RX7 MIX1", NULL, "RX7 MIX1 INP1"},
+ {"RX7 MIX1", NULL, "RX7 MIX1 INP2"},
+ {"RX1 MIX2", NULL, "RX1 MIX1"},
+ {"RX1 MIX2", NULL, "RX1 MIX2 INP1"},
+ {"RX1 MIX2", NULL, "RX1 MIX2 INP2"},
+ {"RX2 MIX2", NULL, "RX2 MIX1"},
+ {"RX2 MIX2", NULL, "RX2 MIX2 INP1"},
+ {"RX2 MIX2", NULL, "RX2 MIX2 INP2"},
+ {"RX3 MIX2", NULL, "RX3 MIX1"},
+ {"RX3 MIX2", NULL, "RX3 MIX2 INP1"},
+ {"RX3 MIX2", NULL, "RX3 MIX2 INP2"},
+
+ {"RX1 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX1 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX1 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX1 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX1 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX1 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX1 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX1 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX1 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX1 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX1 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX1 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX1 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX1 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX1 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX1 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX1 MIX1 INP3", "RX1", "SLIM RX1"},
+ {"RX1 MIX1 INP3", "RX2", "SLIM RX2"},
+ {"RX1 MIX1 INP3", "RX3", "SLIM RX3"},
+ {"RX1 MIX1 INP3", "RX4", "SLIM RX4"},
+ {"RX1 MIX1 INP3", "RX5", "SLIM RX5"},
+ {"RX1 MIX1 INP3", "RX6", "SLIM RX6"},
+ {"RX1 MIX1 INP3", "RX7", "SLIM RX7"},
+ {"RX2 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX2 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX2 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX2 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX2 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX2 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX2 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX2 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX2 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX2 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX2 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX2 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX2 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX2 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX2 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX2 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX3 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX3 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX3 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX3 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX3 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX3 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX3 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX3 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX3 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX3 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX3 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX3 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX3 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX3 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX3 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX3 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX4 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX4 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX4 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX4 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX4 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX4 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX4 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX4 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX4 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX4 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX4 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX4 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX4 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX4 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX4 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX4 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX5 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX5 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX5 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX5 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX5 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX5 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX5 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX5 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX5 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX5 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX5 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX5 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX5 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX5 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX5 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX5 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX6 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX6 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX6 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX6 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX6 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX6 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX6 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX6 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX6 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX6 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX6 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX6 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX6 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX6 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX6 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX6 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX7 MIX1 INP1", "RX1", "SLIM RX1"},
+ {"RX7 MIX1 INP1", "RX2", "SLIM RX2"},
+ {"RX7 MIX1 INP1", "RX3", "SLIM RX3"},
+ {"RX7 MIX1 INP1", "RX4", "SLIM RX4"},
+ {"RX7 MIX1 INP1", "RX5", "SLIM RX5"},
+ {"RX7 MIX1 INP1", "RX6", "SLIM RX6"},
+ {"RX7 MIX1 INP1", "RX7", "SLIM RX7"},
+ {"RX7 MIX1 INP1", "IIR1", "IIR1"},
+ {"RX7 MIX1 INP2", "RX1", "SLIM RX1"},
+ {"RX7 MIX1 INP2", "RX2", "SLIM RX2"},
+ {"RX7 MIX1 INP2", "RX3", "SLIM RX3"},
+ {"RX7 MIX1 INP2", "RX4", "SLIM RX4"},
+ {"RX7 MIX1 INP2", "RX5", "SLIM RX5"},
+ {"RX7 MIX1 INP2", "RX6", "SLIM RX6"},
+ {"RX7 MIX1 INP2", "RX7", "SLIM RX7"},
+ {"RX7 MIX1 INP2", "IIR1", "IIR1"},
+ {"RX1 MIX2 INP1", "IIR1", "IIR1"},
+ {"RX1 MIX2 INP2", "IIR1", "IIR1"},
+ {"RX2 MIX2 INP1", "IIR1", "IIR1"},
+ {"RX2 MIX2 INP2", "IIR1", "IIR1"},
+ {"RX3 MIX2 INP1", "IIR1", "IIR1"},
+ {"RX3 MIX2 INP2", "IIR1", "IIR1"},
+
+ /* Decimator Inputs */
+ {"DEC1 MUX", "DMIC1", "DMIC1"},
+ {"DEC1 MUX", "ADC6", "ADC6"},
+ {"DEC1 MUX", NULL, "CDC_CONN"},
+ {"DEC2 MUX", "DMIC2", "DMIC2"},
+ {"DEC2 MUX", "ADC5", "ADC5"},
+ {"DEC2 MUX", NULL, "CDC_CONN"},
+ {"DEC3 MUX", "DMIC3", "DMIC3"},
+ {"DEC3 MUX", "ADC4", "ADC4"},
+ {"DEC3 MUX", NULL, "CDC_CONN"},
+ {"DEC4 MUX", "DMIC4", "DMIC4"},
+ {"DEC4 MUX", "ADC3", "ADC3"},
+ {"DEC4 MUX", NULL, "CDC_CONN"},
+ {"DEC5 MUX", "DMIC5", "DMIC5"},
+ {"DEC5 MUX", "ADC2", "ADC2"},
+ {"DEC5 MUX", NULL, "CDC_CONN"},
+ {"DEC6 MUX", "DMIC6", "DMIC6"},
+ {"DEC6 MUX", "ADC1", "ADC1"},
+ {"DEC6 MUX", NULL, "CDC_CONN"},
+ {"DEC7 MUX", "DMIC1", "DMIC1"},
+ {"DEC7 MUX", "DMIC6", "DMIC6"},
+ {"DEC7 MUX", "ADC1", "ADC1"},
+ {"DEC7 MUX", "ADC6", "ADC6"},
+ {"DEC7 MUX", NULL, "CDC_CONN"},
+ {"DEC8 MUX", "DMIC2", "DMIC2"},
+ {"DEC8 MUX", "DMIC5", "DMIC5"},
+ {"DEC8 MUX", "ADC2", "ADC2"},
+ {"DEC8 MUX", "ADC5", "ADC5"},
+ {"DEC8 MUX", NULL, "CDC_CONN"},
+ {"DEC9 MUX", "DMIC4", "DMIC4"},
+ {"DEC9 MUX", "DMIC5", "DMIC5"},
+ {"DEC9 MUX", "ADC2", "ADC2"},
+ {"DEC9 MUX", "ADC3", "ADC3"},
+ {"DEC9 MUX", NULL, "CDC_CONN"},
+ {"DEC10 MUX", "DMIC3", "DMIC3"},
+ {"DEC10 MUX", "DMIC6", "DMIC6"},
+ {"DEC10 MUX", "ADC1", "ADC1"},
+ {"DEC10 MUX", "ADC4", "ADC4"},
+ {"DEC10 MUX", NULL, "CDC_CONN"},
+
+ /* ADC Connections */
+ {"ADC1", NULL, "AMIC1"},
+ {"ADC2", NULL, "AMIC2"},
+ {"ADC3", NULL, "AMIC3"},
+ {"ADC4", NULL, "AMIC4"},
+ {"ADC5", NULL, "AMIC5"},
+ {"ADC6", NULL, "AMIC6"},
+
+ /* AUX PGA Connections */
+ {"HPHL_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"HPHL_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"HPHL_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"HPHL_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"HPHR_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"HPHR_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"HPHR_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"HPHR_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"LINEOUT1_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"LINEOUT1_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"LINEOUT1_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"LINEOUT1_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"LINEOUT2_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"LINEOUT2_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"LINEOUT2_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"LINEOUT2_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"LINEOUT3_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"LINEOUT3_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"LINEOUT3_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"LINEOUT3_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"LINEOUT4_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"LINEOUT4_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"LINEOUT4_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"LINEOUT4_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"LINEOUT5_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"LINEOUT5_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"LINEOUT5_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"LINEOUT5_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"EAR_PA_MIXER", "AUX_PGA_L Switch", "AUX_PGA_Left"},
+ {"EAR_PA_MIXER", "AUX_PGA_R Switch", "AUX_PGA_Right"},
+ {"EAR_PA_MIXER", "AUX_PGA_L_INV Switch", "AUX_PGA_Left"},
+ {"EAR_PA_MIXER", "AUX_PGA_R_INV Switch", "AUX_PGA_Right"},
+ {"AUX_PGA_Left", NULL, "AMIC5"},
+ {"AUX_PGA_Right", NULL, "AMIC6"},
+
+ {"IIR1", NULL, "IIR1 INP1 MUX"},
+ {"IIR1 INP1 MUX", "DEC1", "DEC1 MUX"},
+ {"IIR1 INP1 MUX", "DEC2", "DEC2 MUX"},
+ {"IIR1 INP1 MUX", "DEC3", "DEC3 MUX"},
+ {"IIR1 INP1 MUX", "DEC4", "DEC4 MUX"},
+ {"IIR1 INP1 MUX", "DEC5", "DEC5 MUX"},
+ {"IIR1 INP1 MUX", "DEC6", "DEC6 MUX"},
+ {"IIR1 INP1 MUX", "DEC7", "DEC7 MUX"},
+ {"IIR1 INP1 MUX", "DEC8", "DEC8 MUX"},
+ {"IIR1 INP1 MUX", "DEC9", "DEC9 MUX"},
+ {"IIR1 INP1 MUX", "DEC10", "DEC10 MUX"},
+
+ {"MIC BIAS1 Internal1", NULL, "LDO_H"},
+ {"MIC BIAS1 Internal2", NULL, "LDO_H"},
+ {"MIC BIAS1 External", NULL, "LDO_H"},
+ {"MIC BIAS2 Internal1", NULL, "LDO_H"},
+ {"MIC BIAS2 Internal2", NULL, "LDO_H"},
+ {"MIC BIAS2 Internal3", NULL, "LDO_H"},
+ {"MIC BIAS2 External", NULL, "LDO_H"},
+ {"MIC BIAS3 Internal1", NULL, "LDO_H"},
+ {"MIC BIAS3 Internal2", NULL, "LDO_H"},
+ {"MIC BIAS3 External", NULL, "LDO_H"},
+ {"MIC BIAS4 External", NULL, "LDO_H"},
+};
+
+static const struct snd_soc_dapm_route tabla_1_x_lineout_2_to_4_map[] = {
+
+ {"RX4 DSM MUX", "DSM_INV", "RX3 MIX2"},
+ {"RX4 DSM MUX", "CIC_OUT", "RX4 MIX1"},
+
+ {"LINEOUT2 DAC", NULL, "RX4 DSM MUX"},
+
+ {"LINEOUT3 DAC", NULL, "RX5 MIX1"},
+ {"LINEOUT3 DAC GROUND", "Switch", "RX3 MIX2"},
+ {"LINEOUT3 DAC", NULL, "LINEOUT3 DAC GROUND"},
+
+ {"RX6 DSM MUX", "DSM_INV", "RX5 MIX1"},
+ {"RX6 DSM MUX", "CIC_OUT", "RX6 MIX1"},
+
+ {"LINEOUT4 DAC", NULL, "RX6 DSM MUX"},
+ {"LINEOUT4 DAC GROUND", "Switch", "RX4 DSM MUX"},
+ {"LINEOUT4 DAC", NULL, "LINEOUT4 DAC GROUND"},
+};
+
+
+static const struct snd_soc_dapm_route tabla_2_x_lineout_2_to_4_map[] = {
+
+ {"RX4 DSM MUX", "DSM_INV", "RX3 MIX2"},
+ {"RX4 DSM MUX", "CIC_OUT", "RX4 MIX1"},
+
+ {"LINEOUT3 DAC", NULL, "RX4 DSM MUX"},
+
+ {"LINEOUT2 DAC", NULL, "RX5 MIX1"},
+
+ {"RX6 DSM MUX", "DSM_INV", "RX5 MIX1"},
+ {"RX6 DSM MUX", "CIC_OUT", "RX6 MIX1"},
+
+ {"LINEOUT4 DAC", NULL, "RX6 DSM MUX"},
+};
+
+static int tabla_readable(struct snd_soc_codec *ssc, unsigned int reg)
+{
+ int i;
+ struct wcd9xxx *tabla_core = dev_get_drvdata(ssc->dev->parent);
+
+ if (TABLA_IS_1_X(tabla_core->version)) {
+ for (i = 0; i < ARRAY_SIZE(tabla_1_reg_readable); i++) {
+ if (tabla_1_reg_readable[i] == reg)
+ return 1;
+ }
+ } else {
+ for (i = 0; i < ARRAY_SIZE(tabla_2_reg_readable); i++) {
+ if (tabla_2_reg_readable[i] == reg)
+ return 1;
+ }
+ }
+
+ return tabla_reg_readable[reg];
+}
+static bool tabla_is_digital_gain_register(unsigned int reg)
+{
+ bool rtn = false;
+ switch (reg) {
+ case TABLA_A_CDC_RX1_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_RX2_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_RX3_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_RX4_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_RX5_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_RX6_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_RX7_VOL_CTL_B2_CTL:
+ case TABLA_A_CDC_TX1_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX2_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX3_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX4_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX5_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX6_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX7_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX8_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX9_VOL_CTL_GAIN:
+ case TABLA_A_CDC_TX10_VOL_CTL_GAIN:
+ rtn = true;
+ break;
+ default:
+ break;
+ }
+ return rtn;
+}
+static int tabla_volatile(struct snd_soc_codec *ssc, unsigned int reg)
+{
+ /* Registers lower than 0x100 are top level registers which can be
+ * written by the Tabla core driver.
+ */
+
+ if ((reg >= TABLA_A_CDC_MBHC_EN_CTL) || (reg < 0x100))
+ return 1;
+
+ /* IIR Coeff registers are not cacheable */
+ if ((reg >= TABLA_A_CDC_IIR1_COEF_B1_CTL) &&
+ (reg <= TABLA_A_CDC_IIR2_COEF_B5_CTL))
+ return 1;
+
+ /* Digital gain register is not cacheable so we have to write
+ * the setting even it is the same
+ */
+ if (tabla_is_digital_gain_register(reg))
+ return 1;
+
+ /* HPH status registers */
+ if (reg == TABLA_A_RX_HPH_L_STATUS || reg == TABLA_A_RX_HPH_R_STATUS)
+ return 1;
+
+ return 0;
+}
+
+#define TABLA_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+static int tabla_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ BUG_ON(reg > TABLA_MAX_REGISTER);
+
+ if (!tabla_volatile(codec, reg)) {
+ ret = snd_soc_cache_write(codec, reg, value);
+ if (ret != 0)
+ dev_err(codec->dev, "Cache write to %x failed: %d\n",
+ reg, ret);
+ }
+
+ return wcd9xxx_reg_write(codec->control_data, reg, value);
+}
+static unsigned int tabla_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ unsigned int val;
+ int ret;
+
+ BUG_ON(reg > TABLA_MAX_REGISTER);
+
+ if (!tabla_volatile(codec, reg) && tabla_readable(codec, reg) &&
+ reg < codec->driver->reg_cache_size) {
+ ret = snd_soc_cache_read(codec, reg, &val);
+ if (ret >= 0) {
+ return val;
+ } else
+ dev_err(codec->dev, "Cache read from %x failed: %d\n",
+ reg, ret);
+ }
+
+ val = wcd9xxx_reg_read(codec->control_data, reg);
+ return val;
+}
+
+static s16 tabla_get_current_v_ins(struct tabla_priv *tabla, bool hu)
+{
+ s16 v_ins;
+ if ((tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV) &&
+ tabla->mbhc_micbias_switched)
+ v_ins = hu ? (s16)tabla->mbhc_data.adj_v_ins_hu :
+ (s16)tabla->mbhc_data.adj_v_ins_h;
+ else
+ v_ins = hu ? (s16)tabla->mbhc_data.v_ins_hu :
+ (s16)tabla->mbhc_data.v_ins_h;
+ return v_ins;
+}
+
+static s16 tabla_get_current_v_hs_max(struct tabla_priv *tabla)
+{
+ s16 v_hs_max;
+ struct tabla_mbhc_plug_type_cfg *plug_type;
+
+ plug_type = TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla->mbhc_cfg.calibration);
+ if ((tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV) &&
+ tabla->mbhc_micbias_switched)
+ v_hs_max = tabla->mbhc_data.adj_v_hs_max;
+ else
+ v_hs_max = plug_type->v_hs_max;
+ return v_hs_max;
+}
+
+static void tabla_codec_calibrate_hs_polling(struct snd_soc_codec *codec)
+{
+ u8 *n_ready, *n_cic;
+ struct tabla_mbhc_btn_detect_cfg *btn_det;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ const s16 v_ins_hu = tabla_get_current_v_ins(tabla, true);
+
+ btn_det = TABLA_MBHC_CAL_BTN_DET_PTR(tabla->mbhc_cfg.calibration);
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B1_CTL,
+ v_ins_hu & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B2_CTL,
+ (v_ins_hu >> 8) & 0xFF);
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B3_CTL,
+ tabla->mbhc_data.v_b1_hu & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B4_CTL,
+ (tabla->mbhc_data.v_b1_hu >> 8) & 0xFF);
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B5_CTL,
+ tabla->mbhc_data.v_b1_h & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B6_CTL,
+ (tabla->mbhc_data.v_b1_h >> 8) & 0xFF);
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B9_CTL,
+ tabla->mbhc_data.v_brh & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B10_CTL,
+ (tabla->mbhc_data.v_brh >> 8) & 0xFF);
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B11_CTL,
+ tabla->mbhc_data.v_brl & 0xFF);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_VOLT_B12_CTL,
+ (tabla->mbhc_data.v_brl >> 8) & 0xFF);
+
+ n_ready = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_N_READY);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_TIMER_B1_CTL,
+ n_ready[tabla_codec_mclk_index(tabla)]);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_TIMER_B2_CTL,
+ tabla->mbhc_data.npoll);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_TIMER_B3_CTL,
+ tabla->mbhc_data.nbounce_wait);
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_N_CIC);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_TIMER_B6_CTL,
+ n_cic[tabla_codec_mclk_index(tabla)]);
+}
+
+static int tabla_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct wcd9xxx *tabla_core = dev_get_drvdata(dai->codec->dev->parent);
+ pr_debug("%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+ if ((tabla_core != NULL) &&
+ (tabla_core->dev != NULL) &&
+ (tabla_core->dev->parent != NULL))
+ pm_runtime_get_sync(tabla_core->dev->parent);
+
+ return 0;
+}
+
+static void tabla_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct wcd9xxx *tabla_core = dev_get_drvdata(dai->codec->dev->parent);
+ pr_debug("%s(): substream = %s stream = %d\n" , __func__,
+ substream->name, substream->stream);
+ if ((tabla_core != NULL) &&
+ (tabla_core->dev != NULL) &&
+ (tabla_core->dev->parent != NULL)) {
+ pm_runtime_mark_last_busy(tabla_core->dev->parent);
+ pm_runtime_put(tabla_core->dev->parent);
+ }
+}
+
+int tabla_mclk_enable(struct snd_soc_codec *codec, int mclk_enable, bool dapm)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: mclk_enable = %u, dapm = %d\n", __func__, mclk_enable,
+ dapm);
+ if (dapm)
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ if (mclk_enable) {
+ tabla->mclk_enabled = true;
+
+ if (tabla->mbhc_polling_active) {
+ tabla_codec_pause_hs_polling(codec);
+ tabla_codec_disable_clock_block(codec);
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_AUDIO_MODE);
+ tabla_codec_enable_clock_block(codec, 0);
+ tabla_codec_calibrate_hs_polling(codec);
+ tabla_codec_start_hs_polling(codec);
+ } else {
+ tabla_codec_disable_clock_block(codec);
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_AUDIO_MODE);
+ tabla_codec_enable_clock_block(codec, 0);
+ }
+ } else {
+
+ if (!tabla->mclk_enabled) {
+ if (dapm)
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ pr_err("Error, MCLK already diabled\n");
+ return -EINVAL;
+ }
+ tabla->mclk_enabled = false;
+
+ if (tabla->mbhc_polling_active) {
+ tabla_codec_pause_hs_polling(codec);
+ tabla_codec_disable_clock_block(codec);
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_MBHC_MODE);
+ tabla_enable_rx_bias(codec, 1);
+ tabla_codec_enable_clock_block(codec, 1);
+ tabla_codec_calibrate_hs_polling(codec);
+ tabla_codec_start_hs_polling(codec);
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1,
+ 0x05, 0x01);
+ } else {
+ tabla_codec_disable_clock_block(codec);
+ tabla_codec_enable_bandgap(codec,
+ TABLA_BANDGAP_OFF);
+ }
+ }
+ if (dapm)
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ return 0;
+}
+
+static int tabla_set_dai_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ pr_debug("%s\n", __func__);
+ return 0;
+}
+
+static int tabla_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ u8 val = 0;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(dai->codec);
+
+ pr_debug("%s\n", __func__);
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ /* CPU is master */
+ if (tabla->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ if (dai->id == AIF1_CAP)
+ snd_soc_update_bits(dai->codec,
+ TABLA_A_CDC_CLK_TX_I2S_CTL,
+ TABLA_I2S_MASTER_MODE_MASK, 0);
+ else if (dai->id == AIF1_PB)
+ snd_soc_update_bits(dai->codec,
+ TABLA_A_CDC_CLK_RX_I2S_CTL,
+ TABLA_I2S_MASTER_MODE_MASK, 0);
+ }
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* CPU is slave */
+ if (tabla->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ val = TABLA_I2S_MASTER_MODE_MASK;
+ if (dai->id == AIF1_CAP)
+ snd_soc_update_bits(dai->codec,
+ TABLA_A_CDC_CLK_TX_I2S_CTL, val, val);
+ else if (dai->id == AIF1_PB)
+ snd_soc_update_bits(dai->codec,
+ TABLA_A_CDC_CLK_RX_I2S_CTL, val, val);
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int tabla_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(dai->codec);
+ u32 i = 0;
+ if (!tx_slot && !rx_slot) {
+ pr_err("%s: Invalid\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s(): dai_name = %s DAI-ID %x tx_ch %d rx_ch %d\n",
+ __func__, dai->name, dai->id, tx_num, rx_num);
+
+ if (dai->id == AIF1_PB || dai->id == AIF2_PB || dai->id == AIF3_PB) {
+ for (i = 0; i < rx_num; i++) {
+ tabla->dai[dai->id - 1].ch_num[i] = rx_slot[i];
+ tabla->dai[dai->id - 1].ch_act = 0;
+ tabla->dai[dai->id - 1].ch_tot = rx_num;
+ }
+ } else if (dai->id == AIF1_CAP || dai->id == AIF2_CAP ||
+ dai->id == AIF3_CAP) {
+ for (i = 0; i < tx_num; i++) {
+ tabla->dai[dai->id - 1].ch_num[i] = tx_slot[i];
+ tabla->dai[dai->id - 1].ch_act = 0;
+ tabla->dai[dai->id - 1].ch_tot = tx_num;
+ }
+ }
+ return 0;
+}
+
+static int tabla_get_channel_map(struct snd_soc_dai *dai,
+ unsigned int *tx_num, unsigned int *tx_slot,
+ unsigned int *rx_num, unsigned int *rx_slot)
+
+{
+ struct wcd9xxx *tabla = dev_get_drvdata(dai->codec->control_data);
+
+ u32 cnt = 0;
+ u32 tx_ch[SLIM_MAX_TX_PORTS];
+ u32 rx_ch[SLIM_MAX_RX_PORTS];
+
+ if (!rx_slot && !tx_slot) {
+ pr_err("%s: Invalid\n", __func__);
+ return -EINVAL;
+ }
+
+ /* for virtual port, codec driver needs to do
+ * housekeeping, for now should be ok
+ */
+ wcd9xxx_get_channel(tabla, rx_ch, tx_ch);
+ if (dai->id == AIF1_PB) {
+ *rx_num = tabla_dai[dai->id - 1].playback.channels_max;
+ while (cnt < *rx_num) {
+ rx_slot[cnt] = rx_ch[cnt];
+ cnt++;
+ }
+ } else if (dai->id == AIF1_CAP) {
+ *tx_num = tabla_dai[dai->id - 1].capture.channels_max;
+ while (cnt < *tx_num) {
+ tx_slot[cnt] = tx_ch[6 + cnt];
+ cnt++;
+ }
+ } else if (dai->id == AIF2_PB) {
+ *rx_num = tabla_dai[dai->id - 1].playback.channels_max;
+ while (cnt < *rx_num) {
+ rx_slot[cnt] = rx_ch[5 + cnt];
+ cnt++;
+ }
+ } else if (dai->id == AIF2_CAP) {
+ *tx_num = tabla_dai[dai->id - 1].capture.channels_max;
+ tx_slot[0] = tx_ch[cnt];
+ tx_slot[1] = tx_ch[1 + cnt];
+ tx_slot[2] = tx_ch[5 + cnt];
+ tx_slot[3] = tx_ch[3 + cnt];
+
+ } else if (dai->id == AIF3_PB) {
+ *rx_num = tabla_dai[dai->id - 1].playback.channels_max;
+ rx_slot[0] = rx_ch[3];
+ rx_slot[1] = rx_ch[4];
+
+ } else if (dai->id == AIF3_CAP) {
+ *tx_num = tabla_dai[dai->id - 1].capture.channels_max;
+ tx_slot[cnt] = tx_ch[2 + cnt];
+ tx_slot[cnt + 1] = tx_ch[4 + cnt];
+ }
+ pr_debug("%s(): dai_name = %s DAI-ID %x tx_ch %d rx_ch %d\n",
+ __func__, dai->name, dai->id, *tx_num, *rx_num);
+
+
+ return 0;
+}
+
+static int tabla_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(dai->codec);
+ u8 path, shift;
+ u16 tx_fs_reg, rx_fs_reg;
+ u8 tx_fs_rate, rx_fs_rate, rx_state, tx_state;
+ u32 compander_fs;
+
+ pr_debug("%s: dai_name = %s DAI-ID %x rate %d num_ch %d\n", __func__,
+ dai->name, dai->id, params_rate(params),
+ params_channels(params));
+
+ switch (params_rate(params)) {
+ case 8000:
+ tx_fs_rate = 0x00;
+ rx_fs_rate = 0x00;
+ compander_fs = COMPANDER_FS_8KHZ;
+ break;
+ case 16000:
+ tx_fs_rate = 0x01;
+ rx_fs_rate = 0x20;
+ compander_fs = COMPANDER_FS_16KHZ;
+ break;
+ case 32000:
+ tx_fs_rate = 0x02;
+ rx_fs_rate = 0x40;
+ compander_fs = COMPANDER_FS_32KHZ;
+ break;
+ case 48000:
+ tx_fs_rate = 0x03;
+ rx_fs_rate = 0x60;
+ compander_fs = COMPANDER_FS_48KHZ;
+ break;
+ case 96000:
+ tx_fs_rate = 0x04;
+ rx_fs_rate = 0x80;
+ compander_fs = COMPANDER_FS_96KHZ;
+ break;
+ case 192000:
+ tx_fs_rate = 0x05;
+ rx_fs_rate = 0xA0;
+ compander_fs = COMPANDER_FS_192KHZ;
+ break;
+ default:
+ pr_err("%s: Invalid sampling rate %d\n", __func__,
+ params_rate(params));
+ return -EINVAL;
+ }
+
+
+ /**
+ * If current dai is a tx dai, set sample rate to
+ * all the txfe paths that are currently not active
+ */
+ if ((dai->id == AIF1_CAP) || (dai->id == AIF2_CAP) ||
+ (dai->id == AIF3_CAP)) {
+
+ tx_state = snd_soc_read(codec,
+ TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL);
+
+ for (path = 1, shift = 0;
+ path <= NUM_DECIMATORS; path++, shift++) {
+
+ if (path == BITS_PER_REG + 1) {
+ shift = 0;
+ tx_state = snd_soc_read(codec,
+ TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL);
+ }
+
+ if (!(tx_state & (1 << shift))) {
+ tx_fs_reg = TABLA_A_CDC_TX1_CLK_FS_CTL
+ + (BITS_PER_REG*(path-1));
+ snd_soc_update_bits(codec, tx_fs_reg,
+ 0x07, tx_fs_rate);
+ }
+ }
+ if (tabla->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_TX_I2S_CTL,
+ 0x20, 0x20);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_TX_I2S_CTL,
+ 0x20, 0x00);
+ break;
+ default:
+ pr_err("invalid format\n");
+ break;
+ }
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_TX_I2S_CTL,
+ 0x07, tx_fs_rate);
+ } else {
+ tabla->dai[dai->id - 1].rate = params_rate(params);
+ }
+ }
+ /**
+ * TODO: Need to handle case where same RX chain takes 2 or more inputs
+ * with varying sample rates
+ */
+
+ /**
+ * If current dai is a rx dai, set sample rate to
+ * all the rx paths that are currently not active
+ */
+ if (dai->id == AIF1_PB || dai->id == AIF2_PB || dai->id == AIF3_PB) {
+
+ rx_state = snd_soc_read(codec,
+ TABLA_A_CDC_CLK_RX_B1_CTL);
+
+ for (path = 1, shift = 0;
+ path <= NUM_INTERPOLATORS; path++, shift++) {
+
+ if (!(rx_state & (1 << shift))) {
+ rx_fs_reg = TABLA_A_CDC_RX1_B5_CTL
+ + (BITS_PER_REG*(path-1));
+ snd_soc_update_bits(codec, rx_fs_reg,
+ 0xE0, rx_fs_rate);
+ if (comp_rx_path[shift] < COMPANDER_MAX)
+ tabla->comp_fs[comp_rx_path[shift]]
+ = compander_fs;
+ }
+ }
+ if (tabla->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_RX_I2S_CTL,
+ 0x20, 0x20);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_CLK_RX_I2S_CTL,
+ 0x20, 0x00);
+ break;
+ default:
+ pr_err("invalid format\n");
+ break;
+ }
+ snd_soc_update_bits(codec, TABLA_A_CDC_CLK_RX_I2S_CTL,
+ 0x03, (rx_fs_rate >> 0x05));
+ } else {
+ tabla->dai[dai->id - 1].rate = params_rate(params);
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops tabla_dai_ops = {
+ .startup = tabla_startup,
+ .shutdown = tabla_shutdown,
+ .hw_params = tabla_hw_params,
+ .set_sysclk = tabla_set_dai_sysclk,
+ .set_fmt = tabla_set_dai_fmt,
+ .set_channel_map = tabla_set_channel_map,
+ .get_channel_map = tabla_get_channel_map,
+};
+
+static struct snd_soc_dai_driver tabla_dai[] = {
+ {
+ .name = "tabla_rx1",
+ .id = AIF1_PB,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_max = 192000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &tabla_dai_ops,
+ },
+ {
+ .name = "tabla_tx1",
+ .id = AIF1_CAP,
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_max = 192000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 4,
+ },
+ .ops = &tabla_dai_ops,
+ },
+ {
+ .name = "tabla_rx2",
+ .id = AIF2_PB,
+ .playback = {
+ .stream_name = "AIF2 Playback",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &tabla_dai_ops,
+ },
+ {
+ .name = "tabla_tx2",
+ .id = AIF2_CAP,
+ .capture = {
+ .stream_name = "AIF2 Capture",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_max = 192000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 4,
+ },
+ .ops = &tabla_dai_ops,
+ },
+ {
+ .name = "tabla_tx3",
+ .id = AIF3_CAP,
+ .capture = {
+ .stream_name = "AIF3 Capture",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &tabla_dai_ops,
+ },
+ {
+ .name = "tabla_rx3",
+ .id = AIF3_PB,
+ .playback = {
+ .stream_name = "AIF3 Playback",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ },
+ .ops = &tabla_dai_ops,
+ },
+};
+
+static struct snd_soc_dai_driver tabla_i2s_dai[] = {
+ {
+ .name = "tabla_i2s_rx1",
+ .id = 1,
+ .playback = {
+ .stream_name = "AIF1 Playback",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_max = 192000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 4,
+ },
+ .ops = &tabla_dai_ops,
+ },
+ {
+ .name = "tabla_i2s_tx1",
+ .id = 2,
+ .capture = {
+ .stream_name = "AIF1 Capture",
+ .rates = WCD9310_RATES,
+ .formats = TABLA_FORMATS,
+ .rate_max = 192000,
+ .rate_min = 8000,
+ .channels_min = 1,
+ .channels_max = 4,
+ },
+ .ops = &tabla_dai_ops,
+ },
+};
+
+static int tabla_codec_enable_slimrx(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct wcd9xxx *tabla;
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla_p = snd_soc_codec_get_drvdata(codec);
+ u32 j = 0;
+ u32 ret = 0;
+ codec->control_data = dev_get_drvdata(codec->dev->parent);
+ tabla = codec->control_data;
+ /* Execute the callback only if interface type is slimbus */
+ if (tabla_p->intf_type != WCD9XXX_INTERFACE_TYPE_SLIMBUS)
+ return 0;
+
+ pr_debug("%s: %s %d\n", __func__, w->name, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ for (j = 0; j < ARRAY_SIZE(tabla_dai); j++) {
+ if ((tabla_dai[j].id == AIF1_CAP) ||
+ (tabla_dai[j].id == AIF2_CAP) ||
+ (tabla_dai[j].id == AIF3_CAP))
+ continue;
+ if (!strncmp(w->sname,
+ tabla_dai[j].playback.stream_name, 13)) {
+ ++tabla_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (tabla_p->dai[j].ch_act == tabla_p->dai[j].ch_tot)
+ ret = wcd9xxx_cfg_slim_sch_rx(tabla,
+ tabla_p->dai[j].ch_num,
+ tabla_p->dai[j].ch_tot,
+ tabla_p->dai[j].rate);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ for (j = 0; j < ARRAY_SIZE(tabla_dai); j++) {
+ if ((tabla_dai[j].id == AIF1_CAP) ||
+ (tabla_dai[j].id == AIF2_CAP) ||
+ (tabla_dai[j].id == AIF3_CAP))
+ continue;
+ if (!strncmp(w->sname,
+ tabla_dai[j].playback.stream_name, 13)) {
+ --tabla_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (!tabla_p->dai[j].ch_act) {
+ ret = wcd9xxx_close_slim_sch_rx(tabla,
+ tabla_p->dai[j].ch_num,
+ tabla_p->dai[j].ch_tot);
+ usleep_range(15000, 15000);
+ tabla_p->dai[j].rate = 0;
+ memset(tabla_p->dai[j].ch_num, 0, (sizeof(u32)*
+ tabla_p->dai[j].ch_tot));
+ tabla_p->dai[j].ch_tot = 0;
+ }
+ }
+ return ret;
+}
+
+static int tabla_codec_enable_slimtx(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct wcd9xxx *tabla;
+ struct snd_soc_codec *codec = w->codec;
+ struct tabla_priv *tabla_p = snd_soc_codec_get_drvdata(codec);
+ /* index to the DAI ID, for now hardcoding */
+ u32 j = 0;
+ u32 ret = 0;
+
+ codec->control_data = dev_get_drvdata(codec->dev->parent);
+ tabla = codec->control_data;
+
+ /* Execute the callback only if interface type is slimbus */
+ if (tabla_p->intf_type != WCD9XXX_INTERFACE_TYPE_SLIMBUS)
+ return 0;
+
+ pr_debug("%s(): %s %d\n", __func__, w->name, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ for (j = 0; j < ARRAY_SIZE(tabla_dai); j++) {
+ if (tabla_dai[j].id == AIF1_PB ||
+ tabla_dai[j].id == AIF2_PB ||
+ tabla_dai[j].id == AIF3_PB)
+ continue;
+ if (!strncmp(w->sname,
+ tabla_dai[j].capture.stream_name, 13)) {
+ ++tabla_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (tabla_p->dai[j].ch_act == tabla_p->dai[j].ch_tot)
+ ret = wcd9xxx_cfg_slim_sch_tx(tabla,
+ tabla_p->dai[j].ch_num,
+ tabla_p->dai[j].ch_tot,
+ tabla_p->dai[j].rate);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ for (j = 0; j < ARRAY_SIZE(tabla_dai); j++) {
+ if (tabla_dai[j].id == AIF1_PB ||
+ tabla_dai[j].id == AIF2_PB ||
+ tabla_dai[j].id == AIF3_PB)
+ continue;
+ if (!strncmp(w->sname,
+ tabla_dai[j].capture.stream_name, 13)) {
+ --tabla_p->dai[j].ch_act;
+ break;
+ }
+ }
+ if (!tabla_p->dai[j].ch_act) {
+ ret = wcd9xxx_close_slim_sch_tx(tabla,
+ tabla_p->dai[j].ch_num,
+ tabla_p->dai[j].ch_tot);
+ tabla_p->dai[j].rate = 0;
+ memset(tabla_p->dai[j].ch_num, 0, (sizeof(u32)*
+ tabla_p->dai[j].ch_tot));
+ tabla_p->dai[j].ch_tot = 0;
+ }
+ }
+ return ret;
+}
+
+/* Todo: Have seperate dapm widgets for I2S and Slimbus.
+ * Might Need to have callbacks registered only for slimbus
+ */
+static const struct snd_soc_dapm_widget tabla_dapm_widgets[] = {
+ /*RX stuff */
+ SND_SOC_DAPM_OUTPUT("EAR"),
+
+ SND_SOC_DAPM_PGA("EAR PA", TABLA_A_RX_EAR_EN, 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_MIXER("DAC1", TABLA_A_RX_EAR_EN, 6, 0, dac1_switch,
+ ARRAY_SIZE(dac1_switch)),
+
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX1", "AIF1 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX2", "AIF1 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX3", "AIF1 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX4", "AIF3 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX5", "AIF3 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX6", "AIF2 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_AIF_IN_E("SLIM RX7", "AIF2 Playback", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimrx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Headphone */
+ SND_SOC_DAPM_OUTPUT("HEADPHONE"),
+ SND_SOC_DAPM_PGA_E("HPHL", TABLA_A_RX_HPH_CNP_EN, 5, 0, NULL, 0,
+ tabla_hph_pa_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MIXER("HPHL DAC", TABLA_A_RX_HPH_L_DAC_CTL, 7, 0,
+ hphl_switch, ARRAY_SIZE(hphl_switch)),
+
+ SND_SOC_DAPM_PGA_E("HPHR", TABLA_A_RX_HPH_CNP_EN, 4, 0, NULL, 0,
+ tabla_hph_pa_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_DAC_E("HPHR DAC", NULL, TABLA_A_RX_HPH_R_DAC_CTL, 7, 0,
+ tabla_hphr_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Speaker */
+ SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT3"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT4"),
+ SND_SOC_DAPM_OUTPUT("LINEOUT5"),
+
+ SND_SOC_DAPM_PGA_E("LINEOUT1 PA", TABLA_A_RX_LINE_CNP_EN, 0, 0, NULL,
+ 0, tabla_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LINEOUT2 PA", TABLA_A_RX_LINE_CNP_EN, 1, 0, NULL,
+ 0, tabla_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LINEOUT3 PA", TABLA_A_RX_LINE_CNP_EN, 2, 0, NULL,
+ 0, tabla_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LINEOUT4 PA", TABLA_A_RX_LINE_CNP_EN, 3, 0, NULL,
+ 0, tabla_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PGA_E("LINEOUT5 PA", TABLA_A_RX_LINE_CNP_EN, 4, 0, NULL, 0,
+ tabla_codec_enable_lineout, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_DAC_E("LINEOUT1 DAC", NULL, TABLA_A_RX_LINE_1_DAC_CTL, 7, 0
+ , tabla_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("LINEOUT2 DAC", NULL, TABLA_A_RX_LINE_2_DAC_CTL, 7, 0
+ , tabla_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_DAC_E("LINEOUT3 DAC", NULL, TABLA_A_RX_LINE_3_DAC_CTL, 7, 0
+ , tabla_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH("LINEOUT3 DAC GROUND", SND_SOC_NOPM, 0, 0,
+ &lineout3_ground_switch),
+ SND_SOC_DAPM_DAC_E("LINEOUT4 DAC", NULL, TABLA_A_RX_LINE_4_DAC_CTL, 7, 0
+ , tabla_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_SWITCH("LINEOUT4 DAC GROUND", SND_SOC_NOPM, 0, 0,
+ &lineout4_ground_switch),
+ SND_SOC_DAPM_DAC_E("LINEOUT5 DAC", NULL, TABLA_A_RX_LINE_5_DAC_CTL, 7, 0
+ , tabla_lineout_dac_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MIXER_E("RX1 MIX2", TABLA_A_CDC_CLK_RX_B1_CTL, 0, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER_E("RX2 MIX2", TABLA_A_CDC_CLK_RX_B1_CTL, 1, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER_E("RX3 MIX2", TABLA_A_CDC_CLK_RX_B1_CTL, 2, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER_E("RX4 MIX1", TABLA_A_CDC_CLK_RX_B1_CTL, 3, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER_E("RX5 MIX1", TABLA_A_CDC_CLK_RX_B1_CTL, 4, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER_E("RX6 MIX1", TABLA_A_CDC_CLK_RX_B1_CTL, 5, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER_E("RX7 MIX1", TABLA_A_CDC_CLK_RX_B1_CTL, 6, 0, NULL,
+ 0, tabla_codec_reset_interpolator, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
+
+ SND_SOC_DAPM_MIXER("RX1 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX2 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX3 MIX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX_E("RX4 DSM MUX", TABLA_A_CDC_CLK_RX_B1_CTL, 3, 0,
+ &rx4_dsm_mux, tabla_codec_reset_interpolator,
+ SND_SOC_DAPM_PRE_PMU),
+
+ SND_SOC_DAPM_MUX_E("RX6 DSM MUX", TABLA_A_CDC_CLK_RX_B1_CTL, 5, 0,
+ &rx6_dsm_mux, tabla_codec_reset_interpolator,
+ SND_SOC_DAPM_PRE_PMU),
+
+ SND_SOC_DAPM_MIXER("RX1 CHAIN", TABLA_A_CDC_RX1_B6_CTL, 5, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("RX2 CHAIN", TABLA_A_CDC_RX2_B6_CTL, 5, 0, NULL, 0),
+
+ SND_SOC_DAPM_MUX("RX1 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX1 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX1 MIX1 INP3", SND_SOC_NOPM, 0, 0,
+ &rx_mix1_inp3_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx2_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx2_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX3 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx3_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX3 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx3_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX4 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx4_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX4 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx4_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX5 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx5_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX5 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx5_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX6 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx6_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX6 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx6_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX7 MIX1 INP1", SND_SOC_NOPM, 0, 0,
+ &rx7_mix1_inp1_mux),
+ SND_SOC_DAPM_MUX("RX7 MIX1 INP2", SND_SOC_NOPM, 0, 0,
+ &rx7_mix1_inp2_mux),
+ SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+ &rx1_mix2_inp1_mux),
+ SND_SOC_DAPM_MUX("RX1 MIX2 INP2", SND_SOC_NOPM, 0, 0,
+ &rx1_mix2_inp2_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+ &rx2_mix2_inp1_mux),
+ SND_SOC_DAPM_MUX("RX2 MIX2 INP2", SND_SOC_NOPM, 0, 0,
+ &rx2_mix2_inp2_mux),
+ SND_SOC_DAPM_MUX("RX3 MIX2 INP1", SND_SOC_NOPM, 0, 0,
+ &rx3_mix2_inp1_mux),
+ SND_SOC_DAPM_MUX("RX3 MIX2 INP2", SND_SOC_NOPM, 0, 0,
+ &rx3_mix2_inp2_mux),
+
+ SND_SOC_DAPM_SUPPLY("CP", TABLA_A_CP_EN, 0, 0,
+ tabla_codec_enable_charge_pump, SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD),
+
+ SND_SOC_DAPM_SUPPLY("RX_BIAS", SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_rx_bias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ /* TX */
+
+ SND_SOC_DAPM_SUPPLY("CDC_CONN", TABLA_A_CDC_CLK_OTHR_CTL, 2, 0, NULL,
+ 0),
+
+ SND_SOC_DAPM_SUPPLY("LDO_H", TABLA_A_LDO_H_MODE_1, 7, 0,
+ tabla_codec_enable_ldo_h, SND_SOC_DAPM_POST_PMU),
+
+ SND_SOC_DAPM_SUPPLY("COMP1_CLK", SND_SOC_NOPM, 0, 0,
+ tabla_config_compander, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD),
+ SND_SOC_DAPM_SUPPLY("COMP2_CLK", SND_SOC_NOPM, 1, 0,
+ tabla_config_compander, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_POST_PMD),
+
+ SND_SOC_DAPM_INPUT("AMIC1"),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS1 External", TABLA_A_MICB_1_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS1 Internal1", TABLA_A_MICB_1_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS1 Internal2", TABLA_A_MICB_1_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("ADC1", NULL, TABLA_A_TX_1_2_EN, 7, 0,
+ tabla_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_INPUT("AMIC3"),
+ SND_SOC_DAPM_ADC_E("ADC3", NULL, TABLA_A_TX_3_4_EN, 7, 0,
+ tabla_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_INPUT("AMIC4"),
+ SND_SOC_DAPM_ADC_E("ADC4", NULL, TABLA_A_TX_3_4_EN, 3, 0,
+ tabla_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_INPUT("AMIC5"),
+ SND_SOC_DAPM_ADC_E("ADC5", NULL, TABLA_A_TX_5_6_EN, 7, 0,
+ tabla_codec_enable_adc, SND_SOC_DAPM_POST_PMU),
+
+ SND_SOC_DAPM_INPUT("AMIC6"),
+ SND_SOC_DAPM_ADC_E("ADC6", NULL, TABLA_A_TX_5_6_EN, 3, 0,
+ tabla_codec_enable_adc, SND_SOC_DAPM_POST_PMU),
+
+ SND_SOC_DAPM_MUX_E("DEC1 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 0, 0,
+ &dec1_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC2 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 1, 0,
+ &dec2_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC3 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 2, 0,
+ &dec3_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC4 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 3, 0,
+ &dec4_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC5 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 4, 0,
+ &dec5_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC6 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 5, 0,
+ &dec6_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC7 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 6, 0,
+ &dec7_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC8 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B1_CTL, 7, 0,
+ &dec8_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC9 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL, 0, 0,
+ &dec9_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX_E("DEC10 MUX", TABLA_A_CDC_CLK_TX_CLK_EN_B2_CTL, 1, 0,
+ &dec10_mux, tabla_codec_enable_dec,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("ANC1 MUX", SND_SOC_NOPM, 0, 0, &anc1_mux),
+ SND_SOC_DAPM_MUX("ANC2 MUX", SND_SOC_NOPM, 0, 0, &anc2_mux),
+
+ SND_SOC_DAPM_MIXER_E("ANC", SND_SOC_NOPM, 0, 0, NULL, 0,
+ tabla_codec_enable_anc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("ANC1 FB MUX", SND_SOC_NOPM, 0, 0, &anc1_fb_mux),
+
+ SND_SOC_DAPM_INPUT("AMIC2"),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 External", TABLA_A_MICB_2_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 Internal1", TABLA_A_MICB_2_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 Internal2", TABLA_A_MICB_2_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS2 Internal3", TABLA_A_MICB_2_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS3 External", TABLA_A_MICB_3_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS3 Internal1", TABLA_A_MICB_3_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MICBIAS_E("MIC BIAS3 Internal2", TABLA_A_MICB_3_CTL, 7, 0,
+ tabla_codec_enable_micbias, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("ADC2", NULL, TABLA_A_TX_1_2_EN, 3, 0,
+ tabla_codec_enable_adc, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX1 MUX", SND_SOC_NOPM, 0, 0, &sb_tx1_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX1", "AIF2 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX2 MUX", SND_SOC_NOPM, 0, 0, &sb_tx2_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX2", "AIF2 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX3 MUX", SND_SOC_NOPM, 0, 0, &sb_tx3_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX3", "AIF3 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX4 MUX", SND_SOC_NOPM, 0, 0, &sb_tx4_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX4", "AIF2 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX5 MUX", SND_SOC_NOPM, 0, 0, &sb_tx5_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX5", "AIF3 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX6 MUX", SND_SOC_NOPM, 0, 0, &sb_tx6_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX6", "AIF2 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX7 MUX", SND_SOC_NOPM, 0, 0, &sb_tx7_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX7", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX8 MUX", SND_SOC_NOPM, 0, 0, &sb_tx8_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX8", "AIF1 Capture", 0, SND_SOC_NOPM, 0,
+ 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX9 MUX", SND_SOC_NOPM, 0, 0, &sb_tx9_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX9", "AIF1 Capture", NULL, SND_SOC_NOPM,
+ 0, 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_MUX("SLIM TX10 MUX", SND_SOC_NOPM, 0, 0, &sb_tx10_mux),
+ SND_SOC_DAPM_AIF_OUT_E("SLIM TX10", "AIF1 Capture", NULL, SND_SOC_NOPM,
+ 0, 0, tabla_codec_enable_slimtx,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* Digital Mic Inputs */
+ SND_SOC_DAPM_ADC_E("DMIC1", NULL, SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC_E("DMIC2", NULL, SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC_E("DMIC3", NULL, SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC_E("DMIC4", NULL, SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC_E("DMIC5", NULL, SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_ADC_E("DMIC6", NULL, SND_SOC_NOPM, 0, 0,
+ tabla_codec_enable_dmic, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+
+ /* Sidetone */
+ SND_SOC_DAPM_MUX("IIR1 INP1 MUX", SND_SOC_NOPM, 0, 0, &iir1_inp1_mux),
+ SND_SOC_DAPM_PGA("IIR1", TABLA_A_CDC_CLK_SD_CTL, 0, 0, NULL, 0),
+
+ /* AUX PGA */
+ SND_SOC_DAPM_ADC_E("AUX_PGA_Left", NULL, TABLA_A_AUX_L_EN, 7, 0,
+ tabla_codec_enable_aux_pga, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_ADC_E("AUX_PGA_Right", NULL, TABLA_A_AUX_R_EN, 7, 0,
+ tabla_codec_enable_aux_pga, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMD),
+
+ /* Lineout, ear and HPH PA Mixers */
+ SND_SOC_DAPM_MIXER("HPHL_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ hphl_pa_mix, ARRAY_SIZE(hphl_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("HPHR_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ hphr_pa_mix, ARRAY_SIZE(hphr_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("LINEOUT1_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ lineout1_pa_mix, ARRAY_SIZE(lineout1_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("LINEOUT2_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ lineout2_pa_mix, ARRAY_SIZE(lineout2_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("LINEOUT3_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ lineout3_pa_mix, ARRAY_SIZE(lineout3_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("LINEOUT4_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ lineout4_pa_mix, ARRAY_SIZE(lineout4_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("LINEOUT5_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ lineout5_pa_mix, ARRAY_SIZE(lineout5_pa_mix)),
+
+ SND_SOC_DAPM_MIXER("EAR_PA_MIXER", SND_SOC_NOPM, 0, 0,
+ ear_pa_mix, ARRAY_SIZE(ear_pa_mix)),
+};
+
+static short tabla_codec_read_sta_result(struct snd_soc_codec *codec)
+{
+ u8 bias_msb, bias_lsb;
+ short bias_value;
+
+ bias_msb = snd_soc_read(codec, TABLA_A_CDC_MBHC_B3_STATUS);
+ bias_lsb = snd_soc_read(codec, TABLA_A_CDC_MBHC_B2_STATUS);
+ bias_value = (bias_msb << 8) | bias_lsb;
+ return bias_value;
+}
+
+static short tabla_codec_read_dce_result(struct snd_soc_codec *codec)
+{
+ u8 bias_msb, bias_lsb;
+ short bias_value;
+
+ bias_msb = snd_soc_read(codec, TABLA_A_CDC_MBHC_B5_STATUS);
+ bias_lsb = snd_soc_read(codec, TABLA_A_CDC_MBHC_B4_STATUS);
+ bias_value = (bias_msb << 8) | bias_lsb;
+ return bias_value;
+}
+
+static void tabla_turn_onoff_rel_detection(struct snd_soc_codec *codec, bool on)
+{
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x02, on << 1);
+}
+
+static short __tabla_codec_sta_dce(struct snd_soc_codec *codec, int dce,
+ bool override_bypass, bool noreldetection)
+{
+ short bias_value;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ wcd9xxx_disable_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL);
+ if (noreldetection)
+ tabla_turn_onoff_rel_detection(codec, false);
+
+ /* Turn on the override */
+ if (!override_bypass)
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x4, 0x4);
+ if (dce) {
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x4);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x0);
+ usleep_range(tabla->mbhc_data.t_sta_dce,
+ tabla->mbhc_data.t_sta_dce);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x4);
+ usleep_range(tabla->mbhc_data.t_dce,
+ tabla->mbhc_data.t_dce);
+ bias_value = tabla_codec_read_dce_result(codec);
+ } else {
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x2);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x0);
+ usleep_range(tabla->mbhc_data.t_sta_dce,
+ tabla->mbhc_data.t_sta_dce);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x2);
+ usleep_range(tabla->mbhc_data.t_sta,
+ tabla->mbhc_data.t_sta);
+ bias_value = tabla_codec_read_sta_result(codec);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x0);
+ }
+ /* Turn off the override after measuring mic voltage */
+ if (!override_bypass)
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x04, 0x00);
+
+ if (noreldetection)
+ tabla_turn_onoff_rel_detection(codec, true);
+ wcd9xxx_enable_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL);
+
+ return bias_value;
+}
+
+static short tabla_codec_sta_dce(struct snd_soc_codec *codec, int dce,
+ bool norel)
+{
+ return __tabla_codec_sta_dce(codec, dce, false, norel);
+}
+
+/* called only from interrupt which is under codec_resource_lock acquisition */
+static short tabla_codec_setup_hs_polling(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ short bias_value;
+ u8 cfilt_mode;
+
+ pr_debug("%s: enter, mclk_enabled %d\n", __func__, tabla->mclk_enabled);
+ if (!tabla->mbhc_cfg.calibration) {
+ pr_err("Error, no tabla calibration\n");
+ return -ENODEV;
+ }
+
+ if (!tabla->mclk_enabled) {
+ tabla_codec_disable_clock_block(codec);
+ tabla_codec_enable_bandgap(codec, TABLA_BANDGAP_MBHC_MODE);
+ tabla_enable_rx_bias(codec, 1);
+ tabla_codec_enable_clock_block(codec, 1);
+ }
+
+ snd_soc_update_bits(codec, TABLA_A_CLK_BUFF_EN1, 0x05, 0x01);
+
+ /* Make sure CFILT is in fast mode, save current mode */
+ cfilt_mode = snd_soc_read(codec, tabla->mbhc_bias_regs.cfilt_ctl);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.cfilt_ctl, 0x70, 0x00);
+
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg, 0x1F, 0x16);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x2, 0x2);
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x84);
+
+ snd_soc_update_bits(codec, TABLA_A_TX_7_MBHC_EN, 0x80, 0x80);
+ snd_soc_update_bits(codec, TABLA_A_TX_7_MBHC_EN, 0x1F, 0x1C);
+ snd_soc_update_bits(codec, TABLA_A_TX_7_MBHC_TEST_CTL, 0x40, 0x40);
+
+ snd_soc_update_bits(codec, TABLA_A_TX_7_MBHC_EN, 0x80, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x00);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x2, 0x2);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x8, 0x8);
+
+ tabla_codec_calibrate_hs_polling(codec);
+
+ /* don't flip override */
+ bias_value = __tabla_codec_sta_dce(codec, 1, true, true);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.cfilt_ctl, 0x40,
+ cfilt_mode);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x13, 0x00);
+
+ return bias_value;
+}
+
+static int tabla_cancel_btn_work(struct tabla_priv *tabla)
+{
+ int r = 0;
+ struct wcd9xxx *core = dev_get_drvdata(tabla->codec->dev->parent);
+
+ if (cancel_delayed_work_sync(&tabla->mbhc_btn_dwork)) {
+ /* if scheduled mbhc_btn_dwork is canceled from here,
+ * we have to unlock from here instead btn_work */
+ wcd9xxx_unlock_sleep(core);
+ r = 1;
+ }
+ return r;
+}
+
+/* called under codec_resource_lock acquisition */
+void tabla_set_and_turnoff_hph_padac(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ u8 wg_time;
+
+ wg_time = snd_soc_read(codec, TABLA_A_RX_HPH_CNP_WG_TIME) ;
+ wg_time += 1;
+
+ /* If headphone PA is on, check if userspace receives
+ * removal event to sync-up PA's state */
+ if (tabla_is_hph_pa_on(codec)) {
+ pr_debug("%s PA is on, setting PA_OFF_ACK\n", __func__);
+ set_bit(TABLA_HPHL_PA_OFF_ACK, &tabla->hph_pa_dac_state);
+ set_bit(TABLA_HPHR_PA_OFF_ACK, &tabla->hph_pa_dac_state);
+ } else {
+ pr_debug("%s PA is off\n", __func__);
+ }
+
+ if (tabla_is_hph_dac_on(codec, 1))
+ set_bit(TABLA_HPHL_DAC_OFF_ACK, &tabla->hph_pa_dac_state);
+ if (tabla_is_hph_dac_on(codec, 0))
+ set_bit(TABLA_HPHR_DAC_OFF_ACK, &tabla->hph_pa_dac_state);
+
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_CNP_EN, 0x30, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_L_DAC_CTL,
+ 0xC0, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_R_DAC_CTL,
+ 0xC0, 0x00);
+ usleep_range(wg_time * 1000, wg_time * 1000);
+}
+
+static void tabla_clr_and_turnon_hph_padac(struct tabla_priv *tabla)
+{
+ bool pa_turned_on = false;
+ struct snd_soc_codec *codec = tabla->codec;
+ u8 wg_time;
+
+ wg_time = snd_soc_read(codec, TABLA_A_RX_HPH_CNP_WG_TIME) ;
+ wg_time += 1;
+
+ if (test_and_clear_bit(TABLA_HPHR_DAC_OFF_ACK,
+ &tabla->hph_pa_dac_state)) {
+ pr_debug("%s: HPHR clear flag and enable DAC\n", __func__);
+ snd_soc_update_bits(tabla->codec, TABLA_A_RX_HPH_R_DAC_CTL,
+ 0xC0, 0xC0);
+ }
+ if (test_and_clear_bit(TABLA_HPHL_DAC_OFF_ACK,
+ &tabla->hph_pa_dac_state)) {
+ pr_debug("%s: HPHL clear flag and enable DAC\n", __func__);
+ snd_soc_update_bits(tabla->codec, TABLA_A_RX_HPH_L_DAC_CTL,
+ 0xC0, 0xC0);
+ }
+
+ if (test_and_clear_bit(TABLA_HPHR_PA_OFF_ACK,
+ &tabla->hph_pa_dac_state)) {
+ pr_debug("%s: HPHR clear flag and enable PA\n", __func__);
+ snd_soc_update_bits(tabla->codec, TABLA_A_RX_HPH_CNP_EN, 0x10,
+ 1 << 4);
+ pa_turned_on = true;
+ }
+ if (test_and_clear_bit(TABLA_HPHL_PA_OFF_ACK,
+ &tabla->hph_pa_dac_state)) {
+ pr_debug("%s: HPHL clear flag and enable PA\n", __func__);
+ snd_soc_update_bits(tabla->codec, TABLA_A_RX_HPH_CNP_EN, 0x20,
+ 1 << 5);
+ pa_turned_on = true;
+ }
+
+ if (pa_turned_on) {
+ pr_debug("%s: PA was turned off by MBHC and not by DAPM\n",
+ __func__);
+ usleep_range(wg_time * 1000, wg_time * 1000);
+ }
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_report_plug(struct snd_soc_codec *codec, int insertion,
+ enum snd_jack_types jack_type)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ if (!insertion) {
+ /* Report removal */
+ tabla->hph_status &= ~jack_type;
+ if (tabla->mbhc_cfg.headset_jack) {
+ /* cancel possibly scheduled btn work and
+ * report release if we reported button press */
+ if (tabla_cancel_btn_work(tabla)) {
+ pr_debug("%s: button press is canceled\n",
+ __func__);
+ } else if (tabla->buttons_pressed) {
+ pr_debug("%s: Reporting release for reported "
+ "button press %d\n", __func__,
+ jack_type);
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.button_jack, 0,
+ tabla->buttons_pressed);
+ tabla->buttons_pressed &=
+ ~TABLA_JACK_BUTTON_MASK;
+ }
+ pr_debug("%s: Reporting removal %d(%x)\n", __func__,
+ jack_type, tabla->hph_status);
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.headset_jack,
+ tabla->hph_status,
+ TABLA_JACK_MASK);
+ }
+ tabla_set_and_turnoff_hph_padac(codec);
+ hphocp_off_report(tabla, SND_JACK_OC_HPHR,
+ TABLA_IRQ_HPH_PA_OCPR_FAULT);
+ hphocp_off_report(tabla, SND_JACK_OC_HPHL,
+ TABLA_IRQ_HPH_PA_OCPL_FAULT);
+ tabla->current_plug = PLUG_TYPE_NONE;
+ tabla->mbhc_polling_active = false;
+ } else {
+ /* Report insertion */
+ tabla->hph_status |= jack_type;
+
+ if (jack_type == SND_JACK_HEADPHONE)
+ tabla->current_plug = PLUG_TYPE_HEADPHONE;
+ else if (jack_type == SND_JACK_UNSUPPORTED)
+ tabla->current_plug = PLUG_TYPE_GND_MIC_SWAP;
+ else if (jack_type == SND_JACK_HEADSET) {
+ tabla->mbhc_polling_active = true;
+ tabla->current_plug = PLUG_TYPE_HEADSET;
+ }
+ if (tabla->mbhc_cfg.headset_jack) {
+ pr_debug("%s: Reporting insertion %d(%x)\n", __func__,
+ jack_type, tabla->hph_status);
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.headset_jack,
+ tabla->hph_status,
+ TABLA_JACK_MASK);
+ }
+ tabla_clr_and_turnon_hph_padac(tabla);
+ }
+}
+
+static int tabla_codec_enable_hs_detect(struct snd_soc_codec *codec,
+ int insertion, int trigger,
+ bool padac_off)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ int central_bias_enabled = 0;
+ const struct tabla_mbhc_general_cfg *generic =
+ TABLA_MBHC_CAL_GENERAL_PTR(tabla->mbhc_cfg.calibration);
+ const struct tabla_mbhc_plug_detect_cfg *plug_det =
+ TABLA_MBHC_CAL_PLUG_DET_PTR(tabla->mbhc_cfg.calibration);
+
+ if (!tabla->mbhc_cfg.calibration) {
+ pr_err("Error, no tabla calibration\n");
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_INT_CTL, 0x1, 0);
+
+ /* Make sure mic bias and Mic line schmitt trigger
+ * are turned OFF
+ */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg, 0x01, 0x01);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg, 0x90, 0x00);
+
+ if (insertion) {
+ tabla_codec_switch_micbias(codec, 0);
+
+ /* DAPM can manipulate PA/DAC bits concurrently */
+ if (padac_off == true) {
+ tabla_set_and_turnoff_hph_padac(codec);
+ }
+
+ if (trigger & MBHC_USE_HPHL_TRIGGER) {
+ /* Enable HPH Schmitt Trigger */
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x11,
+ 0x11);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x0C,
+ plug_det->hph_current << 2);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x02,
+ 0x02);
+ }
+ if (trigger & MBHC_USE_MB_TRIGGER) {
+ /* enable the mic line schmitt trigger */
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.mbhc_reg,
+ 0x60, plug_det->mic_current << 5);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.mbhc_reg,
+ 0x80, 0x80);
+ usleep_range(plug_det->t_mic_pid, plug_det->t_mic_pid);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.ctl_reg, 0x01,
+ 0x00);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.mbhc_reg,
+ 0x10, 0x10);
+ }
+
+ /* setup for insetion detection */
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_INT_CTL, 0x2, 0);
+ } else {
+ pr_debug("setup for removal detection\n");
+ /* Make sure the HPH schmitt trigger is OFF */
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x12, 0x00);
+
+ /* enable the mic line schmitt trigger */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg,
+ 0x01, 0x00);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg, 0x60,
+ plug_det->mic_current << 5);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg,
+ 0x80, 0x80);
+ usleep_range(plug_det->t_mic_pid, plug_det->t_mic_pid);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg,
+ 0x10, 0x10);
+
+ /* Setup for low power removal detection */
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_INT_CTL, 0x2, 0x2);
+ }
+
+ if (snd_soc_read(codec, TABLA_A_CDC_MBHC_B1_CTL) & 0x4) {
+ /* called called by interrupt */
+ if (!(tabla->clock_active)) {
+ tabla_codec_enable_config_mode(codec, 1);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL,
+ 0x06, 0);
+ usleep_range(generic->t_shutdown_plug_rem,
+ generic->t_shutdown_plug_rem);
+ tabla_codec_enable_config_mode(codec, 0);
+ } else
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL,
+ 0x06, 0);
+ }
+
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.int_rbias, 0x80, 0);
+
+ /* If central bandgap disabled */
+ if (!(snd_soc_read(codec, TABLA_A_PIN_CTL_OE1) & 1)) {
+ snd_soc_update_bits(codec, TABLA_A_PIN_CTL_OE1, 0x3, 0x3);
+ usleep_range(generic->t_bg_fast_settle,
+ generic->t_bg_fast_settle);
+ central_bias_enabled = 1;
+ }
+
+ /* If LDO_H disabled */
+ if (snd_soc_read(codec, TABLA_A_PIN_CTL_OE0) & 0x80) {
+ snd_soc_update_bits(codec, TABLA_A_PIN_CTL_OE0, 0x10, 0);
+ snd_soc_update_bits(codec, TABLA_A_PIN_CTL_OE0, 0x80, 0x80);
+ usleep_range(generic->t_ldoh, generic->t_ldoh);
+ snd_soc_update_bits(codec, TABLA_A_PIN_CTL_OE0, 0x80, 0);
+
+ if (central_bias_enabled)
+ snd_soc_update_bits(codec, TABLA_A_PIN_CTL_OE1, 0x1, 0);
+ }
+
+ snd_soc_update_bits(codec, tabla->reg_addr.micb_4_mbhc, 0x3,
+ tabla->mbhc_cfg.micbias);
+
+ wcd9xxx_enable_irq(codec->control_data, TABLA_IRQ_MBHC_INSERTION);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_INT_CTL, 0x1, 0x1);
+ return 0;
+}
+
+static u16 tabla_codec_v_sta_dce(struct snd_soc_codec *codec, bool dce,
+ s16 vin_mv)
+{
+ struct tabla_priv *tabla;
+ s16 diff, zero;
+ u32 mb_mv, in;
+ u16 value;
+
+ tabla = snd_soc_codec_get_drvdata(codec);
+ mb_mv = tabla->mbhc_data.micb_mv;
+
+ if (mb_mv == 0) {
+ pr_err("%s: Mic Bias voltage is set to zero\n", __func__);
+ return -EINVAL;
+ }
+
+ if (dce) {
+ diff = (tabla->mbhc_data.dce_mb) - (tabla->mbhc_data.dce_z);
+ zero = (tabla->mbhc_data.dce_z);
+ } else {
+ diff = (tabla->mbhc_data.sta_mb) - (tabla->mbhc_data.sta_z);
+ zero = (tabla->mbhc_data.sta_z);
+ }
+ in = (u32) diff * vin_mv;
+
+ value = (u16) (in / mb_mv) + zero;
+ return value;
+}
+
+static s32 tabla_codec_sta_dce_v(struct snd_soc_codec *codec, s8 dce,
+ u16 bias_value)
+{
+ struct tabla_priv *tabla;
+ s16 value, z, mb;
+ s32 mv;
+
+ tabla = snd_soc_codec_get_drvdata(codec);
+ value = bias_value;
+ if (dce) {
+ z = (tabla->mbhc_data.dce_z);
+ mb = (tabla->mbhc_data.dce_mb);
+ mv = (value - z) * (s32)tabla->mbhc_data.micb_mv / (mb - z);
+ } else {
+ z = (tabla->mbhc_data.sta_z);
+ mb = (tabla->mbhc_data.sta_mb);
+ mv = (value - z) * (s32)tabla->mbhc_data.micb_mv / (mb - z);
+ }
+
+ return mv;
+}
+
+static void btn_lpress_fn(struct work_struct *work)
+{
+ struct delayed_work *delayed_work;
+ struct tabla_priv *tabla;
+ short bias_value;
+ int dce_mv, sta_mv;
+ struct wcd9xxx *core;
+
+ pr_debug("%s:\n", __func__);
+
+ delayed_work = to_delayed_work(work);
+ tabla = container_of(delayed_work, struct tabla_priv, mbhc_btn_dwork);
+ core = dev_get_drvdata(tabla->codec->dev->parent);
+
+ if (tabla) {
+ if (tabla->mbhc_cfg.button_jack) {
+ bias_value = tabla_codec_read_sta_result(tabla->codec);
+ sta_mv = tabla_codec_sta_dce_v(tabla->codec, 0,
+ bias_value);
+ bias_value = tabla_codec_read_dce_result(tabla->codec);
+ dce_mv = tabla_codec_sta_dce_v(tabla->codec, 1,
+ bias_value);
+ pr_debug("%s: Reporting long button press event"
+ " STA: %d, DCE: %d\n", __func__,
+ sta_mv, dce_mv);
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.button_jack,
+ tabla->buttons_pressed,
+ tabla->buttons_pressed);
+ }
+ } else {
+ pr_err("%s: Bad tabla private data\n", __func__);
+ }
+
+ pr_debug("%s: leave\n", __func__);
+ wcd9xxx_unlock_sleep(core);
+}
+
+void tabla_mbhc_cal(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla;
+ struct tabla_mbhc_btn_detect_cfg *btn_det;
+ u8 cfilt_mode, bg_mode;
+ u8 ncic, nmeas, navg;
+ u32 mclk_rate;
+ u32 dce_wait, sta_wait;
+ u8 *n_cic;
+ void *calibration;
+
+ tabla = snd_soc_codec_get_drvdata(codec);
+ calibration = tabla->mbhc_cfg.calibration;
+
+ wcd9xxx_disable_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL);
+ tabla_turn_onoff_rel_detection(codec, false);
+
+ /* First compute the DCE / STA wait times
+ * depending on tunable parameters.
+ * The value is computed in microseconds
+ */
+ btn_det = TABLA_MBHC_CAL_BTN_DET_PTR(calibration);
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_N_CIC);
+ ncic = n_cic[tabla_codec_mclk_index(tabla)];
+ nmeas = TABLA_MBHC_CAL_BTN_DET_PTR(calibration)->n_meas;
+ navg = TABLA_MBHC_CAL_GENERAL_PTR(calibration)->mbhc_navg;
+ mclk_rate = tabla->mbhc_cfg.mclk_rate;
+ dce_wait = (1000 * 512 * ncic * (nmeas + 1)) / (mclk_rate / 1000);
+ sta_wait = (1000 * 128 * (navg + 1)) / (mclk_rate / 1000);
+
+ tabla->mbhc_data.t_dce = dce_wait;
+ tabla->mbhc_data.t_sta = sta_wait;
+
+ /* LDOH and CFILT are already configured during pdata handling.
+ * Only need to make sure CFILT and bandgap are in Fast mode.
+ * Need to restore defaults once calculation is done.
+ */
+ cfilt_mode = snd_soc_read(codec, tabla->mbhc_bias_regs.cfilt_ctl);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.cfilt_ctl, 0x40, 0x00);
+ bg_mode = snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x02,
+ 0x02);
+
+ /* Micbias, CFILT, LDOH, MBHC MUX mode settings
+ * to perform ADC calibration
+ */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg, 0x60,
+ tabla->mbhc_cfg.micbias << 5);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg, 0x01, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_LDO_H_MODE_1, 0x60, 0x60);
+ snd_soc_write(codec, TABLA_A_TX_7_MBHC_TEST_CTL, 0x78);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x04, 0x04);
+
+ /* DCE measurement for 0 volts */
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x0A);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x04);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x02);
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x81);
+ usleep_range(100, 100);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x04);
+ usleep_range(tabla->mbhc_data.t_dce, tabla->mbhc_data.t_dce);
+ tabla->mbhc_data.dce_z = tabla_codec_read_dce_result(codec);
+
+ /* DCE measurment for MB voltage */
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x0A);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x02);
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x82);
+ usleep_range(100, 100);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x04);
+ usleep_range(tabla->mbhc_data.t_dce, tabla->mbhc_data.t_dce);
+ tabla->mbhc_data.dce_mb = tabla_codec_read_dce_result(codec);
+
+ /* Sta measuremnt for 0 volts */
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x0A);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x02);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x02);
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x81);
+ usleep_range(100, 100);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x02);
+ usleep_range(tabla->mbhc_data.t_sta, tabla->mbhc_data.t_sta);
+ tabla->mbhc_data.sta_z = tabla_codec_read_sta_result(codec);
+
+ /* STA Measurement for MB Voltage */
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x82);
+ usleep_range(100, 100);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_EN_CTL, 0x02);
+ usleep_range(tabla->mbhc_data.t_sta, tabla->mbhc_data.t_sta);
+ tabla->mbhc_data.sta_mb = tabla_codec_read_sta_result(codec);
+
+ /* Restore default settings. */
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x04, 0x00);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.cfilt_ctl, 0x40,
+ cfilt_mode);
+ snd_soc_update_bits(codec, TABLA_A_BIAS_CENTRAL_BG_CTL, 0x02, bg_mode);
+
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x84);
+ usleep_range(100, 100);
+
+ wcd9xxx_enable_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL);
+ tabla_turn_onoff_rel_detection(codec, true);
+}
+
+void *tabla_mbhc_cal_btn_det_mp(const struct tabla_mbhc_btn_detect_cfg* btn_det,
+ const enum tabla_mbhc_btn_det_mem mem)
+{
+ void *ret = &btn_det->_v_btn_low;
+
+ switch (mem) {
+ case TABLA_BTN_DET_GAIN:
+ ret += sizeof(btn_det->_n_cic);
+ case TABLA_BTN_DET_N_CIC:
+ ret += sizeof(btn_det->_n_ready);
+ case TABLA_BTN_DET_N_READY:
+ ret += sizeof(btn_det->_v_btn_high[0]) * btn_det->num_btn;
+ case TABLA_BTN_DET_V_BTN_HIGH:
+ ret += sizeof(btn_det->_v_btn_low[0]) * btn_det->num_btn;
+ case TABLA_BTN_DET_V_BTN_LOW:
+ /* do nothing */
+ break;
+ default:
+ ret = NULL;
+ }
+
+ return ret;
+}
+
+static s16 tabla_scale_v_micb_vddio(struct tabla_priv *tabla, int v,
+ bool tovddio)
+{
+ int r;
+ int vddio_k, mb_k;
+ vddio_k = tabla_find_k_value(tabla->pdata->micbias.ldoh_v,
+ VDDIO_MICBIAS_MV);
+ mb_k = tabla_find_k_value(tabla->pdata->micbias.ldoh_v,
+ tabla->mbhc_data.micb_mv);
+ if (tovddio)
+ r = v * vddio_k / mb_k;
+ else
+ r = v * mb_k / vddio_k;
+ return r;
+}
+
+static void tabla_mbhc_calc_thres(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla;
+ s16 btn_mv = 0, btn_delta_mv;
+ struct tabla_mbhc_btn_detect_cfg *btn_det;
+ struct tabla_mbhc_plug_type_cfg *plug_type;
+ u16 *btn_high;
+ u8 *n_ready;
+ int i;
+
+ tabla = snd_soc_codec_get_drvdata(codec);
+ btn_det = TABLA_MBHC_CAL_BTN_DET_PTR(tabla->mbhc_cfg.calibration);
+ plug_type = TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla->mbhc_cfg.calibration);
+
+ n_ready = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_N_READY);
+ if (tabla->mbhc_cfg.mclk_rate == TABLA_MCLK_RATE_12288KHZ) {
+ tabla->mbhc_data.npoll = 4;
+ tabla->mbhc_data.nbounce_wait = 30;
+ } else if (tabla->mbhc_cfg.mclk_rate == TABLA_MCLK_RATE_9600KHZ) {
+ tabla->mbhc_data.npoll = 7;
+ tabla->mbhc_data.nbounce_wait = 23;
+ }
+
+ tabla->mbhc_data.t_sta_dce = ((1000 * 256) /
+ (tabla->mbhc_cfg.mclk_rate / 1000) *
+ n_ready[tabla_codec_mclk_index(tabla)]) +
+ 10;
+ tabla->mbhc_data.v_ins_hu =
+ tabla_codec_v_sta_dce(codec, STA, plug_type->v_hs_max);
+ tabla->mbhc_data.v_ins_h =
+ tabla_codec_v_sta_dce(codec, DCE, plug_type->v_hs_max);
+
+ tabla->mbhc_data.v_inval_ins_low = TABLA_MBHC_FAKE_INSERT_LOW;
+ if (tabla->mbhc_cfg.gpio)
+ tabla->mbhc_data.v_inval_ins_high =
+ TABLA_MBHC_FAKE_INSERT_HIGH;
+ else
+ tabla->mbhc_data.v_inval_ins_high =
+ TABLA_MBHC_FAKE_INS_HIGH_NO_GPIO;
+
+ if (tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV) {
+ tabla->mbhc_data.adj_v_hs_max =
+ tabla_scale_v_micb_vddio(tabla, plug_type->v_hs_max, true);
+ tabla->mbhc_data.adj_v_ins_hu =
+ tabla_codec_v_sta_dce(codec, STA,
+ tabla->mbhc_data.adj_v_hs_max);
+ tabla->mbhc_data.adj_v_ins_h =
+ tabla_codec_v_sta_dce(codec, DCE,
+ tabla->mbhc_data.adj_v_hs_max);
+ tabla->mbhc_data.v_inval_ins_low =
+ tabla_scale_v_micb_vddio(tabla,
+ tabla->mbhc_data.v_inval_ins_low,
+ false);
+ tabla->mbhc_data.v_inval_ins_high =
+ tabla_scale_v_micb_vddio(tabla,
+ tabla->mbhc_data.v_inval_ins_high,
+ false);
+ }
+
+ btn_high = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_V_BTN_HIGH);
+ for (i = 0; i < btn_det->num_btn; i++)
+ btn_mv = btn_high[i] > btn_mv ? btn_high[i] : btn_mv;
+
+ tabla->mbhc_data.v_b1_h = tabla_codec_v_sta_dce(codec, DCE, btn_mv);
+ btn_delta_mv = btn_mv + btn_det->v_btn_press_delta_sta;
+ tabla->mbhc_data.v_b1_hu =
+ tabla_codec_v_sta_dce(codec, STA, btn_delta_mv);
+
+ btn_delta_mv = btn_mv + btn_det->v_btn_press_delta_cic;
+
+ tabla->mbhc_data.v_b1_huc =
+ tabla_codec_v_sta_dce(codec, DCE, btn_delta_mv);
+
+ tabla->mbhc_data.v_brh = tabla->mbhc_data.v_b1_h;
+ tabla->mbhc_data.v_brl = TABLA_MBHC_BUTTON_MIN;
+
+ tabla->mbhc_data.v_no_mic =
+ tabla_codec_v_sta_dce(codec, STA, plug_type->v_no_mic);
+}
+
+void tabla_mbhc_init(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla;
+ struct tabla_mbhc_general_cfg *generic;
+ struct tabla_mbhc_btn_detect_cfg *btn_det;
+ int n;
+ u8 *n_cic, *gain;
+ struct wcd9xxx *tabla_core = dev_get_drvdata(codec->dev->parent);
+
+ tabla = snd_soc_codec_get_drvdata(codec);
+ generic = TABLA_MBHC_CAL_GENERAL_PTR(tabla->mbhc_cfg.calibration);
+ btn_det = TABLA_MBHC_CAL_BTN_DET_PTR(tabla->mbhc_cfg.calibration);
+
+ for (n = 0; n < 8; n++) {
+ if ((!TABLA_IS_1_X(tabla_core->version)) || n != 7) {
+ snd_soc_update_bits(codec,
+ TABLA_A_CDC_MBHC_FEATURE_B1_CFG,
+ 0x07, n);
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_FEATURE_B2_CFG,
+ btn_det->c[n]);
+ }
+ }
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B2_CTL, 0x07,
+ btn_det->nc);
+
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_N_CIC);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_TIMER_B6_CTL, 0xFF,
+ n_cic[tabla_codec_mclk_index(tabla)]);
+
+ gain = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_GAIN);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B2_CTL, 0x78,
+ gain[tabla_codec_mclk_index(tabla)] << 3);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_TIMER_B4_CTL, 0x70,
+ generic->mbhc_nsa << 4);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_TIMER_B4_CTL, 0x0F,
+ btn_det->n_meas);
+
+ snd_soc_write(codec, TABLA_A_CDC_MBHC_TIMER_B5_CTL, generic->mbhc_navg);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x80, 0x80);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x78,
+ btn_det->mbhc_nsc << 3);
+
+ snd_soc_update_bits(codec, tabla->reg_addr.micb_4_mbhc, 0x03,
+ TABLA_MICBIAS2);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x02, 0x02);
+
+ snd_soc_update_bits(codec, TABLA_A_MBHC_SCALING_MUX_2, 0xF0, 0xF0);
+}
+
+static bool tabla_mbhc_fw_validate(const struct firmware *fw)
+{
+ u32 cfg_offset;
+ struct tabla_mbhc_imped_detect_cfg *imped_cfg;
+ struct tabla_mbhc_btn_detect_cfg *btn_cfg;
+
+ if (fw->size < TABLA_MBHC_CAL_MIN_SIZE)
+ return false;
+
+ /* previous check guarantees that there is enough fw data up
+ * to num_btn
+ */
+ btn_cfg = TABLA_MBHC_CAL_BTN_DET_PTR(fw->data);
+ cfg_offset = (u32) ((void *) btn_cfg - (void *) fw->data);
+ if (fw->size < (cfg_offset + TABLA_MBHC_CAL_BTN_SZ(btn_cfg)))
+ return false;
+
+ /* previous check guarantees that there is enough fw data up
+ * to start of impedance detection configuration
+ */
+ imped_cfg = TABLA_MBHC_CAL_IMPED_DET_PTR(fw->data);
+ cfg_offset = (u32) ((void *) imped_cfg - (void *) fw->data);
+
+ if (fw->size < (cfg_offset + TABLA_MBHC_CAL_IMPED_MIN_SZ))
+ return false;
+
+ if (fw->size < (cfg_offset + TABLA_MBHC_CAL_IMPED_SZ(imped_cfg)))
+ return false;
+
+ return true;
+}
+
+/* called under codec_resource_lock acquisition */
+static int tabla_determine_button(const struct tabla_priv *priv,
+ const s32 micmv)
+{
+ s16 *v_btn_low, *v_btn_high;
+ struct tabla_mbhc_btn_detect_cfg *btn_det;
+ int i, btn = -1;
+
+ btn_det = TABLA_MBHC_CAL_BTN_DET_PTR(priv->mbhc_cfg.calibration);
+ v_btn_low = tabla_mbhc_cal_btn_det_mp(btn_det, TABLA_BTN_DET_V_BTN_LOW);
+ v_btn_high = tabla_mbhc_cal_btn_det_mp(btn_det,
+ TABLA_BTN_DET_V_BTN_HIGH);
+
+ for (i = 0; i < btn_det->num_btn; i++) {
+ if ((v_btn_low[i] <= micmv) && (v_btn_high[i] >= micmv)) {
+ btn = i;
+ break;
+ }
+ }
+
+ if (btn == -1)
+ pr_debug("%s: couldn't find button number for mic mv %d\n",
+ __func__, micmv);
+
+ return btn;
+}
+
+static int tabla_get_button_mask(const int btn)
+{
+ int mask = 0;
+ switch (btn) {
+ case 0:
+ mask = SND_JACK_BTN_0;
+ break;
+ case 1:
+ mask = SND_JACK_BTN_1;
+ break;
+ case 2:
+ mask = SND_JACK_BTN_2;
+ break;
+ case 3:
+ mask = SND_JACK_BTN_3;
+ break;
+ case 4:
+ mask = SND_JACK_BTN_4;
+ break;
+ case 5:
+ mask = SND_JACK_BTN_5;
+ break;
+ case 6:
+ mask = SND_JACK_BTN_6;
+ break;
+ case 7:
+ mask = SND_JACK_BTN_7;
+ break;
+ }
+ return mask;
+}
+
+static irqreturn_t tabla_dce_handler(int irq, void *data)
+{
+ int i, mask;
+ short dce, sta;
+ s32 mv, mv_s, stamv_s;
+ bool vddio;
+ int btn = -1, meas = 0;
+ struct tabla_priv *priv = data;
+ const struct tabla_mbhc_btn_detect_cfg *d =
+ TABLA_MBHC_CAL_BTN_DET_PTR(priv->mbhc_cfg.calibration);
+ short btnmeas[d->n_btn_meas + 1];
+ struct snd_soc_codec *codec = priv->codec;
+ struct wcd9xxx *core = dev_get_drvdata(priv->codec->dev->parent);
+ int n_btn_meas = d->n_btn_meas;
+ u8 mbhc_status = snd_soc_read(codec, TABLA_A_CDC_MBHC_B1_STATUS) & 0x3E;
+
+ pr_debug("%s: enter\n", __func__);
+
+ TABLA_ACQUIRE_LOCK(priv->codec_resource_lock);
+ if (priv->mbhc_state == MBHC_STATE_POTENTIAL_RECOVERY) {
+ pr_debug("%s: mbhc is being recovered, skip button press\n",
+ __func__);
+ goto done;
+ }
+
+ priv->mbhc_state = MBHC_STATE_POTENTIAL;
+
+ if (!priv->mbhc_polling_active) {
+ pr_warn("%s: mbhc polling is not active, skip button press\n",
+ __func__);
+ goto done;
+ }
+
+ dce = tabla_codec_read_dce_result(codec);
+ mv = tabla_codec_sta_dce_v(codec, 1, dce);
+
+ /* If GPIO interrupt already kicked in, ignore button press */
+ if (priv->in_gpio_handler) {
+ pr_debug("%s: GPIO State Changed, ignore button press\n",
+ __func__);
+ btn = -1;
+ goto done;
+ }
+
+ vddio = (priv->mbhc_data.micb_mv != VDDIO_MICBIAS_MV &&
+ priv->mbhc_micbias_switched);
+ mv_s = vddio ? tabla_scale_v_micb_vddio(priv, mv, false) : mv;
+
+ if (mbhc_status != TABLA_MBHC_STATUS_REL_DETECTION) {
+ if (priv->mbhc_last_resume &&
+ !time_after(jiffies, priv->mbhc_last_resume + HZ)) {
+ pr_debug("%s: Button is already released shortly after "
+ "resume\n", __func__);
+ n_btn_meas = 0;
+ } else {
+ pr_debug("%s: Button is already released without "
+ "resume", __func__);
+ sta = tabla_codec_read_sta_result(codec);
+ stamv_s = tabla_codec_sta_dce_v(codec, 0, sta);
+ if (vddio)
+ stamv_s = tabla_scale_v_micb_vddio(priv,
+ stamv_s,
+ false);
+ btn = tabla_determine_button(priv, mv_s);
+ if (btn != tabla_determine_button(priv, stamv_s))
+ btn = -1;
+ goto done;
+ }
+ }
+
+ /* determine pressed button */
+ btnmeas[meas++] = tabla_determine_button(priv, mv_s);
+ pr_debug("%s: meas %d - DCE %d,%d,%d button %d\n", __func__,
+ meas - 1, dce, mv, mv_s, btnmeas[meas - 1]);
+ if (n_btn_meas == 0)
+ btn = btnmeas[0];
+ for (; ((d->n_btn_meas) && (meas < (d->n_btn_meas + 1))); meas++) {
+ dce = tabla_codec_sta_dce(codec, 1, false);
+ mv = tabla_codec_sta_dce_v(codec, 1, dce);
+ mv_s = vddio ? tabla_scale_v_micb_vddio(priv, mv, false) : mv;
+
+ btnmeas[meas] = tabla_determine_button(priv, mv_s);
+ pr_debug("%s: meas %d - DCE %d,%d,%d button %d\n",
+ __func__, meas, dce, mv, mv_s, btnmeas[meas]);
+ /* if large enough measurements are collected,
+ * start to check if last all n_btn_con measurements were
+ * in same button low/high range */
+ if (meas + 1 >= d->n_btn_con) {
+ for (i = 0; i < d->n_btn_con; i++)
+ if ((btnmeas[meas] < 0) ||
+ (btnmeas[meas] != btnmeas[meas - i]))
+ break;
+ if (i == d->n_btn_con) {
+ /* button pressed */
+ btn = btnmeas[meas];
+ break;
+ } else if ((n_btn_meas - meas) < (d->n_btn_con - 1)) {
+ /* if left measurements are less than n_btn_con,
+ * it's impossible to find button number */
+ break;
+ }
+ }
+ }
+
+ if (btn >= 0) {
+ if (priv->in_gpio_handler) {
+ pr_debug("%s: GPIO already triggered, ignore button "
+ "press\n", __func__);
+ goto done;
+ }
+ mask = tabla_get_button_mask(btn);
+ priv->buttons_pressed |= mask;
+ wcd9xxx_lock_sleep(core);
+ if (schedule_delayed_work(&priv->mbhc_btn_dwork,
+ msecs_to_jiffies(400)) == 0) {
+ WARN(1, "Button pressed twice without release"
+ "event\n");
+ wcd9xxx_unlock_sleep(core);
+ }
+ } else {
+ pr_debug("%s: bogus button press, too short press?\n",
+ __func__);
+ }
+
+ done:
+ pr_debug("%s: leave\n", __func__);
+ TABLA_RELEASE_LOCK(priv->codec_resource_lock);
+ return IRQ_HANDLED;
+}
+
+static int tabla_is_fake_press(struct tabla_priv *priv)
+{
+ int i;
+ int r = 0;
+ struct snd_soc_codec *codec = priv->codec;
+ const int dces = MBHC_NUM_DCE_PLUG_DETECT;
+ s16 mb_v, v_ins_hu, v_ins_h;
+
+ v_ins_hu = tabla_get_current_v_ins(priv, true);
+ v_ins_h = tabla_get_current_v_ins(priv, false);
+
+ for (i = 0; i < dces; i++) {
+ usleep_range(10000, 10000);
+ if (i == 0) {
+ mb_v = tabla_codec_sta_dce(codec, 0, true);
+ pr_debug("%s: STA[0]: %d,%d\n", __func__, mb_v,
+ tabla_codec_sta_dce_v(codec, 0, mb_v));
+ if (mb_v < (s16)priv->mbhc_data.v_b1_hu ||
+ mb_v > v_ins_hu) {
+ r = 1;
+ break;
+ }
+ } else {
+ mb_v = tabla_codec_sta_dce(codec, 1, true);
+ pr_debug("%s: DCE[%d]: %d,%d\n", __func__, i, mb_v,
+ tabla_codec_sta_dce_v(codec, 1, mb_v));
+ if (mb_v < (s16)priv->mbhc_data.v_b1_h ||
+ mb_v > v_ins_h) {
+ r = 1;
+ break;
+ }
+ }
+ }
+
+ return r;
+}
+
+static irqreturn_t tabla_release_handler(int irq, void *data)
+{
+ int ret;
+ struct tabla_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+
+ pr_debug("%s: enter\n", __func__);
+
+ TABLA_ACQUIRE_LOCK(priv->codec_resource_lock);
+ priv->mbhc_state = MBHC_STATE_RELEASE;
+
+ tabla_codec_drive_v_to_micbias(codec, 10000);
+
+ if (priv->buttons_pressed & TABLA_JACK_BUTTON_MASK) {
+ ret = tabla_cancel_btn_work(priv);
+ if (ret == 0) {
+ pr_debug("%s: Reporting long button release event\n",
+ __func__);
+ if (priv->mbhc_cfg.button_jack)
+ tabla_snd_soc_jack_report(priv,
+ priv->mbhc_cfg.button_jack, 0,
+ priv->buttons_pressed);
+ } else {
+ if (tabla_is_fake_press(priv)) {
+ pr_debug("%s: Fake button press interrupt\n",
+ __func__);
+ } else if (priv->mbhc_cfg.button_jack) {
+ if (priv->in_gpio_handler) {
+ pr_debug("%s: GPIO kicked in, ignore\n",
+ __func__);
+ } else {
+ pr_debug("%s: Reporting short button "
+ "press and release\n",
+ __func__);
+ tabla_snd_soc_jack_report(priv,
+ priv->mbhc_cfg.button_jack,
+ priv->buttons_pressed,
+ priv->buttons_pressed);
+ tabla_snd_soc_jack_report(priv,
+ priv->mbhc_cfg.button_jack, 0,
+ priv->buttons_pressed);
+ }
+ }
+ }
+
+ priv->buttons_pressed &= ~TABLA_JACK_BUTTON_MASK;
+ }
+
+ tabla_codec_calibrate_hs_polling(codec);
+
+ if (priv->mbhc_cfg.gpio)
+ msleep(TABLA_MBHC_GPIO_REL_DEBOUNCE_TIME_MS);
+
+ tabla_codec_start_hs_polling(codec);
+
+ pr_debug("%s: leave\n", __func__);
+ TABLA_RELEASE_LOCK(priv->codec_resource_lock);
+ return IRQ_HANDLED;
+}
+
+static void tabla_codec_shutdown_hs_removal_detect(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ const struct tabla_mbhc_general_cfg *generic =
+ TABLA_MBHC_CAL_GENERAL_PTR(tabla->mbhc_cfg.calibration);
+
+ if (!tabla->mclk_enabled && !tabla->mbhc_polling_active)
+ tabla_codec_enable_config_mode(codec, 1);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0x2, 0x2);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x6, 0x0);
+
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg, 0x80, 0x00);
+
+ usleep_range(generic->t_shutdown_plug_rem,
+ generic->t_shutdown_plug_rem);
+
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL, 0xA, 0x8);
+ if (!tabla->mclk_enabled && !tabla->mbhc_polling_active)
+ tabla_codec_enable_config_mode(codec, 0);
+
+ snd_soc_write(codec, TABLA_A_MBHC_SCALING_MUX_1, 0x00);
+}
+
+static void tabla_codec_cleanup_hs_polling(struct snd_soc_codec *codec)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ tabla_codec_shutdown_hs_removal_detect(codec);
+
+ if (!tabla->mclk_enabled) {
+ tabla_codec_disable_clock_block(codec);
+ tabla_codec_enable_bandgap(codec, TABLA_BANDGAP_OFF);
+ }
+
+ tabla->mbhc_polling_active = false;
+ tabla->mbhc_state = MBHC_STATE_NONE;
+}
+
+static irqreturn_t tabla_hphl_ocp_irq(int irq, void *data)
+{
+ struct tabla_priv *tabla = data;
+ struct snd_soc_codec *codec;
+
+ pr_info("%s: received HPHL OCP irq\n", __func__);
+
+ if (tabla) {
+ codec = tabla->codec;
+ if (tabla->hphlocp_cnt++ < TABLA_OCP_ATTEMPT) {
+ pr_info("%s: retry\n", __func__);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10,
+ 0x00);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10,
+ 0x10);
+ } else {
+ wcd9xxx_disable_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPL_FAULT);
+ tabla->hphlocp_cnt = 0;
+ tabla->hph_status |= SND_JACK_OC_HPHL;
+ if (tabla->mbhc_cfg.headset_jack)
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.headset_jack,
+ tabla->hph_status,
+ TABLA_JACK_MASK);
+ }
+ } else {
+ pr_err("%s: Bad tabla private data\n", __func__);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t tabla_hphr_ocp_irq(int irq, void *data)
+{
+ struct tabla_priv *tabla = data;
+ struct snd_soc_codec *codec;
+
+ pr_info("%s: received HPHR OCP irq\n", __func__);
+
+ if (tabla) {
+ codec = tabla->codec;
+ if (tabla->hphrocp_cnt++ < TABLA_OCP_ATTEMPT) {
+ pr_info("%s: retry\n", __func__);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10,
+ 0x00);
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10,
+ 0x10);
+ } else {
+ wcd9xxx_disable_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPR_FAULT);
+ tabla->hphrocp_cnt = 0;
+ tabla->hph_status |= SND_JACK_OC_HPHR;
+ if (tabla->mbhc_cfg.headset_jack)
+ tabla_snd_soc_jack_report(tabla,
+ tabla->mbhc_cfg.headset_jack,
+ tabla->hph_status,
+ TABLA_JACK_MASK);
+ }
+ } else {
+ pr_err("%s: Bad tabla private data\n", __func__);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static bool tabla_is_inval_ins_range(struct snd_soc_codec *codec,
+ s32 mic_volt, bool highhph, bool *highv)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ bool invalid = false;
+ s16 v_hs_max;
+
+ /* Perform this check only when the high voltage headphone
+ * needs to be considered as invalid
+ */
+ v_hs_max = tabla_get_current_v_hs_max(tabla);
+ *highv = mic_volt > v_hs_max;
+ if (!highhph && *highv)
+ invalid = true;
+ else if (mic_volt < tabla->mbhc_data.v_inval_ins_high &&
+ (mic_volt > tabla->mbhc_data.v_inval_ins_low))
+ invalid = true;
+
+ return invalid;
+}
+
+static bool tabla_is_inval_ins_delta(struct snd_soc_codec *codec,
+ int mic_volt, int mic_volt_prev,
+ int threshold)
+{
+ return abs(mic_volt - mic_volt_prev) > threshold;
+}
+
+/* called under codec_resource_lock acquisition */
+void tabla_find_plug_and_report(struct snd_soc_codec *codec,
+ enum tabla_mbhc_plug_type plug_type)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ if (plug_type == PLUG_TYPE_HEADPHONE &&
+ tabla->current_plug == PLUG_TYPE_NONE) {
+ /* Nothing was reported previously
+ * report a headphone or unsupported
+ */
+ tabla_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+ tabla_codec_cleanup_hs_polling(codec);
+ } else if (plug_type == PLUG_TYPE_GND_MIC_SWAP) {
+ if (tabla->current_plug == PLUG_TYPE_HEADSET)
+ tabla_codec_report_plug(codec, 0, SND_JACK_HEADSET);
+ else if (tabla->current_plug == PLUG_TYPE_HEADPHONE)
+ tabla_codec_report_plug(codec, 0, SND_JACK_HEADPHONE);
+
+ tabla_codec_report_plug(codec, 1, SND_JACK_UNSUPPORTED);
+ tabla_codec_cleanup_hs_polling(codec);
+ } else if (plug_type == PLUG_TYPE_HEADSET) {
+ /* If Headphone was reported previously, this will
+ * only report the mic line
+ */
+ tabla_codec_report_plug(codec, 1, SND_JACK_HEADSET);
+ msleep(100);
+ tabla_codec_start_hs_polling(codec);
+ } else if (plug_type == PLUG_TYPE_HIGH_HPH) {
+ if (tabla->current_plug == PLUG_TYPE_NONE)
+ tabla_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+ tabla_codec_cleanup_hs_polling(codec);
+ pr_debug("setup mic trigger for further detection\n");
+ tabla->lpi_enabled = true;
+ tabla_codec_enable_hs_detect(codec, 1,
+ MBHC_USE_MB_TRIGGER |
+ MBHC_USE_HPHL_TRIGGER,
+ false);
+ } else {
+ WARN(1, "Unexpected current plug_type %d, plug_type %d\n",
+ tabla->current_plug, plug_type);
+ }
+}
+
+/* should be called under interrupt context that hold suspend */
+static void tabla_schedule_hs_detect_plug(struct tabla_priv *tabla)
+{
+ pr_debug("%s: scheduling tabla_hs_correct_gpio_plug\n", __func__);
+ tabla->hs_detect_work_stop = false;
+ wcd9xxx_lock_sleep(tabla->codec->control_data);
+ schedule_work(&tabla->hs_correct_plug_work);
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_cancel_hs_detect_plug(struct tabla_priv *tabla)
+{
+ pr_debug("%s: canceling hs_correct_plug_work\n", __func__);
+ tabla->hs_detect_work_stop = true;
+ wmb();
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ if (cancel_work_sync(&tabla->hs_correct_plug_work)) {
+ pr_debug("%s: hs_correct_plug_work is canceled\n", __func__);
+ wcd9xxx_unlock_sleep(tabla->codec->control_data);
+ }
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+}
+
+static bool tabla_hs_gpio_level_remove(struct tabla_priv *tabla)
+{
+ return (gpio_get_value_cansleep(tabla->mbhc_cfg.gpio) !=
+ tabla->mbhc_cfg.gpio_level_insert);
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_hphr_gnd_switch(struct snd_soc_codec *codec, bool on)
+{
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x01, on);
+ if (on)
+ usleep_range(5000, 5000);
+}
+
+/* called under codec_resource_lock acquisition and mbhc override = 1 */
+static enum tabla_mbhc_plug_type
+tabla_codec_get_plug_type(struct snd_soc_codec *codec, bool highhph)
+{
+ int i;
+ bool gndswitch, vddioswitch;
+ int scaled;
+ struct tabla_mbhc_plug_type_cfg *plug_type_ptr;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ const bool vddio = (tabla->mbhc_data.micb_mv != VDDIO_MICBIAS_MV);
+ int num_det = (MBHC_NUM_DCE_PLUG_DETECT + vddio);
+ enum tabla_mbhc_plug_type plug_type[num_det];
+ s16 mb_v[num_det];
+ s32 mic_mv[num_det];
+ bool inval;
+ bool highdelta;
+ bool ahighv = false, highv;
+
+ /* make sure override is on */
+ WARN_ON(!(snd_soc_read(codec, TABLA_A_CDC_MBHC_B1_CTL) & 0x04));
+
+ /* GND and MIC swap detection requires at least 2 rounds of DCE */
+ BUG_ON(num_det < 2);
+
+ plug_type_ptr =
+ TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla->mbhc_cfg.calibration);
+
+ plug_type[0] = PLUG_TYPE_INVALID;
+
+ /* performs DCEs for N times
+ * 1st: check if voltage is in invalid range
+ * 2nd - N-2nd: check voltage range and delta
+ * N-1st: check voltage range, delta with HPHR GND switch
+ * Nth: check voltage range with VDDIO switch if micbias V != vddio V*/
+ for (i = 0; i < num_det; i++) {
+ gndswitch = (i == (num_det - 1 - vddio));
+ vddioswitch = (vddio && ((i == num_det - 1) ||
+ (i == num_det - 2)));
+ if (i == 0) {
+ mb_v[i] = tabla_codec_setup_hs_polling(codec);
+ mic_mv[i] = tabla_codec_sta_dce_v(codec, 1 , mb_v[i]);
+ inval = tabla_is_inval_ins_range(codec, mic_mv[i],
+ highhph, &highv);
+ ahighv |= highv;
+ scaled = mic_mv[i];
+ } else {
+ if (vddioswitch)
+ __tabla_codec_switch_micbias(tabla->codec, 1,
+ false, false);
+ if (gndswitch)
+ tabla_codec_hphr_gnd_switch(codec, true);
+ mb_v[i] = __tabla_codec_sta_dce(codec, 1, true, true);
+ mic_mv[i] = tabla_codec_sta_dce_v(codec, 1 , mb_v[i]);
+ if (vddioswitch)
+ scaled = tabla_scale_v_micb_vddio(tabla,
+ mic_mv[i],
+ false);
+ else
+ scaled = mic_mv[i];
+ /* !gndswitch & vddioswitch means the previous DCE
+ * was done with gndswitch, don't compare with DCE
+ * with gndswitch */
+ highdelta = tabla_is_inval_ins_delta(codec, scaled,
+ mic_mv[i - !gndswitch - vddioswitch],
+ TABLA_MBHC_FAKE_INS_DELTA_SCALED_MV);
+ inval = (tabla_is_inval_ins_range(codec, mic_mv[i],
+ highhph, &highv) ||
+ highdelta);
+ ahighv |= highv;
+ if (gndswitch)
+ tabla_codec_hphr_gnd_switch(codec, false);
+ if (vddioswitch)
+ __tabla_codec_switch_micbias(tabla->codec, 0,
+ false, false);
+ /* claim UNSUPPORTED plug insertion when
+ * good headset is detected but HPHR GND switch makes
+ * delta difference */
+ if (i == (num_det - 2) && highdelta && !ahighv)
+ plug_type[0] = PLUG_TYPE_GND_MIC_SWAP;
+ else if (i == (num_det - 1) && inval)
+ plug_type[0] = PLUG_TYPE_INVALID;
+ }
+ pr_debug("%s: DCE #%d, %04x, V %d, scaled V %d, GND %d, "
+ "VDDIO %d, inval %d\n", __func__,
+ i + 1, mb_v[i] & 0xffff, mic_mv[i], scaled, gndswitch,
+ vddioswitch, inval);
+ /* don't need to run further DCEs */
+ if (ahighv && inval)
+ break;
+ mic_mv[i] = scaled;
+ }
+
+ for (i = 0; (plug_type[0] != PLUG_TYPE_GND_MIC_SWAP && !inval) &&
+ i < num_det; i++) {
+ /*
+ * If we are here, means none of the all
+ * measurements are fake, continue plug type detection.
+ * If all three measurements do not produce same
+ * plug type, restart insertion detection
+ */
+ if (mic_mv[i] < plug_type_ptr->v_no_mic) {
+ plug_type[i] = PLUG_TYPE_HEADPHONE;
+ pr_debug("%s: Detect attempt %d, detected Headphone\n",
+ __func__, i);
+ } else if (highhph && (mic_mv[i] > plug_type_ptr->v_hs_max)) {
+ plug_type[i] = PLUG_TYPE_HIGH_HPH;
+ pr_debug("%s: Detect attempt %d, detected High "
+ "Headphone\n", __func__, i);
+ } else {
+ plug_type[i] = PLUG_TYPE_HEADSET;
+ pr_debug("%s: Detect attempt %d, detected Headset\n",
+ __func__, i);
+ }
+
+ if (i > 0 && (plug_type[i - 1] != plug_type[i])) {
+ pr_err("%s: Detect attempt %d and %d are not same",
+ __func__, i - 1, i);
+ plug_type[0] = PLUG_TYPE_INVALID;
+ inval = true;
+ break;
+ }
+ }
+
+ pr_debug("%s: Detected plug type %d\n", __func__, plug_type[0]);
+ return plug_type[0];
+}
+
+static void tabla_hs_correct_gpio_plug(struct work_struct *work)
+{
+ struct tabla_priv *tabla;
+ struct snd_soc_codec *codec;
+ int retry = 0, pt_gnd_mic_swap_cnt = 0;
+ bool correction = false;
+ enum tabla_mbhc_plug_type plug_type;
+ unsigned long timeout;
+
+ tabla = container_of(work, struct tabla_priv, hs_correct_plug_work);
+ codec = tabla->codec;
+
+ pr_debug("%s: enter\n", __func__);
+ tabla->mbhc_cfg.mclk_cb_fn(codec, 1, false);
+
+ /* Keep override on during entire plug type correction work.
+ *
+ * This is okay under the assumption that any GPIO irqs which use
+ * MBHC block cancel and sync this work so override is off again
+ * prior to GPIO interrupt handler's MBHC block usage.
+ * Also while this correction work is running, we can guarantee
+ * DAPM doesn't use any MBHC block as this work only runs with
+ * headphone detection.
+ */
+ tabla_turn_onoff_override(codec, true);
+
+ timeout = jiffies + msecs_to_jiffies(TABLA_HS_DETECT_PLUG_TIME_MS);
+ while (!time_after(jiffies, timeout)) {
+ ++retry;
+ rmb();
+ if (tabla->hs_detect_work_stop) {
+ pr_debug("%s: stop requested\n", __func__);
+ break;
+ }
+
+ msleep(TABLA_HS_DETECT_PLUG_INERVAL_MS);
+ if (tabla_hs_gpio_level_remove(tabla)) {
+ pr_debug("%s: GPIO value is low\n", __func__);
+ break;
+ }
+
+ /* can race with removal interrupt */
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ plug_type = tabla_codec_get_plug_type(codec, true);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+
+ if (plug_type == PLUG_TYPE_INVALID) {
+ pr_debug("Invalid plug in attempt # %d\n", retry);
+ if (retry == NUM_ATTEMPTS_TO_REPORT &&
+ tabla->current_plug == PLUG_TYPE_NONE) {
+ tabla_codec_report_plug(codec, 1,
+ SND_JACK_HEADPHONE);
+ }
+ } else if (plug_type == PLUG_TYPE_HEADPHONE) {
+ pr_debug("Good headphone detected, continue polling mic\n");
+ if (tabla->current_plug == PLUG_TYPE_NONE)
+ tabla_codec_report_plug(codec, 1,
+ SND_JACK_HEADPHONE);
+ } else {
+ if (plug_type == PLUG_TYPE_GND_MIC_SWAP) {
+ pt_gnd_mic_swap_cnt++;
+ if (pt_gnd_mic_swap_cnt <
+ TABLA_MBHC_GND_MIC_SWAP_THRESHOLD)
+ continue;
+ else if (pt_gnd_mic_swap_cnt >
+ TABLA_MBHC_GND_MIC_SWAP_THRESHOLD) {
+ /* This is due to GND/MIC switch didn't
+ * work, Report unsupported plug */
+ } else if (tabla->mbhc_cfg.swap_gnd_mic) {
+ /* if switch is toggled, check again,
+ * otherwise report unsupported plug */
+ if (tabla->mbhc_cfg.swap_gnd_mic(codec))
+ continue;
+ }
+ } else
+ pt_gnd_mic_swap_cnt = 0;
+
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ /* Turn off override */
+ tabla_turn_onoff_override(codec, false);
+ /* The valid plug also includes PLUG_TYPE_GND_MIC_SWAP
+ */
+ tabla_find_plug_and_report(codec, plug_type);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ pr_debug("Attempt %d found correct plug %d\n", retry,
+ plug_type);
+ correction = true;
+ break;
+ }
+ }
+
+ /* Turn off override */
+ if (!correction)
+ tabla_turn_onoff_override(codec, false);
+
+ tabla->mbhc_cfg.mclk_cb_fn(codec, 0, false);
+ pr_debug("%s: leave\n", __func__);
+ /* unlock sleep */
+ wcd9xxx_unlock_sleep(tabla->codec->control_data);
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_decide_gpio_plug(struct snd_soc_codec *codec)
+{
+ enum tabla_mbhc_plug_type plug_type;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+
+ pr_debug("%s: enter\n", __func__);
+
+ tabla_turn_onoff_override(codec, true);
+ plug_type = tabla_codec_get_plug_type(codec, true);
+ tabla_turn_onoff_override(codec, false);
+
+ if (tabla_hs_gpio_level_remove(tabla)) {
+ pr_debug("%s: GPIO value is low when determining plug\n",
+ __func__);
+ return;
+ }
+
+ if (plug_type == PLUG_TYPE_INVALID ||
+ plug_type == PLUG_TYPE_GND_MIC_SWAP) {
+ tabla_schedule_hs_detect_plug(tabla);
+ } else if (plug_type == PLUG_TYPE_HEADPHONE) {
+ tabla_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+
+ tabla_schedule_hs_detect_plug(tabla);
+ } else {
+ pr_debug("%s: Valid plug found, determine plug type %d\n",
+ __func__, plug_type);
+ tabla_find_plug_and_report(codec, plug_type);
+ }
+}
+
+/* called under codec_resource_lock acquisition */
+static void tabla_codec_detect_plug_type(struct snd_soc_codec *codec)
+{
+ enum tabla_mbhc_plug_type plug_type;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ const struct tabla_mbhc_plug_detect_cfg *plug_det =
+ TABLA_MBHC_CAL_PLUG_DET_PTR(tabla->mbhc_cfg.calibration);
+
+ /* Turn on the override,
+ * tabla_codec_setup_hs_polling requires override on */
+ tabla_turn_onoff_override(codec, true);
+
+ if (plug_det->t_ins_complete > 20)
+ msleep(plug_det->t_ins_complete);
+ else
+ usleep_range(plug_det->t_ins_complete * 1000,
+ plug_det->t_ins_complete * 1000);
+
+ if (tabla->mbhc_cfg.gpio) {
+ /* Turn off the override */
+ tabla_turn_onoff_override(codec, false);
+ if (tabla_hs_gpio_level_remove(tabla))
+ pr_debug("%s: GPIO value is low when determining "
+ "plug\n", __func__);
+ else
+ tabla_codec_decide_gpio_plug(codec);
+ return;
+ }
+
+ plug_type = tabla_codec_get_plug_type(codec, false);
+ tabla_turn_onoff_override(codec, false);
+
+ if (plug_type == PLUG_TYPE_INVALID) {
+ pr_debug("%s: Invalid plug type detected\n", __func__);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_B1_CTL, 0x02, 0x02);
+ tabla_codec_cleanup_hs_polling(codec);
+ tabla_codec_enable_hs_detect(codec, 1,
+ MBHC_USE_MB_TRIGGER |
+ MBHC_USE_HPHL_TRIGGER, false);
+ } else if (plug_type == PLUG_TYPE_GND_MIC_SWAP) {
+ pr_debug("%s: GND-MIC swapped plug type detected\n", __func__);
+ tabla_codec_report_plug(codec, 1, SND_JACK_UNSUPPORTED);
+ tabla_codec_cleanup_hs_polling(codec);
+ tabla_codec_enable_hs_detect(codec, 0, 0, false);
+ } else if (plug_type == PLUG_TYPE_HEADPHONE) {
+ pr_debug("%s: Headphone Detected\n", __func__);
+ tabla_codec_report_plug(codec, 1, SND_JACK_HEADPHONE);
+ tabla_codec_cleanup_hs_polling(codec);
+ tabla_codec_enable_hs_detect(codec, 0, 0, false);
+ } else if (plug_type == PLUG_TYPE_HEADSET) {
+ pr_debug("%s: Headset detected\n", __func__);
+ tabla_codec_report_plug(codec, 1, SND_JACK_HEADSET);
+
+ /* avoid false button press detect */
+ msleep(50);
+ tabla_codec_start_hs_polling(codec);
+ }
+}
+
+/* called only from interrupt which is under codec_resource_lock acquisition */
+static void tabla_hs_insert_irq_gpio(struct tabla_priv *priv, bool is_removal)
+{
+ struct snd_soc_codec *codec = priv->codec;
+
+ if (!is_removal) {
+ pr_debug("%s: MIC trigger insertion interrupt\n", __func__);
+
+ rmb();
+ if (priv->lpi_enabled)
+ msleep(100);
+
+ rmb();
+ if (!priv->lpi_enabled) {
+ pr_debug("%s: lpi is disabled\n", __func__);
+ } else if (gpio_get_value_cansleep(priv->mbhc_cfg.gpio) ==
+ priv->mbhc_cfg.gpio_level_insert) {
+ pr_debug("%s: Valid insertion, "
+ "detect plug type\n", __func__);
+ tabla_codec_decide_gpio_plug(codec);
+ } else {
+ pr_debug("%s: Invalid insertion, "
+ "stop plug detection\n", __func__);
+ }
+ } else {
+ pr_err("%s: GPIO used, invalid MBHC Removal\n", __func__);
+ }
+}
+
+/* called only from interrupt which is under codec_resource_lock acquisition */
+static void tabla_hs_insert_irq_nogpio(struct tabla_priv *priv, bool is_removal,
+ bool is_mb_trigger)
+{
+ int ret;
+ struct snd_soc_codec *codec = priv->codec;
+ struct wcd9xxx *core = dev_get_drvdata(priv->codec->dev->parent);
+
+ if (is_removal) {
+ /* cancel possiblely running hs detect work */
+ tabla_cancel_hs_detect_plug(priv);
+
+ /*
+ * If headphone is removed while playback is in progress,
+ * it is possible that micbias will be switched to VDDIO.
+ */
+ tabla_codec_switch_micbias(codec, 0);
+ if (priv->current_plug == PLUG_TYPE_HEADPHONE)
+ tabla_codec_report_plug(codec, 0, SND_JACK_HEADPHONE);
+ else if (priv->current_plug == PLUG_TYPE_GND_MIC_SWAP)
+ tabla_codec_report_plug(codec, 0, SND_JACK_UNSUPPORTED);
+ else
+ WARN(1, "%s: Unexpected current plug type %d\n",
+ __func__, priv->current_plug);
+ tabla_codec_shutdown_hs_removal_detect(codec);
+ tabla_codec_enable_hs_detect(codec, 1,
+ MBHC_USE_MB_TRIGGER |
+ MBHC_USE_HPHL_TRIGGER,
+ true);
+ } else if (is_mb_trigger && !is_removal) {
+ pr_debug("%s: Waiting for Headphone left trigger\n",
+ __func__);
+ wcd9xxx_lock_sleep(core);
+ if (schedule_delayed_work(&priv->mbhc_insert_dwork,
+ usecs_to_jiffies(1000000)) == 0) {
+ pr_err("%s: mbhc_insert_dwork is already scheduled\n",
+ __func__);
+ wcd9xxx_unlock_sleep(core);
+ }
+ tabla_codec_enable_hs_detect(codec, 1, MBHC_USE_HPHL_TRIGGER,
+ false);
+ } else {
+ ret = cancel_delayed_work(&priv->mbhc_insert_dwork);
+ if (ret != 0) {
+ pr_debug("%s: Complete plug insertion, Detecting plug "
+ "type\n", __func__);
+ tabla_codec_detect_plug_type(codec);
+ wcd9xxx_unlock_sleep(core);
+ } else {
+ wcd9xxx_enable_irq(codec->control_data,
+ TABLA_IRQ_MBHC_INSERTION);
+ pr_err("%s: Error detecting plug insertion\n",
+ __func__);
+ }
+ }
+}
+
+static irqreturn_t tabla_hs_insert_irq(int irq, void *data)
+{
+ bool is_mb_trigger, is_removal;
+ struct tabla_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+
+ pr_debug("%s: enter\n", __func__);
+ TABLA_ACQUIRE_LOCK(priv->codec_resource_lock);
+ wcd9xxx_disable_irq(codec->control_data, TABLA_IRQ_MBHC_INSERTION);
+
+ is_mb_trigger = !!(snd_soc_read(codec, priv->mbhc_bias_regs.mbhc_reg) &
+ 0x10);
+ is_removal = !!(snd_soc_read(codec, TABLA_A_CDC_MBHC_INT_CTL) & 0x02);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_INT_CTL, 0x03, 0x00);
+
+ /* Turn off both HPH and MIC line schmitt triggers */
+ snd_soc_update_bits(codec, priv->mbhc_bias_regs.mbhc_reg, 0x90, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x13, 0x00);
+ snd_soc_update_bits(codec, priv->mbhc_bias_regs.ctl_reg, 0x01, 0x00);
+
+ if (priv->mbhc_cfg.gpio)
+ tabla_hs_insert_irq_gpio(priv, is_removal);
+ else
+ tabla_hs_insert_irq_nogpio(priv, is_removal, is_mb_trigger);
+
+ TABLA_RELEASE_LOCK(priv->codec_resource_lock);
+ return IRQ_HANDLED;
+}
+
+static bool is_valid_mic_voltage(struct snd_soc_codec *codec, s32 mic_mv)
+{
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ const struct tabla_mbhc_plug_type_cfg *plug_type =
+ TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla->mbhc_cfg.calibration);
+ const s16 v_hs_max = tabla_get_current_v_hs_max(tabla);
+
+ return (!(mic_mv > 10 && mic_mv < 80) && (mic_mv > plug_type->v_no_mic)
+ && (mic_mv < v_hs_max)) ? true : false;
+}
+
+/* called under codec_resource_lock acquisition
+ * returns true if mic voltage range is back to normal insertion
+ * returns false either if timedout or removed */
+static bool tabla_hs_remove_settle(struct snd_soc_codec *codec)
+{
+ int i;
+ bool timedout, settled = false;
+ s32 mic_mv[MBHC_NUM_DCE_PLUG_DETECT];
+ short mb_v[MBHC_NUM_DCE_PLUG_DETECT];
+ unsigned long retry = 0, timeout;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ const s16 v_hs_max = tabla_get_current_v_hs_max(tabla);
+
+ timeout = jiffies + msecs_to_jiffies(TABLA_HS_DETECT_PLUG_TIME_MS);
+ while (!(timedout = time_after(jiffies, timeout))) {
+ retry++;
+ if (tabla->mbhc_cfg.gpio && tabla_hs_gpio_level_remove(tabla)) {
+ pr_debug("%s: GPIO indicates removal\n", __func__);
+ break;
+ }
+
+ if (tabla->mbhc_cfg.gpio) {
+ if (retry > 1)
+ msleep(250);
+ else
+ msleep(50);
+ }
+
+ if (tabla->mbhc_cfg.gpio && tabla_hs_gpio_level_remove(tabla)) {
+ pr_debug("%s: GPIO indicates removal\n", __func__);
+ break;
+ }
+
+ for (i = 0; i < MBHC_NUM_DCE_PLUG_DETECT; i++) {
+ mb_v[i] = tabla_codec_sta_dce(codec, 1, true);
+ mic_mv[i] = tabla_codec_sta_dce_v(codec, 1 , mb_v[i]);
+ pr_debug("%s : DCE run %lu, mic_mv = %d(%x)\n",
+ __func__, retry, mic_mv[i], mb_v[i]);
+ }
+
+ if (tabla->mbhc_cfg.gpio && tabla_hs_gpio_level_remove(tabla)) {
+ pr_debug("%s: GPIO indicates removal\n", __func__);
+ break;
+ }
+
+ if (tabla->current_plug == PLUG_TYPE_NONE) {
+ pr_debug("%s : headset/headphone is removed\n",
+ __func__);
+ break;
+ }
+
+ for (i = 0; i < MBHC_NUM_DCE_PLUG_DETECT; i++)
+ if (!is_valid_mic_voltage(codec, mic_mv[i]))
+ break;
+
+ if (i == MBHC_NUM_DCE_PLUG_DETECT) {
+ pr_debug("%s: MIC voltage settled\n", __func__);
+ settled = true;
+ msleep(200);
+ break;
+ }
+
+ /* only for non-GPIO remove irq */
+ if (!tabla->mbhc_cfg.gpio) {
+ for (i = 0; i < MBHC_NUM_DCE_PLUG_DETECT; i++)
+ if (mic_mv[i] < v_hs_max)
+ break;
+ if (i == MBHC_NUM_DCE_PLUG_DETECT) {
+ pr_debug("%s: Headset is removed\n", __func__);
+ break;
+ }
+ }
+ }
+
+ if (timedout)
+ pr_debug("%s: Microphone did not settle in %d seconds\n",
+ __func__, TABLA_HS_DETECT_PLUG_TIME_MS);
+ return settled;
+}
+
+/* called only from interrupt which is under codec_resource_lock acquisition */
+static void tabla_hs_remove_irq_gpio(struct tabla_priv *priv)
+{
+ struct snd_soc_codec *codec = priv->codec;
+
+ if (tabla_hs_remove_settle(codec))
+ tabla_codec_start_hs_polling(codec);
+ pr_debug("%s: remove settle done\n", __func__);
+}
+
+/* called only from interrupt which is under codec_resource_lock acquisition */
+static void tabla_hs_remove_irq_nogpio(struct tabla_priv *priv)
+{
+ short bias_value;
+ bool removed = true;
+ struct snd_soc_codec *codec = priv->codec;
+ const struct tabla_mbhc_general_cfg *generic =
+ TABLA_MBHC_CAL_GENERAL_PTR(priv->mbhc_cfg.calibration);
+ int min_us = TABLA_FAKE_REMOVAL_MIN_PERIOD_MS * 1000;
+
+ if (priv->current_plug != PLUG_TYPE_HEADSET) {
+ pr_debug("%s(): Headset is not inserted, ignore removal\n",
+ __func__);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL,
+ 0x08, 0x08);
+ return;
+ }
+
+ usleep_range(generic->t_shutdown_plug_rem,
+ generic->t_shutdown_plug_rem);
+
+ do {
+ bias_value = tabla_codec_sta_dce(codec, 1, true);
+ pr_debug("%s: DCE %d,%d, %d us left\n", __func__, bias_value,
+ tabla_codec_sta_dce_v(codec, 1, bias_value), min_us);
+ if (bias_value < tabla_get_current_v_ins(priv, false)) {
+ pr_debug("%s: checking false removal\n", __func__);
+ msleep(500);
+ removed = !tabla_hs_remove_settle(codec);
+ pr_debug("%s: headset %sactually removed\n", __func__,
+ removed ? "" : "not ");
+ break;
+ }
+ min_us -= priv->mbhc_data.t_dce;
+ } while (min_us > 0);
+
+ if (removed) {
+ /* cancel possiblely running hs detect work */
+ tabla_cancel_hs_detect_plug(priv);
+ /*
+ * If this removal is not false, first check the micbias
+ * switch status and switch it to LDOH if it is already
+ * switched to VDDIO.
+ */
+ tabla_codec_switch_micbias(codec, 0);
+
+ tabla_codec_report_plug(codec, 0, SND_JACK_HEADSET);
+ tabla_codec_cleanup_hs_polling(codec);
+ tabla_codec_enable_hs_detect(codec, 1,
+ MBHC_USE_MB_TRIGGER |
+ MBHC_USE_HPHL_TRIGGER,
+ true);
+ } else {
+ tabla_codec_start_hs_polling(codec);
+ }
+}
+
+static irqreturn_t tabla_hs_remove_irq(int irq, void *data)
+{
+ struct tabla_priv *priv = data;
+ bool vddio;
+ pr_debug("%s: enter, removal interrupt\n", __func__);
+
+ TABLA_ACQUIRE_LOCK(priv->codec_resource_lock);
+ vddio = (priv->mbhc_data.micb_mv != VDDIO_MICBIAS_MV &&
+ priv->mbhc_micbias_switched);
+ if (vddio)
+ __tabla_codec_switch_micbias(priv->codec, 0, false, true);
+
+ if (priv->mbhc_cfg.gpio)
+ tabla_hs_remove_irq_gpio(priv);
+ else
+ tabla_hs_remove_irq_nogpio(priv);
+
+ /* if driver turned off vddio switch and headset is not removed,
+ * turn on the vddio switch back, if headset is removed then vddio
+ * switch is off by time now and shouldn't be turn on again from here */
+ if (vddio && priv->current_plug == PLUG_TYPE_HEADSET)
+ __tabla_codec_switch_micbias(priv->codec, 1, true, true);
+ TABLA_RELEASE_LOCK(priv->codec_resource_lock);
+
+ return IRQ_HANDLED;
+}
+
+void mbhc_insert_work(struct work_struct *work)
+{
+ struct delayed_work *dwork;
+ struct tabla_priv *tabla;
+ struct snd_soc_codec *codec;
+ struct wcd9xxx *tabla_core;
+
+ dwork = to_delayed_work(work);
+ tabla = container_of(dwork, struct tabla_priv, mbhc_insert_dwork);
+ codec = tabla->codec;
+ tabla_core = dev_get_drvdata(codec->dev->parent);
+
+ pr_debug("%s:\n", __func__);
+
+ /* Turn off both HPH and MIC line schmitt triggers */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.mbhc_reg, 0x90, 0x00);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x13, 0x00);
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg, 0x01, 0x00);
+ wcd9xxx_disable_irq_sync(codec->control_data, TABLA_IRQ_MBHC_INSERTION);
+ tabla_codec_detect_plug_type(codec);
+ wcd9xxx_unlock_sleep(tabla_core);
+}
+
+static void tabla_hs_gpio_handler(struct snd_soc_codec *codec)
+{
+ bool insert;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ bool is_removed = false;
+
+ pr_debug("%s: enter\n", __func__);
+
+ tabla->in_gpio_handler = true;
+ /* Wait here for debounce time */
+ usleep_range(TABLA_GPIO_IRQ_DEBOUNCE_TIME_US,
+ TABLA_GPIO_IRQ_DEBOUNCE_TIME_US);
+
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+
+ /* cancel pending button press */
+ if (tabla_cancel_btn_work(tabla))
+ pr_debug("%s: button press is canceled\n", __func__);
+
+ insert = (gpio_get_value_cansleep(tabla->mbhc_cfg.gpio) ==
+ tabla->mbhc_cfg.gpio_level_insert);
+ if ((tabla->current_plug == PLUG_TYPE_NONE) && insert) {
+ tabla->lpi_enabled = false;
+ wmb();
+
+ /* cancel detect plug */
+ tabla_cancel_hs_detect_plug(tabla);
+
+ /* Disable Mic Bias pull down and HPH Switch to GND */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg, 0x01,
+ 0x00);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x01, 0x00);
+ tabla_codec_detect_plug_type(codec);
+ } else if ((tabla->current_plug != PLUG_TYPE_NONE) && !insert) {
+ tabla->lpi_enabled = false;
+ wmb();
+
+ /* cancel detect plug */
+ tabla_cancel_hs_detect_plug(tabla);
+
+ if (tabla->current_plug == PLUG_TYPE_HEADPHONE) {
+ tabla_codec_report_plug(codec, 0, SND_JACK_HEADPHONE);
+ is_removed = true;
+ } else if (tabla->current_plug == PLUG_TYPE_GND_MIC_SWAP) {
+ tabla_codec_report_plug(codec, 0, SND_JACK_UNSUPPORTED);
+ is_removed = true;
+ } else if (tabla->current_plug == PLUG_TYPE_HEADSET) {
+ tabla_codec_pause_hs_polling(codec);
+ tabla_codec_cleanup_hs_polling(codec);
+ tabla_codec_report_plug(codec, 0, SND_JACK_HEADSET);
+ is_removed = true;
+ }
+
+ if (is_removed) {
+ /* Enable Mic Bias pull down and HPH Switch to GND */
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.ctl_reg, 0x01,
+ 0x01);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x01,
+ 0x01);
+ /* Make sure mic trigger is turned off */
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.ctl_reg,
+ 0x01, 0x01);
+ snd_soc_update_bits(codec,
+ tabla->mbhc_bias_regs.mbhc_reg,
+ 0x90, 0x00);
+ /* Reset MBHC State Machine */
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL,
+ 0x08, 0x08);
+ snd_soc_update_bits(codec, TABLA_A_CDC_MBHC_CLK_CTL,
+ 0x08, 0x00);
+ /* Turn off override */
+ tabla_turn_onoff_override(codec, false);
+ }
+ }
+
+ tabla->in_gpio_handler = false;
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ pr_debug("%s: leave\n", __func__);
+}
+
+static irqreturn_t tabla_mechanical_plug_detect_irq(int irq, void *data)
+{
+ int r = IRQ_HANDLED;
+ struct snd_soc_codec *codec = data;
+
+ if (unlikely(wcd9xxx_lock_sleep(codec->control_data) == false)) {
+ pr_warn("%s: failed to hold suspend\n", __func__);
+ r = IRQ_NONE;
+ } else {
+ tabla_hs_gpio_handler(codec);
+ wcd9xxx_unlock_sleep(codec->control_data);
+ }
+
+ return r;
+}
+
+static int tabla_mbhc_init_and_calibrate(struct tabla_priv *tabla)
+{
+ int ret = 0;
+ struct snd_soc_codec *codec = tabla->codec;
+
+ tabla->mbhc_cfg.mclk_cb_fn(codec, 1, false);
+ tabla_mbhc_init(codec);
+ tabla_mbhc_cal(codec);
+ tabla_mbhc_calc_thres(codec);
+ tabla->mbhc_cfg.mclk_cb_fn(codec, 0, false);
+ tabla_codec_calibrate_hs_polling(codec);
+ if (!tabla->mbhc_cfg.gpio) {
+ ret = tabla_codec_enable_hs_detect(codec, 1,
+ MBHC_USE_MB_TRIGGER |
+ MBHC_USE_HPHL_TRIGGER,
+ false);
+
+ if (IS_ERR_VALUE(ret))
+ pr_err("%s: Failed to setup MBHC detection\n",
+ __func__);
+ } else {
+ /* Enable Mic Bias pull down and HPH Switch to GND */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.ctl_reg,
+ 0x01, 0x01);
+ snd_soc_update_bits(codec, TABLA_A_MBHC_HPH, 0x01, 0x01);
+ INIT_WORK(&tabla->hs_correct_plug_work,
+ tabla_hs_correct_gpio_plug);
+ }
+
+ if (!IS_ERR_VALUE(ret)) {
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL, 0x10, 0x10);
+ wcd9xxx_enable_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPL_FAULT);
+ wcd9xxx_enable_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPR_FAULT);
+
+ if (tabla->mbhc_cfg.gpio) {
+ ret = request_threaded_irq(tabla->mbhc_cfg.gpio_irq,
+ NULL,
+ tabla_mechanical_plug_detect_irq,
+ (IRQF_TRIGGER_RISING |
+ IRQF_TRIGGER_FALLING),
+ "tabla-gpio", codec);
+ if (!IS_ERR_VALUE(ret)) {
+ ret = enable_irq_wake(tabla->mbhc_cfg.gpio_irq);
+ /* Bootup time detection */
+ tabla_hs_gpio_handler(codec);
+ }
+ }
+ }
+
+ return ret;
+}
+
+static void mbhc_fw_read(struct work_struct *work)
+{
+ struct delayed_work *dwork;
+ struct tabla_priv *tabla;
+ struct snd_soc_codec *codec;
+ const struct firmware *fw;
+ int ret = -1, retry = 0;
+
+ dwork = to_delayed_work(work);
+ tabla = container_of(dwork, struct tabla_priv, mbhc_firmware_dwork);
+ codec = tabla->codec;
+
+ while (retry < MBHC_FW_READ_ATTEMPTS) {
+ retry++;
+ pr_info("%s:Attempt %d to request MBHC firmware\n",
+ __func__, retry);
+ ret = request_firmware(&fw, "wcd9310/wcd9310_mbhc.bin",
+ codec->dev);
+
+ if (ret != 0) {
+ usleep_range(MBHC_FW_READ_TIMEOUT,
+ MBHC_FW_READ_TIMEOUT);
+ } else {
+ pr_info("%s: MBHC Firmware read succesful\n", __func__);
+ break;
+ }
+ }
+
+ if (ret != 0) {
+ pr_err("%s: Cannot load MBHC firmware use default cal\n",
+ __func__);
+ } else if (tabla_mbhc_fw_validate(fw) == false) {
+ pr_err("%s: Invalid MBHC cal data size use default cal\n",
+ __func__);
+ release_firmware(fw);
+ } else {
+ tabla->mbhc_cfg.calibration = (void *)fw->data;
+ tabla->mbhc_fw = fw;
+ }
+
+ (void) tabla_mbhc_init_and_calibrate(tabla);
+}
+
+int tabla_hs_detect(struct snd_soc_codec *codec,
+ const struct tabla_mbhc_config *cfg)
+{
+ struct tabla_priv *tabla;
+ int rc = 0;
+
+ if (!codec || !cfg->calibration) {
+ pr_err("Error: no codec or calibration\n");
+ return -EINVAL;
+ }
+
+ if (cfg->mclk_rate != TABLA_MCLK_RATE_12288KHZ) {
+ if (cfg->mclk_rate == TABLA_MCLK_RATE_9600KHZ)
+ pr_err("Error: clock rate %dHz is not yet supported\n",
+ cfg->mclk_rate);
+ else
+ pr_err("Error: unsupported clock rate %d\n",
+ cfg->mclk_rate);
+ return -EINVAL;
+ }
+
+ tabla = snd_soc_codec_get_drvdata(codec);
+ tabla->mbhc_cfg = *cfg;
+ tabla->in_gpio_handler = false;
+ tabla->current_plug = PLUG_TYPE_NONE;
+ tabla->lpi_enabled = false;
+ tabla_get_mbhc_micbias_regs(codec, &tabla->mbhc_bias_regs);
+
+ /* Put CFILT in fast mode by default */
+ snd_soc_update_bits(codec, tabla->mbhc_bias_regs.cfilt_ctl,
+ 0x40, TABLA_CFILT_FAST_MODE);
+ INIT_DELAYED_WORK(&tabla->mbhc_firmware_dwork, mbhc_fw_read);
+ INIT_DELAYED_WORK(&tabla->mbhc_btn_dwork, btn_lpress_fn);
+ INIT_WORK(&tabla->hphlocp_work, hphlocp_off_report);
+ INIT_WORK(&tabla->hphrocp_work, hphrocp_off_report);
+ INIT_DELAYED_WORK(&tabla->mbhc_insert_dwork, mbhc_insert_work);
+
+ if (!tabla->mbhc_cfg.read_fw_bin)
+ rc = tabla_mbhc_init_and_calibrate(tabla);
+ else
+ schedule_delayed_work(&tabla->mbhc_firmware_dwork,
+ usecs_to_jiffies(MBHC_FW_READ_TIMEOUT));
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(tabla_hs_detect);
+
+static unsigned long slimbus_value;
+
+static irqreturn_t tabla_slimbus_irq(int irq, void *data)
+{
+ struct tabla_priv *priv = data;
+ struct snd_soc_codec *codec = priv->codec;
+ int i, j;
+ u8 val;
+
+ for (i = 0; i < WCD9XXX_SLIM_NUM_PORT_REG; i++) {
+ slimbus_value = wcd9xxx_interface_reg_read(codec->control_data,
+ TABLA_SLIM_PGD_PORT_INT_STATUS0 + i);
+ for_each_set_bit(j, &slimbus_value, BITS_PER_BYTE) {
+ val = wcd9xxx_interface_reg_read(codec->control_data,
+ TABLA_SLIM_PGD_PORT_INT_SOURCE0 + i*8 + j);
+ if (val & 0x1)
+ pr_err_ratelimited("overflow error on port %x,"
+ " value %x\n", i*8 + j, val);
+ if (val & 0x2)
+ pr_err_ratelimited("underflow error on port %x,"
+ " value %x\n", i*8 + j, val);
+ }
+ wcd9xxx_interface_reg_write(codec->control_data,
+ TABLA_SLIM_PGD_PORT_INT_CLR0 + i, 0xFF);
+ }
+
+ return IRQ_HANDLED;
+}
+
+static int tabla_handle_pdata(struct tabla_priv *tabla)
+{
+ struct snd_soc_codec *codec = tabla->codec;
+ struct wcd9xxx_pdata *pdata = tabla->pdata;
+ int k1, k2, k3, rc = 0;
+ u8 leg_mode = pdata->amic_settings.legacy_mode;
+ u8 txfe_bypass = pdata->amic_settings.txfe_enable;
+ u8 txfe_buff = pdata->amic_settings.txfe_buff;
+ u8 flag = pdata->amic_settings.use_pdata;
+ u8 i = 0, j = 0;
+ u8 val_txfe = 0, value = 0;
+
+ if (!pdata) {
+ rc = -ENODEV;
+ goto done;
+ }
+
+ /* Make sure settings are correct */
+ if ((pdata->micbias.ldoh_v > TABLA_LDOH_2P85_V) ||
+ (pdata->micbias.bias1_cfilt_sel > TABLA_CFILT3_SEL) ||
+ (pdata->micbias.bias2_cfilt_sel > TABLA_CFILT3_SEL) ||
+ (pdata->micbias.bias3_cfilt_sel > TABLA_CFILT3_SEL) ||
+ (pdata->micbias.bias4_cfilt_sel > TABLA_CFILT3_SEL)) {
+ rc = -EINVAL;
+ goto done;
+ }
+
+ /* figure out k value */
+ k1 = tabla_find_k_value(pdata->micbias.ldoh_v,
+ pdata->micbias.cfilt1_mv);
+ k2 = tabla_find_k_value(pdata->micbias.ldoh_v,
+ pdata->micbias.cfilt2_mv);
+ k3 = tabla_find_k_value(pdata->micbias.ldoh_v,
+ pdata->micbias.cfilt3_mv);
+
+ if (IS_ERR_VALUE(k1) || IS_ERR_VALUE(k2) || IS_ERR_VALUE(k3)) {
+ rc = -EINVAL;
+ goto done;
+ }
+
+ /* Set voltage level and always use LDO */
+ snd_soc_update_bits(codec, TABLA_A_LDO_H_MODE_1, 0x0C,
+ (pdata->micbias.ldoh_v << 2));
+
+ snd_soc_update_bits(codec, TABLA_A_MICB_CFILT_1_VAL, 0xFC,
+ (k1 << 2));
+ snd_soc_update_bits(codec, TABLA_A_MICB_CFILT_2_VAL, 0xFC,
+ (k2 << 2));
+ snd_soc_update_bits(codec, TABLA_A_MICB_CFILT_3_VAL, 0xFC,
+ (k3 << 2));
+
+ snd_soc_update_bits(codec, TABLA_A_MICB_1_CTL, 0x60,
+ (pdata->micbias.bias1_cfilt_sel << 5));
+ snd_soc_update_bits(codec, TABLA_A_MICB_2_CTL, 0x60,
+ (pdata->micbias.bias2_cfilt_sel << 5));
+ snd_soc_update_bits(codec, TABLA_A_MICB_3_CTL, 0x60,
+ (pdata->micbias.bias3_cfilt_sel << 5));
+ snd_soc_update_bits(codec, tabla->reg_addr.micb_4_ctl, 0x60,
+ (pdata->micbias.bias4_cfilt_sel << 5));
+
+ for (i = 0; i < 6; j++, i += 2) {
+ if (flag & (0x01 << i)) {
+ value = (leg_mode & (0x01 << i)) ? 0x10 : 0x00;
+ val_txfe = (txfe_bypass & (0x01 << i)) ? 0x20 : 0x00;
+ val_txfe = val_txfe |
+ ((txfe_buff & (0x01 << i)) ? 0x10 : 0x00);
+ snd_soc_update_bits(codec, TABLA_A_TX_1_2_EN + j * 10,
+ 0x10, value);
+ snd_soc_update_bits(codec,
+ TABLA_A_TX_1_2_TEST_EN + j * 10,
+ 0x30, val_txfe);
+ }
+ if (flag & (0x01 << (i + 1))) {
+ value = (leg_mode & (0x01 << (i + 1))) ? 0x01 : 0x00;
+ val_txfe = (txfe_bypass &
+ (0x01 << (i + 1))) ? 0x02 : 0x00;
+ val_txfe |= (txfe_buff &
+ (0x01 << (i + 1))) ? 0x01 : 0x00;
+ snd_soc_update_bits(codec, TABLA_A_TX_1_2_EN + j * 10,
+ 0x01, value);
+ snd_soc_update_bits(codec,
+ TABLA_A_TX_1_2_TEST_EN + j * 10,
+ 0x03, val_txfe);
+ }
+ }
+ if (flag & 0x40) {
+ value = (leg_mode & 0x40) ? 0x10 : 0x00;
+ value = value | ((txfe_bypass & 0x40) ? 0x02 : 0x00);
+ value = value | ((txfe_buff & 0x40) ? 0x01 : 0x00);
+ snd_soc_update_bits(codec, TABLA_A_TX_7_MBHC_EN,
+ 0x13, value);
+ }
+
+ if (pdata->ocp.use_pdata) {
+ /* not defined in CODEC specification */
+ if (pdata->ocp.hph_ocp_limit == 1 ||
+ pdata->ocp.hph_ocp_limit == 5) {
+ rc = -EINVAL;
+ goto done;
+ }
+ snd_soc_update_bits(codec, TABLA_A_RX_COM_OCP_CTL,
+ 0x0F, pdata->ocp.num_attempts);
+ snd_soc_write(codec, TABLA_A_RX_COM_OCP_COUNT,
+ ((pdata->ocp.run_time << 4) | pdata->ocp.wait_time));
+ snd_soc_update_bits(codec, TABLA_A_RX_HPH_OCP_CTL,
+ 0xE0, (pdata->ocp.hph_ocp_limit << 5));
+ }
+
+ for (i = 0; i < ARRAY_SIZE(pdata->regulator); i++) {
+ if (!strncmp(pdata->regulator[i].name, "CDC_VDDA_RX", 11)) {
+ if (pdata->regulator[i].min_uV == 1800000 &&
+ pdata->regulator[i].max_uV == 1800000) {
+ snd_soc_write(codec, TABLA_A_BIAS_REF_CTL,
+ 0x1C);
+ } else if (pdata->regulator[i].min_uV == 2200000 &&
+ pdata->regulator[i].max_uV == 2200000) {
+ snd_soc_write(codec, TABLA_A_BIAS_REF_CTL,
+ 0x1E);
+ } else {
+ pr_err("%s: unsupported CDC_VDDA_RX voltage "
+ "min %d, max %d\n", __func__,
+ pdata->regulator[i].min_uV,
+ pdata->regulator[i].max_uV);
+ rc = -EINVAL;
+ }
+ break;
+ }
+ }
+done:
+ return rc;
+}
+
+static const struct tabla_reg_mask_val tabla_1_1_reg_defaults[] = {
+
+ /* Tabla 1.1 MICBIAS changes */
+ TABLA_REG_VAL(TABLA_A_MICB_1_INT_RBIAS, 0x24),
+ TABLA_REG_VAL(TABLA_A_MICB_2_INT_RBIAS, 0x24),
+ TABLA_REG_VAL(TABLA_A_MICB_3_INT_RBIAS, 0x24),
+
+ /* Tabla 1.1 HPH changes */
+ TABLA_REG_VAL(TABLA_A_RX_HPH_BIAS_PA, 0x57),
+ TABLA_REG_VAL(TABLA_A_RX_HPH_BIAS_LDO, 0x56),
+
+ /* Tabla 1.1 EAR PA changes */
+ TABLA_REG_VAL(TABLA_A_RX_EAR_BIAS_PA, 0xA6),
+ TABLA_REG_VAL(TABLA_A_RX_EAR_GAIN, 0x02),
+ TABLA_REG_VAL(TABLA_A_RX_EAR_VCM, 0x03),
+
+ /* Tabla 1.1 Lineout_5 Changes */
+ TABLA_REG_VAL(TABLA_A_RX_LINE_5_GAIN, 0x10),
+
+ /* Tabla 1.1 RX Changes */
+ TABLA_REG_VAL(TABLA_A_CDC_RX1_B5_CTL, 0x78),
+ TABLA_REG_VAL(TABLA_A_CDC_RX2_B5_CTL, 0x78),
+ TABLA_REG_VAL(TABLA_A_CDC_RX3_B5_CTL, 0x78),
+ TABLA_REG_VAL(TABLA_A_CDC_RX4_B5_CTL, 0x78),
+ TABLA_REG_VAL(TABLA_A_CDC_RX5_B5_CTL, 0x78),
+ TABLA_REG_VAL(TABLA_A_CDC_RX6_B5_CTL, 0x78),
+ TABLA_REG_VAL(TABLA_A_CDC_RX7_B5_CTL, 0x78),
+
+ /* Tabla 1.1 RX1 and RX2 Changes */
+ TABLA_REG_VAL(TABLA_A_CDC_RX1_B6_CTL, 0xA0),
+ TABLA_REG_VAL(TABLA_A_CDC_RX2_B6_CTL, 0xA0),
+
+ /* Tabla 1.1 RX3 to RX7 Changes */
+ TABLA_REG_VAL(TABLA_A_CDC_RX3_B6_CTL, 0x80),
+ TABLA_REG_VAL(TABLA_A_CDC_RX4_B6_CTL, 0x80),
+ TABLA_REG_VAL(TABLA_A_CDC_RX5_B6_CTL, 0x80),
+ TABLA_REG_VAL(TABLA_A_CDC_RX6_B6_CTL, 0x80),
+ TABLA_REG_VAL(TABLA_A_CDC_RX7_B6_CTL, 0x80),
+
+ /* Tabla 1.1 CLASSG Changes */
+ TABLA_REG_VAL(TABLA_A_CDC_CLSG_FREQ_THRESH_B3_CTL, 0x1B),
+};
+
+static const struct tabla_reg_mask_val tabla_2_0_reg_defaults[] = {
+ /* Tabla 2.0 MICBIAS changes */
+ TABLA_REG_VAL(TABLA_A_MICB_2_MBHC, 0x02),
+};
+
+static const struct tabla_reg_mask_val tabla_1_x_only_reg_2_0_defaults[] = {
+ TABLA_REG_VAL(TABLA_1_A_MICB_4_INT_RBIAS, 0x24),
+};
+
+static const struct tabla_reg_mask_val tabla_2_only_reg_2_0_defaults[] = {
+ TABLA_REG_VAL(TABLA_2_A_MICB_4_INT_RBIAS, 0x24),
+};
+
+static void tabla_update_reg_defaults(struct snd_soc_codec *codec)
+{
+ u32 i;
+ struct wcd9xxx *tabla_core = dev_get_drvdata(codec->dev->parent);
+
+ for (i = 0; i < ARRAY_SIZE(tabla_1_1_reg_defaults); i++)
+ snd_soc_write(codec, tabla_1_1_reg_defaults[i].reg,
+ tabla_1_1_reg_defaults[i].val);
+
+ for (i = 0; i < ARRAY_SIZE(tabla_2_0_reg_defaults); i++)
+ snd_soc_write(codec, tabla_2_0_reg_defaults[i].reg,
+ tabla_2_0_reg_defaults[i].val);
+
+ if (TABLA_IS_1_X(tabla_core->version)) {
+ for (i = 0; i < ARRAY_SIZE(tabla_1_x_only_reg_2_0_defaults);
+ i++)
+ snd_soc_write(codec,
+ tabla_1_x_only_reg_2_0_defaults[i].reg,
+ tabla_1_x_only_reg_2_0_defaults[i].val);
+ } else {
+ for (i = 0; i < ARRAY_SIZE(tabla_2_only_reg_2_0_defaults); i++)
+ snd_soc_write(codec,
+ tabla_2_only_reg_2_0_defaults[i].reg,
+ tabla_2_only_reg_2_0_defaults[i].val);
+ }
+}
+
+static const struct tabla_reg_mask_val tabla_codec_reg_init_val[] = {
+ /* Initialize current threshold to 350MA
+ * number of wait and run cycles to 4096
+ */
+ {TABLA_A_RX_HPH_OCP_CTL, 0xE0, 0x60},
+ {TABLA_A_RX_COM_OCP_COUNT, 0xFF, 0xFF},
+
+ {TABLA_A_QFUSE_CTL, 0xFF, 0x03},
+
+ /* Initialize gain registers to use register gain */
+ {TABLA_A_RX_HPH_L_GAIN, 0x10, 0x10},
+ {TABLA_A_RX_HPH_R_GAIN, 0x10, 0x10},
+ {TABLA_A_RX_LINE_1_GAIN, 0x10, 0x10},
+ {TABLA_A_RX_LINE_2_GAIN, 0x10, 0x10},
+ {TABLA_A_RX_LINE_3_GAIN, 0x10, 0x10},
+ {TABLA_A_RX_LINE_4_GAIN, 0x10, 0x10},
+
+ /* Initialize mic biases to differential mode */
+ {TABLA_A_MICB_1_INT_RBIAS, 0x24, 0x24},
+ {TABLA_A_MICB_2_INT_RBIAS, 0x24, 0x24},
+ {TABLA_A_MICB_3_INT_RBIAS, 0x24, 0x24},
+
+ {TABLA_A_CDC_CONN_CLSG_CTL, 0x3C, 0x14},
+
+ /* Use 16 bit sample size for TX1 to TX6 */
+ {TABLA_A_CDC_CONN_TX_SB_B1_CTL, 0x30, 0x20},
+ {TABLA_A_CDC_CONN_TX_SB_B2_CTL, 0x30, 0x20},
+ {TABLA_A_CDC_CONN_TX_SB_B3_CTL, 0x30, 0x20},
+ {TABLA_A_CDC_CONN_TX_SB_B4_CTL, 0x30, 0x20},
+ {TABLA_A_CDC_CONN_TX_SB_B5_CTL, 0x30, 0x20},
+ {TABLA_A_CDC_CONN_TX_SB_B6_CTL, 0x30, 0x20},
+
+ /* Use 16 bit sample size for TX7 to TX10 */
+ {TABLA_A_CDC_CONN_TX_SB_B7_CTL, 0x60, 0x40},
+ {TABLA_A_CDC_CONN_TX_SB_B8_CTL, 0x60, 0x40},
+ {TABLA_A_CDC_CONN_TX_SB_B9_CTL, 0x60, 0x40},
+ {TABLA_A_CDC_CONN_TX_SB_B10_CTL, 0x60, 0x40},
+
+ /* Use 16 bit sample size for RX */
+ {TABLA_A_CDC_CONN_RX_SB_B1_CTL, 0xFF, 0xAA},
+ {TABLA_A_CDC_CONN_RX_SB_B2_CTL, 0xFF, 0xAA},
+
+ /*enable HPF filter for TX paths */
+ {TABLA_A_CDC_TX1_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX2_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX3_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX4_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX5_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX6_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX7_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX8_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX9_MUX_CTL, 0x8, 0x0},
+ {TABLA_A_CDC_TX10_MUX_CTL, 0x8, 0x0},
+
+ /* config Decimator for DMIC CLK_MODE_1(3.072Mhz@12.88Mhz mclk) */
+ {TABLA_A_CDC_TX1_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX2_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX3_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX4_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX5_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX6_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX7_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX8_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX9_DMIC_CTL, 0x1, 0x1},
+ {TABLA_A_CDC_TX10_DMIC_CTL, 0x1, 0x1},
+
+ /* config DMIC clk to CLK_MODE_1 (3.072Mhz@12.88Mhz mclk) */
+ {TABLA_A_CDC_CLK_DMIC_CTL, 0x2A, 0x2A},
+
+};
+
+static const struct tabla_reg_mask_val tabla_1_x_codec_reg_init_val[] = {
+ /* Initialize mic biases to differential mode */
+ {TABLA_1_A_MICB_4_INT_RBIAS, 0x24, 0x24},
+};
+
+static const struct tabla_reg_mask_val tabla_2_higher_codec_reg_init_val[] = {
+ /* Initialize mic biases to differential mode */
+ {TABLA_2_A_MICB_4_INT_RBIAS, 0x24, 0x24},
+};
+
+static void tabla_codec_init_reg(struct snd_soc_codec *codec)
+{
+ u32 i;
+ struct wcd9xxx *tabla_core = dev_get_drvdata(codec->dev->parent);
+
+ for (i = 0; i < ARRAY_SIZE(tabla_codec_reg_init_val); i++)
+ snd_soc_update_bits(codec, tabla_codec_reg_init_val[i].reg,
+ tabla_codec_reg_init_val[i].mask,
+ tabla_codec_reg_init_val[i].val);
+ if (TABLA_IS_1_X(tabla_core->version)) {
+ for (i = 0; i < ARRAY_SIZE(tabla_1_x_codec_reg_init_val); i++)
+ snd_soc_update_bits(codec,
+ tabla_1_x_codec_reg_init_val[i].reg,
+ tabla_1_x_codec_reg_init_val[i].mask,
+ tabla_1_x_codec_reg_init_val[i].val);
+ } else {
+ for (i = 0; i < ARRAY_SIZE(tabla_2_higher_codec_reg_init_val);
+ i++)
+ snd_soc_update_bits(codec,
+ tabla_2_higher_codec_reg_init_val[i].reg,
+ tabla_2_higher_codec_reg_init_val[i].mask,
+ tabla_2_higher_codec_reg_init_val[i].val);
+ }
+}
+
+static void tabla_update_reg_address(struct tabla_priv *priv)
+{
+ struct wcd9xxx *tabla_core = dev_get_drvdata(priv->codec->dev->parent);
+ struct tabla_reg_address *reg_addr = &priv->reg_addr;
+
+ if (TABLA_IS_1_X(tabla_core->version)) {
+ reg_addr->micb_4_mbhc = TABLA_1_A_MICB_4_MBHC;
+ reg_addr->micb_4_int_rbias = TABLA_1_A_MICB_4_INT_RBIAS;
+ reg_addr->micb_4_ctl = TABLA_1_A_MICB_4_CTL;
+ } else if (TABLA_IS_2_0(tabla_core->version)) {
+ reg_addr->micb_4_mbhc = TABLA_2_A_MICB_4_MBHC;
+ reg_addr->micb_4_int_rbias = TABLA_2_A_MICB_4_INT_RBIAS;
+ reg_addr->micb_4_ctl = TABLA_2_A_MICB_4_CTL;
+ }
+}
+
+#ifdef CONFIG_DEBUG_FS
+static int codec_debug_open(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
+static ssize_t codec_debug_write(struct file *filp,
+ const char __user *ubuf, size_t cnt, loff_t *ppos)
+{
+ char lbuf[32];
+ char *buf;
+ int rc;
+ struct tabla_priv *tabla = filp->private_data;
+
+ if (cnt > sizeof(lbuf) - 1)
+ return -EINVAL;
+
+ rc = copy_from_user(lbuf, ubuf, cnt);
+ if (rc)
+ return -EFAULT;
+
+ lbuf[cnt] = '\0';
+ buf = (char *)lbuf;
+ tabla->no_mic_headset_override = (*strsep(&buf, " ") == '0') ?
+ false : true;
+ return rc;
+}
+
+static ssize_t codec_mbhc_debug_read(struct file *file, char __user *buf,
+ size_t count, loff_t *pos)
+{
+ const int size = 768;
+ char buffer[size];
+ int n = 0;
+ struct tabla_priv *tabla = file->private_data;
+ struct snd_soc_codec *codec = tabla->codec;
+ const struct mbhc_internal_cal_data *p = &tabla->mbhc_data;
+ const s16 v_ins_hu_cur = tabla_get_current_v_ins(tabla, true);
+ const s16 v_ins_h_cur = tabla_get_current_v_ins(tabla, false);
+
+ n = scnprintf(buffer, size - n, "dce_z = %x(%dmv)\n", p->dce_z,
+ tabla_codec_sta_dce_v(codec, 1, p->dce_z));
+ n += scnprintf(buffer + n, size - n, "dce_mb = %x(%dmv)\n",
+ p->dce_mb, tabla_codec_sta_dce_v(codec, 1, p->dce_mb));
+ n += scnprintf(buffer + n, size - n, "sta_z = %x(%dmv)\n",
+ p->sta_z, tabla_codec_sta_dce_v(codec, 0, p->sta_z));
+ n += scnprintf(buffer + n, size - n, "sta_mb = %x(%dmv)\n",
+ p->sta_mb, tabla_codec_sta_dce_v(codec, 0, p->sta_mb));
+ n += scnprintf(buffer + n, size - n, "t_dce = %x\n", p->t_dce);
+ n += scnprintf(buffer + n, size - n, "t_sta = %x\n", p->t_sta);
+ n += scnprintf(buffer + n, size - n, "micb_mv = %dmv\n",
+ p->micb_mv);
+ n += scnprintf(buffer + n, size - n, "v_ins_hu = %x(%dmv)%s\n",
+ p->v_ins_hu,
+ tabla_codec_sta_dce_v(codec, 0, p->v_ins_hu),
+ p->v_ins_hu == v_ins_hu_cur ? "*" : "");
+ n += scnprintf(buffer + n, size - n, "v_ins_h = %x(%dmv)%s\n",
+ p->v_ins_h, tabla_codec_sta_dce_v(codec, 1, p->v_ins_h),
+ p->v_ins_h == v_ins_h_cur ? "*" : "");
+ n += scnprintf(buffer + n, size - n, "adj_v_ins_hu = %x(%dmv)%s\n",
+ p->adj_v_ins_hu,
+ tabla_codec_sta_dce_v(codec, 0, p->adj_v_ins_hu),
+ p->adj_v_ins_hu == v_ins_hu_cur ? "*" : "");
+ n += scnprintf(buffer + n, size - n, "adj_v_ins_h = %x(%dmv)%s\n",
+ p->adj_v_ins_h,
+ tabla_codec_sta_dce_v(codec, 1, p->adj_v_ins_h),
+ p->adj_v_ins_h == v_ins_h_cur ? "*" : "");
+ n += scnprintf(buffer + n, size - n, "v_b1_hu = %x(%dmv)\n",
+ p->v_b1_hu, tabla_codec_sta_dce_v(codec, 0, p->v_b1_hu));
+ n += scnprintf(buffer + n, size - n, "v_b1_h = %x(%dmv)\n",
+ p->v_b1_h, tabla_codec_sta_dce_v(codec, 1, p->v_b1_h));
+ n += scnprintf(buffer + n, size - n, "v_b1_huc = %x(%dmv)\n",
+ p->v_b1_huc,
+ tabla_codec_sta_dce_v(codec, 1, p->v_b1_huc));
+ n += scnprintf(buffer + n, size - n, "v_brh = %x(%dmv)\n",
+ p->v_brh, tabla_codec_sta_dce_v(codec, 1, p->v_brh));
+ n += scnprintf(buffer + n, size - n, "v_brl = %x(%dmv)\n", p->v_brl,
+ tabla_codec_sta_dce_v(codec, 0, p->v_brl));
+ n += scnprintf(buffer + n, size - n, "v_no_mic = %x(%dmv)\n",
+ p->v_no_mic,
+ tabla_codec_sta_dce_v(codec, 0, p->v_no_mic));
+ n += scnprintf(buffer + n, size - n, "npoll = %d\n", p->npoll);
+ n += scnprintf(buffer + n, size - n, "nbounce_wait = %d\n",
+ p->nbounce_wait);
+ n += scnprintf(buffer + n, size - n, "v_inval_ins_low = %d\n",
+ p->v_inval_ins_low);
+ n += scnprintf(buffer + n, size - n, "v_inval_ins_high = %d\n",
+ p->v_inval_ins_high);
+ if (tabla->mbhc_cfg.gpio)
+ n += scnprintf(buffer + n, size - n, "GPIO insert = %d\n",
+ tabla_hs_gpio_level_remove(tabla));
+ buffer[n] = 0;
+
+ return simple_read_from_buffer(buf, count, pos, buffer, n);
+}
+
+static const struct file_operations codec_debug_ops = {
+ .open = codec_debug_open,
+ .write = codec_debug_write,
+};
+
+static const struct file_operations codec_mbhc_debug_ops = {
+ .open = codec_debug_open,
+ .read = codec_mbhc_debug_read,
+};
+#endif
+
+static int tabla_codec_probe(struct snd_soc_codec *codec)
+{
+ struct wcd9xxx *control;
+ struct tabla_priv *tabla;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret = 0;
+ int i;
+ int ch_cnt;
+
+ codec->control_data = dev_get_drvdata(codec->dev->parent);
+ control = codec->control_data;
+
+ tabla = kzalloc(sizeof(struct tabla_priv), GFP_KERNEL);
+ if (!tabla) {
+ dev_err(codec->dev, "Failed to allocate private data\n");
+ return -ENOMEM;
+ }
+ for (i = 0 ; i < NUM_DECIMATORS; i++) {
+ tx_hpf_work[i].tabla = tabla;
+ tx_hpf_work[i].decimator = i + 1;
+ INIT_DELAYED_WORK(&tx_hpf_work[i].dwork,
+ tx_hpf_corner_freq_callback);
+ }
+
+ /* Make sure mbhc micbias register addresses are zeroed out */
+ memset(&tabla->mbhc_bias_regs, 0,
+ sizeof(struct mbhc_micbias_regs));
+ tabla->mbhc_micbias_switched = false;
+
+ /* Make sure mbhc intenal calibration data is zeroed out */
+ memset(&tabla->mbhc_data, 0,
+ sizeof(struct mbhc_internal_cal_data));
+ tabla->mbhc_data.t_sta_dce = DEFAULT_DCE_STA_WAIT;
+ tabla->mbhc_data.t_dce = DEFAULT_DCE_WAIT;
+ tabla->mbhc_data.t_sta = DEFAULT_STA_WAIT;
+ snd_soc_codec_set_drvdata(codec, tabla);
+
+ tabla->mclk_enabled = false;
+ tabla->bandgap_type = TABLA_BANDGAP_OFF;
+ tabla->clock_active = false;
+ tabla->config_mode_active = false;
+ tabla->mbhc_polling_active = false;
+ tabla->mbhc_fake_ins_start = 0;
+ tabla->no_mic_headset_override = false;
+ tabla->hs_polling_irq_prepared = false;
+ mutex_init(&tabla->codec_resource_lock);
+ tabla->codec = codec;
+ tabla->mbhc_state = MBHC_STATE_NONE;
+ tabla->mbhc_last_resume = 0;
+ for (i = 0; i < COMPANDER_MAX; i++) {
+ tabla->comp_enabled[i] = 0;
+ tabla->comp_fs[i] = COMPANDER_FS_48KHZ;
+ }
+ tabla->pdata = dev_get_platdata(codec->dev->parent);
+ tabla->intf_type = wcd9xxx_get_intf_type();
+ tabla->aux_pga_cnt = 0;
+ tabla->aux_l_gain = 0x1F;
+ tabla->aux_r_gain = 0x1F;
+ tabla_update_reg_address(tabla);
+ tabla_update_reg_defaults(codec);
+ tabla_codec_init_reg(codec);
+ ret = tabla_handle_pdata(tabla);
+ if (IS_ERR_VALUE(ret)) {
+ pr_err("%s: bad pdata\n", __func__);
+ goto err_pdata;
+ }
+
+// snd_soc_add_codec_controls(codec, tabla_snd_controls,
+// ARRAY_SIZE(tabla_snd_controls));
+ if (TABLA_IS_1_X(control->version))
+ snd_soc_add_codec_controls(codec, tabla_1_x_snd_controls,
+ ARRAY_SIZE(tabla_1_x_snd_controls));
+ else
+ snd_soc_add_codec_controls(codec, tabla_2_higher_snd_controls,
+ ARRAY_SIZE(tabla_2_higher_snd_controls));
+
+// snd_soc_dapm_new_controls(dapm, tabla_dapm_widgets,
+// ARRAY_SIZE(tabla_dapm_widgets));
+ if (TABLA_IS_1_X(control->version))
+ snd_soc_dapm_new_controls(dapm, tabla_1_x_dapm_widgets,
+ ARRAY_SIZE(tabla_1_x_dapm_widgets));
+ else
+ snd_soc_dapm_new_controls(dapm, tabla_2_higher_dapm_widgets,
+ ARRAY_SIZE(tabla_2_higher_dapm_widgets));
+
+ if (tabla->intf_type == WCD9XXX_INTERFACE_TYPE_I2C) {
+ snd_soc_dapm_new_controls(dapm, tabla_dapm_i2s_widgets,
+ ARRAY_SIZE(tabla_dapm_i2s_widgets));
+ snd_soc_dapm_add_routes(dapm, audio_i2s_map,
+ ARRAY_SIZE(audio_i2s_map));
+ }
+// snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+ if (TABLA_IS_1_X(control->version)) {
+ snd_soc_dapm_add_routes(dapm, tabla_1_x_lineout_2_to_4_map,
+ ARRAY_SIZE(tabla_1_x_lineout_2_to_4_map));
+ } else if (TABLA_IS_2_0(control->version)) {
+ snd_soc_dapm_add_routes(dapm, tabla_2_x_lineout_2_to_4_map,
+ ARRAY_SIZE(tabla_2_x_lineout_2_to_4_map));
+ } else {
+ pr_err("%s : ERROR. Unsupported Tabla version 0x%2x\n",
+ __func__, control->version);
+ goto err_pdata;
+ }
+
+ snd_soc_dapm_sync(dapm);
+
+ ret = wcd9xxx_request_irq(codec->control_data, TABLA_IRQ_MBHC_INSERTION,
+ tabla_hs_insert_irq, "Headset insert detect", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_MBHC_INSERTION);
+ goto err_insert_irq;
+ }
+ wcd9xxx_disable_irq(codec->control_data, TABLA_IRQ_MBHC_INSERTION);
+
+ ret = wcd9xxx_request_irq(codec->control_data, TABLA_IRQ_MBHC_REMOVAL,
+ tabla_hs_remove_irq, "Headset remove detect", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_MBHC_REMOVAL);
+ goto err_remove_irq;
+ }
+
+ ret = wcd9xxx_request_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL,
+ tabla_dce_handler, "DC Estimation detect", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_MBHC_POTENTIAL);
+ goto err_potential_irq;
+ }
+
+ ret = wcd9xxx_request_irq(codec->control_data, TABLA_IRQ_MBHC_RELEASE,
+ tabla_release_handler, "Button Release detect", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_MBHC_RELEASE);
+ goto err_release_irq;
+ }
+
+ ret = wcd9xxx_request_irq(codec->control_data, TABLA_IRQ_SLIMBUS,
+ tabla_slimbus_irq, "SLIMBUS Slave", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_SLIMBUS);
+ goto err_slimbus_irq;
+ }
+
+ for (i = 0; i < WCD9XXX_SLIM_NUM_PORT_REG; i++)
+ wcd9xxx_interface_reg_write(codec->control_data,
+ TABLA_SLIM_PGD_PORT_INT_EN0 + i, 0xFF);
+
+ ret = wcd9xxx_request_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPL_FAULT, tabla_hphl_ocp_irq,
+ "HPH_L OCP detect", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_HPH_PA_OCPL_FAULT);
+ goto err_hphl_ocp_irq;
+ }
+ wcd9xxx_disable_irq(codec->control_data, TABLA_IRQ_HPH_PA_OCPL_FAULT);
+
+ ret = wcd9xxx_request_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPR_FAULT, tabla_hphr_ocp_irq,
+ "HPH_R OCP detect", tabla);
+ if (ret) {
+ pr_err("%s: Failed to request irq %d\n", __func__,
+ TABLA_IRQ_HPH_PA_OCPR_FAULT);
+ goto err_hphr_ocp_irq;
+ }
+ wcd9xxx_disable_irq(codec->control_data, TABLA_IRQ_HPH_PA_OCPR_FAULT);
+ for (i = 0; i < ARRAY_SIZE(tabla_dai); i++) {
+ switch (tabla_dai[i].id) {
+ case AIF1_PB:
+ ch_cnt = tabla_dai[i].playback.channels_max;
+ break;
+ case AIF1_CAP:
+ ch_cnt = tabla_dai[i].capture.channels_max;
+ break;
+ case AIF2_PB:
+ ch_cnt = tabla_dai[i].playback.channels_max;
+ break;
+ case AIF2_CAP:
+ ch_cnt = tabla_dai[i].capture.channels_max;
+ break;
+ case AIF3_PB:
+ ch_cnt = tabla_dai[i].playback.channels_max;
+ break;
+ case AIF3_CAP:
+ ch_cnt = tabla_dai[i].capture.channels_max;
+ break;
+ default:
+ continue;
+ }
+ tabla->dai[i].ch_num = kzalloc((sizeof(unsigned int)*
+ ch_cnt), GFP_KERNEL);
+ }
+
+#ifdef CONFIG_DEBUG_FS
+ if (ret == 0) {
+ tabla->debugfs_poke =
+ debugfs_create_file("TRRS", S_IFREG | S_IRUGO, NULL, tabla,
+ &codec_debug_ops);
+ tabla->debugfs_mbhc =
+ debugfs_create_file("tabla_mbhc", S_IFREG | S_IRUGO,
+ NULL, tabla, &codec_mbhc_debug_ops);
+ }
+#endif
+ codec->ignore_pmdown_time = 1;
+ return ret;
+
+err_hphr_ocp_irq:
+ wcd9xxx_free_irq(codec->control_data,
+ TABLA_IRQ_HPH_PA_OCPL_FAULT, tabla);
+err_hphl_ocp_irq:
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_SLIMBUS, tabla);
+err_slimbus_irq:
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_RELEASE, tabla);
+err_release_irq:
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL, tabla);
+err_potential_irq:
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_REMOVAL, tabla);
+err_remove_irq:
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_INSERTION, tabla);
+err_insert_irq:
+err_pdata:
+ mutex_destroy(&tabla->codec_resource_lock);
+ kfree(tabla);
+ return ret;
+}
+static int tabla_codec_remove(struct snd_soc_codec *codec)
+{
+ int i;
+ struct tabla_priv *tabla = snd_soc_codec_get_drvdata(codec);
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_SLIMBUS, tabla);
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_RELEASE, tabla);
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_POTENTIAL, tabla);
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_REMOVAL, tabla);
+ wcd9xxx_free_irq(codec->control_data, TABLA_IRQ_MBHC_INSERTION, tabla);
+ TABLA_ACQUIRE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_disable_clock_block(codec);
+ TABLA_RELEASE_LOCK(tabla->codec_resource_lock);
+ tabla_codec_enable_bandgap(codec, TABLA_BANDGAP_OFF);
+ if (tabla->mbhc_fw)
+ release_firmware(tabla->mbhc_fw);
+ for (i = 0; i < ARRAY_SIZE(tabla_dai); i++)
+ kfree(tabla->dai[i].ch_num);
+ mutex_destroy(&tabla->codec_resource_lock);
+#ifdef CONFIG_DEBUG_FS
+ debugfs_remove(tabla->debugfs_poke);
+ debugfs_remove(tabla->debugfs_mbhc);
+#endif
+ kfree(tabla);
+ return 0;
+}
+static struct snd_soc_codec_driver soc_codec_dev_tabla = {
+ .probe = tabla_codec_probe,
+ .remove = tabla_codec_remove,
+ .read = tabla_read,
+ .write = tabla_write,
+ .readable_register = tabla_readable,
+ .volatile_register = tabla_volatile,
+
+ .reg_cache_size = TABLA_CACHE_SIZE,
+ .reg_cache_default = tabla_reg_defaults,
+ .reg_word_size = 1,
+ .controls = tabla_snd_controls,
+ .num_controls = ARRAY_SIZE(tabla_snd_controls),
+ .dapm_widgets = tabla_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tabla_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+#ifdef CONFIG_PM
+static int tabla_suspend(struct device *dev)
+{
+ dev_dbg(dev, "%s: system suspend\n", __func__);
+ return 0;
+}
+
+static int tabla_resume(struct device *dev)
+{
+ struct platform_device *pdev = to_platform_device(dev);
+ struct tabla_priv *tabla = platform_get_drvdata(pdev);
+ dev_dbg(dev, "%s: system resume\n", __func__);
+ tabla->mbhc_last_resume = jiffies;
+ return 0;
+}
+
+static const struct dev_pm_ops tabla_pm_ops = {
+ .suspend = tabla_suspend,
+ .resume = tabla_resume,
+};
+#endif
+
+static int __devinit tabla_probe(struct platform_device *pdev)
+{
+ int ret = 0;
+ pr_err("tabla_probe\n");
+ if (wcd9xxx_get_intf_type() == WCD9XXX_INTERFACE_TYPE_SLIMBUS)
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_tabla,
+ tabla_dai, ARRAY_SIZE(tabla_dai));
+ else if (wcd9xxx_get_intf_type() == WCD9XXX_INTERFACE_TYPE_I2C)
+ ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_tabla,
+ tabla_i2s_dai, ARRAY_SIZE(tabla_i2s_dai));
+ return ret;
+}
+static int __devexit tabla_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+static struct platform_driver tabla_codec_driver = {
+ .probe = tabla_probe,
+ .remove = tabla_remove,
+ .driver = {
+ .name = "tabla_codec",
+ .owner = THIS_MODULE,
+#ifdef CONFIG_PM
+ .pm = &tabla_pm_ops,
+#endif
+ },
+};
+
+static struct platform_driver tabla1x_codec_driver = {
+ .probe = tabla_probe,
+ .remove = tabla_remove,
+ .driver = {
+ .name = "tabla1x_codec",
+ .owner = THIS_MODULE,
+#ifdef CONFIG_PM
+ .pm = &tabla_pm_ops,
+#endif
+ },
+};
+
+static int __init tabla_codec_init(void)
+{
+ int rtn = platform_driver_register(&tabla_codec_driver);
+ if (rtn == 0) {
+ rtn = platform_driver_register(&tabla1x_codec_driver);
+ if (rtn != 0)
+ platform_driver_unregister(&tabla_codec_driver);
+ }
+ return rtn;
+}
+
+static void __exit tabla_codec_exit(void)
+{
+ platform_driver_unregister(&tabla1x_codec_driver);
+ platform_driver_unregister(&tabla_codec_driver);
+}
+
+module_init(tabla_codec_init);
+module_exit(tabla_codec_exit);
+
+MODULE_DESCRIPTION("Tabla codec driver");
+MODULE_VERSION("1.0");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wcd9310.h b/sound/soc/codecs/wcd9310.h
new file mode 100644
index 0000000..1cca360
--- /dev/null
+++ b/sound/soc/codecs/wcd9310.h
@@ -0,0 +1,253 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <linux/mfd/wcd9xxx/wcd9xxx-slimslave.h>
+
+#define TABLA_NUM_REGISTERS 0x400
+#define TABLA_MAX_REGISTER (TABLA_NUM_REGISTERS-1)
+#define TABLA_CACHE_SIZE TABLA_NUM_REGISTERS
+#define TABLA_1_X_ONLY_REGISTERS 3
+#define TABLA_2_HIGHER_ONLY_REGISTERS 3
+
+#define TABLA_REG_VAL(reg, val) {reg, 0, val}
+
+#define DEFAULT_DCE_STA_WAIT 55
+#define DEFAULT_DCE_WAIT 60000
+#define DEFAULT_STA_WAIT 5000
+#define VDDIO_MICBIAS_MV 1800
+
+#define STA 0
+#define DCE 1
+
+#define TABLA_JACK_BUTTON_MASK (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \
+ SND_JACK_BTN_2 | SND_JACK_BTN_3 | \
+ SND_JACK_BTN_4 | SND_JACK_BTN_5 | \
+ SND_JACK_BTN_6 | SND_JACK_BTN_7)
+
+extern const u8 tabla_reg_readable[TABLA_CACHE_SIZE];
+extern const u32 tabla_1_reg_readable[TABLA_1_X_ONLY_REGISTERS];
+extern const u32 tabla_2_reg_readable[TABLA_2_HIGHER_ONLY_REGISTERS];
+extern const u8 tabla_reg_defaults[TABLA_CACHE_SIZE];
+
+enum tabla_micbias_num {
+ TABLA_MICBIAS1 = 0,
+ TABLA_MICBIAS2,
+ TABLA_MICBIAS3,
+ TABLA_MICBIAS4,
+};
+
+enum tabla_pid_current {
+ TABLA_PID_MIC_2P5_UA,
+ TABLA_PID_MIC_5_UA,
+ TABLA_PID_MIC_10_UA,
+ TABLA_PID_MIC_20_UA,
+};
+
+struct tabla_reg_mask_val {
+ u16 reg;
+ u8 mask;
+ u8 val;
+};
+
+enum tabla_mbhc_clk_freq {
+ TABLA_MCLK_12P2MHZ = 0,
+ TABLA_MCLK_9P6MHZ,
+ TABLA_NUM_CLK_FREQS,
+};
+
+enum tabla_mbhc_analog_pwr_cfg {
+ TABLA_ANALOG_PWR_COLLAPSED = 0,
+ TABLA_ANALOG_PWR_ON,
+ TABLA_NUM_ANALOG_PWR_CONFIGS,
+};
+
+enum tabla_mbhc_btn_det_mem {
+ TABLA_BTN_DET_V_BTN_LOW,
+ TABLA_BTN_DET_V_BTN_HIGH,
+ TABLA_BTN_DET_N_READY,
+ TABLA_BTN_DET_N_CIC,
+ TABLA_BTN_DET_GAIN
+};
+
+struct tabla_mbhc_general_cfg {
+ u8 t_ldoh;
+ u8 t_bg_fast_settle;
+ u8 t_shutdown_plug_rem;
+ u8 mbhc_nsa;
+ u8 mbhc_navg;
+ u8 v_micbias_l;
+ u8 v_micbias;
+ u8 mbhc_reserved;
+ u16 settle_wait;
+ u16 t_micbias_rampup;
+ u16 t_micbias_rampdown;
+ u16 t_supply_bringup;
+} __packed;
+
+struct tabla_mbhc_plug_detect_cfg {
+ u32 mic_current;
+ u32 hph_current;
+ u16 t_mic_pid;
+ u16 t_ins_complete;
+ u16 t_ins_retry;
+ u16 v_removal_delta;
+ u8 micbias_slow_ramp;
+ u8 reserved0;
+ u8 reserved1;
+ u8 reserved2;
+} __packed;
+
+struct tabla_mbhc_plug_type_cfg {
+ u8 av_detect;
+ u8 mono_detect;
+ u8 num_ins_tries;
+ u8 reserved0;
+ s16 v_no_mic;
+ s16 v_av_min;
+ s16 v_av_max;
+ s16 v_hs_min;
+ s16 v_hs_max;
+ u16 reserved1;
+} __packed;
+
+
+struct tabla_mbhc_btn_detect_cfg {
+ s8 c[8];
+ u8 nc;
+ u8 n_meas;
+ u8 mbhc_nsc;
+ u8 n_btn_meas;
+ u8 n_btn_con;
+ u8 num_btn;
+ u8 reserved0;
+ u8 reserved1;
+ u16 t_poll;
+ u16 t_bounce_wait;
+ u16 t_rel_timeout;
+ s16 v_btn_press_delta_sta;
+ s16 v_btn_press_delta_cic;
+ u16 t_btn0_timeout;
+ s16 _v_btn_low[0]; /* v_btn_low[num_btn] */
+ s16 _v_btn_high[0]; /* v_btn_high[num_btn] */
+ u8 _n_ready[TABLA_NUM_CLK_FREQS];
+ u8 _n_cic[TABLA_NUM_CLK_FREQS];
+ u8 _gain[TABLA_NUM_CLK_FREQS];
+} __packed;
+
+struct tabla_mbhc_imped_detect_cfg {
+ u8 _hs_imped_detect;
+ u8 _n_rload;
+ u8 _hph_keep_on;
+ u8 _repeat_rload_calc;
+ u16 _t_dac_ramp_time;
+ u16 _rhph_high;
+ u16 _rhph_low;
+ u16 _rload[0]; /* rload[n_rload] */
+ u16 _alpha[0]; /* alpha[n_rload] */
+ u16 _beta[3];
+} __packed;
+
+struct tabla_mbhc_config {
+ struct snd_soc_jack *headset_jack;
+ struct snd_soc_jack *button_jack;
+ bool read_fw_bin;
+ /* void* calibration contains:
+ * struct tabla_mbhc_general_cfg generic;
+ * struct tabla_mbhc_plug_detect_cfg plug_det;
+ * struct tabla_mbhc_plug_type_cfg plug_type;
+ * struct tabla_mbhc_btn_detect_cfg btn_det;
+ * struct tabla_mbhc_imped_detect_cfg imped_det;
+ * Note: various size depends on btn_det->num_btn
+ */
+ void *calibration;
+ enum tabla_micbias_num micbias;
+ int (*mclk_cb_fn) (struct snd_soc_codec*, int, bool);
+ unsigned int mclk_rate;
+ unsigned int gpio;
+ unsigned int gpio_irq;
+ int gpio_level_insert;
+ /* swap_gnd_mic returns true if extern GND/MIC swap switch toggled */
+ bool (*swap_gnd_mic) (struct snd_soc_codec *);
+};
+
+extern int tabla_hs_detect(struct snd_soc_codec *codec,
+ const struct tabla_mbhc_config *cfg);
+
+struct anc_header {
+ u32 reserved[3];
+ u32 num_anc_slots;
+};
+
+extern int tabla_mclk_enable(struct snd_soc_codec *codec, int mclk_enable,
+ bool dapm);
+
+extern void *tabla_mbhc_cal_btn_det_mp(const struct tabla_mbhc_btn_detect_cfg
+ *btn_det,
+ const enum tabla_mbhc_btn_det_mem mem);
+
+#define TABLA_MBHC_CAL_SIZE(buttons, rload) ( \
+ sizeof(enum tabla_micbias_num) + \
+ sizeof(struct tabla_mbhc_general_cfg) + \
+ sizeof(struct tabla_mbhc_plug_detect_cfg) + \
+ ((sizeof(s16) + sizeof(s16)) * buttons) + \
+ sizeof(struct tabla_mbhc_plug_type_cfg) + \
+ sizeof(struct tabla_mbhc_btn_detect_cfg) + \
+ sizeof(struct tabla_mbhc_imped_detect_cfg) + \
+ ((sizeof(u16) + sizeof(u16)) * rload) \
+ )
+
+#define TABLA_MBHC_CAL_GENERAL_PTR(cali) ( \
+ (struct tabla_mbhc_general_cfg *) cali)
+#define TABLA_MBHC_CAL_PLUG_DET_PTR(cali) ( \
+ (struct tabla_mbhc_plug_detect_cfg *) \
+ &(TABLA_MBHC_CAL_GENERAL_PTR(cali)[1]))
+#define TABLA_MBHC_CAL_PLUG_TYPE_PTR(cali) ( \
+ (struct tabla_mbhc_plug_type_cfg *) \
+ &(TABLA_MBHC_CAL_PLUG_DET_PTR(cali)[1]))
+#define TABLA_MBHC_CAL_BTN_DET_PTR(cali) ( \
+ (struct tabla_mbhc_btn_detect_cfg *) \
+ &(TABLA_MBHC_CAL_PLUG_TYPE_PTR(cali)[1]))
+#define TABLA_MBHC_CAL_IMPED_DET_PTR(cali) ( \
+ (struct tabla_mbhc_imped_detect_cfg *) \
+ (((void *)&TABLA_MBHC_CAL_BTN_DET_PTR(cali)[1]) + \
+ (TABLA_MBHC_CAL_BTN_DET_PTR(cali)->num_btn * \
+ (sizeof(TABLA_MBHC_CAL_BTN_DET_PTR(cali)->_v_btn_low[0]) + \
+ sizeof(TABLA_MBHC_CAL_BTN_DET_PTR(cali)->_v_btn_high[0])))) \
+ )
+
+/* minimum size of calibration data assuming there is only one button and
+ * one rload.
+ */
+#define TABLA_MBHC_CAL_MIN_SIZE ( \
+ sizeof(struct tabla_mbhc_general_cfg) + \
+ sizeof(struct tabla_mbhc_plug_detect_cfg) + \
+ sizeof(struct tabla_mbhc_plug_type_cfg) + \
+ sizeof(struct tabla_mbhc_btn_detect_cfg) + \
+ sizeof(struct tabla_mbhc_imped_detect_cfg) + \
+ (sizeof(u16) * 2))
+
+#define TABLA_MBHC_CAL_BTN_SZ(cfg_ptr) ( \
+ sizeof(struct tabla_mbhc_btn_detect_cfg) + \
+ (cfg_ptr->num_btn * (sizeof(cfg_ptr->_v_btn_low[0]) + \
+ sizeof(cfg_ptr->_v_btn_high[0]))))
+
+#define TABLA_MBHC_CAL_IMPED_MIN_SZ ( \
+ sizeof(struct tabla_mbhc_imped_detect_cfg) + \
+ sizeof(u16) * 2)
+
+#define TABLA_MBHC_CAL_IMPED_SZ(cfg_ptr) ( \
+ sizeof(struct tabla_mbhc_imped_detect_cfg) + \
+ (cfg_ptr->_n_rload * (sizeof(cfg_ptr->_rload[0]) + \
+ sizeof(cfg_ptr->_alpha[0]))))
+
+
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 6c028c4..cd2dfe8 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -643,6 +643,8 @@
SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0),
+
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
@@ -867,9 +869,11 @@
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
@@ -886,9 +890,11 @@
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
diff --git a/sound/soc/msm/Kconfig b/sound/soc/msm/Kconfig
new file mode 100644
index 0000000..36cbb23
--- /dev/null
+++ b/sound/soc/msm/Kconfig
@@ -0,0 +1,158 @@
+menu "MSM SoC Audio support"
+
+#7201 7625 variants
+config SND_MSM_DAI_SOC
+ tristate
+
+config SND_MSM_SOC_MSM7K
+ tristate
+
+config SND_MSM_SOC
+ tristate "SoC Audio for the MSM series chips"
+ depends on ARCH_MSM7X27
+ select SND_MSM_DAI_SOC
+ select SND_MSM_SOC_MSM7K
+ default n
+ help
+ To add support for ALSA PCM driver for MSM board.
+
+#7630 Variants
+config SND_MSM7KV2_DAI_SOC
+ tristate
+
+config SND_MSM_SOC_MSM7KV2
+ tristate
+
+config SND_MSM7KV2_SOC
+ tristate "SoC Audio for the MSM7KV2 chip"
+ depends on ARCH_MSM7X30 && SND_SOC && MSM7KV2_AUDIO
+ select SND_MSM_SOC_MSM7KV2
+ select SND_MSM7KV2_DAI_SOC
+ default n
+ help
+ To add support for ALSA PCM driver for QSD8k board.
+
+config SND_MSM_MVS7x30_SOC
+ tristate
+
+config SND_MSM_MVS_DAI_SOC
+ tristate
+
+config SND_MVS_SOC
+ tristate "SoC Mvs support for MSM7X30"
+ depends on SND_MSM7KV2_SOC
+ select SND_MSM_MVS7x30_SOC
+ select SND_MSM_MVS_DAI_SOC
+ default n
+ help
+ To support Mvs packet capture/playback
+
+#8660 Variants
+config SND_SOC_MSM8X60_PCM
+ tristate
+
+config SND_SOC_MSM8X60_DAI
+ tristate
+
+config SND_SOC_MSM8X60
+ tristate "SoC Audio over DSP support for MSM8660"
+ depends on ARCH_MSM8X60 && SND_SOC && MSM8X60_AUDIO
+ select SND_SOC_MSM8X60_PCM
+ select SND_SOC_MSM8X60_DAI
+ select SND_SOC_MSM_QDSP6_INTF
+ default y
+ help
+ To add support for SoC audio on MSM8X60. This driver
+ Adds support for audio over DSP. The driver adds Kcontrols
+ to do device switch/routing and volume control support for all
+ audio sessions. The kcontols also does sesion management for
+ voice calls
+
+config SND_SOC_MSM_HOSTLESS_PCM
+ tristate
+
+config SND_SOC_LPASS_PCM
+ tristate
+
+config SND_SOC_MSM8660_LPAIF
+ tristate
+
+config SND_VOIP_PCM
+ tristate
+
+config SND_SOC_MSM_QDSP6_HDMI_AUDIO
+ tristate "Soc QDSP6 HDMI Audio DAI driver"
+ depends on FB_MSM_HDMI_MSM_PANEL
+ default n
+ help
+ To support HDMI Audio on MSM8960 over QDSP6.
+
+config MSM_8x60_VOIP
+ tristate "SoC Machine driver for voip"
+ depends on SND_SOC_MSM8X60
+ select SND_MSM_MVS_DAI_SOC
+ select SND_VOIP_PCM
+ default n
+ help
+ To support ALSA VOIP driver for MSM8x60 target.
+ This driver communicates with QDSP6, for getting
+ uplink and downlink voice packets.
+
+config SND_SOC_MSM_QDSP6_INTF
+ bool "SoC Q6 audio driver for MSM8960"
+ depends on MSM_QDSP6_APR
+ default n
+ help
+ To add support for SoC audio on MSM8960.
+
+config SND_SOC_VOICE
+ bool "SoC Q6 voice driver for MSM8960"
+ depends on SND_SOC_MSM_QDSP6_INTF
+ default n
+ help
+ To add support for SoC voice on MSM8960.
+
+config SND_SOC_QDSP6
+ tristate "SoC ALSA audio driver for QDSP6"
+ select SND_SOC_MSM_QDSP6_INTF
+ default n
+ help
+ To add support for MSM QDSP6 Soc Audio.
+
+config SND_SOC_MSM8960
+ tristate "SoC Machine driver for MSM8960 and APQ8064 boards"
+ depends on ARCH_MSM8960 || ARCH_APQ8064
+ select SND_SOC_VOICE
+ select SND_SOC_QDSP6
+ select SND_SOC_MSM_STUB
+ select SND_SOC_WCD9310
+ select SND_SOC_WCD9304
+ select SND_SOC_MSM_HOSTLESS_PCM
+ select SND_SOC_MSM_QDSP6_HDMI_AUDIO
+ select SND_SOC_CS8427
+ default n
+ help
+ To add support for SoC audio on MSM8960 and APQ8064 boards
+
+config SND_SOC_MDM9615
+ tristate "SoC Machine driver for MDM9615 boards"
+ depends on ARCH_MSM9615
+ select SND_SOC_VOICE
+ select SND_SOC_QDSP6
+ select SND_SOC_MSM_STUB
+ select SND_SOC_WCD9310
+ select SND_SOC_MSM_HOSTLESS_PCM
+ select SND_DYNAMIC_MINORS
+ help
+ To add support for SoC audio on MDM9615 boards
+
+config SND_SOC_MSM8660_APQ
+ tristate "Soc Machine driver for APQ8060 WM8903 codec"
+ depends on ARCH_MSM8X60
+ select SND_SOC_QDSP6
+ select SND_SOC_WM8903
+ select SND_SOC_MSM_STUB
+ default n
+ help
+ To add support for SoC audio on APQ8060 board
+endmenu
diff --git a/sound/soc/msm/Makefile b/sound/soc/msm/Makefile
new file mode 100644
index 0000000..0df028c
--- /dev/null
+++ b/sound/soc/msm/Makefile
@@ -0,0 +1,86 @@
+# MSM CPU/CODEC DAI Support
+snd-soc-msm-dai-objs := msm-dai.o
+obj-$(CONFIG_SND_MSM_DAI_SOC) += snd-soc-msm-dai.o
+
+snd-soc-msm7kv2-dai-objs := msm7kv2-dai.o
+obj-$(CONFIG_SND_MSM7KV2_DAI_SOC) += snd-soc-msm7kv2-dai.o
+
+# MSM Platform Support
+snd-soc-msm-objs := msm-pcm.o msm7k-pcm.o
+obj-$(CONFIG_SND_MSM_SOC) += snd-soc-msm.o
+
+snd-soc-msmv2-objs := msm7kv2-dsp.o msm7kv2-pcm.o
+obj-$(CONFIG_SND_MSM7KV2_SOC) += snd-soc-msmv2.o
+
+# MSM Machine Support
+snd-soc-msm7k-objs := msm7201.o
+obj-$(CONFIG_SND_MSM_SOC_MSM7K) += snd-soc-msm7k.o
+
+snd-soc-msm7kv2-objs := msm7x30.o
+obj-$(CONFIG_SND_MSM_SOC_MSM7KV2) += snd-soc-msm7kv2.o
+
+# 8660 ALSA Support
+snd-soc-msm8x60-dai-objs := msm8x60-dai.o
+obj-$(CONFIG_SND_SOC_MSM8X60_DAI) += snd-soc-msm8x60-dai.o
+
+snd-soc-msm8x60-pcm-objs := msm8x60-pcm.o
+obj-$(CONFIG_SND_SOC_MSM8X60_PCM) += snd-soc-msm8x60-pcm.o
+
+snd-soc-msm8x60-objs := msm8x60.o
+obj-$(CONFIG_SND_SOC_MSM8X60) += snd-soc-msm8x60.o
+
+
+#MVS Support
+snd-soc-msm-mvs-dai-objs := mvs-dai.o
+obj-$(CONFIG_SND_MSM_MVS_DAI_SOC) += snd-soc-msm-mvs-dai.o
+
+snd-soc-msm-mvs-objs := msm-mvs.o
+obj-$(CONFIG_SND_MVS_SOC) += snd-soc-msm-mvs.o
+
+# 8660 ALSA Support
+snd-soc-lpass-objs := lpass-i2s.o lpass-dma.o
+obj-$(CONFIG_SND_SOC_MSM8660_LPAIF) += snd-soc-lpass.o
+
+snd-soc-lpass-pcm-objs := lpass-pcm.o
+obj-$(CONFIG_SND_SOC_LPASS_PCM) += snd-soc-lpass-pcm.o
+
+#8660 VOIP Driver Support
+
+snd-soc-msm-voip-objs := msm-voip.o
+obj-$(CONFIG_SND_VOIP_PCM) += snd-soc-msm-voip.o
+
+snd-soc-lpass-dma-objs := lpass-dma.o
+obj-$(CONFIG_SND_SOC_MSM8X60) += snd-soc-lpass-dma.o
+
+# for MSM 8960 sound card driver
+
+obj-$(CONFIG_SND_SOC_MSM_QDSP6_INTF) += qdsp6/
+
+snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-multi-ch-pcm-q6.o msm-pcm-routing.o msm-dai-fe.o msm-compr-q6.o msm-dai-stub.o
+obj-$(CONFIG_SND_SOC_MSM_QDSP6_HDMI_AUDIO) += msm-dai-q6-hdmi.o
+obj-$(CONFIG_SND_SOC_VOICE) += msm-pcm-voice.o msm-pcm-voip.o
+snd-soc-qdsp6-objs += msm-pcm-lpa.o msm-pcm-afe.o
+obj-$(CONFIG_SND_SOC_QDSP6) += snd-soc-qdsp6.o
+
+snd-soc-msm8960-objs := msm8960.o apq8064.o msm8930.o
+obj-$(CONFIG_SND_SOC_MSM8960) += snd-soc-msm8960.o
+
+# Generic MSM drivers
+snd-soc-hostless-pcm-objs := msm-pcm-hostless.o
+obj-$(CONFIG_SND_SOC_MSM_HOSTLESS_PCM) += snd-soc-hostless-pcm.o
+
+snd-soc-msm8660-apq-objs := msm8660-apq-wm8903.o
+obj-$(CONFIG_SND_SOC_MSM8660_APQ) += snd-soc-msm8660-apq.o
+
+# for MDM 9615 sound card driver
+snd-soc-mdm9615-objs := mdm9615.o
+obj-$(CONFIG_SND_SOC_MDM9615) += snd-soc-mdm9615.o
+
+# for MSM 8974 sound card driver
+obj-$(CONFIG_SND_SOC_MSM_QDSP6V2_INTF) += qdsp6v2/
+snd-soc-msm8974-objs := msm8974.o
+obj-$(CONFIG_SND_SOC_MSM8974) += snd-soc-msm8974.o
+
+snd-soc-qdsp6v2-objs := msm-dai-fe.o msm-dai-stub.o
+obj-$(CONFIG_SND_SOC_QDSP6V2) += snd-soc-qdsp6v2.o
+
diff --git a/sound/soc/msm/apq8064.c b/sound/soc/msm/apq8064.c
new file mode 100644
index 0000000..353ffd6
--- /dev/null
+++ b/sound/soc/msm/apq8064.c
@@ -0,0 +1,1870 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <mach/socinfo.h>
+#include "msm-pcm-routing.h"
+#include "../codecs/wcd9310.h"
+
+/* 8064 machine driver */
+
+#define PM8921_GPIO_BASE NR_GPIO_IRQS
+#define PM8921_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio - 1 + PM8921_GPIO_BASE)
+
+#define MSM8064_SPK_ON 1
+#define MSM8064_SPK_OFF 0
+
+#define MSM_SLIM_0_RX_MAX_CHANNELS 2
+#define MSM_SLIM_0_TX_MAX_CHANNELS 4
+
+#define BTSCO_RATE_8KHZ 8000
+#define BTSCO_RATE_16KHZ 16000
+
+#define BOTTOM_SPK_AMP_POS 0x1
+#define BOTTOM_SPK_AMP_NEG 0x2
+#define TOP_SPK_AMP_POS 0x4
+#define TOP_SPK_AMP_NEG 0x8
+
+#define GPIO_AUX_PCM_DOUT 43
+#define GPIO_AUX_PCM_DIN 44
+#define GPIO_AUX_PCM_SYNC 45
+#define GPIO_AUX_PCM_CLK 46
+
+#define TABLA_EXT_CLK_RATE 12288000
+
+#define TABLA_MBHC_DEF_BUTTONS 8
+#define TABLA_MBHC_DEF_RLOADS 5
+
+#define JACK_DETECT_GPIO 38
+
+/* Shared channel numbers for Slimbus ports that connect APQ to MDM. */
+enum {
+ SLIM_1_RX_1 = 145, /* BT-SCO and USB TX */
+ SLIM_1_TX_1 = 146, /* BT-SCO and USB RX */
+ SLIM_3_RX_1 = 151, /* External echo-cancellation ref */
+ SLIM_3_RX_2 = 152, /* External echo-cancellation ref */
+ SLIM_3_TX_1 = 147, /* HDMI RX */
+ SLIM_4_TX_1 = 148, /* In-call recording RX */
+ SLIM_4_TX_2 = 149, /* In-call recording RX */
+ SLIM_4_RX_1 = 150, /* In-call music delivery TX */
+};
+
+enum {
+ INCALL_REC_MONO,
+ INCALL_REC_STEREO,
+};
+
+static u32 top_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(18);
+static u32 bottom_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(19);
+static int msm_spk_control;
+static int msm_ext_bottom_spk_pamp;
+static int msm_ext_top_spk_pamp;
+static int msm_slim_0_rx_ch = 1;
+static int msm_slim_0_tx_ch = 1;
+static int msm_slim_3_rx_ch = 1;
+
+static int msm_btsco_rate = BTSCO_RATE_8KHZ;
+static int msm_btsco_ch = 1;
+
+static int rec_mode = INCALL_REC_MONO;
+
+static struct clk *codec_clk;
+static int clk_users;
+
+static int msm_headset_gpios_configured;
+
+static struct snd_soc_jack hs_jack;
+static struct snd_soc_jack button_jack;
+
+static int apq8064_hs_detect_use_gpio = -1;
+module_param(apq8064_hs_detect_use_gpio, int, 0444);
+MODULE_PARM_DESC(apq8064_hs_detect_use_gpio, "Use GPIO for headset detection");
+
+static bool apq8064_hs_detect_use_firmware;
+module_param(apq8064_hs_detect_use_firmware, bool, 0444);
+MODULE_PARM_DESC(apq8064_hs_detect_use_firmware, "Use firmware for headset "
+ "detection");
+
+static int msm_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm);
+
+static struct tabla_mbhc_config mbhc_cfg = {
+ .headset_jack = &hs_jack,
+ .button_jack = &button_jack,
+ .read_fw_bin = false,
+ .calibration = NULL,
+ .micbias = TABLA_MICBIAS2,
+ .mclk_cb_fn = msm_enable_codec_ext_clk,
+ .mclk_rate = TABLA_EXT_CLK_RATE,
+ .gpio = 0,
+ .gpio_irq = 0,
+ .gpio_level_insert = 1,
+};
+
+static void msm_enable_ext_spk_amp_gpio(u32 spk_amp_gpio)
+{
+ int ret = 0;
+
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .
+ function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ if (spk_amp_gpio == bottom_spk_pamp_gpio) {
+
+ ret = gpio_request(bottom_spk_pamp_gpio, "BOTTOM_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting BOTTOM SPK AMP GPIO %u\n",
+ __func__, bottom_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(bottom_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Bottom Spk Ampl"
+ " gpio %u\n", __func__, bottom_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Bottom spkr amp gpio\n", __func__);
+ gpio_direction_output(bottom_spk_pamp_gpio, 1);
+ }
+
+ } else if (spk_amp_gpio == top_spk_pamp_gpio) {
+
+ ret = gpio_request(top_spk_pamp_gpio, "TOP_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting GPIO %d\n", __func__,
+ top_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(top_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Top Spk Ampl"
+ " gpio %u\n", __func__, top_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Top spkr amp gpio\n", __func__);
+ gpio_direction_output(top_spk_pamp_gpio, 1);
+ }
+ } else {
+ pr_err("%s: ERROR : Invalid External Speaker Ampl GPIO."
+ " gpio = %u\n", __func__, spk_amp_gpio);
+ return;
+ }
+}
+
+static void msm_ext_spk_power_amp_on(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Bottom Speaker Ampl already "
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm_ext_bottom_spk_pamp |= spk;
+
+ if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ msm_enable_ext_spk_amp_gpio(bottom_spk_pamp_gpio);
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Bottom Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Top Speaker Ampl already"
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm_ext_top_spk_pamp |= spk;
+
+ if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ msm_enable_ext_spk_amp_gpio(top_spk_pamp_gpio);
+ pr_debug("%s: sleeping 4 ms after turning on "
+ " external Top Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm_ext_spk_power_amp_off(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if (!msm_ext_bottom_spk_pamp)
+ return;
+
+ gpio_direction_output(bottom_spk_pamp_gpio, 0);
+ gpio_free(bottom_spk_pamp_gpio);
+ msm_ext_bottom_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Bottom"
+ " Speaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if (!msm_ext_top_spk_pamp)
+ return;
+
+ gpio_direction_output(top_spk_pamp_gpio, 0);
+ gpio_free(top_spk_pamp_gpio);
+ msm_ext_top_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Top"
+ " Spkaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+ } else {
+
+ pr_err("%s: ERROR : Invalid Ext Spk Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm_ext_control(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
+ if (msm_spk_control == MSM8064_SPK_ON) {
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+ } else {
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Neg");
+ }
+
+ snd_soc_dapm_sync(dapm);
+}
+
+static int msm_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
+ ucontrol->value.integer.value[0] = msm_spk_control;
+ return 0;
+}
+static int msm_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm_spk_control = ucontrol->value.integer.value[0];
+ msm_ext_control(codec);
+ return 1;
+}
+static int msm_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm_ext_spk_power_amp_on(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm_ext_spk_power_amp_on(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+
+ } else {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm_ext_spk_power_amp_off(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm_ext_spk_power_amp_off(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ }
+ return 0;
+}
+
+static int msm_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm)
+{
+ pr_debug("%s: enable = %d\n", __func__, enable);
+ if (enable) {
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users != 1)
+ return 0;
+
+ if (codec_clk) {
+ clk_set_rate(codec_clk, TABLA_EXT_CLK_RATE);
+ clk_prepare_enable(codec_clk);
+ tabla_mclk_enable(codec, 1, dapm);
+ } else {
+ pr_err("%s: Error setting Tabla MCLK\n", __func__);
+ clk_users--;
+ return -EINVAL;
+ }
+ } else {
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users == 0)
+ return 0;
+ clk_users--;
+ if (!clk_users) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+ tabla_mclk_enable(codec, 0, dapm);
+ clk_disable_unprepare(codec_clk);
+ }
+ }
+ return 0;
+}
+
+static int msm_mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+
+ if (clk_users != 1)
+ return 0;
+
+ if (codec_clk) {
+ clk_set_rate(codec_clk, 12288000);
+ clk_prepare_enable(codec_clk);
+ tabla_mclk_enable(w->codec, 1, true);
+
+ } else {
+ pr_err("%s: Error setting Tabla MCLK\n", __func__);
+ clk_users--;
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+
+ if (clk_users == 0)
+ return 0;
+
+ clk_users--;
+
+ if (!clk_users) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+
+ tabla_mclk_enable(w->codec, 0, true);
+ clk_disable_unprepare(codec_clk);
+ }
+ break;
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget apq8064_dapm_widgets[] = {
+
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ msm_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Pos", msm_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Neg", msm_spkramp_event),
+
+ SND_SOC_DAPM_SPK("Ext Spk Top Pos", msm_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Top Neg", msm_spkramp_event),
+
+ /************ Analog MICs ************/
+ /**
+ * Analog mic7 (Front Top) on Liquid.
+ * Used as Handset mic on CDP.
+ */
+ SND_SOC_DAPM_MIC("Analog mic7", NULL),
+
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
+
+ /*********** Digital Mics ***************/
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic2", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic3", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic4", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic5", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic6", NULL),
+};
+
+static const struct snd_soc_dapm_route apq8064_common_audio_map[] = {
+
+ {"RX_BIAS", NULL, "MCLK"},
+ {"LDO_H", NULL, "MCLK"},
+
+ {"HEADPHONE", NULL, "LDO_H"},
+
+ /* Speaker path */
+ {"Ext Spk Bottom Pos", NULL, "LINEOUT1"},
+ {"Ext Spk Bottom Neg", NULL, "LINEOUT3"},
+
+ {"Ext Spk Top Pos", NULL, "LINEOUT2"},
+ {"Ext Spk Top Neg", NULL, "LINEOUT4"},
+
+ /************ Analog MIC Paths ************/
+
+ /* Headset Mic */
+ {"AMIC2", NULL, "MIC BIAS2 External"},
+ {"MIC BIAS2 External", NULL, "Headset Mic"},
+
+ /* Headset ANC microphones */
+ {"AMIC3", NULL, "MIC BIAS3 Internal1"},
+ {"MIC BIAS3 Internal1", NULL, "ANCRight Headset Mic"},
+
+ {"AMIC4", NULL, "MIC BIAS1 Internal2"},
+ {"MIC BIAS1 Internal2", NULL, "ANCLeft Headset Mic"},
+};
+
+static const struct snd_soc_dapm_route apq8064_mtp_audio_map[] = {
+
+ /************ Digital MIC Paths ************/
+
+ /*
+ * Digital Mic1 (Front bottom Left) on MTP.
+ * Conncted to DMIC1 Input on Tabla codec.
+ */
+ {"DMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic1"},
+
+ /**
+ * Digital Mic2 (Front bottom right) on MTP.
+ * Conncted to DMIC2 Input on Tabla codec.
+ */
+ {"DMIC2", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic2"},
+
+ /**
+ * Digital Mic3 (Back bottom) on MTP.
+ * Conncted to DMIC3 Input on Tabla codec.
+ */
+ {"DMIC3", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic3"},
+
+ /**
+ * Digital Mic4 (Back top) on MTP.
+ * Conncted to DMIC4 Input on Tabla codec.
+ */
+ {"DMIC4", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic4"},
+
+ /**
+ * Digital Mic5 (Top front Mic) on MTP.
+ * Conncted to DMIC6 Input on Tabla codec.
+ */
+ {"DMIC6", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic5"},
+
+};
+
+static const struct snd_soc_dapm_route apq8064_liquid_cdp_audio_map[] = {
+
+ /************ Analog MIC Paths ************/
+ /**
+ * Analog mic7 (Front Top Mic) on Liquid.
+ * Used as Handset mic on CDP.
+ * Not there on MTP.
+ */
+ {"AMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Analog mic7"},
+
+
+ /************ Digital MIC Paths ************/
+ /**
+ * The digital Mic routes are setup considering
+ * Liquid as default device.
+ */
+
+ /**
+ * Digital Mic1 (Front bottom left corner) on Liquid.
+ * Digital Mic2 (Front bottom right) on MTP.
+ * Digital Mic GM1 on CDP mainboard.
+ * Conncted to DMIC2 Input on Tabla codec.
+ */
+ {"DMIC2", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic1"},
+
+ /**
+ * Digital Mic2 (Front left side) on Liquid.
+ * Digital Mic GM2 on CDP mainboard.
+ * Not there on MTP.
+ * Conncted to DMIC3 Input on Tabla codec.
+ */
+ {"DMIC3", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic2"},
+
+ /**
+ * Digital Mic3. Front bottom left of middle on Liquid.
+ * Digital Mic5 (Top front Mic) on MTP.
+ * Digital Mic GM5 on CDP mainboard.
+ * Conncted to DMIC6 Input on Tabla codec.
+ */
+ {"DMIC6", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic3"},
+
+ /**
+ * Digital Mic4. Back bottom on Liquid.
+ * Digital Mic GM3 on CDP mainboard.
+ * Top Front Mic on MTP.
+ * Conncted to DMIC5 Input on Tabla codec.
+ */
+ {"DMIC5", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic4"},
+
+ /**
+ * Digital Mic5. Front bottom right of middle on Liquid.
+ * Digital Mic GM6 on CDP mainboard.
+ * Not there on MTP.
+ * Conncted to DMIC4 Input on Tabla codec.
+ */
+ {"DMIC4", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic5"},
+
+ /* Digital Mic6 (Front bottom right corner) on Liquid.
+ * Digital Mic1 (Front bottom Left) on MTP.
+ * Digital Mic GM4 on CDP.
+ * Conncted to DMIC1 Input on Tabla codec.
+ */
+ {"DMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic6"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *slim0_rx_ch_text[] = {"One", "Two"};
+static const char *slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
+
+static const struct soc_enum msm_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
+ SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
+};
+
+static const char *btsco_rate_text[] = {"8000", "16000"};
+static const struct soc_enum msm_btsco_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, btsco_rate_text),
+};
+
+static int msm_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
+ msm_slim_0_rx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_0_rx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
+ msm_slim_0_rx_ch);
+ return 1;
+}
+
+static int msm_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
+ msm_slim_0_tx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_0_tx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
+ msm_slim_0_tx_ch);
+ return 1;
+}
+
+static int msm_slim_3_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_3_rx_ch = %d\n", __func__,
+ msm_slim_3_rx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_3_rx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_3_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_3_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_3_rx_ch = %d\n", __func__,
+ msm_slim_3_rx_ch);
+ return 1;
+}
+
+static int msm_btsco_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_btsco_rate = %d", __func__,
+ msm_btsco_rate);
+ ucontrol->value.integer.value[0] = msm_btsco_rate;
+ return 0;
+}
+
+static int msm_btsco_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ msm_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ case 1:
+ msm_btsco_rate = BTSCO_RATE_16KHZ;
+ break;
+ default:
+ msm_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ }
+ pr_debug("%s: msm_btsco_rate = %d\n", __func__,
+ msm_btsco_rate);
+ return 0;
+}
+
+static int msm_incall_rec_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = rec_mode;
+ return 0;
+}
+
+static int msm_incall_rec_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ rec_mode = ucontrol->value.integer.value[0];
+ pr_debug("%s: rec_mode:%d\n", __func__, rec_mode);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new tabla_msm_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", msm_enum[0], msm_get_spk,
+ msm_set_spk),
+ SOC_ENUM_EXT("SLIM_0_RX Channels", msm_enum[1],
+ msm_slim_0_rx_ch_get, msm_slim_0_rx_ch_put),
+ SOC_ENUM_EXT("SLIM_0_TX Channels", msm_enum[2],
+ msm_slim_0_tx_ch_get, msm_slim_0_tx_ch_put),
+ SOC_ENUM_EXT("Internal BTSCO SampleRate", msm_btsco_enum[0],
+ msm_btsco_rate_get, msm_btsco_rate_put),
+ SOC_SINGLE_EXT("Incall Rec Mode", SND_SOC_NOPM, 0, 1, 0,
+ msm_incall_rec_mode_get, msm_incall_rec_mode_put),
+ SOC_ENUM_EXT("SLIM_3_RX Channels", msm_enum[1],
+ msm_slim_3_rx_ch_get, msm_slim_3_rx_ch_put),
+};
+
+static void *def_tabla_mbhc_cal(void)
+{
+ void *tabla_cal;
+ struct tabla_mbhc_btn_detect_cfg *btn_cfg;
+ u16 *btn_low, *btn_high;
+ u8 *n_ready, *n_cic, *gain;
+
+ tabla_cal = kzalloc(TABLA_MBHC_CAL_SIZE(TABLA_MBHC_DEF_BUTTONS,
+ TABLA_MBHC_DEF_RLOADS),
+ GFP_KERNEL);
+ if (!tabla_cal) {
+ pr_err("%s: out of memory\n", __func__);
+ return NULL;
+ }
+
+#define S(X, Y) ((TABLA_MBHC_CAL_GENERAL_PTR(tabla_cal)->X) = (Y))
+ S(t_ldoh, 100);
+ S(t_bg_fast_settle, 100);
+ S(t_shutdown_plug_rem, 255);
+ S(mbhc_nsa, 4);
+ S(mbhc_navg, 4);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_DET_PTR(tabla_cal)->X) = (Y))
+ S(mic_current, TABLA_PID_MIC_5_UA);
+ S(hph_current, TABLA_PID_MIC_5_UA);
+ S(t_mic_pid, 100);
+ S(t_ins_complete, 250);
+ S(t_ins_retry, 200);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla_cal)->X) = (Y))
+ S(v_no_mic, 30);
+ S(v_hs_max, 1550);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal)->X) = (Y))
+ S(c[0], 62);
+ S(c[1], 124);
+ S(nc, 1);
+ S(n_meas, 3);
+ S(mbhc_nsc, 11);
+ S(n_btn_meas, 1);
+ S(n_btn_con, 2);
+ S(num_btn, TABLA_MBHC_DEF_BUTTONS);
+ S(v_btn_press_delta_sta, 100);
+ S(v_btn_press_delta_cic, 50);
+#undef S
+ btn_cfg = TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal);
+ btn_low = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_LOW);
+ btn_high = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_HIGH);
+ btn_low[0] = -50;
+ btn_high[0] = 10;
+ btn_low[1] = 11;
+ btn_high[1] = 38;
+ btn_low[2] = 39;
+ btn_high[2] = 64;
+ btn_low[3] = 65;
+ btn_high[3] = 91;
+ btn_low[4] = 92;
+ btn_high[4] = 115;
+ btn_low[5] = 116;
+ btn_high[5] = 141;
+ btn_low[6] = 142;
+ btn_high[6] = 163;
+ btn_low[7] = 164;
+ btn_high[7] = 250;
+ n_ready = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_READY);
+ n_ready[0] = 48;
+ n_ready[1] = 38;
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_CIC);
+ n_cic[0] = 60;
+ n_cic[1] = 47;
+ gain = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_GAIN);
+ gain[0] = 11;
+ gain[1] = 9;
+
+ return tabla_cal;
+}
+
+static int msm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ unsigned int rx_ch_cnt = 0, tx_ch_cnt = 0;
+ unsigned int user_set_tx_ch = 0;
+
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ pr_debug("%s: rx_0_ch=%d\n", __func__, msm_slim_0_rx_ch);
+
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
+ msm_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai, 0, 0,
+ msm_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+ } else {
+
+ if (codec_dai->id == 2)
+ user_set_tx_ch = msm_slim_0_tx_ch;
+ else if (codec_dai->id == 4)
+ user_set_tx_ch = params_channels(params);
+ else if (codec_dai->id == 5) {
+ /* DAI 5 is used for external EC reference from codec.
+ * Since Rx is fed as reference for EC, the config of
+ * this DAI is based on that of the Rx path.
+ */
+ user_set_tx_ch = msm_slim_0_rx_ch;
+ }
+
+ pr_debug("%s: %s_tx_dai_id_%d_ch=%d\n", __func__,
+ codec_dai->name, codec_dai->id, user_set_tx_ch);
+
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai,
+ user_set_tx_ch, tx_ch, 0 , 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai,
+ user_set_tx_ch, tx_ch, 0, 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+
+
+ }
+end:
+ return ret;
+}
+
+static int msm_stubrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ rtd->pmdown_time = 0;
+
+ return 0;
+}
+
+static int msm_slimbus_1_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch = SLIM_1_RX_1, tx_ch = SLIM_1_TX_1;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_debug("%s: APQ BT/USB TX -> SLIMBUS_1_RX -> MDM TX shared ch %d\n",
+ __func__, rx_ch);
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0, 1, &rx_ch);
+ if (ret < 0) {
+ pr_err("%s: Erorr %d setting SLIM_1 RX channel map\n",
+ __func__, ret);
+
+ goto end;
+ }
+ } else {
+ pr_debug("%s: MDM RX -> SLIMBUS_1_TX -> APQ BT/USB Rx shared ch %d\n",
+ __func__, tx_ch);
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 1, &tx_ch, 0, 0);
+ if (ret < 0) {
+ pr_err("%s: Erorr %d setting SLIM_1 TX channel map\n",
+ __func__, ret);
+
+ goto end;
+ }
+ }
+
+end:
+ return ret;
+}
+
+static int msm_slimbus_3_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch[2] = {SLIM_3_RX_1, SLIM_3_RX_2};
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_debug("%s: slim_3_rx_ch %d, sch %d %d\n",
+ __func__, msm_slim_3_rx_ch,
+ rx_ch[0], rx_ch[1]);
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
+ msm_slim_3_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: Erorr %d setting SLIM_3 RX channel map\n",
+ __func__, ret);
+
+ goto end;
+ }
+ } else {
+ pr_err("%s: SLIMBUS_3_TX not defined for this DAI\n", __func__);
+ }
+
+end:
+ return ret;
+}
+
+static int msm_slimbus_4_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch = SLIM_4_RX_1, tx_ch[2];
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_debug("%s: APQ Incall Playback SLIMBUS_4_RX -> MDM TX shared ch %d\n",
+ __func__, rx_ch);
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0, 1, &rx_ch);
+ if (ret < 0) {
+ pr_err("%s: Erorr %d setting SLIM_4 RX channel map\n",
+ __func__, ret);
+
+ }
+ } else {
+ if (rec_mode == INCALL_REC_STEREO) {
+ tx_ch[0] = SLIM_4_TX_1;
+ tx_ch[1] = SLIM_4_TX_2;
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 2,
+ tx_ch, 0, 0);
+ } else {
+ tx_ch[0] = SLIM_4_TX_1;
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 1,
+ tx_ch, 0, 0);
+ }
+ pr_debug("%s: Incall Record shared tx_ch[0]:%d, tx_ch[1]:%d\n",
+ __func__, tx_ch[0], tx_ch[1]);
+
+ if (ret < 0) {
+ pr_err("%s: Erorr %d setting SLIM_4 TX channel map\n",
+ __func__, ret);
+
+ }
+ }
+
+ return ret;
+}
+
+static int msm_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+ uint32_t revision;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("%s(), dev_name%s\n", __func__, dev_name(cpu_dai->dev));
+
+ /*if (machine_is_msm_liquid()) {
+ top_spk_pamp_gpio = (PM8921_GPIO_PM_TO_SYS(19));
+ bottom_spk_pamp_gpio = (PM8921_GPIO_PM_TO_SYS(18));
+ }*/
+
+ snd_soc_dapm_new_controls(dapm, apq8064_dapm_widgets,
+ ARRAY_SIZE(apq8064_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, apq8064_common_audio_map,
+ ARRAY_SIZE(apq8064_common_audio_map));
+
+ if (machine_is_apq8064_mtp()) {
+ snd_soc_dapm_add_routes(dapm, apq8064_mtp_audio_map,
+ ARRAY_SIZE(apq8064_mtp_audio_map));
+ } else {
+ snd_soc_dapm_add_routes(dapm, apq8064_liquid_cdp_audio_map,
+ ARRAY_SIZE(apq8064_liquid_cdp_audio_map));
+ }
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+
+ snd_soc_dapm_sync(dapm);
+
+ err = snd_soc_jack_new(codec, "Headset Jack",
+ (SND_JACK_HEADSET | SND_JACK_OC_HPHL | SND_JACK_OC_HPHR),
+ &hs_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ err = snd_soc_jack_new(codec, "Button Jack",
+ TABLA_JACK_BUTTON_MASK, &button_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ codec_clk = clk_get(cpu_dai->dev, "osr_clk");
+
+ /* APQ8064 Rev 1.1 CDP and Liquid have mechanical switch */
+ revision = socinfo_get_version();
+ if (apq8064_hs_detect_use_gpio != -1) {
+ if (apq8064_hs_detect_use_gpio == 1)
+ pr_debug("%s: MBHC mechanical is enabled by request\n",
+ __func__);
+ else if (apq8064_hs_detect_use_gpio == 0)
+ pr_debug("%s: MBHC mechanical is disabled by request\n",
+ __func__);
+ else
+ pr_warn("%s: Invalid hs_detect_use_gpio %d\n", __func__,
+ apq8064_hs_detect_use_gpio);
+ } else if (SOCINFO_VERSION_MAJOR(revision) == 0) {
+ pr_warn("%s: Unknown HW revision detected %d.%d\n", __func__,
+ SOCINFO_VERSION_MAJOR(revision),
+ SOCINFO_VERSION_MINOR(revision));
+ } else if ((SOCINFO_VERSION_MAJOR(revision) == 1 &&
+ SOCINFO_VERSION_MINOR(revision) >= 1 &&
+ (machine_is_apq8064_cdp() ||
+ machine_is_apq8064_liquid())) ||
+ SOCINFO_VERSION_MAJOR(revision) > 1) {
+ pr_debug("%s: MBHC mechanical switch available APQ8064 "
+ "detected\n", __func__);
+ apq8064_hs_detect_use_gpio = 1;
+ }
+
+ if (apq8064_hs_detect_use_gpio == 1) {
+ pr_debug("%s: Using MBHC mechanical switch\n", __func__);
+ mbhc_cfg.gpio = JACK_DETECT_GPIO;
+ mbhc_cfg.gpio_irq = gpio_to_irq(JACK_DETECT_GPIO);
+ err = gpio_request(mbhc_cfg.gpio, "MBHC_HS_DETECT");
+ if (err < 0) {
+ pr_err("%s: gpio_request %d failed %d\n", __func__,
+ mbhc_cfg.gpio, err);
+ return err;
+ }
+ gpio_direction_input(JACK_DETECT_GPIO);
+ } else
+ pr_debug("%s: Not using MBHC mechanical switch\n", __func__);
+
+ mbhc_cfg.read_fw_bin = apq8064_hs_detect_use_firmware;
+
+ err = tabla_hs_detect(codec, &mbhc_cfg);
+
+ return err;
+}
+
+static int msm_slim_0_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm_slim_0_rx_ch;
+
+ return 0;
+}
+
+static int msm_slim_0_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm_slim_0_tx_ch;
+
+ return 0;
+}
+
+static int msm_slim_3_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm_slim_3_rx_ch;
+
+ return 0;
+}
+
+static int msm_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+
+ return 0;
+}
+
+static int msm_hdmi_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s channels->min %u channels->max %u ()\n", __func__,
+ channels->min, channels->max);
+
+ rate->min = rate->max = 48000;
+
+ return 0;
+}
+
+static int msm_btsco_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = msm_btsco_rate;
+ channels->min = channels->max = msm_btsco_ch;
+
+ return 0;
+}
+static int msm_auxpcm_be_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* PCM only supports mono output with 8khz sample rate */
+ rate->min = rate->max = 8000;
+ channels->min = channels->max = 1;
+
+ return 0;
+}
+
+static int msm_aux_pcm_get_gpios(void)
+{
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ ret = gpio_request(GPIO_AUX_PCM_DOUT, "AUX PCM DOUT");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DOUT",
+ __func__, GPIO_AUX_PCM_DOUT);
+ goto fail_dout;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_DIN, "AUX PCM DIN");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DIN",
+ __func__, GPIO_AUX_PCM_DIN);
+ goto fail_din;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_SYNC, "AUX PCM SYNC");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM SYNC",
+ __func__, GPIO_AUX_PCM_SYNC);
+ goto fail_sync;
+ }
+ ret = gpio_request(GPIO_AUX_PCM_CLK, "AUX PCM CLK");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM CLK",
+ __func__, GPIO_AUX_PCM_CLK);
+ goto fail_clk;
+ }
+
+ return 0;
+
+fail_clk:
+ gpio_free(GPIO_AUX_PCM_SYNC);
+fail_sync:
+ gpio_free(GPIO_AUX_PCM_DIN);
+fail_din:
+ gpio_free(GPIO_AUX_PCM_DOUT);
+fail_dout:
+
+ return ret;
+}
+
+static int msm_aux_pcm_free_gpios(void)
+{
+ gpio_free(GPIO_AUX_PCM_DIN);
+ gpio_free(GPIO_AUX_PCM_DOUT);
+ gpio_free(GPIO_AUX_PCM_SYNC);
+ gpio_free(GPIO_AUX_PCM_CLK);
+
+ return 0;
+}
+static int msm_startup(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+ return 0;
+}
+
+static int msm_auxpcm_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ ret = msm_aux_pcm_get_gpios();
+ if (ret < 0) {
+ pr_err("%s: Aux PCM GPIO request failed\n", __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static void msm_auxpcm_shutdown(struct snd_pcm_substream *substream)
+{
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ msm_aux_pcm_free_gpios();
+}
+
+static void msm_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+}
+
+static struct snd_soc_ops msm_be_ops = {
+ .startup = msm_startup,
+ .hw_params = msm_hw_params,
+ .shutdown = msm_shutdown,
+};
+
+static struct snd_soc_ops msm_auxpcm_be_ops = {
+ .startup = msm_auxpcm_startup,
+ .shutdown = msm_auxpcm_shutdown,
+};
+
+static struct snd_soc_ops msm_slimbus_1_be_ops = {
+ .startup = msm_startup,
+ .hw_params = msm_slimbus_1_hw_params,
+ .shutdown = msm_shutdown,
+};
+
+static struct snd_soc_ops msm_slimbus_3_be_ops = {
+ .startup = msm_startup,
+ .hw_params = msm_slimbus_3_hw_params,
+ .shutdown = msm_shutdown,
+};
+
+static struct snd_soc_ops msm_slimbus_4_be_ops = {
+ .startup = msm_startup,
+ .hw_params = msm_slimbus_4_hw_params,
+ .shutdown = msm_shutdown,
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm_dai[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MSM8960 Media1",
+ .stream_name = "MultiMedia1",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MSM8960 Media2",
+ .stream_name = "MultiMedia2",
+ .cpu_dai_name = "MultiMedia2",
+ .platform_name = "msm-multi-ch-pcm-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+ {
+ .name = "Circuit-Switch Voice",
+ .stream_name = "CS-Voice",
+ .cpu_dai_name = "CS-VOICE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_CS_VOICE,
+ },
+ {
+ .name = "MSM VoIP",
+ .stream_name = "VoIP",
+ .cpu_dai_name = "VoIP",
+ .platform_name = "msm-voip-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_VOIP,
+ },
+ {
+ .name = "MSM8960 LPA",
+ .stream_name = "LPA",
+ .cpu_dai_name = "MultiMedia3",
+ .platform_name = "msm-pcm-lpa",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
+ },
+ /* Hostless PMC purpose */
+ {
+ .name = "SLIMBUS_0 Hostless",
+ .stream_name = "SLIMBUS_0 Hostless",
+ .cpu_dai_name = "SLIMBUS0_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ /* .be_id = do not care */
+ },
+ {
+ .name = "INT_FM Hostless",
+ .stream_name = "INT_FM Hostless",
+ .cpu_dai_name = "INT_FM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ /* .be_id = do not care */
+ },
+ {
+ .name = "MSM AFE-PCM RX",
+ .stream_name = "AFE-PROXY RX",
+ .cpu_dai_name = "msm-dai-q6.241",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = "MSM AFE-PCM TX",
+ .stream_name = "AFE-PROXY TX",
+ .cpu_dai_name = "msm-dai-q6.240",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "MSM8960 Compr",
+ .stream_name = "COMPR",
+ .cpu_dai_name = "MultiMedia4",
+ .platform_name = "msm-compr-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA4,
+ },
+ {
+ .name = "AUXPCM Hostless",
+ .stream_name = "AUXPCM Hostless",
+ .cpu_dai_name = "AUXPCM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* HDMI Hostless */
+ {
+ .name = "HDMI_RX_HOSTLESS",
+ .stream_name = "HDMI_RX_HOSTLESS",
+ .cpu_dai_name = "HDMI_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = "Voice Stub",
+ .stream_name = "Voice Stub",
+ .cpu_dai_name = "VOICE_STUB",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* Backend DAI Links */
+ {
+ .name = LPASS_BE_SLIMBUS_0_RX,
+ .stream_name = "Slimbus Playback",
+ .cpu_dai_name = "msm-dai-q6.16384",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ .init = &msm_audrx_init,
+ .be_hw_params_fixup = msm_slim_0_rx_be_hw_params_fixup,
+ .ops = &msm_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_0_TX,
+ .stream_name = "Slimbus Capture",
+ .cpu_dai_name = "msm-dai-q6.16385",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ .be_hw_params_fixup = msm_slim_0_tx_be_hw_params_fixup,
+ .ops = &msm_be_ops,
+ },
+ /* Backend BT/FM DAI Links */
+ {
+ .name = LPASS_BE_INT_BT_SCO_RX,
+ .stream_name = "Internal BT-SCO Playback",
+ .cpu_dai_name = "msm-dai-q6.12288",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ .be_hw_params_fixup = msm_btsco_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_BT_SCO_TX,
+ .stream_name = "Internal BT-SCO Capture",
+ .cpu_dai_name = "msm-dai-q6.12289",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ .be_hw_params_fixup = msm_btsco_be_hw_params_fixup,
+ },
+ {
+ .name = LPASS_BE_INT_FM_RX,
+ .stream_name = "Internal FM Playback",
+ .cpu_dai_name = "msm-dai-q6.12292",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_RX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_FM_TX,
+ .stream_name = "Internal FM Capture",
+ .cpu_dai_name = "msm-dai-q6.12293",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_TX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ },
+ /* HDMI BACK END DAI Link */
+ {
+ .name = LPASS_BE_HDMI,
+ .stream_name = "HDMI Playback",
+ .cpu_dai_name = "msm-dai-q6-hdmi.8",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_HDMI_RX,
+ .be_hw_params_fixup = msm_hdmi_be_hw_params_fixup,
+ },
+ /* Backend AFE DAI Links */
+ {
+ .name = LPASS_BE_AFE_PCM_RX,
+ .stream_name = "AFE Playback",
+ .cpu_dai_name = "msm-dai-q6.224",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_RX,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_AFE_PCM_TX,
+ .stream_name = "AFE Capture",
+ .cpu_dai_name = "msm-dai-q6.225",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_TX,
+ },
+ /* AUX PCM Backend DAI Links */
+ {
+ .name = LPASS_BE_AUXPCM_RX,
+ .stream_name = "AUX PCM Playback",
+ .cpu_dai_name = "msm-dai-q6.2",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_RX,
+ .be_hw_params_fixup = msm_auxpcm_be_params_fixup,
+ .ops = &msm_auxpcm_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_AUXPCM_TX,
+ .stream_name = "AUX PCM Capture",
+ .cpu_dai_name = "msm-dai-q6.3",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_TX,
+ .be_hw_params_fixup = msm_auxpcm_be_params_fixup,
+ },
+ {
+ .name = LPASS_BE_STUB_RX,
+ .stream_name = "Stub Playback",
+ .cpu_dai_name = "msm-dai-stub",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_rx2",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_EXTPROC_RX,
+ .be_hw_params_fixup = msm_slim_0_rx_be_hw_params_fixup,
+ .init = &msm_stubrx_init,
+ .ops = &msm_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_STUB_TX,
+ .stream_name = "Stub Capture",
+ .cpu_dai_name = "msm-dai-stub",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_EXTPROC_TX,
+ .be_hw_params_fixup = msm_slim_0_tx_be_hw_params_fixup,
+ .ops = &msm_be_ops,
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_1_RX,
+ .stream_name = "Slimbus1 Playback",
+ .cpu_dai_name = "msm-dai-q6.16386",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ .be_hw_params_fixup = msm_btsco_be_hw_params_fixup,
+ .ops = &msm_slimbus_1_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_1_TX,
+ .stream_name = "Slimbus1 Capture",
+ .cpu_dai_name = "msm-dai-q6.16387",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_1_TX,
+ .be_hw_params_fixup = msm_btsco_be_hw_params_fixup,
+ .ops = &msm_slimbus_1_be_ops,
+ },
+ {
+ .name = "SLIMBUS_2 Hostless",
+ .stream_name = "SLIMBUS_2 Hostless",
+ .cpu_dai_name = "msm-dai-q6.16389",
+ .platform_name = "msm-pcm-hostless",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx2",
+ .ignore_suspend = 1,
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ops = &msm_be_ops,
+ },
+
+ /* Incall Music Back End DAI Link */
+ {
+ .name = LPASS_BE_SLIMBUS_4_RX,
+ .stream_name = "Slimbus4 Playback",
+ .cpu_dai_name = "msm-dai-q6.16392",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_4_RX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ .ops = &msm_slimbus_4_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ /* Incall Record Back End DAI Link */
+ {
+ .name = LPASS_BE_SLIMBUS_4_TX,
+ .stream_name = "Slimbus4 Capture",
+ .cpu_dai_name = "msm-dai-q6.16393",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_4_TX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ .ops = &msm_slimbus_4_be_ops,
+ },
+ {
+ .name = LPASS_BE_STUB_1_TX,
+ .stream_name = "Stub1 Capture",
+ .cpu_dai_name = "msm-dai-stub",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx3",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_EXTPROC_EC_TX,
+ /* This BE is used for external EC reference from codec. Since
+ * Rx is fed as reference for EC, the config of this DAI is
+ * based on that of the Rx path.
+ */
+ .be_hw_params_fixup = msm_slim_0_rx_be_hw_params_fixup,
+ .ops = &msm_be_ops,
+ },
+ {
+
+ .name = LPASS_BE_SLIMBUS_3_RX,
+ .stream_name = "Slimbus3 Playback",
+ .cpu_dai_name = "msm-dai-q6.16390",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_3_RX,
+ .be_hw_params_fixup = msm_slim_3_rx_be_hw_params_fixup,
+ .ops = &msm_slimbus_3_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+};
+
+struct snd_soc_card snd_soc_card_msm = {
+ .name = "apq8064-tabla-snd-card",
+ .dai_link = msm_dai,
+ .num_links = ARRAY_SIZE(msm_dai),
+ .controls = tabla_msm_controls,
+ .num_controls = ARRAY_SIZE(tabla_msm_controls),
+};
+
+static struct platform_device *msm_snd_device;
+
+static int msm_configure_headset_mic_gpios(void)
+{
+ int ret;
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ ret = gpio_request(PM8921_GPIO_PM_TO_SYS(23), "AV_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+
+ ret = pm8xxx_gpio_config(PM8921_GPIO_PM_TO_SYS(23), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(23));
+ else
+ gpio_direction_output(PM8921_GPIO_PM_TO_SYS(23), 0);
+
+ ret = gpio_request(PM8921_GPIO_PM_TO_SYS(35), "US_EURO_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(35));
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+ ret = pm8xxx_gpio_config(PM8921_GPIO_PM_TO_SYS(35), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(35));
+ else
+ gpio_direction_output(PM8921_GPIO_PM_TO_SYS(35), 0);
+
+ return 0;
+}
+static void msm_free_headset_mic_gpios(void)
+{
+ if (msm_headset_gpios_configured) {
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ gpio_free(PM8921_GPIO_PM_TO_SYS(35));
+ }
+}
+
+static int __init msm_audio_init(void)
+{
+ int ret;
+
+ if (!cpu_is_apq8064() || (socinfo_get_id() == 130)) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return -ENODEV;
+ }
+
+ mbhc_cfg.calibration = def_tabla_mbhc_cal();
+ if (!mbhc_cfg.calibration) {
+ pr_err("Calibration data allocation failed\n");
+ return -ENOMEM;
+ }
+
+ msm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(msm_snd_device, &snd_soc_card_msm);
+ ret = platform_device_add(msm_snd_device);
+ if (ret) {
+ platform_device_put(msm_snd_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+
+ if (msm_configure_headset_mic_gpios()) {
+ pr_err("%s Fail to configure headset mic gpios\n", __func__);
+ msm_headset_gpios_configured = 0;
+ } else
+ msm_headset_gpios_configured = 1;
+
+ return ret;
+
+}
+module_init(msm_audio_init);
+
+static void __exit msm_audio_exit(void)
+{
+ if (!cpu_is_apq8064() || (socinfo_get_id() == 130)) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return ;
+ }
+ msm_free_headset_mic_gpios();
+ platform_device_unregister(msm_snd_device);
+ if (mbhc_cfg.gpio)
+ gpio_free(mbhc_cfg.gpio);
+ kfree(mbhc_cfg.calibration);
+}
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC msm");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/lpass-dma.c b/sound/soc/msm/lpass-dma.c
new file mode 100644
index 0000000..66c1836
--- /dev/null
+++ b/sound/soc/msm/lpass-dma.c
@@ -0,0 +1,488 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/debugfs.h>
+#include <linux/delay.h>
+#include <linux/uaccess.h>
+#include <linux/android_pmem.h>
+#include <linux/irq.h>
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#include <linux/slab.h>
+#include <linux/msm_audio.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <mach/msm_iomap-8x60.h>
+#include <mach/audio_dma_msm8k.h>
+#include <sound/dai.h>
+#include "lpass-pcm.h"
+
+struct dai_baseinfo {
+ void __iomem *base;
+};
+
+static struct dai_baseinfo dai_info;
+
+struct dai_drv {
+ u8 *buffer;
+ u32 buffer_phys;
+ int channels;
+ irqreturn_t (*callback) (int intrsrc, void *private_data);
+ void *private_data;
+ int in_use;
+ u32 buffer_len;
+ u32 period_len;
+ u32 master_mode;
+};
+
+static struct dai_drv *dai[MAX_CHANNELS];
+static spinlock_t dai_lock;
+
+static int dai_find_dma_channel(uint32_t intrsrc)
+{
+ int i, dma_channel = 0;
+ pr_debug("%s\n", __func__);
+
+ for (i = 0; i <= 27; i += 3) {
+ if (intrsrc & (1 << i)) {
+ dma_channel = i / 3;
+ break;
+ }
+ }
+ return dma_channel;
+}
+
+void register_dma_irq_handler(int dma_ch,
+ irqreturn_t (*callback) (int intrsrc, void *private_data),
+ void *private_data)
+{
+ pr_debug("%s\n", __func__);
+ dai[dma_ch]->callback = callback;
+ dai[dma_ch]->private_data = private_data;
+}
+
+void unregister_dma_irq_handler(int dma_ch)
+{
+ pr_debug("%s\n", __func__);
+ dai[dma_ch]->callback = NULL;
+ dai[dma_ch]->private_data = NULL;
+}
+
+static irqreturn_t dai_irq_handler(int irq, void *data)
+{
+ unsigned long flag;
+ uint32_t intrsrc;
+ uint32_t dma_ch = 0;
+ irqreturn_t ret = IRQ_HANDLED;
+
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&dai_lock, flag);
+ intrsrc = readl(dai_info.base + LPAIF_IRQ_STAT(0));
+ writel(intrsrc, dai_info.base + LPAIF_IRQ_CLEAR(0));
+ mb();
+ while (intrsrc) {
+ dma_ch = dai_find_dma_channel(intrsrc);
+
+ if (!dai[dma_ch]->callback)
+ goto handled;
+ if (!dai[dma_ch]->private_data)
+ goto handled;
+ ret = dai[dma_ch]->callback(intrsrc,
+ dai[dma_ch]->private_data);
+ intrsrc &= ~(0x7 << (dma_ch * 3));
+ }
+handled:
+ spin_unlock_irqrestore(&dai_lock, flag);
+ return ret;
+}
+
+void dai_print_state(uint32_t dma_ch)
+{
+ int i = 0;
+ unsigned long *ptrmem = (unsigned long *)dai_info.base;
+
+ for (i = 0; i < 4; i++, ++ptrmem)
+ pr_debug("[0x%08x]=0x%08x\n", (unsigned int)ptrmem,
+ (unsigned int)*ptrmem);
+
+ ptrmem = (unsigned long *)(dai_info.base
+ + DMA_CH_CTL_BASE + DMA_CH_INDEX(dma_ch));
+ for (i = 0; i < 10; i++, ++ptrmem)
+ pr_debug("[0x%08x]=0x%08x\n", (unsigned int)ptrmem,
+ (unsigned int) *ptrmem);
+}
+
+static int dai_enable_irq(uint32_t dma_ch)
+{
+ int ret;
+ pr_debug("%s\n", __func__);
+ ret = request_irq(LPASS_SCSS_AUDIO_IF_OUT0_IRQ, dai_irq_handler,
+ IRQF_TRIGGER_RISING | IRQF_SHARED, "msm-i2s",
+ (void *) (dma_ch+1));
+ if (ret < 0) {
+ pr_debug("Request Irq Failed err = %d\n", ret);
+ return ret;
+ }
+ return ret;
+}
+
+static void dai_config_dma(uint32_t dma_ch)
+{
+ pr_debug("%s dma_ch = %u\n", __func__, dma_ch);
+
+ writel(dai[dma_ch]->buffer_phys,
+ dai_info.base + LPAIF_DMA_BASE(dma_ch));
+ writel(((dai[dma_ch]->buffer_len >> 2) - 1),
+ dai_info.base + LPAIF_DMA_BUFF_LEN(dma_ch));
+ writel(((dai[dma_ch]->period_len >> 2) - 1),
+ dai_info.base + LPAIF_DMA_PER_LEN(dma_ch));
+ mb();
+}
+
+static void dai_enable_codec(uint32_t dma_ch, int codec)
+{
+ uint32_t intrVal;
+ uint32_t i2sctl;
+ pr_debug("%s\n", __func__);
+
+ intrVal = readl(dai_info.base + LPAIF_IRQ_EN(0));
+ intrVal = intrVal | (7 << (dma_ch * 3));
+ writel(intrVal, dai_info.base + LPAIF_IRQ_EN(0));
+ if (codec == DAI_SPKR) {
+ writel(0x0813, dai_info.base + LPAIF_DMA_CTL(dma_ch));
+ i2sctl = 0x4400;
+ i2sctl |= (dai[dma_ch]->master_mode ? WS_SRC_INT : WS_SRC_EXT);
+ writel(i2sctl, dai_info.base + LPAIF_I2S_CTL_OFFSET(DAI_SPKR));
+ } else if (codec == DAI_MIC) {
+ writel(0x81b, dai_info.base + LPAIF_DMA_CTL(dma_ch));
+ i2sctl = 0x0110;
+ i2sctl |= (dai[dma_ch]->master_mode ? WS_SRC_INT : WS_SRC_EXT);
+ writel(i2sctl, dai_info.base + LPAIF_I2S_CTL_OFFSET(DAI_MIC));
+ }
+}
+
+static void dai_disable_codec(uint32_t dma_ch, int codec)
+{
+ uint32_t intrVal = 0;
+ uint32_t intrVal1 = 0;
+ unsigned long flag = 0x0;
+
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&dai_lock, flag);
+
+ intrVal1 = readl(dai_info.base + LPAIF_I2S_CTL_OFFSET(codec));
+
+ if (codec == DAI_SPKR)
+ intrVal1 = intrVal1 & ~(1 << 14);
+ else if (codec == DAI_MIC)
+ intrVal1 = intrVal1 & ~(1 << 8);
+
+ writel(intrVal1, dai_info.base + LPAIF_I2S_CTL_OFFSET(codec));
+ intrVal = 0x0;
+ writel(intrVal, dai_info.base + LPAIF_DMA_CTL(dma_ch));
+
+ spin_unlock_irqrestore(&dai_lock, flag);
+}
+
+int dai_open(uint32_t dma_ch)
+{
+
+ pr_debug("%s\n", __func__);
+ if (!dai_info.base) {
+ pr_debug("%s failed as no msm-dai device\n", __func__);
+ return -ENODEV;
+ }
+ if (dma_ch >= MAX_CHANNELS) {
+ pr_debug("%s over max channesl %d\n", __func__, dma_ch);
+ return -ENODEV;
+ }
+ return 0;
+}
+
+void dai_close(uint32_t dma_ch)
+{
+ pr_debug("%s\n", __func__);
+ if ((dma_ch >= 0) && (dma_ch < 5))
+ dai_disable_codec(dma_ch, DAI_SPKR);
+ else
+ dai_disable_codec(dma_ch, DAI_MIC);
+ free_irq(LPASS_SCSS_AUDIO_IF_OUT0_IRQ, (void *) (dma_ch + 1));
+}
+
+void dai_set_master_mode(uint32_t dma_ch, int mode)
+{
+ if (dma_ch < MAX_CHANNELS)
+ dai[dma_ch]->master_mode = mode;
+ else
+ pr_err("%s: invalid dma channel\n", __func__);
+}
+
+int dai_set_params(uint32_t dma_ch, struct dai_dma_params *params)
+{
+ pr_debug("%s\n", __func__);
+ dai[dma_ch]->buffer = params->buffer;
+ dai[dma_ch]->buffer_phys = params->src_start;
+ dai[dma_ch]->channels = params->channels;
+ dai[dma_ch]->buffer_len = params->buffer_size;
+ dai[dma_ch]->period_len = params->period_size;
+ mb();
+ dai_config_dma(dma_ch);
+ return dma_ch;
+}
+
+int dai_start(uint32_t dma_ch)
+{
+ unsigned long flag = 0x0;
+
+ spin_lock_irqsave(&dai_lock, flag);
+ dai_enable_irq(dma_ch);
+ if ((dma_ch >= 0) && (dma_ch < 5))
+ dai_enable_codec(dma_ch, DAI_SPKR);
+ else
+ dai_enable_codec(dma_ch, DAI_MIC);
+ spin_unlock_irqrestore(&dai_lock, flag);
+ dai_print_state(dma_ch);
+ return 0;
+}
+
+#define HDMI_BURST_INCR4 (1 << 11)
+#define HDMI_WPSCNT (1 << 8)
+#define HDMI_AUDIO_INTF (5 << 4)
+#define HDMI_FIFO_WATER_MARK (7 << 1)
+#define HDMI_ENABLE (1)
+
+int dai_start_hdmi(uint32_t dma_ch)
+{
+ unsigned long flag = 0x0;
+ uint32_t val;
+
+ pr_debug("%s dma_ch = %u\n", __func__, dma_ch);
+
+ spin_lock_irqsave(&dai_lock, flag);
+
+ dai_enable_irq(dma_ch);
+
+ if ((dma_ch >= 0) && (dma_ch < 5)) {
+
+ val = readl(dai_info.base + LPAIF_IRQ_EN(0));
+ val = val | (7 << (dma_ch * 3));
+ writel(val, dai_info.base + LPAIF_IRQ_EN(0));
+ mb();
+
+
+ val = (HDMI_BURST_INCR4 | HDMI_WPSCNT | HDMI_AUDIO_INTF |
+ HDMI_FIFO_WATER_MARK | HDMI_ENABLE);
+
+ writel(val, dai_info.base + LPAIF_DMA_CTL(dma_ch));
+ }
+ spin_unlock_irqrestore(&dai_lock, flag);
+
+ mb();
+ dai_print_state(dma_ch);
+ return 0;
+}
+
+int wait_for_dma_cnt_stop(uint32_t dma_ch)
+{
+ uint32_t dma_per_cnt_reg_val, dma_per_cnt, prev_dma_per_cnt;
+ uint32_t i;
+
+ pr_info("%s dma_ch %u\n", __func__, dma_ch);
+
+ dma_per_cnt_reg_val = readl_relaxed(dai_info.base +
+ LPAIF_DMA_PER_CNT(dma_ch));
+
+ dma_per_cnt =
+ ((LPAIF_DMA_PER_CNT_PER_CNT_MASK & dma_per_cnt_reg_val) >>
+ LPAIF_DMA_PER_CNT_PER_CNT_SHIFT) -
+ ((LPAIF_DMA_PER_CNT_FIFO_WORDCNT_MASK & dma_per_cnt_reg_val) >>
+ LPAIF_DMA_PER_CNT_FIFO_WORDCNT_SHIFT);
+
+ prev_dma_per_cnt = dma_per_cnt;
+
+ i = 1;
+ pr_info("%s: i = %u dma_per_cnt_reg_val 0x%08x , dma_per_cnt %u\n",
+ __func__, i, dma_per_cnt_reg_val, dma_per_cnt);
+
+ while (i <= 50) {
+ msleep(50);
+
+ dma_per_cnt_reg_val = readl_relaxed(dai_info.base +
+ LPAIF_DMA_PER_CNT(dma_ch));
+
+ dma_per_cnt =
+ ((LPAIF_DMA_PER_CNT_PER_CNT_MASK & dma_per_cnt_reg_val) >>
+ LPAIF_DMA_PER_CNT_PER_CNT_SHIFT) -
+ ((LPAIF_DMA_PER_CNT_FIFO_WORDCNT_MASK & dma_per_cnt_reg_val) >>
+ LPAIF_DMA_PER_CNT_FIFO_WORDCNT_SHIFT);
+
+ i++;
+
+ pr_info("%s: i = %u dma_per_cnt_reg_val 0x%08x , dma_per_cnt %u\n",
+ __func__, i, dma_per_cnt_reg_val, dma_per_cnt);
+
+ if (prev_dma_per_cnt == dma_per_cnt)
+ break;
+
+ prev_dma_per_cnt = dma_per_cnt;
+ }
+ return 0;
+}
+
+void dai_stop_hdmi(uint32_t dma_ch)
+{
+ unsigned long flag = 0x0;
+ uint32_t intrVal;
+ uint32_t int_mask = 0x00000007;
+
+ pr_debug("%s dma_ch %u\n", __func__, dma_ch);
+
+ spin_lock_irqsave(&dai_lock, flag);
+
+ free_irq(LPASS_SCSS_AUDIO_IF_OUT0_IRQ, (void *) (dma_ch + 1));
+
+
+ intrVal = 0x0;
+ writel(intrVal, dai_info.base + LPAIF_DMA_CTL(dma_ch));
+
+ mb();
+
+ intrVal = readl(dai_info.base + LPAIF_IRQ_EN(0));
+
+ int_mask = ((int_mask) << (dma_ch * 3));
+ int_mask = ~int_mask;
+
+ intrVal = intrVal & int_mask;
+ writel(intrVal, dai_info.base + LPAIF_IRQ_EN(0));
+
+ mb();
+
+ spin_unlock_irqrestore(&dai_lock, flag);
+}
+
+int dai_stop(uint32_t dma_ch)
+{
+ pr_debug("%s\n", __func__);
+ return 0;
+}
+
+
+uint32_t dai_get_dma_pos(uint32_t dma_ch)
+{
+
+ uint32_t addr;
+
+ pr_debug("%s\n", __func__);
+ addr = readl(dai_info.base + LPAIF_DMA_CURR_ADDR(dma_ch));
+
+ return addr;
+}
+
+static int __devinit dai_probe(struct platform_device *pdev)
+{
+ int rc = 0;
+ int i = 0;
+ struct resource *src;
+ src = platform_get_resource_byname(pdev, IORESOURCE_MEM, "msm-dai");
+ if (!src) {
+ rc = -ENODEV;
+ pr_debug("%s Error rc=%d\n", __func__, rc);
+ goto error;
+ }
+ for (i = 0; i <= MAX_CHANNELS; i++) {
+ dai[i] = kzalloc(sizeof(struct dai_drv), GFP_KERNEL);
+ if (!dai[0]) {
+ pr_debug("Allocation failed for dma_channel = 0\n");
+ return -ENODEV;
+ }
+ }
+ dai_info.base = ioremap(src->start, (src->end - src->start) + 1);
+ pr_debug("%s: msm-dai: 0x%08x\n", __func__,
+ (unsigned int)dai_info.base);
+ spin_lock_init(&dai_lock);
+error:
+ return rc;
+}
+
+static int dai_remove(struct platform_device *pdev)
+{
+ iounmap(dai_info.base);
+ return 0;
+}
+
+static struct platform_driver dai_driver = {
+ .probe = dai_probe,
+ .remove = dai_remove,
+ .driver = {
+ .name = "msm-dai",
+ .owner = THIS_MODULE
+ },
+};
+
+static struct resource msm_lpa_resources[] = {
+ {
+ .start = MSM_LPA_PHYS,
+ .end = MSM_LPA_END,
+ .flags = IORESOURCE_MEM,
+ .name = "msm-dai",
+ },
+};
+
+static struct platform_device *codec_device;
+
+static int msm_dai_dev_register(const char *name)
+{
+ int ret = 0;
+
+ pr_debug("%s : called\n", __func__);
+ codec_device = platform_device_alloc(name, -1);
+ if (codec_device == NULL) {
+ pr_debug("Failed to allocate %s\n", name);
+ return -ENODEV;
+ }
+
+ platform_set_drvdata(codec_device, (void *)&dai_info);
+ platform_device_add_resources(codec_device, &msm_lpa_resources[0],
+ ARRAY_SIZE(msm_lpa_resources));
+ ret = platform_device_add(codec_device);
+ if (ret != 0) {
+ pr_debug("Failed to register %s: %d\n", name, ret);
+ platform_device_put(codec_device);
+ }
+ return ret;
+}
+
+static int __init dai_init(void)
+{
+ if (msm_dai_dev_register("msm-dai")) {
+ pr_notice("dai_init: msm-dai Failed");
+ return -ENODEV;
+ }
+ return platform_driver_register(&dai_driver);
+}
+
+static void __exit dai_exit(void)
+{
+ platform_driver_unregister(&dai_driver);
+ platform_device_put(codec_device);
+}
+
+module_init(dai_init);
+module_exit(dai_exit);
+
+MODULE_DESCRIPTION("MSM I2S driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/lpass-i2s.c b/sound/soc/msm/lpass-i2s.c
new file mode 100644
index 0000000..9583c52
--- /dev/null
+++ b/sound/soc/msm/lpass-i2s.c
@@ -0,0 +1,139 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/dai.h>
+
+static int msm_cpu_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ uint32_t dma_ch = dai->id;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ ret = dai_open(dma_ch);
+ return ret;
+
+}
+
+static void msm_cpu_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ uint32_t dma_ch = dai->id;
+
+ pr_debug("%s\n", __func__);
+ dai_close(dma_ch);
+}
+
+static int msm_cpu_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ pr_debug("%s\n", __func__);
+ return 0;
+}
+
+static int msm_cpu_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ uint32_t dma_ch = dai->id;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ dai_set_master_mode(dma_ch, 1); /* CPU is master */
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ dai_set_master_mode(dma_ch, 0); /* CPU is slave */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops msm_cpu_dai_ops = {
+ .startup = msm_cpu_dai_startup,
+ .shutdown = msm_cpu_dai_shutdown,
+ .trigger = msm_cpu_dai_trigger,
+ .set_fmt = msm_cpu_dai_fmt,
+
+};
+
+
+#define MSM_DAI_SPEAKER_BUILDER(link_id) \
+{ \
+ .name = "msm-speaker-dai-"#link_id, \
+ .id = (link_id), \
+ .playback = { \
+ .rates = SNDRV_PCM_RATE_8000_96000, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ .channels_min = 1, \
+ .channels_max = 2, \
+ .rate_max = 96000, \
+ .rate_min = 8000, \
+ }, \
+ .ops = &msm_cpu_dai_ops, \
+}
+
+
+#define MSM_DAI_MIC_BUILDER(link_id) \
+{ \
+ .name = "msm-mic-dai-"#link_id, \
+ .id = (link_id), \
+ .capture = { \
+ .rates = SNDRV_PCM_RATE_8000_96000, \
+ .formats = SNDRV_PCM_FMTBIT_S16_LE, \
+ .rate_min = 8000, \
+ .rate_max = 96000, \
+ .channels_min = 1, \
+ .channels_max = 2, \
+ }, \
+ .ops = &msm_cpu_dai_ops, \
+}
+
+
+struct snd_soc_dai msm_cpu_dai[] = {
+ MSM_DAI_SPEAKER_BUILDER(0),
+ MSM_DAI_SPEAKER_BUILDER(1),
+ MSM_DAI_SPEAKER_BUILDER(2),
+ MSM_DAI_SPEAKER_BUILDER(3),
+ MSM_DAI_SPEAKER_BUILDER(4),
+ MSM_DAI_MIC_BUILDER(5),
+ MSM_DAI_MIC_BUILDER(6),
+ MSM_DAI_MIC_BUILDER(7),
+};
+EXPORT_SYMBOL_GPL(msm_cpu_dai);
+
+static int __init msm_cpu_dai_init(void)
+{
+ return snd_soc_register_dais(msm_cpu_dai, ARRAY_SIZE(msm_cpu_dai));
+}
+module_init(msm_cpu_dai_init);
+
+static void __exit msm_cpu_dai_exit(void)
+{
+ snd_soc_unregister_dais(msm_cpu_dai, ARRAY_SIZE(msm_cpu_dai));
+}
+module_exit(msm_cpu_dai_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM CPU DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/lpass-pcm.c b/sound/soc/msm/lpass-pcm.c
new file mode 100644
index 0000000..efd7c06
--- /dev/null
+++ b/sound/soc/msm/lpass-pcm.c
@@ -0,0 +1,369 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/audio_dma_msm8k.h>
+#include <sound/dai.h>
+#include "lpass-pcm.h"
+
+static const struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .period_bytes_min = 32,
+ .period_bytes_max = DMASZ/4,
+ .buffer_bytes_max = DMASZ,
+ .rate_max = 96000,
+ .rate_min = 8000,
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .periods_min = 4,
+ .periods_max = 512,
+ .fifo_size = 0,
+};
+
+struct msm_pcm_data {
+ spinlock_t lock;
+ int ch;
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+
+ pr_debug("%s\n", __func__);
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static irqreturn_t msm_pcm_irq(int intrsrc, void *data)
+{
+ struct snd_pcm_substream *substream = data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = (struct msm_audio *)runtime->private_data;
+ int dma_ch = 0;
+ unsigned int has_xrun, pending;
+ int ret = IRQ_NONE;
+
+ if (prtd)
+ dma_ch = prtd->dma_ch;
+ else
+ return ret;
+
+ pr_debug("msm8660-pcm: msm_pcm_irq called\n");
+ pending = (intrsrc
+ & (UNDER_CH(dma_ch) | PER_CH(dma_ch) | ERR_CH(dma_ch)));
+ has_xrun = (pending & UNDER_CH(dma_ch));
+
+ if (unlikely(has_xrun) &&
+ substream->runtime &&
+ snd_pcm_running(substream)) {
+ pr_err("xrun\n");
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ ret = IRQ_HANDLED;
+ pending &= ~UNDER_CH(dma_ch);
+ }
+
+
+ if (pending & PER_CH(dma_ch)) {
+ ret = IRQ_HANDLED;
+ if (likely(substream->runtime &&
+ snd_pcm_running(substream))) {
+ /* end of buffer missed? loop back */
+ if (++prtd->period_index >= runtime->periods)
+ prtd->period_index = 0;
+ snd_pcm_period_elapsed(substream);
+ pr_debug("period elapsed\n");
+ }
+ pending &= ~PER_CH(dma_ch);
+ }
+
+ if (unlikely(pending
+ & (UNDER_CH(dma_ch) & PER_CH(dma_ch) & ERR_CH(dma_ch)))) {
+ if (pending & UNDER_CH(dma_ch))
+ pr_err("msm8660-pcm: DMA %x Underflow\n",
+ dma_ch);
+ if (pending & ERR_CH(dma_ch))
+ pr_err("msm8660-pcm: DMA %x Master Error\n",
+ dma_ch);
+
+ }
+ return ret;
+}
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = (struct msm_audio *)runtime->private_data;
+ struct dai_dma_params dma_params;
+ int dma_ch = 0;
+
+ if (prtd)
+ dma_ch = prtd->dma_ch;
+ else
+ return 0;
+
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ pr_debug("%s:prtd->pcm_size = %d\n", __func__, prtd->pcm_size);
+ pr_debug("%s:prtd->pcm_count = %d\n", __func__, prtd->pcm_count);
+
+ if (prtd->enabled)
+ return 0;
+
+ dma_params.src_start = runtime->dma_addr;
+ dma_params.buffer = (u8 *)runtime->dma_area;
+ dma_params.buffer_size = prtd->pcm_size;
+ dma_params.period_size = prtd->pcm_count;
+ dma_params.channels = runtime->channels;
+
+ dai_set_params(dma_ch, &dma_params);
+ register_dma_irq_handler(dma_ch, msm_pcm_irq, (void *)substream);
+
+ prtd->enabled = 1;
+ return 0;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = (struct msm_audio *)runtime->private_data;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ dai_start(prtd->dma_ch);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ dai_stop(prtd->dma_ch);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = (struct msm_audio *)runtime->private_data;
+ snd_pcm_uframes_t offset = 0;
+
+ pr_debug("%s: period_index =%d\n", __func__, prtd->period_index);
+ offset = prtd->period_index * runtime->period_size;
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+ return offset;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct msm_audio *prtd = NULL;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ snd_soc_set_runtime_hwparams(substream, &msm_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+
+ if (ret < 0) {
+ pr_err("Error setting hw_constraint\n");
+ goto err;
+ }
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_err("Error snd_pcm_hw_constraint_list failed\n");
+
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+
+ if (prtd == NULL) {
+ pr_err("Error allocating prtd\n");
+ ret = -ENOMEM;
+ goto err;
+ }
+ prtd->dma_ch = cpu_dai->id;
+ prtd->enabled = 0;
+ runtime->dma_bytes = msm_pcm_hardware.buffer_bytes_max;
+ runtime->private_data = prtd;
+err:
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = (struct msm_audio *)runtime->private_data;
+ int dma_ch = 0;
+
+ if (prtd)
+ dma_ch = prtd->dma_ch;
+ else
+ return 0;
+
+ pr_debug("%s\n", __func__);
+ unregister_dma_irq_handler(dma_ch);
+ kfree(runtime->private_data);
+ return 0;
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vms)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ pr_debug("%s\n", __func__);
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ pr_debug("%s: snd_msm_audio_hw_params runtime->dma_addr 0x(%x)\n",
+ __func__, (unsigned int)runtime->dma_addr);
+ pr_debug("%s: snd_msm_audio_hw_params runtime->dma_area 0x(%x)\n",
+ __func__, (unsigned int)runtime->dma_area);
+ pr_debug("%s: snd_msm_audio_hw_params runtime->dma_bytes 0x(%x)\n",
+ __func__, (unsigned int)runtime->dma_bytes);
+
+ return dma_mmap_coherent(substream->pcm->card->dev, vms,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = msm_pcm_hw_params,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int pcm_preallocate_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = msm_pcm_hardware.buffer_bytes_max;
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void msm_pcm_free_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!stream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+static u64 msm_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int msm_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &msm_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (dai->playback.channels_min) {
+ ret = pcm_preallocate_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ }
+ if (dai->capture.channels_min) {
+ ret = pcm_preallocate_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ return ret;
+ }
+ return ret;
+}
+
+struct snd_soc_platform msm8660_soc_platform = {
+ .name = "msm8660-pcm-audio",
+ .pcm_ops = &msm_pcm_ops,
+ .pcm_new = msm_pcm_new,
+ .pcm_free = msm_pcm_free_buffers,
+};
+EXPORT_SYMBOL_GPL(msm8660_soc_platform);
+
+static int __init msm_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&msm8660_soc_platform);
+}
+static void __exit msm_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&msm8660_soc_platform);
+}
+module_init(msm_soc_platform_init);
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("MSM PCM module");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/lpass-pcm.h b/sound/soc/msm/lpass-pcm.h
new file mode 100644
index 0000000..3bec9a7
--- /dev/null
+++ b/sound/soc/msm/lpass-pcm.h
@@ -0,0 +1,45 @@
+/* Copyright (c) 2010, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+
+#define USE_CHANNELS_MIN 1
+#define USE_CHANNELS_MAX 2
+#define NUM_DMAS 9
+#define DMASZ 16384
+#define MAX_CHANNELS 9
+
+#define MSM_LPA_PHYS 0x28100000
+#define MSM_LPA_END 0x2810DFFF
+
+
+struct msm_audio {
+ struct snd_pcm_substream *substream;
+
+ /* data allocated for various buffers */
+ char *data;
+ dma_addr_t phys;
+
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ int enabled;
+ int period;
+ int dma_ch;
+ int period_index;
+ int start;
+};
+
+extern struct snd_soc_dai msm_cpu_dai[NUM_DMAS];
+extern struct snd_soc_platform msm8660_soc_platform;
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/mdm9615.c b/sound/soc/msm/mdm9615.c
new file mode 100644
index 0000000..41a9c48
--- /dev/null
+++ b/sound/soc/msm/mdm9615.c
@@ -0,0 +1,2041 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8018.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <mach/socinfo.h>
+#include "msm-pcm-routing.h"
+#include "../codecs/wcd9310.h"
+#include <mach/gpiomux.h>
+
+/* 9615 machine driver */
+
+#define PM8018_GPIO_BASE NR_GPIO_IRQS
+#define PM8018_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio - 1 + PM8018_GPIO_BASE)
+
+#define MDM9615_SPK_ON 1
+#define MDM9615_SPK_OFF 0
+
+#define MDM9615_SLIM_0_RX_MAX_CHANNELS 2
+#define MDM9615_SLIM_0_TX_MAX_CHANNELS 4
+
+#define BTSCO_RATE_8KHZ 8000
+#define BTSCO_RATE_16KHZ 16000
+
+#define BOTTOM_SPK_AMP_POS 0x1
+#define BOTTOM_SPK_AMP_NEG 0x2
+#define TOP_SPK_AMP_POS 0x4
+#define TOP_SPK_AMP_NEG 0x8
+
+#define GPIO_AUX_PCM_DOUT 63
+#define GPIO_AUX_PCM_DIN 64
+#define GPIO_AUX_PCM_SYNC 65
+#define GPIO_AUX_PCM_CLK 66
+
+#define TABLA_EXT_CLK_RATE 12288000
+
+#define TABLA_MBHC_DEF_BUTTONS 8
+#define TABLA_MBHC_DEF_RLOADS 5
+
+/*
+ * Added for I2S
+ */
+#define GPIO_SPKR_I2S_MCLK 24
+#define GPIO_PRIM_I2S_SCK 20
+#define GPIO_PRIM_I2S_DOUT 23
+#define GPIO_PRIM_I2S_WS 21
+#define GPIO_PRIM_I2S_DIN 22
+#define GPIO_SEC_I2S_SCK 25
+#define GPIO_SEC_I2S_WS 26
+#define GPIO_SEC_I2S_DOUT 28
+#define GPIO_SEC_I2S_DIN 27
+
+static struct gpiomux_setting cdc_i2s_mclk = {
+ .func = GPIOMUX_FUNC_1,
+ .drv = GPIOMUX_DRV_8MA,
+ .pull = GPIOMUX_PULL_NONE,
+};
+
+static struct gpiomux_setting cdc_i2s_sclk = {
+ .func = GPIOMUX_FUNC_1,
+ .drv = GPIOMUX_DRV_8MA,
+ .pull = GPIOMUX_PULL_NONE,
+};
+
+static struct gpiomux_setting cdc_i2s_dout = {
+ .func = GPIOMUX_FUNC_1,
+ .drv = GPIOMUX_DRV_8MA,
+ .pull = GPIOMUX_PULL_NONE,
+};
+
+static struct gpiomux_setting cdc_i2s_ws = {
+ .func = GPIOMUX_FUNC_1,
+ .drv = GPIOMUX_DRV_8MA,
+ .pull = GPIOMUX_PULL_NONE,
+};
+
+static struct gpiomux_setting cdc_i2s_din = {
+ .func = GPIOMUX_FUNC_1,
+ .drv = GPIOMUX_DRV_8MA,
+ .pull = GPIOMUX_PULL_NONE,
+};
+
+
+static struct msm_gpiomux_config msm9615_audio_prim_i2s_codec_configs[] = {
+ {
+ .gpio = GPIO_SPKR_I2S_MCLK,
+ .settings = {
+ [GPIOMUX_SUSPENDED] = &cdc_i2s_mclk,
+ },
+ },
+ {
+ .gpio = GPIO_PRIM_I2S_SCK,
+ .settings = {
+ [GPIOMUX_SUSPENDED] = &cdc_i2s_sclk,
+ },
+ },
+ {
+ .gpio = GPIO_PRIM_I2S_DOUT,
+ .settings = {
+ [GPIOMUX_SUSPENDED] = &cdc_i2s_dout,
+ },
+ },
+ {
+ .gpio = GPIO_PRIM_I2S_WS,
+ .settings = {
+ [GPIOMUX_SUSPENDED] = &cdc_i2s_ws,
+ },
+ },
+ {
+ .gpio = GPIO_PRIM_I2S_DIN,
+ .settings = {
+ [GPIOMUX_SUSPENDED] = &cdc_i2s_din,
+ },
+ },
+};
+
+/* Physical address for LPA CSR
+ * LPA SIF mux registers. These are
+ * ioremap( ) for Virtual address.
+ */
+#define LPASS_CSR_BASE 0x28000000
+#define LPA_IF_BASE 0x28100000
+#define SIF_MUX_REG_BASE (LPASS_CSR_BASE + 0x00000000)
+#define LPA_IF_REG_BASE (LPA_IF_BASE + 0x00000000)
+#define LPASS_SIF_MUX_ADDR (SIF_MUX_REG_BASE + 0x00004000)
+#define LPAIF_SPARE_ADDR (LPA_IF_REG_BASE + 0x00000070)
+/* SIF & SPARE MUX Values */
+#define MSM_SIF_FUNC_PCM 0
+#define MSM_SIF_FUNC_I2S_MIC 1
+#define MSM_SIF_FUNC_I2S_SPKR 2
+#define MSM_LPAIF_SPARE_DISABLE 0x0
+#define MSM_LPAIF_SPARE_BOTH_ENABLE 0x3
+
+/* I2S INTF CTL */
+#define MSM_INTF_PRIM 0
+#define MSM_INTF_SECN 1
+#define MSM_INTF_BOTH 2
+
+/* I2S Dir CTL */
+#define MSM_DIR_RX 0
+#define MSM_DIR_TX 1
+#define MSM_DIR_BOTH 2
+#define MSM_DIR_MAX 3
+
+/* I2S HW Params */
+#define NO_OF_BITS_PER_SAMPLE 16
+#define I2S_MIC_SCLK_RATE 1536000
+static int msm9615_i2s_rx_ch = 1;
+static int msm9615_i2s_tx_ch = 1;
+static int msm9615_i2s_spk_control;
+/* SIF mux bit mask & shift */
+#define LPASS_SIF_MUX_CTL_PRI_MUX_SEL_BMSK 0x30000
+#define LPASS_SIF_MUX_CTL_PRI_MUX_SEL_SHFT 0x10
+#define LPASS_SIF_MUX_CTL_SEC_MUX_SEL_BMSK 0x3
+#define LPASS_SIF_MUX_CTL_SEC_MUX_SEL_SHFT 0x0
+
+#define LPAIF_SPARE_MUX_CTL_SEC_MUX_SEL_BMSK 0x3
+#define LPAIF_SPARE_MUX_CTL_SEC_MUX_SEL_SHFT 0x2
+#define LPAIF_SPARE_MUX_CTL_PRI_MUX_SEL_BMSK 0x3
+#define LPAIF_SPARE_MUX_CTL_PRI_MUX_SEL_SHFT 0x0
+
+static u32 spare_shadow;
+static u32 sif_shadow;
+
+
+struct msm_i2s_mux_ctl {
+ const u8 sifconfig;
+ const u8 spareconfig;
+};
+struct msm_clk {
+ struct clk *osr_clk;
+ struct clk *bit_clk;
+ int clk_enable;
+};
+struct msm_i2s_clk {
+ struct msm_clk rx_clk;
+ struct msm_clk tx_clk;
+};
+struct msm_i2s_ctl {
+ struct msm_i2s_clk prim_clk;
+ struct msm_i2s_clk sec_clk;
+ struct msm_i2s_mux_ctl mux_ctl[MSM_DIR_MAX];
+ u8 intf_status[MSM_INTF_BOTH][MSM_DIR_BOTH];
+ void *sif_virt_addr;
+ void *spare_virt_addr;
+};
+static struct msm_i2s_ctl msm9x15_i2s_ctl = {
+ {{NULL, NULL, 0}, {NULL, NULL, 0} }, /* prim_clk */
+ {{NULL, NULL, 0}, {NULL, NULL, 0} }, /* sec_clk */
+ /* mux_ctl */
+ {
+ /* Rx path only */
+ { MSM_SIF_FUNC_I2S_SPKR, MSM_LPAIF_SPARE_DISABLE },
+ /* Tx path only */
+ { MSM_SIF_FUNC_I2S_MIC, MSM_LPAIF_SPARE_DISABLE },
+ /* Rx + Tx path only */
+ { MSM_SIF_FUNC_I2S_SPKR, MSM_LPAIF_SPARE_BOTH_ENABLE },
+ },
+ /* intf_status */
+ {
+ /* Prim I2S */
+ {0, 0},
+ /* Sec I2S */
+ {0, 0}
+ },
+ /* sif_virt_addr */
+ NULL,
+ /* spare_virt_addr */
+ NULL,
+};
+
+enum msm9x15_set_i2s_clk {
+ MSM_I2S_CLK_SET_FALSE,
+ MSM_I2S_CLK_SET_TRUE,
+ MSM_I2S_CLK_SET_RATE0,
+};
+/*
+ * Added for I2S
+ */
+
+static u32 top_spk_pamp_gpio = PM8018_GPIO_PM_TO_SYS(3);
+static u32 bottom_spk_pamp_gpio = PM8018_GPIO_PM_TO_SYS(5);
+static int mdm9615_spk_control;
+static int mdm9615_ext_bottom_spk_pamp;
+static int mdm9615_ext_top_spk_pamp;
+static int mdm9615_slim_0_rx_ch = 1;
+static int mdm9615_slim_0_tx_ch = 1;
+
+static int mdm9615_btsco_rate = BTSCO_RATE_8KHZ;
+static int mdm9615_btsco_ch = 1;
+
+static struct clk *codec_clk;
+static int clk_users;
+
+static int mdm9615_headset_gpios_configured;
+
+static struct snd_soc_jack hs_jack;
+static struct snd_soc_jack button_jack;
+
+static int mdm9615_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm);
+static struct tabla_mbhc_config mbhc_cfg = {
+ .headset_jack = &hs_jack,
+ .button_jack = &button_jack,
+ .read_fw_bin = false,
+ .calibration = NULL,
+ .micbias = TABLA_MICBIAS2,
+ .mclk_cb_fn = mdm9615_enable_codec_ext_clk,
+ .mclk_rate = TABLA_EXT_CLK_RATE,
+ .gpio = 0,
+ .gpio_irq = 0,
+ .gpio_level_insert = 1,
+};
+
+static void mdm9615_enable_ext_spk_amp_gpio(u32 spk_amp_gpio)
+{
+ int ret = 0;
+
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ if (spk_amp_gpio == bottom_spk_pamp_gpio) {
+
+ ret = gpio_request(bottom_spk_pamp_gpio, "BOTTOM_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting BOTTOM SPK AMP GPIO %u\n",
+ __func__, bottom_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(bottom_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Bottom Spk Ampl"
+ " gpio %u\n", __func__, bottom_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Bottom spkr amp gpio\n", __func__);
+ gpio_direction_output(bottom_spk_pamp_gpio, 1);
+ }
+
+ } else if (spk_amp_gpio == top_spk_pamp_gpio) {
+
+ ret = gpio_request(top_spk_pamp_gpio, "TOP_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting GPIO %d\n", __func__,
+ top_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(top_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Top Spk Ampl"
+ " gpio %u\n", __func__, top_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Top spkr amp gpio\n", __func__);
+ gpio_direction_output(top_spk_pamp_gpio, 1);
+ }
+ } else {
+ pr_err("%s: ERROR : Invalid External Speaker Ampl GPIO."
+ " gpio = %u\n", __func__, spk_amp_gpio);
+ return;
+ }
+}
+
+static void mdm9615_ext_spk_power_amp_on(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if ((mdm9615_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (mdm9615_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Bottom Speaker Ampl already "
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ mdm9615_ext_bottom_spk_pamp |= spk;
+
+ if ((mdm9615_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (mdm9615_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ mdm9615_enable_ext_spk_amp_gpio(bottom_spk_pamp_gpio);
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Bottom Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if ((mdm9615_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (mdm9615_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Top Speaker Ampl already"
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ mdm9615_ext_top_spk_pamp |= spk;
+
+ if ((mdm9615_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (mdm9615_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ mdm9615_enable_ext_spk_amp_gpio(top_spk_pamp_gpio);
+ pr_debug("%s: sleeping 4 ms after turning on "
+ " external Top Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void mdm9615_ext_spk_power_amp_off(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if (!mdm9615_ext_bottom_spk_pamp)
+ return;
+
+ gpio_direction_output(bottom_spk_pamp_gpio, 0);
+ gpio_free(bottom_spk_pamp_gpio);
+ mdm9615_ext_bottom_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Bottom"
+ " Speaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if (!mdm9615_ext_top_spk_pamp)
+ return;
+
+ gpio_direction_output(top_spk_pamp_gpio, 0);
+ gpio_free(top_spk_pamp_gpio);
+ mdm9615_ext_top_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Top"
+ " Spkaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+ } else {
+
+ pr_err("%s: ERROR : Invalid Ext Spk Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void mdm9615_ext_control(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ pr_debug("%s: mdm9615_spk_control = %d", __func__, mdm9615_spk_control);
+ if (mdm9615_spk_control == MDM9615_SPK_ON) {
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+ } else {
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Neg");
+ }
+
+ snd_soc_dapm_sync(dapm);
+}
+
+static int mdm9615_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: mdm9615_spk_control = %d", __func__, mdm9615_spk_control);
+ ucontrol->value.integer.value[0] = mdm9615_spk_control;
+ return 0;
+}
+static int mdm9615_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (mdm9615_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ mdm9615_spk_control = ucontrol->value.integer.value[0];
+ mdm9615_ext_control(codec);
+ return 1;
+}
+static int mdm9615_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ mdm9615_ext_spk_power_amp_on(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ mdm9615_ext_spk_power_amp_on(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ mdm9615_ext_spk_power_amp_on(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ mdm9615_ext_spk_power_amp_on(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+
+ } else {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ mdm9615_ext_spk_power_amp_off(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ mdm9615_ext_spk_power_amp_off(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ mdm9615_ext_spk_power_amp_off(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ mdm9615_ext_spk_power_amp_off(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ }
+ return 0;
+}
+static int mdm9615_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm)
+{
+ pr_debug("%s: enable = %d\n", __func__, enable);
+ if (enable) {
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users != 1)
+ return 0;
+ if (IS_ERR(codec_clk)) {
+
+ pr_err("%s: Error setting Tabla MCLK\n", __func__);
+ clk_users--;
+ return -EINVAL;
+ }
+ clk_set_rate(codec_clk, TABLA_EXT_CLK_RATE);
+ clk_prepare_enable(codec_clk);
+ tabla_mclk_enable(codec, 1, dapm);
+ } else {
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users == 0)
+ return 0;
+ clk_users--;
+ if (!clk_users) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+ tabla_mclk_enable(codec, 0, dapm);
+ clk_disable_unprepare(codec_clk);
+ }
+ }
+ return 0;
+}
+
+static int mdm9615_mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return mdm9615_enable_codec_ext_clk(w->codec, 1, true);
+ case SND_SOC_DAPM_POST_PMD:
+ return mdm9615_enable_codec_ext_clk(w->codec, 0, true);
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget mdm9615_dapm_widgets[] = {
+
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ mdm9615_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Pos", mdm9615_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Neg", mdm9615_spkramp_event),
+
+ SND_SOC_DAPM_SPK("Ext Spk Top Pos", mdm9615_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Top Neg", mdm9615_spkramp_event),
+
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic2", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic3", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic4", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic5", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic6", NULL),
+
+};
+
+static const struct snd_soc_dapm_route common_audio_map[] = {
+
+ {"RX_BIAS", NULL, "MCLK"},
+ {"LDO_H", NULL, "MCLK"},
+
+ /* Speaker path */
+ {"Ext Spk Bottom Pos", NULL, "LINEOUT1"},
+ {"Ext Spk Bottom Neg", NULL, "LINEOUT3"},
+
+ {"Ext Spk Top Pos", NULL, "LINEOUT2"},
+ {"Ext Spk Top Neg", NULL, "LINEOUT4"},
+
+ /* Microphone path */
+ {"AMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Handset Mic"},
+
+ {"AMIC2", NULL, "MIC BIAS2 External"},
+ {"MIC BIAS2 External", NULL, "Headset Mic"},
+
+ /**
+ * AMIC3 and AMIC4 inputs are connected to ANC microphones
+ * These mics are biased differently on CDP and FLUID
+ * routing entries below are based on bias arrangement
+ * on FLUID.
+ */
+ {"AMIC3", NULL, "MIC BIAS3 Internal1"},
+ {"MIC BIAS3 Internal1", NULL, "ANCRight Headset Mic"},
+
+ {"AMIC4", NULL, "MIC BIAS1 Internal2"},
+ {"MIC BIAS1 Internal2", NULL, "ANCLeft Headset Mic"},
+
+ {"HEADPHONE", NULL, "LDO_H"},
+
+ /**
+ * The digital Mic routes are setup considering
+ * fluid as default device.
+ */
+
+ /**
+ * Digital Mic1. Front Bottom left Digital Mic on Fluid and MTP.
+ * Digital Mic GM5 on CDP mainboard.
+ * Conncted to DMIC2 Input on Tabla codec.
+ */
+ {"DMIC2", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic1"},
+
+ /**
+ * Digital Mic2. Front Bottom right Digital Mic on Fluid and MTP.
+ * Digital Mic GM6 on CDP mainboard.
+ * Conncted to DMIC1 Input on Tabla codec.
+ */
+ {"DMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic2"},
+
+ /**
+ * Digital Mic3. Back Bottom Digital Mic on Fluid.
+ * Digital Mic GM1 on CDP mainboard.
+ * Conncted to DMIC4 Input on Tabla codec.
+ */
+ {"DMIC4", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic3"},
+
+ /**
+ * Digital Mic4. Back top Digital Mic on Fluid.
+ * Digital Mic GM2 on CDP mainboard.
+ * Conncted to DMIC3 Input on Tabla codec.
+ */
+ {"DMIC3", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic4"},
+
+ /**
+ * Digital Mic5. Front top Digital Mic on Fluid.
+ * Digital Mic GM3 on CDP mainboard.
+ * Conncted to DMIC5 Input on Tabla codec.
+ */
+ {"DMIC5", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic5"},
+
+ /* Tabla digital Mic6 - back bottom digital Mic on Liquid and
+ * bottom mic on CDP. FLUID/MTP do not have dmic6 installed.
+ */
+ {"DMIC6", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic6"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *slim0_rx_ch_text[] = {"One", "Two"};
+static const char *slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
+
+static const struct soc_enum mdm9615_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
+ SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
+};
+
+static const char *btsco_rate_text[] = {"8000", "16000"};
+static const struct soc_enum mdm9615_btsco_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, btsco_rate_text),
+};
+
+static int mdm9615_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: mdm9615_slim_0_rx_ch = %d\n", __func__,
+ mdm9615_slim_0_rx_ch);
+ ucontrol->value.integer.value[0] = mdm9615_slim_0_rx_ch - 1;
+ return 0;
+}
+
+static int mdm9615_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ mdm9615_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: mdm9615_slim_0_rx_ch = %d\n", __func__,
+ mdm9615_slim_0_rx_ch);
+ return 1;
+}
+
+static int mdm9615_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: mdm9615_slim_0_tx_ch = %d\n", __func__,
+ mdm9615_slim_0_tx_ch);
+ ucontrol->value.integer.value[0] = mdm9615_slim_0_tx_ch - 1;
+ return 0;
+}
+
+static int mdm9615_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ mdm9615_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: mdm9615_slim_0_tx_ch = %d\n", __func__,
+ mdm9615_slim_0_tx_ch);
+ return 1;
+}
+
+static int mdm9615_btsco_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: mdm9615_btsco_rate = %d", __func__, mdm9615_btsco_rate);
+ ucontrol->value.integer.value[0] = mdm9615_btsco_rate;
+ return 0;
+}
+
+static int mdm9615_btsco_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ mdm9615_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ case 1:
+ mdm9615_btsco_rate = BTSCO_RATE_16KHZ;
+ break;
+ default:
+ mdm9615_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ }
+ pr_debug("%s: mdm9615_btsco_rate = %d\n", __func__, mdm9615_btsco_rate);
+ return 0;
+}
+
+static const struct snd_kcontrol_new tabla_mdm9615_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", mdm9615_enum[0], mdm9615_get_spk,
+ mdm9615_set_spk),
+ SOC_ENUM_EXT("SLIM_0_RX Channels", mdm9615_enum[1],
+ mdm9615_slim_0_rx_ch_get, mdm9615_slim_0_rx_ch_put),
+ SOC_ENUM_EXT("SLIM_0_TX Channels", mdm9615_enum[2],
+ mdm9615_slim_0_tx_ch_get, mdm9615_slim_0_tx_ch_put),
+ SOC_ENUM_EXT("Internal BTSCO SampleRate", mdm9615_btsco_enum[0],
+ mdm9615_btsco_rate_get, mdm9615_btsco_rate_put),
+};
+
+static void *def_tabla_mbhc_cal(void)
+{
+ void *tabla_cal;
+ struct tabla_mbhc_btn_detect_cfg *btn_cfg;
+ u16 *btn_low, *btn_high;
+ u8 *n_ready, *n_cic, *gain;
+
+ tabla_cal = kzalloc(TABLA_MBHC_CAL_SIZE(TABLA_MBHC_DEF_BUTTONS,
+ TABLA_MBHC_DEF_RLOADS),
+ GFP_KERNEL);
+ if (!tabla_cal) {
+ pr_err("%s: out of memory\n", __func__);
+ return NULL;
+ }
+
+#define S(X, Y) ((TABLA_MBHC_CAL_GENERAL_PTR(tabla_cal)->X) = (Y))
+ S(t_ldoh, 100);
+ S(t_bg_fast_settle, 100);
+ S(t_shutdown_plug_rem, 255);
+ S(mbhc_nsa, 4);
+ S(mbhc_navg, 4);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_DET_PTR(tabla_cal)->X) = (Y))
+ S(mic_current, TABLA_PID_MIC_5_UA);
+ S(hph_current, TABLA_PID_MIC_5_UA);
+ S(t_mic_pid, 100);
+ S(t_ins_complete, 250);
+ S(t_ins_retry, 200);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla_cal)->X) = (Y))
+ S(v_no_mic, 30);
+ S(v_hs_max, 1550);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal)->X) = (Y))
+ S(c[0], 62);
+ S(c[1], 124);
+ S(nc, 1);
+ S(n_meas, 3);
+ S(mbhc_nsc, 11);
+ S(n_btn_meas, 1);
+ S(n_btn_con, 2);
+ S(num_btn, TABLA_MBHC_DEF_BUTTONS);
+ S(v_btn_press_delta_sta, 100);
+ S(v_btn_press_delta_cic, 50);
+#undef S
+ btn_cfg = TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal);
+ btn_low = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_LOW);
+ btn_high = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_HIGH);
+ btn_low[0] = -50;
+ btn_high[0] = 10;
+ btn_low[1] = 11;
+ btn_high[1] = 38;
+ btn_low[2] = 39;
+ btn_high[2] = 64;
+ btn_low[3] = 65;
+ btn_high[3] = 91;
+ btn_low[4] = 92;
+ btn_high[4] = 115;
+ btn_low[5] = 116;
+ btn_high[5] = 141;
+ btn_low[6] = 142;
+ btn_high[6] = 163;
+ btn_low[7] = 164;
+ btn_high[7] = 250;
+ n_ready = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_READY);
+ n_ready[0] = 48;
+ n_ready[1] = 38;
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_CIC);
+ n_cic[0] = 60;
+ n_cic[1] = 47;
+ gain = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_GAIN);
+ gain[0] = 11;
+ gain[1] = 9;
+
+ return tabla_cal;
+}
+
+static int msm9615_i2s_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm9615_i2s_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm9615_i2s_spk_control = ucontrol->value.integer.value[0];
+ mdm9615_ext_control(codec);
+ return 1;
+}
+
+static int mdm9615_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ unsigned int rx_ch_cnt = 0, tx_ch_cnt = 0;
+
+ pr_debug("%s: ch=%d\n", __func__,
+ mdm9615_slim_0_rx_ch);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
+ mdm9615_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai, 0, 0,
+ mdm9615_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+ } else {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(cpu_dai,
+ mdm9615_slim_0_tx_ch, tx_ch, 0 , 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai,
+ mdm9615_slim_0_tx_ch, tx_ch, 0, 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+ }
+end:
+ return ret;
+}
+
+static int msm9615_i2s_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm9615_i2s_rx_ch = %d\n", __func__,
+ msm9615_i2s_rx_ch);
+ ucontrol->value.integer.value[0] = msm9615_i2s_rx_ch - 1;
+ return 0;
+}
+
+static int msm9615_i2s_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm9615_i2s_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm9615_i2s_rx_ch = %d\n", __func__,
+ msm9615_i2s_rx_ch);
+ return 1;
+}
+
+static int msm9615_i2s_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm9615_i2s_tx_ch = %d\n", __func__,
+ msm9615_i2s_tx_ch);
+ ucontrol->value.integer.value[0] = msm9615_i2s_tx_ch - 1;
+ return 0;
+}
+
+static int msm9615_i2s_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm9615_i2s_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm9615_i2s_tx_ch = %d\n", __func__,
+ msm9615_i2s_tx_ch);
+ return 1;
+}
+
+static int msm9615_i2s_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm9615_spk_control = %d", __func__, mdm9615_spk_control);
+ ucontrol->value.integer.value[0] = msm9615_i2s_spk_control;
+ return 0;
+}
+
+static const struct snd_kcontrol_new tabla_msm9615_i2s_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", mdm9615_enum[0], msm9615_i2s_get_spk,
+ msm9615_i2s_set_spk),
+ SOC_ENUM_EXT("PRI_RX Channels", mdm9615_enum[1],
+ msm9615_i2s_rx_ch_get, msm9615_i2s_rx_ch_put),
+ SOC_ENUM_EXT("PRI_TX Channels", mdm9615_enum[2],
+ msm9615_i2s_tx_ch_get, msm9615_i2s_tx_ch_put),
+};
+
+static int msm9615_i2s_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ snd_soc_dapm_new_controls(dapm, mdm9615_dapm_widgets,
+ ARRAY_SIZE(mdm9615_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, common_audio_map,
+ ARRAY_SIZE(common_audio_map));
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+
+ snd_soc_dapm_sync(dapm);
+
+ err = snd_soc_jack_new(codec, "Headset Jack",
+ (SND_JACK_HEADSET | SND_JACK_OC_HPHL|
+ SND_JACK_OC_HPHR), &hs_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+ err = snd_soc_jack_new(codec, "Button Jack",
+ TABLA_JACK_BUTTON_MASK, &button_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+ codec_clk = clk_get(cpu_dai->dev, "osr_clk");
+ err = tabla_hs_detect(codec, &mbhc_cfg);
+ return err;
+}
+
+static int msm9615_i2s_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm9615_i2s_rx_ch;
+
+ return 0;
+}
+
+static int msm9615_i2s_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ rate->min = rate->max = 48000;
+
+ channels->min = channels->max = msm9615_i2s_tx_ch;
+
+ return 0;
+}
+
+static int mdm9615_i2s_free_gpios(u8 i2s_intf, u8 i2s_dir)
+{
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ if (i2s_intf == MSM_INTF_PRIM) {
+ if (i2s_dir == MSM_DIR_RX)
+ gpio_free(GPIO_PRIM_I2S_DOUT);
+ if (i2s_dir == MSM_DIR_TX)
+ gpio_free(GPIO_PRIM_I2S_DIN);
+ if (pintf->intf_status[i2s_intf][MSM_DIR_TX] == 0 &&
+ pintf->intf_status[i2s_intf][MSM_DIR_RX] == 0) {
+ gpio_free(GPIO_PRIM_I2S_SCK);
+ gpio_free(GPIO_PRIM_I2S_WS);
+ }
+ } else if (i2s_intf == MSM_INTF_SECN) {
+ if (i2s_dir == MSM_DIR_RX)
+ gpio_free(GPIO_SEC_I2S_DOUT);
+ if (i2s_dir == MSM_DIR_TX)
+ gpio_free(GPIO_SEC_I2S_DIN);
+ if (pintf->intf_status[i2s_intf][MSM_DIR_TX] == 0 &&
+ pintf->intf_status[i2s_intf][MSM_DIR_RX] == 0) {
+ gpio_free(GPIO_SEC_I2S_WS);
+ gpio_free(GPIO_SEC_I2S_SCK);
+ }
+ }
+ return 0;
+}
+
+int msm9615_i2s_intf_dir_sel(const char *cpu_dai_name,
+ u8 *i2s_intf, u8 *i2s_dir)
+{
+ int ret = 0;
+ if (i2s_intf == NULL || i2s_dir == NULL || cpu_dai_name == NULL) {
+ ret = 1;
+ goto err;
+ }
+ if (!strncmp(cpu_dai_name, "msm-dai-q6.0", 12)) {
+ *i2s_intf = MSM_INTF_PRIM;
+ *i2s_dir = MSM_DIR_RX;
+ } else if (!strncmp(cpu_dai_name, "msm-dai-q6.1", 12)) {
+ *i2s_intf = MSM_INTF_PRIM;
+ *i2s_dir = MSM_DIR_TX;
+ } else if (!strncmp(cpu_dai_name, "msm-dai-q6.4", 12)) {
+ *i2s_intf = MSM_INTF_SECN;
+ *i2s_dir = MSM_DIR_RX;
+ } else if (!strncmp(cpu_dai_name, "msm-dai-q6.5", 12)) {
+ *i2s_intf = MSM_INTF_SECN;
+ *i2s_dir = MSM_DIR_TX;
+ } else {
+ pr_err("Error in I2S cpu dai name\n");
+ ret = 1;
+ }
+err:
+ return ret;
+}
+
+int msm9615_enable_i2s_gpio(u8 i2s_intf, u8 i2s_dir)
+{
+ u8 ret = 0;
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ if (i2s_intf == MSM_INTF_PRIM) {
+ if (i2s_dir == MSM_DIR_TX) {
+ ret = gpio_request(GPIO_PRIM_I2S_DIN, "I2S_PRIM_DIN");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_PRIM_I2S_DIN);
+ goto err;
+ }
+ } else if (i2s_dir == MSM_DIR_RX) {
+ ret = gpio_request(GPIO_PRIM_I2S_DOUT,
+ "I2S_PRIM_DOUT");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_PRIM_I2S_DOUT);
+ goto err;
+ }
+ } else if (pintf->intf_status[i2s_intf][MSM_DIR_TX] == 0 &&
+ pintf->intf_status[i2s_intf][MSM_DIR_RX] == 0) {
+ ret = gpio_request(GPIO_PRIM_I2S_SCK, "I2S_PRIM_SCK");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_PRIM_I2S_SCK);
+ goto err;
+ }
+ ret = gpio_request(GPIO_PRIM_I2S_WS, "I2S_PRIM_WS");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_PRIM_I2S_WS);
+ goto err;
+ }
+ }
+ } else if (i2s_intf == MSM_INTF_SECN) {
+ if (i2s_dir == MSM_DIR_RX) {
+ ret = gpio_request(GPIO_SEC_I2S_DOUT, "I2S_SEC_DOUT");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_SEC_I2S_DOUT);
+ goto err;
+ }
+ } else if (i2s_dir == MSM_DIR_TX) {
+ ret = gpio_request(GPIO_SEC_I2S_DIN, "I2S_SEC_DIN");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_SEC_I2S_DIN);
+ goto err;
+ }
+ } else if (pintf->intf_status[i2s_intf][MSM_DIR_TX] == 0 &&
+ pintf->intf_status[i2s_intf][MSM_DIR_RX] == 0) {
+ ret = gpio_request(GPIO_SEC_I2S_SCK, "I2S_SEC_SCK");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_SEC_I2S_SCK);
+ goto err;
+ }
+ ret = gpio_request(GPIO_SEC_I2S_WS, "I2S_SEC_WS");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__, GPIO_SEC_I2S_WS);
+ goto err;
+ }
+ }
+ }
+err:
+ return ret;
+}
+
+static int msm9615_set_i2s_osr_bit_clk(struct snd_soc_dai *cpu_dai,
+ u8 i2s_intf, u8 i2s_dir,
+ enum msm9x15_set_i2s_clk enable)
+{
+
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ struct msm_i2s_clk *pclk = &pintf->prim_clk;
+ struct msm_clk *clk_ctl = &pclk->rx_clk;
+ u8 ret = 0;
+ pr_debug("Dev name %s Intf =%d, Dir = %d, Enable=%d\n",
+ cpu_dai->name, i2s_intf, i2s_dir, enable);
+ if (i2s_intf == MSM_INTF_PRIM)
+ pclk = &pintf->prim_clk;
+ else if (i2s_intf == MSM_INTF_SECN)
+ pclk = &pintf->sec_clk;
+
+ if (i2s_dir == MSM_DIR_TX)
+ clk_ctl = &pclk->tx_clk;
+ else if (i2s_dir == MSM_DIR_RX)
+ clk_ctl = &pclk->rx_clk;
+
+ if (enable == MSM_I2S_CLK_SET_TRUE ||
+ enable == MSM_I2S_CLK_SET_RATE0) {
+ if (clk_ctl->clk_enable != 0) {
+ pr_info("%s: I2S Clk is already enabled"
+ "clk users %d\n", __func__,
+ clk_ctl->clk_enable);
+ ret = 0;
+ goto err;
+ }
+ clk_ctl->osr_clk = clk_get(cpu_dai->dev, "osr_clk");
+ if (IS_ERR(clk_ctl->osr_clk)) {
+ pr_err("%s: Fail to get OSR CLK\n", __func__);
+ ret = -EINVAL;
+ goto err;
+ }
+ ret = clk_prepare(clk_ctl->osr_clk);
+ if (ret != 0) {
+ pr_err("Unable to prepare i2s_spkr_osr_clk\n");
+ goto err;
+ }
+ clk_set_rate(clk_ctl->osr_clk, TABLA_EXT_CLK_RATE);
+ ret = clk_enable(clk_ctl->osr_clk);
+ if (ret != 0) {
+ pr_err("Fail to enable i2s_spkr_osr_clk\n");
+ clk_unprepare(clk_ctl->osr_clk);
+ goto err;
+ }
+ clk_ctl->bit_clk = clk_get(cpu_dai->dev, "bit_clk");
+ if (IS_ERR(clk_ctl->bit_clk)) {
+ pr_err("Fail to get i2s_spkr_bit_clk\n");
+ clk_disable(clk_ctl->osr_clk);
+ clk_unprepare(clk_ctl->osr_clk);
+ clk_put(clk_ctl->osr_clk);
+ ret = -EINVAL;
+ goto err;
+ }
+ ret = clk_prepare(clk_ctl->bit_clk);
+ if (ret != 0) {
+ clk_disable(clk_ctl->osr_clk);
+ clk_unprepare(clk_ctl->osr_clk);
+ clk_put(clk_ctl->osr_clk);
+ pr_err("Fail to prepare i2s_spkr_osr_clk\n");
+ goto err;
+ }
+ if (enable == MSM_I2S_CLK_SET_RATE0)
+ clk_set_rate(clk_ctl->bit_clk, 0);
+ else
+ clk_set_rate(clk_ctl->bit_clk, 8);
+ ret = clk_enable(clk_ctl->bit_clk);
+ if (ret != 0) {
+ clk_disable(clk_ctl->osr_clk);
+ clk_unprepare(clk_ctl->osr_clk);
+ clk_put(clk_ctl->osr_clk);
+ clk_unprepare(clk_ctl->bit_clk);
+ pr_err("Unable to enable i2s_spkr_osr_clk\n");
+ goto err;
+ }
+ clk_ctl->clk_enable++;
+ } else if (enable == MSM_I2S_CLK_SET_FALSE &&
+ clk_ctl->clk_enable != 0) {
+ clk_disable(clk_ctl->osr_clk);
+ clk_disable(clk_ctl->bit_clk);
+ clk_unprepare(clk_ctl->osr_clk);
+ clk_unprepare(clk_ctl->bit_clk);
+ clk_put(clk_ctl->bit_clk);
+ clk_put(clk_ctl->osr_clk);
+ clk_ctl->bit_clk = NULL;
+ clk_ctl->osr_clk = NULL;
+ clk_ctl->clk_enable--;
+ ret = 0;
+ }
+err:
+ return ret;
+}
+
+void msm9615_config_i2s_sif_mux(u8 value)
+{
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ sif_shadow = 0x00000;
+ sif_shadow = (sif_shadow & LPASS_SIF_MUX_CTL_PRI_MUX_SEL_BMSK) |
+ (value << LPASS_SIF_MUX_CTL_PRI_MUX_SEL_SHFT);
+ iowrite32(sif_shadow, pintf->sif_virt_addr);
+ /* Dont read SIF register. Device crashes. */
+ pr_debug("%s() SIF Reg = 0x%x\n", __func__, sif_shadow);
+}
+
+void msm9615_config_i2s_spare_mux(u8 value, u8 i2s_intf)
+{
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ if (i2s_intf == MSM_INTF_PRIM) {
+ /* Configure Primary SIF */
+ spare_shadow = (spare_shadow & LPAIF_SPARE_MUX_CTL_PRI_MUX_SEL_BMSK
+ ) | (value << LPAIF_SPARE_MUX_CTL_PRI_MUX_SEL_SHFT);
+ }
+ if (i2s_intf == MSM_INTF_SECN) {
+ /*Secondary interface configuration*/
+ spare_shadow = (spare_shadow & LPAIF_SPARE_MUX_CTL_SEC_MUX_SEL_BMSK
+ ) | (value << LPAIF_SPARE_MUX_CTL_SEC_MUX_SEL_SHFT);
+ }
+ iowrite32(spare_shadow, pintf->spare_virt_addr);
+ /* Dont read SPARE register. Device crashes. */
+ pr_debug("%s( ): SPARE Reg =0x%x\n", __func__, spare_shadow);
+}
+
+static int msm9615_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int rate = params_rate(params);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ struct msm_i2s_clk *pclk = &pintf->prim_clk;
+ struct msm_clk *clk_ctl = &pclk->rx_clk;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int bit_clk_set = 0;
+ u8 i2s_intf, i2s_dir;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (!msm9615_i2s_intf_dir_sel(cpu_dai->name,
+ &i2s_intf, &i2s_dir)) {
+ bit_clk_set = TABLA_EXT_CLK_RATE /
+ (rate * 2 * NO_OF_BITS_PER_SAMPLE);
+ if (bit_clk_set != 8) {
+ if (i2s_intf == MSM_INTF_PRIM)
+ pclk = &pintf->prim_clk;
+ else if (i2s_intf == MSM_INTF_SECN)
+ pclk = &pintf->sec_clk;
+ clk_ctl = &pclk->rx_clk;
+ pr_debug("%s( ): New rate = %d",
+ __func__, bit_clk_set);
+ clk_set_rate(clk_ctl->bit_clk, bit_clk_set);
+ }
+ }
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ bit_clk_set = I2S_MIC_SCLK_RATE / (rate * 2 *
+ NO_OF_BITS_PER_SAMPLE);
+ /* Not required to modify TX rate.
+ * Speaker clock are looped back
+ * to Mic.
+ */
+ }
+ return 1;
+}
+
+static int msm9615_i2s_startup(struct snd_pcm_substream *substream)
+{
+ u8 ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ u8 i2s_intf, i2s_dir;
+ if (!msm9615_i2s_intf_dir_sel(cpu_dai->name, &i2s_intf, &i2s_dir)) {
+ pr_debug("%s( ): cpu name = %s intf =%d dir = %d\n",
+ __func__, cpu_dai->name, i2s_intf, i2s_dir);
+ pr_debug("%s( ): Enable status Rx =%d Tx = %d\n", __func__,
+ pintf->intf_status[i2s_intf][MSM_DIR_RX],
+ pintf->intf_status[i2s_intf][MSM_DIR_TX]);
+ msm9615_enable_i2s_gpio(i2s_intf, i2s_dir);
+ if (i2s_dir == MSM_DIR_TX) {
+ if (pintf->intf_status[i2s_intf][MSM_DIR_RX] > 0) {
+ /* This means that Rx is enabled before */
+ ret = msm9615_set_i2s_osr_bit_clk(cpu_dai,
+ i2s_intf, i2s_dir,
+ MSM_I2S_CLK_SET_RATE0);
+ if (ret != 0) {
+ pr_err("%s: Fail enable I2S clock\n",
+ __func__);
+ return -EINVAL;
+ }
+ msm9615_config_i2s_sif_mux(
+ pintf->mux_ctl[MSM_DIR_BOTH].sifconfig);
+ msm9615_config_i2s_spare_mux(
+ pintf->mux_ctl[MSM_DIR_BOTH].spareconfig,
+ i2s_intf);
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ pr_err("set fmt cpu dai failed\n");
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ pr_err("set fmt codec dai failed\n");
+ } else if (pintf->intf_status[i2s_intf][i2s_dir] == 0) {
+ /* This means that Rx is
+ * not enabled before.
+ * only Tx will be used.
+ */
+ ret = msm9615_set_i2s_osr_bit_clk(cpu_dai,
+ i2s_intf, i2s_dir,
+ MSM_I2S_CLK_SET_TRUE);
+ if (ret != 0) {
+ pr_err("%s: Fail Tx I2S clock\n",
+ __func__);
+ return -EINVAL;
+ }
+ msm9615_config_i2s_sif_mux(
+ pintf->mux_ctl[MSM_DIR_TX].sifconfig);
+ msm9615_config_i2s_spare_mux(
+ pintf->mux_ctl[MSM_DIR_TX].spareconfig,
+ i2s_intf);
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ pr_err("set fmt cpu dai failed\n");
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ pr_err("set fmt codec dai failed\n");
+ }
+ } else if (i2s_dir == MSM_DIR_RX) {
+ if (pintf->intf_status[i2s_intf][MSM_DIR_TX] > 0) {
+ pr_err("%s: Error shutdown Tx first\n",
+ __func__);
+ return -EINVAL;
+ } else if (pintf->intf_status[i2s_intf][i2s_dir]
+ == 0) {
+ ret = msm9615_set_i2s_osr_bit_clk(cpu_dai,
+ i2s_intf, i2s_dir,
+ MSM_I2S_CLK_SET_TRUE);
+ if (ret != 0) {
+ pr_err("%s: Fail Rx I2S clock\n",
+ __func__);
+ return -EINVAL;
+ }
+ msm9615_config_i2s_sif_mux(
+ pintf->mux_ctl[MSM_DIR_RX].sifconfig);
+ msm9615_config_i2s_spare_mux(
+ pintf->mux_ctl[MSM_DIR_RX].spareconfig,
+ i2s_intf);
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ pr_err("set fmt cpu dai failed\n");
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ pr_err("set fmt codec dai failed\n");
+ }
+ }
+ pintf->intf_status[i2s_intf][i2s_dir]++;
+ } else {
+ pr_err("%s: Err in i2s_intf_dir_sel\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("Exit %s() Enable status Rx =%d Tx = %d\n", __func__,
+ pintf->intf_status[i2s_intf][MSM_DIR_RX],
+ pintf->intf_status[i2s_intf][MSM_DIR_TX]);
+ return ret;
+}
+
+static void msm9615_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct msm_i2s_ctl *pintf = &msm9x15_i2s_ctl;
+ u8 i2s_intf = 0, i2s_dir = 0, ret = 0;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ pr_debug("%s( ): Enable status Rx =%d Tx = %d\n",
+ __func__, pintf->intf_status[i2s_intf][MSM_DIR_RX],
+ pintf->intf_status[i2s_intf][MSM_DIR_TX]);
+ if (!msm9615_i2s_intf_dir_sel(cpu_dai->name, &i2s_intf, &i2s_dir)) {
+ pr_debug("%s( ): intf =%d dir = %d\n", __func__,
+ i2s_intf, i2s_dir);
+ if (i2s_dir == MSM_DIR_RX)
+ if (pintf->intf_status[i2s_intf][MSM_DIR_TX] > 0)
+ pr_err("%s: Shutdown Tx First then by RX\n",
+ __func__);
+ ret = msm9615_set_i2s_osr_bit_clk(cpu_dai, i2s_intf, i2s_dir,
+ MSM_I2S_CLK_SET_FALSE);
+ if (ret != 0)
+ pr_err("%s: Cannot disable I2S clock\n",
+ __func__);
+ pintf->intf_status[i2s_intf][i2s_dir]--;
+ mdm9615_i2s_free_gpios(i2s_intf, i2s_dir);
+ }
+ pr_debug("%s( ): Enable status Rx =%d Tx = %d\n", __func__,
+ pintf->intf_status[i2s_intf][MSM_DIR_RX],
+ pintf->intf_status[i2s_intf][MSM_DIR_TX]);
+}
+
+static struct snd_soc_ops msm9615_i2s_be_ops = {
+ .startup = msm9615_i2s_startup,
+ .shutdown = msm9615_i2s_shutdown,
+ .hw_params = msm9615_i2s_hw_params,
+};
+
+static int mdm9615_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("%s(), dev_name%s\n", __func__, dev_name(cpu_dai->dev));
+
+ rtd->pmdown_time = 0;
+
+ snd_soc_dapm_new_controls(dapm, mdm9615_dapm_widgets,
+ ARRAY_SIZE(mdm9615_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, common_audio_map,
+ ARRAY_SIZE(common_audio_map));
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+
+ snd_soc_dapm_sync(dapm);
+
+ err = snd_soc_jack_new(codec, "Headset Jack",
+ (SND_JACK_HEADSET | SND_JACK_OC_HPHL |
+ SND_JACK_OC_HPHR),
+ &hs_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ err = snd_soc_jack_new(codec, "Button Jack",
+ TABLA_JACK_BUTTON_MASK, &button_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+ codec_clk = clk_get(cpu_dai->dev, "osr_clk");
+
+ err = tabla_hs_detect(codec, &mbhc_cfg);
+
+ return err;
+}
+
+static int mdm9615_slim_0_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = mdm9615_slim_0_rx_ch;
+
+ return 0;
+}
+
+static int mdm9615_slim_0_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = mdm9615_slim_0_tx_ch;
+
+ return 0;
+}
+
+static int mdm9615_btsco_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = mdm9615_btsco_rate;
+ channels->min = channels->max = mdm9615_btsco_ch;
+
+ return 0;
+}
+static int mdm9615_auxpcm_be_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* PCM only supports mono output with 8khz sample rate */
+ rate->min = rate->max = 8000;
+ channels->min = channels->max = 1;
+
+ return 0;
+}
+static int mdm9615_aux_pcm_get_gpios(void)
+{
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ ret = gpio_request(GPIO_AUX_PCM_DOUT, "AUX PCM DOUT");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DOUT",
+ __func__, GPIO_AUX_PCM_DOUT);
+ goto fail_dout;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_DIN, "AUX PCM DIN");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DIN",
+ __func__, GPIO_AUX_PCM_DIN);
+ goto fail_din;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_SYNC, "AUX PCM SYNC");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM SYNC",
+ __func__, GPIO_AUX_PCM_SYNC);
+ goto fail_sync;
+ }
+ ret = gpio_request(GPIO_AUX_PCM_CLK, "AUX PCM CLK");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM CLK",
+ __func__, GPIO_AUX_PCM_CLK);
+ goto fail_clk;
+ }
+
+ return 0;
+
+fail_clk:
+ gpio_free(GPIO_AUX_PCM_SYNC);
+fail_sync:
+ gpio_free(GPIO_AUX_PCM_DIN);
+fail_din:
+ gpio_free(GPIO_AUX_PCM_DOUT);
+fail_dout:
+
+ return ret;
+}
+
+static int mdm9615_aux_pcm_free_gpios(void)
+{
+ gpio_free(GPIO_AUX_PCM_DIN);
+ gpio_free(GPIO_AUX_PCM_DOUT);
+ gpio_free(GPIO_AUX_PCM_SYNC);
+ gpio_free(GPIO_AUX_PCM_CLK);
+
+ return 0;
+}
+
+static int mdm9615_startup(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+ return 0;
+}
+
+static int mdm9615_auxpcm_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ ret = mdm9615_aux_pcm_get_gpios();
+ if (ret < 0) {
+ pr_err("%s: Aux PCM GPIO request failed\n", __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static void mdm9615_auxpcm_shutdown(struct snd_pcm_substream *substream)
+{
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ mdm9615_aux_pcm_free_gpios();
+}
+
+static void mdm9615_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+}
+
+static struct snd_soc_ops mdm9615_be_ops = {
+ .startup = mdm9615_startup,
+ .hw_params = mdm9615_hw_params,
+ .shutdown = mdm9615_shutdown,
+};
+
+static struct snd_soc_ops mdm9615_auxpcm_be_ops = {
+ .startup = mdm9615_auxpcm_startup,
+ .shutdown = mdm9615_auxpcm_shutdown,
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link mdm9615_dai_common[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MDM9615 Media1",
+ .stream_name = "MultiMedia1",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MDM9615 Media2",
+ .stream_name = "MultiMedia2",
+ .cpu_dai_name = "MultiMedia2",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+ {
+ .name = "Circuit-Switch Voice",
+ .stream_name = "CS-Voice",
+ .cpu_dai_name = "CS-VOICE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .be_id = MSM_FRONTEND_DAI_CS_VOICE,
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "MSM VoIP",
+ .stream_name = "VoIP",
+ .cpu_dai_name = "VoIP",
+ .platform_name = "msm-voip-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .be_id = MSM_FRONTEND_DAI_VOIP,
+ },
+ /* Hostless PMC purpose */
+ {
+ .name = "SLIMBUS_0 Hostless",
+ .stream_name = "SLIMBUS_0 Hostless",
+ .cpu_dai_name = "SLIMBUS0_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ /* .be_id = do not care */
+ },
+ {
+ .name = "MSM AFE-PCM RX",
+ .stream_name = "AFE-PROXY RX",
+ .cpu_dai_name = "msm-dai-q6.241",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "MSM AFE-PCM TX",
+ .stream_name = "AFE-PROXY TX",
+ .cpu_dai_name = "msm-dai-q6.240",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "AUXPCM Hostless",
+ .stream_name = "AUXPCM Hostless",
+ .cpu_dai_name = "AUXPCM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "VoLTE",
+ .stream_name = "VoLTE",
+ .cpu_dai_name = "VoLTE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .be_id = MSM_FRONTEND_DAI_VOLTE,
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ },
+ /* Backend BT DAI Links */
+ {
+ .name = LPASS_BE_INT_BT_SCO_RX,
+ .stream_name = "Internal BT-SCO Playback",
+ .cpu_dai_name = "msm-dai-q6.12288",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ .be_hw_params_fixup = mdm9615_btsco_be_hw_params_fixup,
+ },
+ {
+ .name = LPASS_BE_INT_BT_SCO_TX,
+ .stream_name = "Internal BT-SCO Capture",
+ .cpu_dai_name = "msm-dai-q6.12289",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ .be_hw_params_fixup = mdm9615_btsco_be_hw_params_fixup,
+ },
+
+ /* Backend AFE DAI Links */
+ {
+ .name = LPASS_BE_AFE_PCM_RX,
+ .stream_name = "AFE Playback",
+ .cpu_dai_name = "msm-dai-q6.224",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_RX,
+ },
+ {
+ .name = LPASS_BE_AFE_PCM_TX,
+ .stream_name = "AFE Capture",
+ .cpu_dai_name = "msm-dai-q6.225",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_TX,
+ },
+ /* AUX PCM Backend DAI Links */
+ {
+ .name = LPASS_BE_AUXPCM_RX,
+ .stream_name = "AUX PCM Playback",
+ .cpu_dai_name = "msm-dai-q6.2",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_RX,
+ .be_hw_params_fixup = mdm9615_auxpcm_be_params_fixup,
+ .ops = &mdm9615_auxpcm_be_ops,
+ },
+ {
+ .name = LPASS_BE_AUXPCM_TX,
+ .stream_name = "AUX PCM Capture",
+ .cpu_dai_name = "msm-dai-q6.3",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_TX,
+ .be_hw_params_fixup = mdm9615_auxpcm_be_params_fixup,
+ },
+
+};
+
+static struct snd_soc_dai_link mdm9615_dai_i2s_tabla[] = {
+ /* Backend I2S DAI Links */
+ {
+ .name = LPASS_BE_PRI_I2S_RX,
+ .stream_name = "Primary I2S Playback",
+ .cpu_dai_name = "msm-dai-q6.0",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_i2s_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_PRI_I2S_RX,
+ .init = &msm9615_i2s_audrx_init,
+ .be_hw_params_fixup = msm9615_i2s_rx_be_hw_params_fixup,
+ .ops = &msm9615_i2s_be_ops,
+ },
+ {
+ .name = LPASS_BE_PRI_I2S_TX,
+ .stream_name = "Primary I2S Capture",
+ .cpu_dai_name = "msm-dai-q6.1",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_i2s_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_PRI_I2S_TX,
+ .be_hw_params_fixup = msm9615_i2s_tx_be_hw_params_fixup,
+ .ops = &msm9615_i2s_be_ops,
+ },
+};
+
+static struct snd_soc_dai_link mdm9615_dai_slimbus_tabla[] = {
+ /* Backend SlimBus DAI Links */
+ {
+ .name = LPASS_BE_SLIMBUS_0_RX,
+ .stream_name = "Slimbus Playback",
+ .cpu_dai_name = "msm-dai-q6.16384",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ .init = &mdm9615_audrx_init,
+ .be_hw_params_fixup = mdm9615_slim_0_rx_be_hw_params_fixup,
+ .ops = &mdm9615_be_ops,
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_0_TX,
+ .stream_name = "Slimbus Capture",
+ .cpu_dai_name = "msm-dai-q6.16385",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ .be_hw_params_fixup = mdm9615_slim_0_tx_be_hw_params_fixup,
+ .ops = &mdm9615_be_ops,
+ },
+};
+
+static struct snd_soc_dai_link mdm9615_i2s_dai[
+ ARRAY_SIZE(mdm9615_dai_common) +
+ ARRAY_SIZE(mdm9615_dai_i2s_tabla)];
+
+static struct snd_soc_dai_link mdm9615_slimbus_dai[
+ ARRAY_SIZE(mdm9615_dai_common) +
+ ARRAY_SIZE(mdm9615_dai_slimbus_tabla)];
+
+
+static struct snd_soc_card snd_soc_card_mdm9615 = {
+ .name = "mdm9615-tabla-snd-card",
+ .controls = tabla_mdm9615_controls,
+ .num_controls = ARRAY_SIZE(tabla_mdm9615_controls),
+};
+
+static struct platform_device *mdm9615_snd_device;
+
+static int mdm9615_configure_headset_mic_gpios(void)
+{
+ int ret;
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ ret = gpio_request(PM8018_GPIO_PM_TO_SYS(23), "AV_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8018_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+
+ ret = pm8xxx_gpio_config(PM8018_GPIO_PM_TO_SYS(23), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8018_GPIO_PM_TO_SYS(23));
+ else
+ gpio_direction_output(PM8018_GPIO_PM_TO_SYS(23), 0);
+
+ ret = gpio_request(PM8018_GPIO_PM_TO_SYS(35), "US_EURO_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8018_GPIO_PM_TO_SYS(35));
+ gpio_free(PM8018_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+ ret = pm8xxx_gpio_config(PM8018_GPIO_PM_TO_SYS(35), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8018_GPIO_PM_TO_SYS(35));
+ else
+ gpio_direction_output(PM8018_GPIO_PM_TO_SYS(35), 0);
+
+ return 0;
+}
+static void mdm9615_free_headset_mic_gpios(void)
+{
+ if (mdm9615_headset_gpios_configured) {
+ gpio_free(PM8018_GPIO_PM_TO_SYS(23));
+ gpio_free(PM8018_GPIO_PM_TO_SYS(35));
+ }
+}
+
+void __init install_codec_i2s_gpio(void)
+{
+ msm_gpiomux_install(msm9615_audio_prim_i2s_codec_configs,
+ ARRAY_SIZE(msm9615_audio_prim_i2s_codec_configs));
+}
+static int __init mdm9615_audio_init(void)
+{
+ int ret;
+
+ if (!cpu_is_msm9615()) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return -ENODEV ;
+ }
+
+ mbhc_cfg.calibration = def_tabla_mbhc_cal();
+ if (!mbhc_cfg.calibration) {
+ pr_err("Calibration data allocation failed\n");
+ return -ENOMEM;
+ }
+
+ mdm9615_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!mdm9615_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+ pr_err("%s: Interface Type = %d\n", __func__,
+ wcd9xxx_get_intf_type());
+ if (wcd9xxx_get_intf_type() == WCD9XXX_INTERFACE_TYPE_SLIMBUS) {
+ memcpy(mdm9615_slimbus_dai, mdm9615_dai_common,
+ sizeof(mdm9615_dai_common));
+ memcpy(mdm9615_slimbus_dai + ARRAY_SIZE(mdm9615_dai_common),
+ mdm9615_dai_slimbus_tabla,
+ sizeof(mdm9615_dai_slimbus_tabla));
+ snd_soc_card_mdm9615.dai_link = mdm9615_slimbus_dai;
+ snd_soc_card_mdm9615.num_links =
+ ARRAY_SIZE(mdm9615_slimbus_dai);
+ } else if (wcd9xxx_get_intf_type() == WCD9XXX_INTERFACE_TYPE_I2C) {
+ install_codec_i2s_gpio();
+ memcpy(mdm9615_i2s_dai, mdm9615_dai_common,
+ sizeof(mdm9615_dai_common));
+ memcpy(mdm9615_i2s_dai + ARRAY_SIZE(mdm9615_dai_common),
+ mdm9615_dai_i2s_tabla,
+ sizeof(mdm9615_dai_i2s_tabla));
+ snd_soc_card_mdm9615.dai_link = mdm9615_i2s_dai;
+ snd_soc_card_mdm9615.num_links =
+ ARRAY_SIZE(mdm9615_i2s_dai);
+ }
+ platform_set_drvdata(mdm9615_snd_device, &snd_soc_card_mdm9615);
+ ret = platform_device_add(mdm9615_snd_device);
+ if (ret) {
+ platform_device_put(mdm9615_snd_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+ if (mdm9615_configure_headset_mic_gpios()) {
+ pr_err("%s Fail to configure headset mic gpios\n", __func__);
+ mdm9615_headset_gpios_configured = 0;
+ } else
+ mdm9615_headset_gpios_configured = 1;
+
+ msm9x15_i2s_ctl.sif_virt_addr = ioremap(LPASS_SIF_MUX_ADDR, 4);
+ msm9x15_i2s_ctl.spare_virt_addr = ioremap(LPAIF_SPARE_ADDR, 4);
+
+ return ret;
+
+}
+module_init(mdm9615_audio_init);
+
+static void __exit mdm9615_audio_exit(void)
+{
+ if (!cpu_is_msm9615()) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return ;
+ }
+ mdm9615_free_headset_mic_gpios();
+ platform_device_unregister(mdm9615_snd_device);
+ kfree(mbhc_cfg.calibration);
+ iounmap(msm9x15_i2s_ctl.sif_virt_addr);
+ iounmap(msm9x15_i2s_ctl.spare_virt_addr);
+
+}
+module_exit(mdm9615_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC MDM9615");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/mpq8064.c b/sound/soc/msm/mpq8064.c
new file mode 100644
index 0000000..0995300
--- /dev/null
+++ b/sound/soc/msm/mpq8064.c
@@ -0,0 +1,1470 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+#include <asm/mach-types.h>
+#include <mach/socinfo.h>
+#include "msm-pcm-routing.h"
+#include "../codecs/wcd9310.h"
+
+/* 8064 machine driver */
+
+#define PM8921_GPIO_BASE NR_GPIO_IRQS
+#define PM8921_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio - 1 + PM8921_GPIO_BASE)
+
+#define MPQ8064_SPK_ON 1
+#define MPQ8064_SPK_OFF 0
+
+#define MSM_SLIM_0_RX_MAX_CHANNELS 2
+#define MSM_SLIM_0_TX_MAX_CHANNELS 4
+
+#define BOTTOM_SPK_AMP_POS 0x1
+#define BOTTOM_SPK_AMP_NEG 0x2
+#define TOP_SPK_AMP_POS 0x4
+#define TOP_SPK_AMP_NEG 0x8
+
+#define TABLA_EXT_CLK_RATE 12288000
+
+#define TABLA_MBHC_DEF_BUTTONS 8
+#define TABLA_MBHC_DEF_RLOADS 5
+
+#define GPIO_SEC_I2S_RX_SCK 47
+#define GPIO_SEC_I2S_RX_WS 48
+#define GPIO_SEC_I2S_RX_DOUT 49
+#define GPIO_SEC_I2S_RX_MCLK 50
+#define I2S_MCLK_RATE 1536000
+
+#define GPIO_MI2S_WS 27
+#define GPIO_MI2S_SCLK 28
+#define GPIO_MI2S_DOUT3 29
+#define GPIO_MI2S_DOUT2 30
+#define GPIO_MI2S_DOUT1 31
+#define GPIO_MI2S_DOUT0 32
+#define GPIO_MI2S_MCLK 33
+
+static struct clk *sec_i2s_rx_osr_clk;
+static struct clk *sec_i2s_rx_bit_clk;
+
+struct request_gpio {
+ unsigned gpio_no;
+ char *gpio_name;
+};
+
+static struct request_gpio sec_i2s_rx_gpio[] = {
+ {
+ .gpio_no = GPIO_SEC_I2S_RX_MCLK,
+ .gpio_name = "SEC_I2S_RX_MCLK",
+ },
+ {
+ .gpio_no = GPIO_SEC_I2S_RX_SCK,
+ .gpio_name = "SEC_I2S_RX_SCK",
+ },
+ {
+ .gpio_no = GPIO_SEC_I2S_RX_WS,
+ .gpio_name = "SEC_I2S_RX_WS",
+ },
+ {
+ .gpio_no = GPIO_SEC_I2S_RX_DOUT,
+ .gpio_name = "SEC_I2S_RX_DOUT",
+ },
+};
+
+static struct request_gpio mi2s_gpio[] = {
+ {
+ .gpio_no = GPIO_MI2S_WS,
+ .gpio_name = "MI2S_WS",
+ },
+ {
+ .gpio_no = GPIO_MI2S_SCLK,
+ .gpio_name = "MI2S_SCLK",
+ },
+ {
+ .gpio_no = GPIO_MI2S_DOUT3,
+ .gpio_name = "MI2S_DOUT3",
+ },
+ {
+ .gpio_no = GPIO_MI2S_DOUT2,
+ .gpio_name = "MI2S_DOUT2",
+ },
+ {
+ .gpio_no = GPIO_MI2S_DOUT1,
+ .gpio_name = "MI2S_DOUT1",
+ },
+ {
+ .gpio_no = GPIO_MI2S_DOUT0,
+ .gpio_name = "MI2S_DOUT0",
+ },
+ {
+ .gpio_no = GPIO_MI2S_MCLK,
+ .gpio_name = "MI2S_MCLK",
+ },
+};
+
+static struct clk *mi2s_bit_clk;
+
+
+
+static u32 top_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(18);
+static u32 bottom_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(19);
+static int msm_spk_control;
+static int msm_ext_bottom_spk_pamp;
+static int msm_ext_top_spk_pamp;
+static int msm_slim_0_rx_ch = 1;
+static int msm_slim_0_tx_ch = 1;
+static int msm_hdmi_rx_ch = 2;
+
+static struct clk *codec_clk;
+static int clk_users;
+
+static int msm_headset_gpios_configured;
+
+static struct snd_soc_jack hs_jack;
+static struct snd_soc_jack button_jack;
+
+static int msm_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm);
+
+static struct tabla_mbhc_config mbhc_cfg = {
+ .headset_jack = &hs_jack,
+ .button_jack = &button_jack,
+ .read_fw_bin = false,
+ .calibration = NULL,
+ .micbias = TABLA_MICBIAS2,
+ .mclk_cb_fn = msm_enable_codec_ext_clk,
+ .mclk_rate = TABLA_EXT_CLK_RATE,
+ .gpio = 0, /* MBHC GPIO is not configured */
+ .gpio_irq = 0,
+ .gpio_level_insert = 1,
+};
+
+static void msm_enable_ext_spk_amp_gpio(u32 spk_amp_gpio)
+{
+ int ret = 0;
+
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .
+ function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ if (spk_amp_gpio == bottom_spk_pamp_gpio) {
+
+ ret = gpio_request(bottom_spk_pamp_gpio, "BOTTOM_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting BOTTOM SPK AMP GPIO %u\n",
+ __func__, bottom_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(bottom_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Bottom Spk Ampl"
+ " gpio %u\n", __func__, bottom_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Bottom spkr amp gpio\n", __func__);
+ gpio_direction_output(bottom_spk_pamp_gpio, 1);
+ }
+
+ } else if (spk_amp_gpio == top_spk_pamp_gpio) {
+
+ ret = gpio_request(top_spk_pamp_gpio, "TOP_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting GPIO %d\n", __func__,
+ top_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(top_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Top Spk Ampl"
+ " gpio %u\n", __func__, top_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Top spkr amp gpio\n", __func__);
+ gpio_direction_output(top_spk_pamp_gpio, 1);
+ }
+ } else {
+ pr_err("%s: ERROR : Invalid External Speaker Ampl GPIO."
+ " gpio = %u\n", __func__, spk_amp_gpio);
+ return;
+ }
+}
+
+static void msm_ext_spk_power_amp_on(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Bottom Speaker Ampl already "
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm_ext_bottom_spk_pamp |= spk;
+
+ if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ msm_enable_ext_spk_amp_gpio(bottom_spk_pamp_gpio);
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Bottom Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Top Speaker Ampl already"
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm_ext_top_spk_pamp |= spk;
+
+ if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ msm_enable_ext_spk_amp_gpio(top_spk_pamp_gpio);
+ pr_debug("%s: sleeping 4 ms after turning on "
+ " external Top Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm_ext_spk_power_amp_off(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if (!msm_ext_bottom_spk_pamp)
+ return;
+
+ gpio_direction_output(bottom_spk_pamp_gpio, 0);
+ gpio_free(bottom_spk_pamp_gpio);
+ msm_ext_bottom_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Bottom"
+ " Speaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if (!msm_ext_top_spk_pamp)
+ return;
+
+ gpio_direction_output(top_spk_pamp_gpio, 0);
+ gpio_free(top_spk_pamp_gpio);
+ msm_ext_top_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Top"
+ " Spkaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+ } else {
+
+ pr_err("%s: ERROR : Invalid Ext Spk Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm_ext_control(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
+ if (msm_spk_control == MPQ8064_SPK_ON) {
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+ } else {
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Neg");
+ }
+
+ snd_soc_dapm_sync(dapm);
+}
+
+static int msm_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
+ ucontrol->value.integer.value[0] = msm_spk_control;
+ return 0;
+}
+static int msm_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm_spk_control = ucontrol->value.integer.value[0];
+ msm_ext_control(codec);
+ return 1;
+}
+static int msm_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm_ext_spk_power_amp_on(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm_ext_spk_power_amp_on(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+
+ } else {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm_ext_spk_power_amp_off(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm_ext_spk_power_amp_off(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ }
+ return 0;
+}
+
+static int msm_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm)
+{
+ pr_debug("%s: enable = %d\n", __func__, enable);
+ if (enable) {
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users != 1)
+ return 0;
+
+ if (codec_clk) {
+ clk_set_rate(codec_clk, TABLA_EXT_CLK_RATE);
+ clk_prepare_enable(codec_clk);
+ tabla_mclk_enable(codec, 1, dapm);
+ } else {
+ pr_err("%s: Error setting Tabla MCLK\n", __func__);
+ clk_users--;
+ return -EINVAL;
+ }
+ } else {
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users == 0)
+ return 0;
+ clk_users--;
+ if (!clk_users) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+ clk_disable_unprepare(codec_clk);
+ tabla_mclk_enable(codec, 0, dapm);
+ }
+ }
+ return 0;
+}
+
+static int msm_mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+
+ if (clk_users != 1)
+ return 0;
+
+ if (codec_clk) {
+ clk_set_rate(codec_clk, 12288000);
+ clk_prepare_enable(codec_clk);
+ tabla_mclk_enable(w->codec, 1, true);
+
+ } else {
+ pr_err("%s: Error setting Tabla MCLK\n", __func__);
+ clk_users--;
+ return -EINVAL;
+ }
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+
+ if (clk_users == 0)
+ return 0;
+
+ clk_users--;
+
+ if (!clk_users) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+
+ clk_disable_unprepare(codec_clk);
+ tabla_mclk_enable(w->codec, 0, true);
+ }
+ break;
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget msm_dapm_widgets[] = {
+
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ msm_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Pos", msm_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Neg", msm_spkramp_event),
+
+ SND_SOC_DAPM_SPK("Ext Spk Top Pos", msm_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Top Neg", msm_spkramp_event),
+
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route common_audio_map[] = {
+
+ {"RX_BIAS", NULL, "MCLK"},
+ {"LDO_H", NULL, "MCLK"},
+
+ /* Speaker path */
+ {"Ext Spk Bottom Pos", NULL, "LINEOUT1"},
+ {"Ext Spk Bottom Neg", NULL, "LINEOUT3"},
+
+ {"Ext Spk Top Pos", NULL, "LINEOUT2"},
+ {"Ext Spk Top Neg", NULL, "LINEOUT4"},
+
+ /* Microphone path */
+ {"AMIC1", NULL, "MIC BIAS1 Internal1"},
+ {"MIC BIAS1 Internal1", NULL, "Handset Mic"},
+
+ {"AMIC2", NULL, "MIC BIAS2 External"},
+ {"MIC BIAS2 External", NULL, "Headset Mic"},
+
+ /**
+ * AMIC3 and AMIC4 inputs are connected to ANC microphones
+ * These mics are biased differently on CDP and FLUID
+ * routing entries below are based on bias arrangement
+ * on FLUID.
+ */
+ {"AMIC3", NULL, "MIC BIAS3 Internal1"},
+ {"MIC BIAS3 Internal1", NULL, "ANCRight Headset Mic"},
+
+ {"AMIC4", NULL, "MIC BIAS1 Internal2"},
+ {"MIC BIAS1 Internal2", NULL, "ANCLeft Headset Mic"},
+
+ {"HEADPHONE", NULL, "LDO_H"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *slim0_rx_ch_text[] = {"One", "Two"};
+static const char *slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
+static const char *hdmi_rx_ch_text[] = {"Two", "Three", "Four", "Five", "Six"};
+
+
+static const struct soc_enum msm_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
+ SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
+ SOC_ENUM_SINGLE_EXT(5, hdmi_rx_ch_text),
+
+};
+
+static int msm_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
+ msm_slim_0_rx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_0_rx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
+ msm_slim_0_rx_ch);
+ return 1;
+}
+
+static int msm_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
+ msm_slim_0_tx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_0_tx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
+ msm_slim_0_tx_ch);
+ return 1;
+}
+
+static int msm_hdmi_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_hdmi_rx_ch = %d\n", __func__,
+ msm_hdmi_rx_ch);
+ ucontrol->value.integer.value[0] = msm_hdmi_rx_ch - 2;
+ return 0;
+}
+
+static int msm_hdmi_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_hdmi_rx_ch = ucontrol->value.integer.value[0] + 2;
+
+ pr_debug("%s: msm_hdmi_rx_ch = %d\n", __func__,
+ msm_hdmi_rx_ch);
+ return 1;
+}
+
+
+static const struct snd_kcontrol_new tabla_msm_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", msm_enum[0], msm_get_spk,
+ msm_set_spk),
+ SOC_ENUM_EXT("SLIM_0_RX Channels", msm_enum[1],
+ msm_slim_0_rx_ch_get, msm_slim_0_rx_ch_put),
+ SOC_ENUM_EXT("SLIM_0_TX Channels", msm_enum[2],
+ msm_slim_0_tx_ch_get, msm_slim_0_tx_ch_put),
+ SOC_ENUM_EXT("HDMI_RX Channels", msm_enum[3],
+ msm_hdmi_rx_ch_get, msm_hdmi_rx_ch_put),
+
+};
+
+static void *def_tabla_mbhc_cal(void)
+{
+ void *tabla_cal;
+ struct tabla_mbhc_btn_detect_cfg *btn_cfg;
+ u16 *btn_low, *btn_high;
+ u8 *n_ready, *n_cic, *gain;
+
+ tabla_cal = kzalloc(TABLA_MBHC_CAL_SIZE(TABLA_MBHC_DEF_BUTTONS,
+ TABLA_MBHC_DEF_RLOADS),
+ GFP_KERNEL);
+ if (!tabla_cal) {
+ pr_err("%s: out of memory\n", __func__);
+ return NULL;
+ }
+
+#define S(X, Y) ((TABLA_MBHC_CAL_GENERAL_PTR(tabla_cal)->X) = (Y))
+ S(t_ldoh, 100);
+ S(t_bg_fast_settle, 100);
+ S(t_shutdown_plug_rem, 255);
+ S(mbhc_nsa, 4);
+ S(mbhc_navg, 4);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_DET_PTR(tabla_cal)->X) = (Y))
+ S(mic_current, TABLA_PID_MIC_5_UA);
+ S(hph_current, TABLA_PID_MIC_5_UA);
+ S(t_mic_pid, 100);
+ S(t_ins_complete, 250);
+ S(t_ins_retry, 200);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla_cal)->X) = (Y))
+ S(v_no_mic, 30);
+ S(v_hs_max, 1550);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal)->X) = (Y))
+ S(c[0], 62);
+ S(c[1], 124);
+ S(nc, 1);
+ S(n_meas, 3);
+ S(mbhc_nsc, 11);
+ S(n_btn_meas, 1);
+ S(n_btn_con, 2);
+ S(num_btn, TABLA_MBHC_DEF_BUTTONS);
+ S(v_btn_press_delta_sta, 100);
+ S(v_btn_press_delta_cic, 50);
+#undef S
+ btn_cfg = TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal);
+ btn_low = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_LOW);
+ btn_high = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_HIGH);
+ btn_low[0] = -50;
+ btn_high[0] = 10;
+ btn_low[1] = 11;
+ btn_high[1] = 38;
+ btn_low[2] = 39;
+ btn_high[2] = 64;
+ btn_low[3] = 65;
+ btn_high[3] = 91;
+ btn_low[4] = 92;
+ btn_high[4] = 115;
+ btn_low[5] = 116;
+ btn_high[5] = 141;
+ btn_low[6] = 142;
+ btn_high[6] = 163;
+ btn_low[7] = 164;
+ btn_high[7] = 250;
+ n_ready = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_READY);
+ n_ready[0] = 48;
+ n_ready[1] = 38;
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_CIC);
+ n_cic[0] = 60;
+ n_cic[1] = 47;
+ gain = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_GAIN);
+ gain[0] = 11;
+ gain[1] = 9;
+
+ return tabla_cal;
+}
+
+static int msm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ unsigned int rx_ch_cnt = 0, tx_ch_cnt = 0;
+
+ pr_debug("%s: ch=%d\n", __func__,
+ msm_slim_0_rx_ch);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
+ msm_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai, 0, 0,
+ msm_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+ } else {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(cpu_dai,
+ msm_slim_0_tx_ch, tx_ch, 0 , 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai,
+ msm_slim_0_tx_ch, tx_ch, 0, 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+
+
+ }
+end:
+ return ret;
+}
+
+static int msm_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("%s(), dev_name%s\n", __func__, dev_name(cpu_dai->dev));
+
+ snd_soc_dapm_new_controls(dapm, msm_dapm_widgets,
+ ARRAY_SIZE(msm_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, common_audio_map,
+ ARRAY_SIZE(common_audio_map));
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+
+ snd_soc_dapm_sync(dapm);
+
+ err = snd_soc_jack_new(codec, "Headset Jack",
+ (SND_JACK_HEADSET | SND_JACK_OC_HPHL | SND_JACK_OC_HPHR),
+ &hs_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ err = snd_soc_jack_new(codec, "Button Jack",
+ TABLA_JACK_BUTTON_MASK, &button_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ codec_clk = clk_get(cpu_dai->dev, "osr_clk");
+
+ err = tabla_hs_detect(codec, &mbhc_cfg);
+
+ return err;
+}
+
+static int msm_slim_0_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm_slim_0_rx_ch;
+
+ return 0;
+}
+
+static int msm_slim_0_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm_slim_0_tx_ch;
+
+ return 0;
+}
+
+static int msm_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+
+ return 0;
+}
+
+static int msm_hdmi_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s channels->min %u channels->max %u ()\n", __func__,
+ channels->min, channels->max);
+
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm_hdmi_rx_ch;
+
+ return 0;
+}
+
+static int msm_mi2s_free_gpios(void)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(mi2s_gpio); i++)
+ gpio_free(mi2s_gpio[i].gpio_no);
+ return 0;
+}
+
+static void msm_mi2s_shutdown(struct snd_pcm_substream *substream)
+{
+ if (mi2s_bit_clk) {
+ clk_disable(mi2s_bit_clk);
+ clk_put(mi2s_bit_clk);
+ mi2s_bit_clk = NULL;
+ }
+ msm_mi2s_free_gpios();
+}
+
+static int configure_mi2s_gpio(void)
+{
+ int rtn;
+ int i;
+ int j;
+ for (i = 0; i < ARRAY_SIZE(mi2s_gpio); i++) {
+ rtn = gpio_request(mi2s_gpio[i].gpio_no,
+ mi2s_gpio[i].gpio_name);
+ pr_debug("%s: gpio = %d, gpio name = %s, rtn = %d\n",
+ __func__,
+ mi2s_gpio[i].gpio_no,
+ mi2s_gpio[i].gpio_name,
+ rtn);
+ if (rtn) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__,
+ mi2s_gpio[i].gpio_no);
+ for (j = i; j >= 0; j--)
+ gpio_free(mi2s_gpio[j].gpio_no);
+ goto err;
+ }
+ }
+err:
+ return rtn;
+}
+static int msm_mi2s_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ configure_mi2s_gpio();
+ mi2s_bit_clk = clk_get(cpu_dai->dev, "bit_clk");
+ if (IS_ERR(mi2s_bit_clk))
+ return PTR_ERR(mi2s_bit_clk);
+ clk_set_rate(mi2s_bit_clk, 0);
+ ret = clk_enable(mi2s_bit_clk);
+ if (IS_ERR_VALUE(ret)) {
+ pr_err("Unable to enable mi2s_bit_clk\n");
+ clk_put(mi2s_bit_clk);
+ return ret;
+ }
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM);
+ if (IS_ERR_VALUE(ret))
+ pr_err("set format for CPU dai failed\n");
+ return ret;
+}
+
+static int msm_startup(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+ return 0;
+}
+
+static void msm_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+}
+
+static struct snd_soc_ops msm_be_ops = {
+ .startup = msm_startup,
+ .hw_params = msm_hw_params,
+ .shutdown = msm_shutdown,
+};
+
+static int mpq8064_sec_i2s_rx_free_gpios(void)
+{
+ int i;
+ for (i = 0; i < ARRAY_SIZE(sec_i2s_rx_gpio); i++)
+ gpio_free(sec_i2s_rx_gpio[i].gpio_no);
+ return 0;
+}
+
+static int mpq8064_sec_i2s_rx_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+
+ int rate = params_rate(params);
+ int bit_clk_set = 0;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bit_clk_set = I2S_MCLK_RATE/(rate * 2 * 16);
+ clk_set_rate(sec_i2s_rx_bit_clk, bit_clk_set);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bit_clk_set = I2S_MCLK_RATE/(rate * 2 * 24);
+ clk_set_rate(sec_i2s_rx_bit_clk, bit_clk_set);
+ break;
+ default:
+ pr_err("wrong format\n");
+ break;
+ }
+ }
+ return 0;
+}
+
+static void mpq8064_sec_i2s_rx_shutdown(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (sec_i2s_rx_bit_clk) {
+ clk_disable_unprepare(sec_i2s_rx_bit_clk);
+ clk_put(sec_i2s_rx_bit_clk);
+ sec_i2s_rx_bit_clk = NULL;
+ }
+ if (sec_i2s_rx_osr_clk) {
+ clk_disable_unprepare(sec_i2s_rx_osr_clk);
+ clk_put(sec_i2s_rx_osr_clk);
+ sec_i2s_rx_osr_clk = NULL;
+ }
+ mpq8064_sec_i2s_rx_free_gpios();
+ }
+ pr_info("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+}
+
+static int configure_sec_i2s_rx_gpio(void)
+{
+ int rtn;
+ int i;
+ int j;
+ for (i = 0; i < ARRAY_SIZE(sec_i2s_rx_gpio); i++) {
+ rtn = gpio_request(sec_i2s_rx_gpio[i].gpio_no,
+ sec_i2s_rx_gpio[i].gpio_name);
+ pr_debug("%s: gpio = %d, gpio name = %s, rtn = %d\n",
+ __func__,
+ sec_i2s_rx_gpio[i].gpio_no,
+ sec_i2s_rx_gpio[i].gpio_name,
+ rtn);
+ if (rtn) {
+ pr_err("%s: Failed to request gpio %d\n",
+ __func__,
+ sec_i2s_rx_gpio[i].gpio_no);
+ for (j = i; j >= 0; j--)
+ gpio_free(sec_i2s_rx_gpio[j].gpio_no);
+
+ goto err;
+ }
+ }
+err:
+ return rtn;
+}
+
+static int mpq8064_sec_i2s_rx_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ configure_sec_i2s_rx_gpio();
+ sec_i2s_rx_osr_clk = clk_get(cpu_dai->dev, "osr_clk");
+ if (IS_ERR(sec_i2s_rx_osr_clk)) {
+ pr_err("Failed to get sec_i2s_rx_osr_clk\n");
+ return PTR_ERR(sec_i2s_rx_osr_clk);
+ }
+ clk_set_rate(sec_i2s_rx_osr_clk, I2S_MCLK_RATE);
+ clk_prepare_enable(sec_i2s_rx_osr_clk);
+ sec_i2s_rx_bit_clk = clk_get(cpu_dai->dev, "bit_clk");
+ if (IS_ERR(sec_i2s_rx_bit_clk)) {
+ pr_err("Failed to get sec i2s osr_clk\n");
+ clk_disable_unprepare(sec_i2s_rx_osr_clk);
+ clk_put(sec_i2s_rx_osr_clk);
+ return PTR_ERR(sec_i2s_rx_bit_clk);
+ }
+ clk_set_rate(sec_i2s_rx_bit_clk, 1);
+ ret = clk_prepare_enable(sec_i2s_rx_bit_clk);
+ if (ret != 0) {
+ pr_err("Unable to enable sec i2s rx_bit_clk\n");
+ clk_put(sec_i2s_rx_bit_clk);
+ clk_disable_unprepare(sec_i2s_rx_osr_clk);
+ clk_put(sec_i2s_rx_osr_clk);
+ return ret;
+ }
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ pr_err("set format for codec dai failed\n");
+ }
+ pr_debug("%s: ret = %d\n", __func__, ret);
+ pr_info("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+ return ret;
+}
+
+static struct snd_soc_ops mpq8064_sec_i2s_rx_be_ops = {
+ .startup = mpq8064_sec_i2s_rx_startup,
+ .shutdown = mpq8064_sec_i2s_rx_shutdown,
+ .hw_params = mpq8064_sec_i2s_rx_hw_params,
+};
+
+static struct snd_soc_ops msm_mi2s_tx_be_ops = {
+ .startup = msm_mi2s_startup,
+ .shutdown = msm_mi2s_shutdown,
+};
+
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm_dai[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MSM8960 Media1",
+ .stream_name = "MultiMedia1",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MSM8960 Media2",
+ .stream_name = "MultiMedia2",
+ .cpu_dai_name = "MultiMedia2",
+ .platform_name = "msm-multi-ch-pcm-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+ {
+ .name = "Circuit-Switch Voice",
+ .stream_name = "CS-Voice",
+ .cpu_dai_name = "CS-VOICE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .be_id = MSM_FRONTEND_DAI_CS_VOICE,
+ },
+ {
+ .name = "MSM VoIP",
+ .stream_name = "VoIP",
+ .cpu_dai_name = "VoIP",
+ .platform_name = "msm-voip-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_VOIP,
+ },
+ {
+ .name = "MSM8960 LPA",
+ .stream_name = "LPA",
+ .cpu_dai_name = "MultiMedia3",
+ .platform_name = "msm-pcm-lpa",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
+ },
+ /* Hostless PMC purpose */
+ {
+ .name = "SLIMBUS_0 Hostless",
+ .stream_name = "SLIMBUS_0 Hostless",
+ .cpu_dai_name = "SLIMBUS0_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "INT_FM Hostless",
+ .stream_name = "INT_FM Hostless",
+ .cpu_dai_name = "INT_FM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "MSM AFE-PCM RX",
+ .stream_name = "AFE-PROXY RX",
+ .cpu_dai_name = "msm-dai-q6.241",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "MSM AFE-PCM TX",
+ .stream_name = "AFE-PROXY TX",
+ .cpu_dai_name = "msm-dai-q6.240",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "MSM8960 Compr",
+ .stream_name = "COMPR",
+ .cpu_dai_name = "MultiMedia4",
+ .platform_name = "msm-compr-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA4,
+ },
+ {
+ .name = "Voice Stub",
+ .stream_name = "Voice Stub",
+ .cpu_dai_name = "VOICE_STUB",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* HDMI Hostless */
+ {
+ .name = "HDMI_RX_HOSTLESS",
+ .stream_name = "HDMI_RX_HOSTLESS",
+ .cpu_dai_name = "HDMI_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* MI2S TX Hostless */
+ {
+ .name = "MI2S_TX Hostless",
+ .stream_name = "MI2S_TX Hostless",
+ .cpu_dai_name = "MI2S_TX_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* Secondary I2S RX Hostless */
+ {
+ .name = "SEC_I2S_RX Hostless",
+ .stream_name = "SEC_I2S_RX Hostless",
+ .cpu_dai_name = "SEC_I2S_RX_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_BESPOKE, SND_SOC_DPCM_TRIGGER_BESPOKE},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* Backend DAI Links */
+ {
+ .name = LPASS_BE_SLIMBUS_0_RX,
+ .stream_name = "Slimbus Playback",
+ .cpu_dai_name = "msm-dai-q6.16384",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ .init = &msm_audrx_init,
+ .be_hw_params_fixup = msm_slim_0_rx_be_hw_params_fixup,
+ .ops = &msm_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_0_TX,
+ .stream_name = "Slimbus Capture",
+ .cpu_dai_name = "msm-dai-q6.16385",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ .be_hw_params_fixup = msm_slim_0_tx_be_hw_params_fixup,
+ .ops = &msm_be_ops,
+ },
+ {
+ .name = LPASS_BE_SEC_I2S_RX,
+ .stream_name = "Secondary I2S Playback",
+ .cpu_dai_name = "msm-dai-q6.4",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "cs8427-spdif.5-0014",
+ .codec_dai_name = "spdif_rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SEC_I2S_RX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ .ops = &mpq8064_sec_i2s_rx_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_FM_RX,
+ .stream_name = "Internal FM Playback",
+ .cpu_dai_name = "msm-dai-q6.12292",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_RX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_FM_TX,
+ .stream_name = "Internal FM Capture",
+ .cpu_dai_name = "msm-dai-q6.12293",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_TX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ },
+ /* HDMI BACK END DAI Link */
+ {
+ .name = LPASS_BE_HDMI,
+ .stream_name = "HDMI Playback",
+ .cpu_dai_name = "msm-dai-q6-hdmi.8",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_HDMI_RX,
+ .be_hw_params_fixup = msm_hdmi_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_MI2S_TX,
+ .stream_name = "MI2S Capture",
+ .cpu_dai_name = "msm-dai-q6-mi2s",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_MI2S_TX,
+ .be_hw_params_fixup = msm_be_hw_params_fixup,
+ .ops = &msm_mi2s_tx_be_ops,
+ },
+ /* Backend AFE DAI Links */
+ {
+ .name = LPASS_BE_AFE_PCM_RX,
+ .stream_name = "AFE Playback",
+ .cpu_dai_name = "msm-dai-q6.224",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_RX,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_AFE_PCM_TX,
+ .stream_name = "AFE Capture",
+ .cpu_dai_name = "msm-dai-q6.225",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_TX,
+ },
+};
+
+
+static struct snd_soc_card snd_soc_card_msm = {
+ .name = "mpq8064-tabla-snd-card",
+ .dai_link = msm_dai,
+ .num_links = ARRAY_SIZE(msm_dai),
+ .controls = tabla_msm_controls,
+ .num_controls = ARRAY_SIZE(tabla_msm_controls),
+};
+
+static struct platform_device *msm_snd_device;
+
+static int msm_configure_headset_mic_gpios(void)
+{
+ int ret;
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ ret = gpio_request(PM8921_GPIO_PM_TO_SYS(23), "AV_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+
+ ret = pm8xxx_gpio_config(PM8921_GPIO_PM_TO_SYS(23), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(23));
+ else
+ gpio_direction_output(PM8921_GPIO_PM_TO_SYS(23), 0);
+
+ ret = gpio_request(PM8921_GPIO_PM_TO_SYS(35), "US_EURO_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(35));
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+ ret = pm8xxx_gpio_config(PM8921_GPIO_PM_TO_SYS(35), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(35));
+ else
+ gpio_direction_output(PM8921_GPIO_PM_TO_SYS(35), 0);
+
+ return 0;
+}
+static void msm_free_headset_mic_gpios(void)
+{
+ if (msm_headset_gpios_configured) {
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ gpio_free(PM8921_GPIO_PM_TO_SYS(35));
+ }
+}
+
+static int __init msm_audio_init(void)
+{
+ int ret;
+
+ if (socinfo_get_id() != 130) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return -ENODEV;
+ }
+
+ mbhc_cfg.calibration = def_tabla_mbhc_cal();
+ if (!mbhc_cfg.calibration) {
+ pr_err("Calibration data allocation failed\n");
+ return -ENOMEM;
+ }
+
+ msm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(msm_snd_device, &snd_soc_card_msm);
+ ret = platform_device_add(msm_snd_device);
+ if (ret) {
+ platform_device_put(msm_snd_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+
+ if (msm_configure_headset_mic_gpios()) {
+ pr_err("%s Fail to configure headset mic gpios\n", __func__);
+ msm_headset_gpios_configured = 0;
+ } else
+ msm_headset_gpios_configured = 1;
+
+ return ret;
+
+}
+module_init(msm_audio_init);
+
+static void __exit msm_audio_exit(void)
+{
+ if (socinfo_get_id() != 130) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return ;
+ }
+ msm_free_headset_mic_gpios();
+ platform_device_unregister(msm_snd_device);
+ kfree(mbhc_cfg.calibration);
+}
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC mpq8064");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-compr-q6.c b/sound/soc/msm/msm-compr-q6.c
new file mode 100644
index 0000000..8061b06
--- /dev/null
+++ b/sound/soc/msm/msm-compr-q6.c
@@ -0,0 +1,753 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6asm.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <sound/timer.h>
+
+#include "msm-compr-q6.h"
+#include "msm-pcm-routing.h"
+
+struct snd_msm {
+ struct msm_audio *prtd;
+ unsigned volume;
+};
+static struct snd_msm compressed_audio = {NULL, 0x2000} ;
+
+static struct audio_locks the_locks;
+
+static struct snd_pcm_hardware msm_compr_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 1200 * 1024 * 2,
+ .period_bytes_min = 4800,
+ .period_bytes_max = 1200 * 1024,
+ .periods_min = 2,
+ .periods_max = 512,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void compr_event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct compr_audio *compr = priv;
+ struct msm_audio *prtd = &compr->prtd;
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_aio_write_param param;
+ struct audio_buffer *buf = NULL;
+ int i = 0;
+
+ pr_debug("%s opcode =%08x\n", __func__, opcode);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE: {
+ uint32_t *ptrmem = (uint32_t *)¶m;
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ else
+ if (substream->timer_running)
+ snd_timer_interrupt(substream->timer, 1);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start)) {
+ atomic_set(&prtd->pending_buffer, 1);
+ break;
+ } else
+ atomic_set(&prtd->pending_buffer, 0);
+
+ if (runtime->status->hw_ptr >= runtime->control->appl_ptr)
+ break;
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ pr_debug("%s:writing buffer[%d] from 0x%08x\n",
+ __func__, prtd->out_head,
+ ((unsigned int)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count)));
+
+ param.paddr = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
+ i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1) & (runtime->periods - 1);
+ break;
+ }
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN: {
+ if (!atomic_read(&prtd->pending_buffer))
+ break;
+ pr_debug("%s:writing %d bytes"
+ " of buffer[%d] to dsp\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s:writing buffer[%d] from 0x%08x\n",
+ __func__, prtd->out_head,
+ ((unsigned int)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count)));
+ param.paddr = (unsigned long)buf[prtd->out_head].phys;
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[prtd->out_head].phys;
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1)
+ & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ }
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ pr_debug("ASM_STREAM_CMD_FLUSH\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_compr_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct asm_aac_cfg aac_cfg;
+ struct asm_wma_cfg wma_cfg;
+ struct asm_wmapro_cfg wma_pro_cfg;
+ int ret;
+
+ pr_debug("compressed stream prepare\n");
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ prtd->out_head = 0;
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ if (prtd->enabled)
+ return 0;
+
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_MP3:
+ ret = q6asm_media_format_block(prtd->audio_client,
+ compr->codec);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+ break;
+ case SND_AUDIOCODEC_AAC:
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
+ aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
+ aac_cfg.format = 0x03;
+ aac_cfg.ch_cfg = runtime->channels;
+ aac_cfg.sample_rate = runtime->rate;
+ ret = q6asm_media_format_block_aac(prtd->audio_client,
+ &aac_cfg);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ case SND_AUDIOCODEC_AC3_PASS_THROUGH:
+ pr_debug("compressd playback, no need to send"
+ " the decoder params\n");
+ break;
+ case SND_AUDIOCODEC_WMA:
+ pr_debug("SND_AUDIOCODEC_WMA\n");
+ memset(&wma_cfg, 0x0, sizeof(struct asm_wma_cfg));
+ wma_cfg.format_tag = compr->info.codec_param.codec.format;
+ wma_cfg.ch_cfg = runtime->channels;
+ wma_cfg.sample_rate = runtime->rate;
+ wma_cfg.avg_bytes_per_sec =
+ compr->info.codec_param.codec.bit_rate/8;
+ wma_cfg.block_align = compr->info.codec_param.codec.align;
+ wma_cfg.valid_bits_per_sample =
+ compr->info.codec_param.codec.options.wma.bits_per_sample;
+ wma_cfg.ch_mask =
+ compr->info.codec_param.codec.options.wma.channelmask;
+ wma_cfg.encode_opt =
+ compr->info.codec_param.codec.options.wma.encodeopt;
+ ret = q6asm_media_format_block_wma(prtd->audio_client,
+ &wma_cfg);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ case SND_AUDIOCODEC_WMA_PRO:
+ pr_debug("SND_AUDIOCODEC_WMA_PRO\n");
+ memset(&wma_pro_cfg, 0x0, sizeof(struct asm_wmapro_cfg));
+ wma_pro_cfg.format_tag = compr->info.codec_param.codec.format;
+ wma_pro_cfg.ch_cfg = compr->info.codec_param.codec.ch_in;
+ wma_pro_cfg.sample_rate = runtime->rate;
+ wma_pro_cfg.avg_bytes_per_sec =
+ compr->info.codec_param.codec.bit_rate/8;
+ wma_pro_cfg.block_align = compr->info.codec_param.codec.align;
+ wma_pro_cfg.valid_bits_per_sample =
+ compr->info.codec_param.codec.options.wma.bits_per_sample;
+ wma_pro_cfg.ch_mask =
+ compr->info.codec_param.codec.options.wma.channelmask;
+ wma_pro_cfg.encode_opt =
+ compr->info.codec_param.codec.options.wma.encodeopt;
+ ret = q6asm_media_format_block_wmapro(prtd->audio_client,
+ &wma_pro_cfg);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->pcm_irq_pos = 0;
+ if (compr->info.codec_param.codec.id ==
+ SND_AUDIOCODEC_AC3_PASS_THROUGH) {
+ msm_pcm_routing_reg_psthr_stream(
+ soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+ }
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->start, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static void populate_codec_list(struct compr_audio *compr,
+ struct snd_pcm_runtime *runtime)
+{
+ pr_debug("%s\n", __func__);
+ /* MP3 Block */
+ compr->info.compr_cap.num_codecs = 1;
+ compr->info.compr_cap.min_fragment_size = runtime->hw.period_bytes_min;
+ compr->info.compr_cap.max_fragment_size = runtime->hw.period_bytes_max;
+ compr->info.compr_cap.min_fragments = runtime->hw.periods_min;
+ compr->info.compr_cap.max_fragments = runtime->hw.periods_max;
+ compr->info.compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
+ compr->info.compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
+ compr->info.compr_cap.codecs[2] = SND_AUDIOCODEC_AC3_PASS_THROUGH;
+ compr->info.compr_cap.codecs[3] = SND_AUDIOCODEC_WMA;
+ compr->info.compr_cap.codecs[4] = SND_AUDIOCODEC_WMA_PRO;
+ /* Add new codecs here */
+}
+
+static int msm_compr_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr;
+ struct msm_audio *prtd;
+ int ret = 0;
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return -EINVAL;
+
+ pr_debug("%s\n", __func__);
+ compr = kzalloc(sizeof(struct compr_audio), GFP_KERNEL);
+ if (compr == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd = &compr->prtd;
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)compr_event_handler, compr);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ runtime->hw = msm_compr_hardware_playback;
+
+ pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ atomic_set(&prtd->pending_buffer, 1);
+ compr->codec = FORMAT_MP3;
+ populate_codec_list(compr, runtime);
+ runtime->private_data = compr;
+ compressed_audio.prtd = &compr->prtd;
+ ret = compressed_set_volume(compressed_audio.volume);
+ if (ret < 0)
+ pr_err("%s : Set Volume failed : %d", __func__, ret);
+
+ ret = q6asm_set_softpause(compressed_audio.prtd->audio_client,
+ &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(compressed_audio.prtd->audio_client,
+ &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ return 0;
+}
+
+int compressed_set_volume(unsigned volume)
+{
+ int rc = 0;
+ if (compressed_audio.prtd && compressed_audio.prtd->audio_client) {
+ rc = q6asm_set_volume(compressed_audio.prtd->audio_client,
+ volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed"
+ " rc=%d\n", __func__, rc);
+ }
+ }
+ compressed_audio.volume = volume;
+ return rc;
+}
+
+static int msm_compr_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ int dir = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ atomic_set(&prtd->pending_buffer, 0);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ compressed_audio.prtd = NULL;
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_compr_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_compr_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = EINVAL;
+ return ret;
+}
+static int msm_compr_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_compr_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = EINVAL;
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_compr_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_compr_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+ return result;
+}
+
+static int msm_compr_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ pr_debug("%s\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ return -EINVAL;
+
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_AC3_PASS_THROUGH:
+ ret = q6asm_open_write_compressed(prtd->audio_client,
+ compr->codec);
+ if (ret < 0) {
+ pr_err("%s: compressed Session out open failed\n",
+ __func__);
+ return -ENOMEM;
+ }
+ break;
+ default:
+ ret = q6asm_open_write(prtd->audio_client, compr->codec);
+ if (ret < 0) {
+ pr_err("%s: Session out open failed\n", __func__);
+ return -ENOMEM;
+ }
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ break;
+ }
+ ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed "
+ "rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static int msm_compr_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int rc = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ uint64_t timestamp;
+ uint64_t temp;
+
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP: {
+ struct snd_compr_tstamp tstamp;
+ pr_debug("SNDRV_COMPRESS_TSTAMP\n");
+
+ memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
+ timestamp = q6asm_get_session_time(prtd->audio_client);
+ if (timestamp < 0) {
+ pr_err("%s: Get Session Time return value =%lld\n",
+ __func__, timestamp);
+ return -EAGAIN;
+ }
+ temp = (timestamp * 2 * runtime->channels);
+ temp = temp * (runtime->rate/1000);
+ temp = div_u64(temp, 1000);
+ tstamp.sampling_rate = runtime->rate;
+ tstamp.timestamp = timestamp;
+ pr_debug("%s: bytes_consumed:,"
+ "timestamp = %lld,\n", __func__,
+ tstamp.timestamp);
+ if (copy_to_user((void *) arg, &tstamp,
+ sizeof(struct snd_compr_tstamp)))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_COMPRESS_GET_CAPS:
+ pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
+ if (copy_to_user((void *) arg, &compr->info.compr_cap,
+ sizeof(struct snd_compr_caps))) {
+ rc = -EFAULT;
+ pr_err("%s: ERROR: copy to user\n", __func__);
+ return rc;
+ }
+ return 0;
+ case SNDRV_COMPRESS_SET_PARAMS:
+ pr_debug("SNDRV_COMPRESS_SET_PARAMS: ");
+ if (copy_from_user(&compr->info.codec_param, (void *) arg,
+ sizeof(struct snd_compr_params))) {
+ rc = -EFAULT;
+ pr_err("%s: ERROR: copy from user\n", __func__);
+ return rc;
+ }
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_MP3:
+ /* For MP3 we dont need any other parameter */
+ pr_debug("SND_AUDIOCODEC_MP3\n");
+ compr->codec = FORMAT_MP3;
+ break;
+ case SND_AUDIOCODEC_AAC:
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ compr->codec = FORMAT_MPEG4_AAC;
+ break;
+ case SND_AUDIOCODEC_AC3_PASS_THROUGH:
+ pr_debug("SND_AUDIOCODEC_AC3_PASS_THROUGH\n");
+ compr->codec = FORMAT_AC3;
+ break;
+ case SND_AUDIOCODEC_WMA:
+ pr_debug("SND_AUDIOCODEC_WMA\n");
+ compr->codec = FORMAT_WMA_V9;
+ break;
+ case SND_AUDIOCODEC_WMA_PRO:
+ pr_debug("SND_AUDIOCODEC_WMA_PRO\n");
+ compr->codec = FORMAT_WMA_V10PRO;
+ break;
+ default:
+ pr_debug("FORMAT_LINEAR_PCM\n");
+ compr->codec = FORMAT_LINEAR_PCM;
+ break;
+ }
+ return 0;
+ case SNDRV_PCM_IOCTL1_RESET:
+ prtd->cmd_ack = 0;
+ rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ if (rc < 0)
+ pr_err("%s: flush cmd failed rc=%d\n", __func__, rc);
+ rc = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (rc < 0)
+ pr_err("Flush cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ break;
+ default:
+ break;
+ }
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static struct snd_pcm_ops msm_compr_ops = {
+ .open = msm_compr_open,
+ .hw_params = msm_compr_hw_params,
+ .close = msm_compr_close,
+ .ioctl = msm_compr_ioctl,
+ .prepare = msm_compr_prepare,
+ .trigger = msm_compr_trigger,
+ .pointer = msm_compr_pointer,
+ .mmap = msm_compr_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_compr_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_compr_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_compr_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_compr_driver = {
+ .driver = {
+ .name = "msm-compr-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_compr_probe,
+ .remove = __devexit_p(msm_compr_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_compr_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_compr_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-compr-q6.h b/sound/soc/msm/msm-compr-q6.h
new file mode 100644
index 0000000..cb7f714
--- /dev/null
+++ b/sound/soc/msm/msm-compr-q6.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef _MSM_COMPR_H
+#define _MSM_COMPR_H
+#include <sound/apr_audio.h>
+#include <sound/q6asm.h>
+#include <sound/compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+
+#include "msm-pcm-q6.h"
+
+struct compr_info {
+ struct snd_compr_caps compr_cap;
+ struct snd_compr_codec_caps codec_caps;
+ struct snd_compr_params codec_param;
+};
+
+struct compr_audio {
+ struct msm_audio prtd;
+ struct compr_info info;
+ uint32_t codec;
+};
+
+#endif /*_MSM_COMPR_H*/
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c
new file mode 100644
index 0000000..c8d3a71c
--- /dev/null
+++ b/sound/soc/msm/msm-dai-fe.c
@@ -0,0 +1,412 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+static struct snd_soc_dai_ops msm_fe_dai_ops = {};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static int multimedia_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops msm_fe_Multimedia_dai_ops = {
+ .startup = multimedia_startup,
+};
+
+static struct snd_soc_dai_driver msm_fe_dais[] = {
+ {
+ .playback = {
+ .stream_name = "Multimedia1 Playback",
+ .aif_name = "MM_DL1",
+ .rates = (SNDRV_PCM_RATE_8000_48000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "Multimedia1 Capture",
+ .aif_name = "MM_UL1",
+ .rates = (SNDRV_PCM_RATE_8000_48000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 4,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_Multimedia_dai_ops,
+ .name = "MultiMedia1",
+ },
+ {
+ .playback = {
+ .stream_name = "Multimedia2 Playback",
+ .aif_name = "MM_DL2",
+ .rates = (SNDRV_PCM_RATE_8000_48000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 6,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "Multimedia2 Capture",
+ .aif_name = "MM_UL2",
+ .rates = (SNDRV_PCM_RATE_8000_48000|
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_Multimedia_dai_ops,
+ .name = "MultiMedia2",
+ },
+ {
+ .playback = {
+ .stream_name = "Voice Playback",
+ .aif_name = "CS-VOICE_DL1",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "Voice Capture",
+ .aif_name = "CS-VOICE_UL1",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "CS-VOICE",
+ },
+ {
+ .playback = {
+ .stream_name = "VoIP Playback",
+ .aif_name = "VOIP_DL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_SPECIAL,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "VoIP Capture",
+ .aif_name = "VOIP_UL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_SPECIAL,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "VoIP",
+ },
+ {
+ .playback = {
+ .stream_name = "MultiMedia3 Playback",
+ .aif_name = "MM_DL3",
+ .rates = (SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_Multimedia_dai_ops,
+ .name = "MultiMedia3",
+ },
+ {
+ .playback = {
+ .stream_name = "MultiMedia4 Playback",
+ .aif_name = "MM_DL4",
+ .rates = (SNDRV_PCM_RATE_8000_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_Multimedia_dai_ops,
+ .name = "MultiMedia4",
+ },
+ /* FE DAIs created for hostless operation purpose */
+ {
+ .playback = {
+ .stream_name = "SLIMBUS0 Hostless Playback",
+ .aif_name = "SLIM0_DL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "SLIMBUS0 Hostless Capture",
+ .aif_name = "SLIM0_UL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "SLIMBUS0_HOSTLESS",
+ },
+ {
+ .playback = {
+ .stream_name = "INT_FM Hostless Playback",
+ .aif_name = "INTFM_DL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "INT_FM Hostless Capture",
+ .aif_name = "INTFM_UL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "INT_FM_HOSTLESS",
+ },
+ {
+ .playback = {
+ .stream_name = "AFE-PROXY Playback",
+ .aif_name = "PCM_RX",
+ .rates = (SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "AFE-PROXY Capture",
+ .aif_name = "PCM_TX",
+ .rates = (SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "AFE-PROXY",
+ },
+ {
+ .playback = {
+ .stream_name = "HDMI_Rx Hostless Playback",
+ .aif_name = "HDMI_DL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "HDMI_HOSTLESS"
+ },
+ {
+ .playback = {
+ .stream_name = "AUXPCM Hostless Playback",
+ .aif_name = "AUXPCM_DL_HL",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ },
+ .capture = {
+ .stream_name = "AUXPCM Hostless Capture",
+ .aif_name = "AUXPCM_UL_HL",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "AUXPCM_HOSTLESS",
+ },
+ {
+ .playback = {
+ .stream_name = "Voice Stub Playback",
+ .aif_name = "VOICE_STUB_DL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "Voice Stub Capture",
+ .aif_name = "VOICE_STUB_UL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "VOICE_STUB",
+ },
+ {
+ .playback = {
+ .stream_name = "VoLTE Playback",
+ .aif_name = "VoLTE_DL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .stream_name = "VoLTE Capture",
+ .aif_name = "VoLTE_UL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "VoLTE",
+ },
+ {
+ .capture = {
+ .stream_name = "MI2S_TX Hostless Capture",
+ .aif_name = "MI2S_UL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "MI2S_TX_HOSTLESS",
+ },
+ {
+ .playback = {
+ .stream_name = "SEC_I2S_RX Hostless Playback",
+ .aif_name = "SEC_I2S_DL_HL",
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_fe_dai_ops,
+ .name = "SEC_I2S_RX_HOSTLESS",
+ },
+};
+
+static __devinit int msm_fe_dai_dev_probe(struct platform_device *pdev)
+{
+ dev_dbg(&pdev->dev, "%s: dev name %s\n", __func__,
+ dev_name(&pdev->dev));
+ return snd_soc_register_dais(&pdev->dev, msm_fe_dais,
+ ARRAY_SIZE(msm_fe_dais));
+}
+
+static __devexit int msm_fe_dai_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_fe_dai_driver = {
+ .probe = msm_fe_dai_dev_probe,
+ .remove = msm_fe_dai_dev_remove,
+ .driver = {
+ .name = "msm-dai-fe",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_fe_dai_init(void)
+{
+ return platform_driver_register(&msm_fe_dai_driver);
+}
+module_init(msm_fe_dai_init);
+
+static void __exit msm_fe_dai_exit(void)
+{
+ platform_driver_unregister(&msm_fe_dai_driver);
+}
+module_exit(msm_fe_dai_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM Frontend DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-dai-q6-hdmi.c b/sound/soc/msm/msm-dai-q6-hdmi.c
new file mode 100644
index 0000000..dfb090e
--- /dev/null
+++ b/sound/soc/msm/msm-dai-q6-hdmi.c
@@ -0,0 +1,326 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/bitops.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/apr_audio.h>
+#include <sound/q6afe.h>
+#include <sound/q6adm.h>
+#include <sound/msm-dai-q6.h>
+#include <mach/msm_hdmi_audio.h>
+
+
+enum {
+ STATUS_PORT_STARTED, /* track if AFE port has started */
+ STATUS_MAX
+};
+
+struct msm_dai_q6_hdmi_dai_data {
+ DECLARE_BITMAP(status_mask, STATUS_MAX);
+ u32 rate;
+ u32 channels;
+ union afe_port_config port_config;
+};
+
+static int msm_dai_q6_hdmi_format_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ struct msm_dai_q6_hdmi_dai_data *dai_data = kcontrol->private_data;
+ int value = ucontrol->value.integer.value[0];
+ dai_data->port_config.hdmi_multi_ch.data_type = value;
+ pr_debug("%s: value = %d\n", __func__, value);
+ return 0;
+}
+
+static int msm_dai_q6_hdmi_format_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ struct msm_dai_q6_hdmi_dai_data *dai_data = kcontrol->private_data;
+ ucontrol->value.integer.value[0] =
+ dai_data->port_config.hdmi_multi_ch.data_type;
+ return 0;
+}
+
+
+/* HDMI format field for AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG command
+ * 0: linear PCM
+ * 1: non-linear PCM
+ */
+static const char *hdmi_format[] = {
+ "LPCM",
+ "Compr"
+};
+
+static const struct soc_enum hdmi_config_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, hdmi_format),
+};
+
+static const struct snd_kcontrol_new hdmi_config_controls[] = {
+ SOC_ENUM_EXT("HDMI RX Format", hdmi_config_enum[0],
+ msm_dai_q6_hdmi_format_get,
+ msm_dai_q6_hdmi_format_put),
+};
+
+/* Current implementation assumes hw_param is called once
+ * This may not be the case but what to do when ADM and AFE
+ * port are already opened and parameter changes
+ */
+static int msm_dai_q6_hdmi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ u32 channel_allocation = 0;
+ u32 level_shift = 0; /* 0dB */
+ bool down_mix = FALSE;
+
+ dai_data->channels = params_channels(params);
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.hdmi_multi_ch.reserved = 0;
+
+ switch (dai_data->channels) {
+ case 2:
+ channel_allocation = 0;
+ hdmi_msm_audio_info_setup(1, MSM_HDMI_AUDIO_CHANNEL_2,
+ channel_allocation, level_shift, down_mix);
+ dai_data->port_config.hdmi_multi_ch.channel_allocation =
+ channel_allocation;
+ break;
+ case 6:
+ channel_allocation = 0x0B;
+ hdmi_msm_audio_info_setup(1, MSM_HDMI_AUDIO_CHANNEL_6,
+ channel_allocation, level_shift, down_mix);
+ dai_data->port_config.hdmi_multi_ch.channel_allocation =
+ channel_allocation;
+ break;
+ default:
+ dev_err(dai->dev, "invalid Channels = %u\n",
+ dai_data->channels);
+ return -EINVAL;
+ }
+ dev_dbg(dai->dev, "%s() num_ch = %u rate =%u"
+ " channel_allocation = %u data type = %d\n", __func__,
+ dai_data->channels,
+ dai_data->rate,
+ dai_data->port_config.hdmi_multi_ch.channel_allocation,
+ dai_data->port_config.hdmi_multi_ch.data_type);
+
+ return 0;
+}
+
+
+static void msm_dai_q6_hdmi_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ pr_info("%s: afe port not started. dai_data->status_mask"
+ " = %ld\n", __func__, *dai_data->status_mask);
+ return;
+ }
+
+ rc = afe_close(dai->id); /* can block */
+
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+
+ pr_debug("%s: dai_data->status_mask = %ld\n", __func__,
+ *dai_data->status_mask);
+
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+}
+
+
+static int msm_dai_q6_hdmi_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ /* PORT START should be set if prepare called in active state */
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+ }
+ return rc;
+}
+
+static int msm_dai_q6_hdmi_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ /* Start/stop port without waiting for Q6 AFE response. Need to have
+ * native q6 AFE driver propagates AFE response in order to handle
+ * port start/stop command error properly if error does arise.
+ */
+ pr_debug("%s:port:%d cmd:%d dai_data->status_mask = %ld",
+ __func__, dai->id, cmd, *dai_data->status_mask);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ afe_port_start_nowait(dai->id, &dai_data->port_config,
+ dai_data->rate);
+
+ set_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ afe_port_stop_nowait(dai->id);
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ break;
+
+ default:
+ dev_err(dai->dev, "invalid Trigger command = %d\n", cmd);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int msm_dai_q6_hdmi_dai_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data;
+ const struct snd_kcontrol_new *kcontrol;
+ int rc = 0;
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_hdmi_dai_data),
+ GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+
+ kcontrol = &hdmi_config_controls[0];
+
+ rc = snd_ctl_add(dai->card->snd_card,
+ snd_ctl_new1(kcontrol, dai_data));
+ return rc;
+}
+
+static int msm_dai_q6_hdmi_dai_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_hdmi_dai_data *dai_data;
+ int rc;
+
+ dai_data = dev_get_drvdata(dai->dev);
+
+ /* If AFE port is still up, close it */
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ rc = afe_close(dai->id); /* can block */
+
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ kfree(dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops msm_dai_q6_hdmi_ops = {
+ .prepare = msm_dai_q6_hdmi_prepare,
+ .trigger = msm_dai_q6_hdmi_trigger,
+ .hw_params = msm_dai_q6_hdmi_hw_params,
+ .shutdown = msm_dai_q6_hdmi_shutdown,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_hdmi_hdmi_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 2,
+ .channels_max = 6,
+ .rate_max = 48000,
+ .rate_min = 48000,
+ },
+ .ops = &msm_dai_q6_hdmi_ops,
+ .probe = msm_dai_q6_hdmi_dai_probe,
+ .remove = msm_dai_q6_hdmi_dai_remove,
+};
+
+
+/* To do: change to register DAIs as batch */
+static __devinit int msm_dai_q6_hdmi_dev_probe(struct platform_device *pdev)
+{
+ int rc = 0;
+
+ dev_dbg(&pdev->dev, "dev name %s dev-id %d\n",
+ dev_name(&pdev->dev), pdev->id);
+
+ switch (pdev->id) {
+ case HDMI_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_hdmi_hdmi_rx_dai);
+ break;
+ default:
+ dev_err(&pdev->dev, "invalid device ID %d\n", pdev->id);
+ rc = -ENODEV;
+ break;
+ }
+ return rc;
+}
+
+static __devexit int msm_dai_q6_hdmi_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_dai_q6_hdmi_driver = {
+ .probe = msm_dai_q6_hdmi_dev_probe,
+ .remove = msm_dai_q6_hdmi_dev_remove,
+ .driver = {
+ .name = "msm-dai-q6-hdmi",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_dai_q6_hdmi_init(void)
+{
+ return platform_driver_register(&msm_dai_q6_hdmi_driver);
+}
+module_init(msm_dai_q6_hdmi_init);
+
+static void __exit msm_dai_q6_hdmi_exit(void)
+{
+ platform_driver_unregister(&msm_dai_q6_hdmi_driver);
+}
+module_exit(msm_dai_q6_hdmi_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM DSP HDMI DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-dai-q6.c b/sound/soc/msm/msm-dai-q6.c
new file mode 100644
index 0000000..284320d
--- /dev/null
+++ b/sound/soc/msm/msm-dai-q6.c
@@ -0,0 +1,1707 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wcd9xxx/core.h>
+#include <linux/bitops.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/apr_audio.h>
+#include <sound/q6afe.h>
+#include <sound/msm-dai-q6.h>
+#include <sound/pcm_params.h>
+#include <mach/clk.h>
+
+enum {
+ STATUS_PORT_STARTED, /* track if AFE port has started */
+ STATUS_MAX
+};
+
+struct msm_dai_q6_dai_data {
+ DECLARE_BITMAP(status_mask, STATUS_MAX);
+ u32 rate;
+ u32 channels;
+ u32 bitwidth;
+ union afe_port_config port_config;
+};
+
+struct msm_dai_q6_mi2s_dai_data {
+ struct msm_dai_q6_dai_data tx_dai;
+ struct msm_dai_q6_dai_data rx_dai;
+ struct snd_pcm_hw_constraint_list rate_constraint;
+ struct snd_pcm_hw_constraint_list bitwidth_constraint;
+};
+
+static struct clk *pcm_clk;
+static DEFINE_MUTEX(aux_pcm_mutex);
+static int aux_pcm_count;
+static struct msm_dai_auxpcm_pdata *auxpcm_plat_data;
+
+static int msm_dai_q6_mi2s_format_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ struct msm_dai_q6_dai_data *dai_data = kcontrol->private_data;
+ int value = ucontrol->value.integer.value[0];
+ dai_data->port_config.mi2s.format = value;
+ pr_debug("%s: value = %d, channel = %d, line = %d\n",
+ __func__, value, dai_data->port_config.mi2s.channel,
+ dai_data->port_config.mi2s.line);
+ return 0;
+}
+
+static int msm_dai_q6_mi2s_format_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ struct msm_dai_q6_dai_data *dai_data = kcontrol->private_data;
+ ucontrol->value.integer.value[0] = dai_data->port_config.mi2s.format ;
+ return 0;
+}
+
+
+/* MI2S format field for AFE_PORT_CMD_I2S_CONFIG command
+ * 0: linear PCM
+ * 1: non-linear PCM
+ * 2: PCM data in IEC 60968 container
+ * 3: compressed data in IEC 60958 container
+ */
+static const char *mi2s_format[] = {
+ "LPCM",
+ "Compr",
+ "60958-LPCM",
+ "60958-Compr"};
+
+static const struct soc_enum mi2s_config_enum[] = {
+ SOC_ENUM_SINGLE_EXT(4, mi2s_format),
+};
+
+static const struct snd_kcontrol_new mi2s_config_controls[] = {
+ SOC_ENUM_EXT("MI2S RX Format", mi2s_config_enum[0],
+ msm_dai_q6_mi2s_format_get,
+ msm_dai_q6_mi2s_format_put),
+ SOC_ENUM_EXT("SEC RX Format", mi2s_config_enum[0],
+ msm_dai_q6_mi2s_format_get,
+ msm_dai_q6_mi2s_format_put),
+ SOC_ENUM_EXT("MI2S TX Format", mi2s_config_enum[0],
+ msm_dai_q6_mi2s_format_get,
+ msm_dai_q6_mi2s_format_put),
+};
+
+static u8 num_of_bits_set(u8 sd_line_mask)
+{
+ u8 num_bits_set = 0;
+
+ while (sd_line_mask) {
+ num_bits_set++;
+ sd_line_mask = sd_line_mask & (sd_line_mask - 1);
+ }
+ return num_bits_set;
+}
+
+static int msm_dai_q6_mi2s_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+
+ dev_dbg(dai->dev, "%s: cnst list %p\n", __func__,
+ mi2s_dai_data->rate_constraint.list);
+
+ if (mi2s_dai_data->rate_constraint.list) {
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &mi2s_dai_data->rate_constraint);
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ &mi2s_dai_data->bitwidth_constraint);
+ }
+
+ return 0;
+}
+
+static int msm_dai_q6_mi2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+ struct msm_dai_q6_dai_data *dai_data =
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &mi2s_dai_data->rx_dai : &mi2s_dai_data->tx_dai);
+
+ dai_data->channels = params_channels(params);
+ switch (dai_data->channels) {
+ case 2:
+ dai_data->port_config.mi2s.channel = MSM_AFE_STEREO;
+ break;
+ case 1:
+ dai_data->port_config.mi2s.channel = MSM_AFE_MONO;
+ break;
+ default:
+ pr_warn("greater than stereo has not been validated");
+ break;
+ }
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.mi2s.bitwidth = 16;
+ dai_data->bitwidth = 16;
+ if (!mi2s_dai_data->rate_constraint.list) {
+ mi2s_dai_data->rate_constraint.list = &dai_data->rate;
+ mi2s_dai_data->bitwidth_constraint.list = &dai_data->bitwidth;
+ }
+ return 0;
+}
+
+static int msm_dai_q6_mi2s_get_lineconfig(u16 sd_lines, u16 *config_ptr,
+ unsigned int *ch_cnt)
+{
+ u8 num_of_sd_lines;
+
+ num_of_sd_lines = num_of_bits_set(sd_lines);
+
+ switch (num_of_sd_lines) {
+ case 0:
+ pr_debug("%s: no line is assigned\n", __func__);
+ break;
+ case 1:
+ switch (sd_lines) {
+ case MSM_MI2S_SD0:
+ *config_ptr = AFE_I2S_SD0;
+ break;
+ case MSM_MI2S_SD1:
+ *config_ptr = AFE_I2S_SD1;
+ break;
+ case MSM_MI2S_SD2:
+ *config_ptr = AFE_I2S_SD2;
+ break;
+ case MSM_MI2S_SD3:
+ *config_ptr = AFE_I2S_SD3;
+ break;
+ default:
+ pr_err("%s: invalid SD line\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ case 2:
+ switch (sd_lines) {
+ case MSM_MI2S_SD0 | MSM_MI2S_SD1:
+ *config_ptr = AFE_I2S_QUAD01;
+ break;
+ case MSM_MI2S_SD2 | MSM_MI2S_SD3:
+ *config_ptr = AFE_I2S_QUAD23;
+ break;
+ default:
+ pr_err("%s: invalid SD line\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ case 3:
+ switch (sd_lines) {
+ case MSM_MI2S_SD0 | MSM_MI2S_SD1 | MSM_MI2S_SD2:
+ *config_ptr = AFE_I2S_6CHS;
+ break;
+ default:
+ pr_err("%s: invalid SD lines\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ case 4:
+ switch (sd_lines) {
+ case MSM_MI2S_SD0 | MSM_MI2S_SD1 | MSM_MI2S_SD2 | MSM_MI2S_SD3:
+ *config_ptr = AFE_I2S_8CHS;
+ break;
+ default:
+ pr_err("%s: invalid SD lines\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ default:
+ pr_err("%s: invalid SD lines\n", __func__);
+ goto error_invalid_data;
+ }
+
+ *ch_cnt = num_of_sd_lines;
+
+ return 0;
+
+error_invalid_data:
+ return -EINVAL;
+}
+
+static int msm_dai_q6_mi2s_platform_data_validation(
+ struct platform_device *pdev, struct snd_soc_dai_driver *dai_driver)
+{
+ struct msm_dai_q6_mi2s_dai_data *dai_data = dev_get_drvdata(&pdev->dev);
+ struct msm_mi2s_pdata *mi2s_pdata =
+ (struct msm_mi2s_pdata *) pdev->dev.platform_data;
+ u16 sdline_config;
+ unsigned int ch_cnt;
+ int rc = 0;
+
+ if ((mi2s_pdata->rx_sd_lines & mi2s_pdata->tx_sd_lines) ||
+ (!mi2s_pdata->rx_sd_lines && !mi2s_pdata->tx_sd_lines)) {
+ dev_err(&pdev->dev,
+ "error sd line conflict or no line assigned\n");
+ rc = -EINVAL;
+ goto rtn;
+ }
+
+ rc = msm_dai_q6_mi2s_get_lineconfig(mi2s_pdata->rx_sd_lines,
+ &sdline_config, &ch_cnt);
+
+ if (IS_ERR_VALUE(rc)) {
+ dev_err(&pdev->dev, "invalid MI2S RX sd line config\n");
+ goto rtn;
+ }
+
+ if (ch_cnt) {
+ dai_data->rx_dai.port_config.mi2s.line = sdline_config;
+ dai_driver->playback.channels_min = 1;
+ dai_driver->playback.channels_max = ch_cnt << 1;
+ } else {
+ dai_driver->playback.channels_min = 0;
+ dai_driver->playback.channels_max = 0;
+ }
+ rc = msm_dai_q6_mi2s_get_lineconfig(mi2s_pdata->tx_sd_lines,
+ &sdline_config, &ch_cnt);
+
+ if (IS_ERR_VALUE(rc)) {
+ dev_err(&pdev->dev, "invalid MI2S TX sd line config\n");
+ goto rtn;
+ }
+
+ if (ch_cnt) {
+ dai_data->tx_dai.port_config.mi2s.line = sdline_config;
+ dai_driver->capture.channels_min = 1;
+ dai_driver->capture.channels_max = ch_cnt << 1;
+ } else {
+ dai_driver->capture.channels_min = 0;
+ dai_driver->capture.channels_max = 0;
+ }
+
+ dev_info(&pdev->dev, "%s: playback sdline %x capture sdline %x\n",
+ __func__, dai_data->rx_dai.port_config.mi2s.line,
+ dai_data->tx_dai.port_config.mi2s.line);
+ dev_info(&pdev->dev, "%s: playback ch_max %d capture ch_mx %d\n",
+ __func__, dai_driver->playback.channels_max,
+ dai_driver->capture.channels_max);
+rtn:
+ return rc;
+}
+
+static int msm_dai_q6_mi2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+
+ if (test_bit(STATUS_PORT_STARTED, mi2s_dai_data->rx_dai.status_mask) ||
+ test_bit(STATUS_PORT_STARTED, mi2s_dai_data->rx_dai.status_mask)) {
+ dev_err(dai->dev, "%s: err chg i2s mode while dai running",
+ __func__);
+ return -EPERM;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ mi2s_dai_data->rx_dai.port_config.mi2s.ws = 1;
+ mi2s_dai_data->tx_dai.port_config.mi2s.ws = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ mi2s_dai_data->rx_dai.port_config.mi2s.ws = 0;
+ mi2s_dai_data->tx_dai.port_config.mi2s.ws = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int msm_dai_q6_mi2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+ struct msm_dai_q6_dai_data *dai_data =
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &mi2s_dai_data->rx_dai : &mi2s_dai_data->tx_dai);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ /* PORT START should be set if prepare called in active state */
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+ }
+ return rc;
+}
+
+static int msm_dai_q6_mi2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+ struct msm_dai_q6_dai_data *dai_data =
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &mi2s_dai_data->rx_dai : &mi2s_dai_data->tx_dai);
+ u16 port_id = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ MI2S_RX : MI2S_TX);
+ int rc = 0;
+
+ dev_dbg(dai->dev, "%s: cmd:%d dai_data->status_mask = %ld",
+ __func__, cmd, *dai_data->status_mask);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ afe_port_start_nowait(port_id,
+ &dai_data->port_config, dai_data->rate);
+ set_bit(STATUS_PORT_STARTED,
+ dai_data->status_mask);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ afe_port_stop_nowait(port_id);
+ clear_bit(STATUS_PORT_STARTED,
+ dai_data->status_mask);
+ }
+ break;
+
+ default:
+ rc = -EINVAL;
+ }
+
+ return rc;
+}
+
+static void msm_dai_q6_mi2s_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+ struct msm_dai_q6_dai_data *dai_data =
+ (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &mi2s_dai_data->rx_dai : &mi2s_dai_data->tx_dai);
+ u16 port_id = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ MI2S_RX : MI2S_TX);
+ int rc = 0;
+
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ rc = afe_close(port_id);
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+
+ if (!test_bit(STATUS_PORT_STARTED, mi2s_dai_data->rx_dai.status_mask) &&
+ !test_bit(STATUS_PORT_STARTED, mi2s_dai_data->rx_dai.status_mask)) {
+ mi2s_dai_data->rate_constraint.list = NULL;
+ mi2s_dai_data->bitwidth_constraint.list = NULL;
+ }
+
+}
+
+static int msm_dai_q6_cdc_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->channels = params_channels(params);
+ switch (dai_data->channels) {
+ case 2:
+ dai_data->port_config.mi2s.channel = MSM_AFE_STEREO;
+ break;
+ case 1:
+ dai_data->port_config.mi2s.channel = MSM_AFE_MONO;
+ break;
+ default:
+ return -EINVAL;
+ break;
+ }
+ dai_data->rate = params_rate(params);
+
+ dev_dbg(dai->dev, " channel %d sample rate %d entered\n",
+ dai_data->channels, dai_data->rate);
+
+ /* Q6 only supports 16 as now */
+ dai_data->port_config.mi2s.bitwidth = 16;
+ dai_data->port_config.mi2s.line = 1;
+ return 0;
+}
+
+static int msm_dai_q6_cdc_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ dai_data->port_config.mi2s.ws = 1; /* CPU is master */
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ dai_data->port_config.mi2s.ws = 0; /* CPU is slave */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static int msm_dai_q6_slim_bus_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->channels = params_channels(params);
+ dai_data->rate = params_rate(params);
+
+ /* Q6 only supports 16 as now */
+ dai_data->port_config.slim_sch.bit_width = 16;
+ dai_data->port_config.slim_sch.data_format = 0;
+ dai_data->port_config.slim_sch.num_channels = dai_data->channels;
+ dai_data->port_config.slim_sch.reserved = 0;
+
+ dev_dbg(dai->dev, "%s:slimbus_dev_id[%hu] bit_wd[%hu] format[%hu]\n"
+ "num_channel %hu slave_ch_mapping[0] %hu\n"
+ "slave_port_mapping[1] %hu slave_port_mapping[2] %hu\n"
+ "slave_port_mapping[3] %hu\n sample_rate %d\n", __func__,
+ dai_data->port_config.slim_sch.slimbus_dev_id,
+ dai_data->port_config.slim_sch.bit_width,
+ dai_data->port_config.slim_sch.data_format,
+ dai_data->port_config.slim_sch.num_channels,
+ dai_data->port_config.slim_sch.slave_ch_mapping[0],
+ dai_data->port_config.slim_sch.slave_ch_mapping[1],
+ dai_data->port_config.slim_sch.slave_ch_mapping[2],
+ dai_data->port_config.slim_sch.slave_ch_mapping[3],
+ dai_data->rate);
+
+ return 0;
+}
+
+static int msm_dai_q6_bt_fm_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->channels = params_channels(params);
+ dai_data->rate = params_rate(params);
+
+ dev_dbg(dai->dev, "channels %d sample rate %d entered\n",
+ dai_data->channels, dai_data->rate);
+
+ memset(&dai_data->port_config, 0, sizeof(dai_data->port_config));
+
+ return 0;
+}
+static int msm_dai_q6_auxpcm_hw_params(
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct msm_dai_auxpcm_pdata *auxpcm_pdata =
+ (struct msm_dai_auxpcm_pdata *) dai->dev->platform_data;
+
+ if (params_channels(params) != 1) {
+ dev_err(dai->dev, "AUX PCM supports only mono stream\n");
+ return -EINVAL;
+ }
+ dai_data->channels = params_channels(params);
+
+ dai_data->rate = params_rate(params);
+ switch (dai_data->rate) {
+ case 8000:
+ dai_data->port_config.pcm.mode = auxpcm_pdata->mode_8k.mode;
+ dai_data->port_config.pcm.sync = auxpcm_pdata->mode_8k.sync;
+ dai_data->port_config.pcm.frame = auxpcm_pdata->mode_8k.frame;
+ dai_data->port_config.pcm.quant = auxpcm_pdata->mode_8k.quant;
+ dai_data->port_config.pcm.slot = auxpcm_pdata->mode_8k.slot;
+ dai_data->port_config.pcm.data = auxpcm_pdata->mode_8k.data;
+ break;
+ case 16000:
+ dai_data->port_config.pcm.mode = auxpcm_pdata->mode_16k.mode;
+ dai_data->port_config.pcm.sync = auxpcm_pdata->mode_16k.sync;
+ dai_data->port_config.pcm.frame = auxpcm_pdata->mode_16k.frame;
+ dai_data->port_config.pcm.quant = auxpcm_pdata->mode_16k.quant;
+ dai_data->port_config.pcm.slot = auxpcm_pdata->mode_16k.slot;
+ dai_data->port_config.pcm.data = auxpcm_pdata->mode_16k.data;
+ break;
+ default:
+ dev_err(dai->dev, "AUX PCM supports only 8kHz and 16kHz sampling rate\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int msm_dai_q6_afe_rtproxy_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.rtproxy.num_ch =
+ params_channels(params);
+
+ pr_debug("channel %d entered,dai_id: %d,rate: %d\n",
+ dai_data->port_config.rtproxy.num_ch, dai->id, dai_data->rate);
+
+ dai_data->port_config.rtproxy.bitwidth = 16; /* Q6 only supports 16 */
+ dai_data->port_config.rtproxy.interleaved = 1;
+ dai_data->port_config.rtproxy.frame_sz = params_period_bytes(params);
+ dai_data->port_config.rtproxy.jitter =
+ dai_data->port_config.rtproxy.frame_sz/2;
+ dai_data->port_config.rtproxy.lw_mark = 0;
+ dai_data->port_config.rtproxy.hw_mark = 0;
+ dai_data->port_config.rtproxy.rsvd = 0;
+
+ return 0;
+}
+
+/* Current implementation assumes hw_param is called once
+ * This may not be the case but what to do when ADM and AFE
+ * port are already opened and parameter changes
+ */
+static int msm_dai_q6_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int rc = 0;
+
+ switch (dai->id) {
+ case PRIMARY_I2S_TX:
+ case PRIMARY_I2S_RX:
+ case SECONDARY_I2S_RX:
+ rc = msm_dai_q6_cdc_hw_params(params, dai, substream->stream);
+ break;
+
+ case SLIMBUS_0_RX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ rc = msm_dai_q6_slim_bus_hw_params(params, dai,
+ substream->stream);
+ break;
+ case INT_BT_SCO_RX:
+ case INT_BT_SCO_TX:
+ case INT_FM_RX:
+ case INT_FM_TX:
+ rc = msm_dai_q6_bt_fm_hw_params(params, dai, substream->stream);
+ break;
+ case RT_PROXY_DAI_001_TX:
+ case RT_PROXY_DAI_001_RX:
+ case RT_PROXY_DAI_002_TX:
+ case RT_PROXY_DAI_002_RX:
+ rc = msm_dai_q6_afe_rtproxy_hw_params(params, dai);
+ break;
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ rc = 0;
+ break;
+ default:
+ dev_err(dai->dev, "invalid AFE port ID\n");
+ rc = -EINVAL;
+ break;
+ }
+
+ return rc;
+}
+
+static void msm_dai_q6_auxpcm_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int rc = 0;
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (aux_pcm_count == 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count is 0. Just"
+ " return\n", __func__, dai->id);
+ mutex_unlock(&aux_pcm_mutex);
+ return;
+ }
+
+ aux_pcm_count--;
+
+ if (aux_pcm_count > 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d\n",
+ __func__, dai->id, aux_pcm_count);
+ mutex_unlock(&aux_pcm_mutex);
+ return;
+ } else if (aux_pcm_count < 0) {
+ dev_err(dai->dev, "%s(): ERROR: dai->id %d"
+ " aux_pcm_count = %d < 0\n",
+ __func__, dai->id, aux_pcm_count);
+ aux_pcm_count = 0;
+ mutex_unlock(&aux_pcm_mutex);
+ return;
+ }
+
+ pr_debug("%s: dai->id = %d aux_pcm_count = %d\n", __func__,
+ dai->id, aux_pcm_count);
+
+ clk_disable_unprepare(pcm_clk);
+ rc = afe_close(PCM_RX); /* can block */
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close PCM_RX AFE port\n");
+
+ rc = afe_close(PCM_TX);
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AUX PCM TX port\n");
+
+ mutex_unlock(&aux_pcm_mutex);
+}
+
+static void msm_dai_q6_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ pr_debug("%s, stop pseudo port:%d\n",
+ __func__, dai->id);
+ rc = afe_stop_pseudo_port(dai->id);
+ break;
+ default:
+ rc = afe_close(dai->id); /* can block */
+ break;
+ }
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+ pr_debug("%s: dai_data->status_mask = %ld\n", __func__,
+ *dai_data->status_mask);
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+}
+
+static int msm_dai_q6_auxpcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+ struct msm_dai_auxpcm_pdata *auxpcm_pdata =
+ (struct msm_dai_auxpcm_pdata *) dai->dev->platform_data;
+ unsigned long pcm_clk_rate;
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (aux_pcm_count == 2) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count is 2. Just"
+ " return.\n", __func__, dai->id);
+ mutex_unlock(&aux_pcm_mutex);
+ return 0;
+ } else if (aux_pcm_count > 2) {
+ dev_err(dai->dev, "%s(): ERROR: dai->id %d"
+ " aux_pcm_count = %d > 2\n",
+ __func__, dai->id, aux_pcm_count);
+ mutex_unlock(&aux_pcm_mutex);
+ return 0;
+ }
+
+ aux_pcm_count++;
+ if (aux_pcm_count == 2) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d after "
+ " increment\n", __func__, dai->id, aux_pcm_count);
+ mutex_unlock(&aux_pcm_mutex);
+ return 0;
+ }
+
+ pr_debug("%s:dai->id:%d aux_pcm_count = %d. opening afe\n",
+ __func__, dai->id, aux_pcm_count);
+
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+
+ /*
+ * For AUX PCM Interface the below sequence of clk
+ * settings and afe_open is a strict requirement.
+ *
+ * Also using afe_open instead of afe_port_start_nowait
+ * to make sure the port is open before deasserting the
+ * clock line. This is required because pcm register is
+ * not written before clock deassert. Hence the hw does
+ * not get updated with new setting if the below clock
+ * assert/deasset and afe_open sequence is not followed.
+ */
+
+ clk_reset(pcm_clk, CLK_RESET_ASSERT);
+
+ afe_open(PCM_RX, &dai_data->port_config, dai_data->rate);
+
+ afe_open(PCM_TX, &dai_data->port_config, dai_data->rate);
+ if (dai_data->rate == 8000) {
+ pcm_clk_rate = auxpcm_pdata->mode_8k.pcm_clk_rate;
+ } else if (dai_data->rate == 16000) {
+ pcm_clk_rate = auxpcm_pdata->mode_8k.pcm_clk_rate;
+ } else {
+ dev_err(dai->dev, "%s: Invalid AUX PCM rate %d\n", __func__,
+ dai_data->rate);
+ return -EINVAL;
+ }
+
+ rc = clk_set_rate(pcm_clk, pcm_clk_rate);
+ if (rc < 0) {
+ pr_err("%s: clk_set_rate failed\n", __func__);
+ return rc;
+ }
+
+ clk_prepare_enable(pcm_clk);
+ clk_reset(pcm_clk, CLK_RESET_DEASSERT);
+
+ mutex_unlock(&aux_pcm_mutex);
+
+ return rc;
+}
+
+static int msm_dai_q6_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ /* PORT START should be set if prepare called in active state */
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+ }
+ return rc;
+}
+
+static int msm_dai_q6_auxpcm_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ int rc = 0;
+
+ pr_debug("%s:port:%d cmd:%d aux_pcm_count= %d",
+ __func__, dai->id, cmd, aux_pcm_count);
+
+ switch (cmd) {
+
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /* afe_open will be called from prepare */
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ return 0;
+
+ default:
+ rc = -EINVAL;
+ }
+
+ return rc;
+
+}
+
+static int msm_dai_q6_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ /* Start/stop port without waiting for Q6 AFE response. Need to have
+ * native q6 AFE driver propagates AFE response in order to handle
+ * port start/stop command error properly if error does arise.
+ */
+ pr_debug("%s:port:%d cmd:%d dai_data->status_mask = %ld",
+ __func__, dai->id, cmd, *dai_data->status_mask);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ afe_pseudo_port_start_nowait(dai->id);
+ break;
+ default:
+ afe_port_start_nowait(dai->id,
+ &dai_data->port_config, dai_data->rate);
+ break;
+ }
+ set_bit(STATUS_PORT_STARTED,
+ dai_data->status_mask);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ afe_pseudo_port_stop_nowait(dai->id);
+ break;
+ default:
+ afe_port_stop_nowait(dai->id);
+ break;
+ }
+ clear_bit(STATUS_PORT_STARTED,
+ dai_data->status_mask);
+ }
+ break;
+
+ default:
+ rc = -EINVAL;
+ }
+
+ return rc;
+}
+static int msm_dai_q6_dai_auxpcm_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc = 0;
+
+ struct msm_dai_auxpcm_pdata *auxpcm_pdata =
+ (struct msm_dai_auxpcm_pdata *) dai->dev->platform_data;
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (!auxpcm_plat_data)
+ auxpcm_plat_data = auxpcm_pdata;
+ else if (auxpcm_plat_data != auxpcm_pdata) {
+
+ dev_err(dai->dev, "AUX PCM RX and TX devices does not have"
+ " same platform data\n");
+ return -EINVAL;
+ }
+
+ /*
+ * The clk name for AUX PCM operation is passed as platform
+ * data to the cpu driver, since cpu drive is unaware of any
+ * boarc specific configuration.
+ */
+ if (!pcm_clk) {
+
+ pcm_clk = clk_get(dai->dev, auxpcm_pdata->clk);
+
+ if (IS_ERR(pcm_clk)) {
+ pr_err("%s: could not get pcm_clk\n", __func__);
+ pcm_clk = NULL;
+ return -ENODEV;
+ }
+ }
+
+ mutex_unlock(&aux_pcm_mutex);
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_dai_data), GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+
+ pr_debug("%s : probe done for dai->id %d\n", __func__, dai->id);
+ return rc;
+}
+
+static int msm_dai_q6_dai_auxpcm_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc;
+
+ dai_data = dev_get_drvdata(dai->dev);
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (aux_pcm_count == 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count is 0. clean"
+ " up and return\n", __func__, dai->id);
+ goto done;
+ }
+
+ aux_pcm_count--;
+
+ if (aux_pcm_count > 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d\n",
+ __func__, dai->id, aux_pcm_count);
+ goto done;
+ } else if (aux_pcm_count < 0) {
+ dev_err(dai->dev, "%s(): ERROR: dai->id %d"
+ " aux_pcm_count = %d < 0\n",
+ __func__, dai->id, aux_pcm_count);
+ goto done;
+ }
+
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d."
+ "closing afe\n",
+ __func__, dai->id, aux_pcm_count);
+
+ rc = afe_close(PCM_RX); /* can block */
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AUX PCM RX AFE port\n");
+
+ rc = afe_close(PCM_TX);
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AUX PCM TX AFE port\n");
+
+done:
+ kfree(dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ mutex_unlock(&aux_pcm_mutex);
+
+ return 0;
+}
+
+static int msm_dai_q6_dai_mi2s_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+ struct snd_kcontrol *kcontrol = NULL;
+ int rc = 0;
+
+ if (mi2s_dai_data->rx_dai.port_config.mi2s.line) {
+ kcontrol = snd_ctl_new1(&mi2s_config_controls[0],
+ &mi2s_dai_data->rx_dai);
+ rc = snd_ctl_add(dai->card->snd_card, kcontrol);
+
+ if (IS_ERR_VALUE(rc)) {
+ dev_err(dai->dev, "%s: err add RX fmt ctl\n", __func__);
+ goto rtn;
+ }
+ }
+
+ if (mi2s_dai_data->tx_dai.port_config.mi2s.line) {
+ rc = snd_ctl_add(dai->card->snd_card,
+ snd_ctl_new1(&mi2s_config_controls[2],
+ &mi2s_dai_data->tx_dai));
+
+ if (IS_ERR_VALUE(rc)) {
+ if (kcontrol)
+ snd_ctl_remove(dai->card->snd_card, kcontrol);
+ dev_err(dai->dev, "%s: err add TX fmt ctl\n", __func__);
+ }
+ }
+
+rtn:
+ return rc;
+}
+
+static int msm_dai_q6_dai_mi2s_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_mi2s_dai_data *mi2s_dai_data =
+ dev_get_drvdata(dai->dev);
+ int rc;
+
+ /* If AFE port is still up, close it */
+ if (test_bit(STATUS_PORT_STARTED, mi2s_dai_data->rx_dai.status_mask)) {
+ rc = afe_close(MI2S_RX); /* can block */
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close MI2S_RX port\n");
+ clear_bit(STATUS_PORT_STARTED,
+ mi2s_dai_data->rx_dai.status_mask);
+ }
+ if (test_bit(STATUS_PORT_STARTED, mi2s_dai_data->tx_dai.status_mask)) {
+ rc = afe_close(MI2S_TX); /* can block */
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close MI2S_TX port\n");
+ clear_bit(STATUS_PORT_STARTED,
+ mi2s_dai_data->tx_dai.status_mask);
+ }
+ kfree(mi2s_dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ return 0;
+}
+
+static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc = 0;
+ const struct snd_kcontrol_new *kcontrol;
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_dai_data),
+ GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+ if (dai->id == SECONDARY_I2S_RX) {
+ kcontrol = &mi2s_config_controls[1];
+ rc = snd_ctl_add(dai->card->snd_card,
+ snd_ctl_new1(kcontrol, dai_data));
+ }
+
+ return rc;
+}
+
+static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc;
+
+ dai_data = dev_get_drvdata(dai->dev);
+
+ /* If AFE port is still up, close it */
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ pr_debug("%s, stop pseudo port:%d\n",
+ __func__, dai->id);
+ rc = afe_stop_pseudo_port(dai->id);
+ break;
+ default:
+ rc = afe_close(dai->id); /* can block */
+ }
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ kfree(dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ return 0;
+}
+
+static int msm_dai_q6_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ int rc = 0;
+
+ dev_dbg(dai->dev, "enter %s, id = %d fmt[%d]\n", __func__,
+ dai->id, fmt);
+ switch (dai->id) {
+ case PRIMARY_I2S_TX:
+ case PRIMARY_I2S_RX:
+ case SECONDARY_I2S_RX:
+ rc = msm_dai_q6_cdc_set_fmt(dai, fmt);
+ break;
+ default:
+ dev_err(dai->dev, "invalid cpu_dai set_fmt\n");
+ rc = -EINVAL;
+ break;
+ }
+
+ return rc;
+}
+
+static int msm_dai_q6_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ int rc = 0;
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ unsigned int i = 0;
+
+ dev_dbg(dai->dev, "%s: dai_id = %d\n", __func__, dai->id);
+ switch (dai->id) {
+ case SLIMBUS_0_RX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_4_RX:
+ /* channel number to be between 128 and 255. For RX port
+ * use channel numbers from 138 to 144, for TX port
+ * use channel numbers from 128 to 137
+ * For ports between MDM-APQ use channel numbers from 145
+ */
+ if (!rx_slot)
+ return -EINVAL;
+ for (i = 0; i < rx_num; i++) {
+ dai_data->port_config.slim_sch.slave_ch_mapping[i] =
+ rx_slot[i];
+ pr_debug("%s: find number of channels[%d] ch[%d]\n",
+ __func__, i,
+ rx_slot[i]);
+ }
+ dai_data->port_config.slim_sch.num_channels = rx_num;
+ pr_debug("%s:SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
+ (dai->id - SLIMBUS_0_RX) / 2,
+ rx_num, dai_data->port_config.slim_sch.slave_ch_mapping[0],
+ dai_data->port_config.slim_sch.slave_ch_mapping[1]);
+
+ break;
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_4_TX:
+ /* channel number to be between 128 and 255. For RX port
+ * use channel numbers from 138 to 144, for TX port
+ * use channel numbers from 128 to 137
+ * For ports between MDM-APQ use channel numbers from 145
+ */
+ if (!tx_slot)
+ return -EINVAL;
+ for (i = 0; i < tx_num; i++) {
+ dai_data->port_config.slim_sch.slave_ch_mapping[i] =
+ tx_slot[i];
+ pr_debug("%s: find number of channels[%d] ch[%d]\n",
+ __func__, i, tx_slot[i]);
+ }
+ dai_data->port_config.slim_sch.num_channels = tx_num;
+ pr_debug("%s:SLIMBUS_%d_TX cnt[%d] ch[%d %d]\n", __func__,
+ (dai->id - SLIMBUS_0_TX) / 2,
+ tx_num, dai_data->port_config.slim_sch.slave_ch_mapping[0],
+ dai_data->port_config.slim_sch.slave_ch_mapping[1]);
+ break;
+ default:
+ dev_err(dai->dev, "invalid cpu_dai set_fmt\n");
+ rc = -EINVAL;
+ break;
+ }
+ return rc;
+}
+
+static struct snd_soc_dai_ops msm_dai_q6_mi2s_ops = {
+ .startup = msm_dai_q6_mi2s_startup,
+ .prepare = msm_dai_q6_mi2s_prepare,
+ .trigger = msm_dai_q6_mi2s_trigger,
+ .hw_params = msm_dai_q6_mi2s_hw_params,
+ .shutdown = msm_dai_q6_mi2s_shutdown,
+ .set_fmt = msm_dai_q6_mi2s_set_fmt,
+};
+
+static struct snd_soc_dai_ops msm_dai_q6_ops = {
+ .prepare = msm_dai_q6_prepare,
+ .trigger = msm_dai_q6_trigger,
+ .hw_params = msm_dai_q6_hw_params,
+ .shutdown = msm_dai_q6_shutdown,
+ .set_fmt = msm_dai_q6_set_fmt,
+ .set_channel_map = msm_dai_q6_set_channel_map,
+};
+
+static struct snd_soc_dai_ops msm_dai_q6_auxpcm_ops = {
+ .prepare = msm_dai_q6_auxpcm_prepare,
+ .trigger = msm_dai_q6_auxpcm_trigger,
+ .hw_params = msm_dai_q6_auxpcm_hw_params,
+ .shutdown = msm_dai_q6_auxpcm_shutdown,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_i2s_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 4,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_i2s_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_afe_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_afe_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 4,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_voice_playback_tx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_incall_record_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_bt_sco_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 16000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_bt_sco_tx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 16000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_fm_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_fm_tx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_aux_pcm_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 16000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_auxpcm_ops,
+ .probe = msm_dai_q6_dai_auxpcm_probe,
+ .remove = msm_dai_q6_dai_auxpcm_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_aux_pcm_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 16000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_auxpcm_ops,
+ .probe = msm_dai_q6_dai_auxpcm_probe,
+ .remove = msm_dai_q6_dai_auxpcm_remove,
+};
+
+/* Channel min and max are initialized base on platform data */
+static struct snd_soc_dai_driver msm_dai_q6_mi2s_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_mi2s_ops,
+ .probe = msm_dai_q6_dai_mi2s_probe,
+ .remove = msm_dai_q6_dai_mi2s_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_1_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_1_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_2_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_2_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 4,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_3_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+/* To do: change to register DAIs as batch */
+static __devinit int msm_dai_q6_dev_probe(struct platform_device *pdev)
+{
+ int rc = 0;
+
+ dev_dbg(&pdev->dev, "dev name %s\n", dev_name(&pdev->dev));
+
+ switch (pdev->id) {
+ case PRIMARY_I2S_RX:
+ case SECONDARY_I2S_RX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_i2s_rx_dai);
+ break;
+ case PRIMARY_I2S_TX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_i2s_tx_dai);
+ break;
+ case PCM_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_aux_pcm_rx_dai);
+ break;
+ case PCM_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_aux_pcm_tx_dai);
+ break;
+
+ case SLIMBUS_0_RX:
+ case SLIMBUS_4_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_rx_dai);
+ break;
+ case SLIMBUS_0_TX:
+ case SLIMBUS_4_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_tx_dai);
+ break;
+ case SLIMBUS_1_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_1_rx_dai);
+ break;
+ case SLIMBUS_1_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_1_tx_dai);
+ break;
+ case SLIMBUS_2_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_2_rx_dai);
+ break;
+ case SLIMBUS_2_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_2_tx_dai);
+ break;
+ case SLIMBUS_3_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_3_rx_dai);
+ break;
+ case INT_BT_SCO_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_bt_sco_rx_dai);
+ break;
+ case INT_BT_SCO_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_bt_sco_tx_dai);
+ break;
+ case INT_FM_RX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_fm_rx_dai);
+ break;
+ case INT_FM_TX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_fm_tx_dai);
+ break;
+ case RT_PROXY_DAI_001_RX:
+ case RT_PROXY_DAI_002_RX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_afe_rx_dai);
+ break;
+ case RT_PROXY_DAI_001_TX:
+ case RT_PROXY_DAI_002_TX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_afe_tx_dai);
+ break;
+ case VOICE_PLAYBACK_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_voice_playback_tx_dai);
+ break;
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_incall_record_dai);
+ break;
+ default:
+ rc = -ENODEV;
+ break;
+ }
+ return rc;
+}
+
+static __devexit int msm_dai_q6_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static __devinit int msm_dai_q6_mi2s_dev_probe(struct platform_device *pdev)
+{
+ struct msm_dai_q6_mi2s_dai_data *dai_data;
+ int rc = 0;
+
+ dev_dbg(&pdev->dev, "%s: pdev %p dev %p\n", __func__, pdev, &pdev->dev);
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_mi2s_dai_data),
+ GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(&pdev->dev, "fail to allocate dai data\n");
+ rc = -ENOMEM;
+ goto rtn;
+ } else
+ dev_set_drvdata(&pdev->dev, dai_data);
+
+ rc = msm_dai_q6_mi2s_platform_data_validation(pdev,
+ &msm_dai_q6_mi2s_dai);
+ if (IS_ERR_VALUE(rc))
+ goto err_pdata;
+
+ dai_data->rate_constraint.count = 1;
+ dai_data->bitwidth_constraint.count = 1;
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_mi2s_dai);
+
+ if (IS_ERR_VALUE(rc))
+ goto err_pdata;
+
+ return 0;
+
+err_pdata:
+ kfree(dai_data);
+rtn:
+ return rc;
+}
+
+static __devexit int msm_dai_q6_mi2s_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_dai_q6_driver = {
+ .probe = msm_dai_q6_dev_probe,
+ .remove = msm_dai_q6_dev_remove,
+ .driver = {
+ .name = "msm-dai-q6",
+ .owner = THIS_MODULE,
+ },
+};
+
+static struct platform_driver msm_dai_q6_mi2s_driver = {
+ .probe = msm_dai_q6_mi2s_dev_probe,
+ .remove = msm_dai_q6_mi2s_dev_remove,
+ .driver = {
+ .name = "msm-dai-q6-mi2s",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_dai_q6_init(void)
+{
+ int rc1, rc2;
+
+ rc1 = platform_driver_register(&msm_dai_q6_mi2s_driver);
+
+ if (IS_ERR_VALUE(rc1))
+ pr_err("%s: fail to register mi2s dai driver\n", __func__);
+
+ rc2 = platform_driver_register(&msm_dai_q6_driver);
+
+ if (IS_ERR_VALUE(rc2))
+ pr_err("%s: fail to register mi2s dai driver\n", __func__);
+
+ return (IS_ERR_VALUE(rc1) && IS_ERR_VALUE(rc2)) ? -1 : 0;
+}
+module_init(msm_dai_q6_init);
+
+static void __exit msm_dai_q6_exit(void)
+{
+ platform_driver_unregister(&msm_dai_q6_driver);
+}
+module_exit(msm_dai_q6_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM DSP DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-dai-stub.c b/sound/soc/msm/msm-dai-stub.c
new file mode 100644
index 0000000..b2bfa2c
--- /dev/null
+++ b/sound/soc/msm/msm-dai-stub.c
@@ -0,0 +1,102 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+static int msm_dai_stub_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ pr_debug("%s:\n", __func__);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops msm_dai_stub_ops = {
+ .set_channel_map = msm_dai_stub_set_channel_map,
+};
+
+static struct snd_soc_dai_driver msm_dai_stub_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_stub_ops,
+};
+
+static __devinit int msm_dai_stub_dev_probe(struct platform_device *pdev)
+{
+ int rc = 0;
+
+ dev_dbg(&pdev->dev, "dev name %s\n", dev_name(&pdev->dev));
+
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_stub_dai);
+
+ return rc;
+}
+
+static __devexit int msm_dai_stub_dev_remove(struct platform_device *pdev)
+{
+ pr_debug("%s:\n", __func__);
+
+ snd_soc_unregister_dai(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver msm_dai_stub_driver = {
+ .probe = msm_dai_stub_dev_probe,
+ .remove = msm_dai_stub_dev_remove,
+ .driver = {
+ .name = "msm-dai-stub",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_dai_stub_init(void)
+{
+ pr_debug("%s:\n", __func__);
+
+ return platform_driver_register(&msm_dai_stub_driver);
+}
+module_init(msm_dai_stub_init);
+
+static void __exit msm_dai_stub_exit(void)
+{
+ pr_debug("%s:\n", __func__);
+
+ platform_driver_unregister(&msm_dai_stub_driver);
+}
+module_exit(msm_dai_stub_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM Stub DSP DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-dai.c b/sound/soc/msm/msm-dai.c
new file mode 100644
index 0000000..61e4675
--- /dev/null
+++ b/sound/soc/msm/msm-dai.c
@@ -0,0 +1,150 @@
+/* sound/soc/msm/msm-dai.c
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2011, Code Aurora Forum. All rights reserved.
+ *
+ * Derived from msm-pcm.c and msm7201.c.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include "msm-pcm.h"
+
+static struct snd_soc_dai_driver msm_pcm_codec_dais[] = {
+{
+ .name = "msm-codec-dai",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_max = USE_CHANNELS_MAX,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .formats = USE_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_max = USE_CHANNELS_MAX,
+ .rate_min = USE_RATE_MIN,
+ .rates = USE_RATE,
+ .formats = USE_FORMATS,
+ },
+},
+};
+
+static struct snd_soc_dai_driver msm_pcm_cpu_dais[] = {
+{
+ .name = "msm-cpu-dai",
+ .playback = {
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .formats = USE_FORMATS,
+ },
+ .capture = {
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .rate_min = USE_RATE_MIN,
+ .rates = USE_RATE,
+ .formats = USE_FORMATS,
+ },
+},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_msm = {
+ .compress_type = SND_SOC_FLAT_COMPRESSION,
+};
+
+static __devinit int asoc_msm_codec_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_msm,
+ msm_pcm_codec_dais, ARRAY_SIZE(msm_pcm_codec_dais));
+}
+
+static int __devexit asoc_msm_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static __devinit int asoc_pcm_cpu_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_dai(&pdev->dev, msm_pcm_cpu_dais);
+}
+
+static int __devexit asoc_pcm_cpu_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_codec_dai_driver = {
+ .probe = asoc_msm_codec_probe,
+ .remove = __devexit_p(asoc_msm_codec_remove),
+ .driver = {
+ .name = "msm-codec-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static struct platform_driver asoc_cpu_dai_driver = {
+ .probe = asoc_pcm_cpu_probe,
+ .remove = __devexit_p(asoc_pcm_cpu_remove),
+ .driver = {
+ .name = "msm-cpu-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_codec_dai_init(void)
+{
+ return platform_driver_register(&asoc_codec_dai_driver);
+}
+
+static void __exit msm_codec_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_codec_dai_driver);
+}
+
+static int __init msm_cpu_dai_init(void)
+{
+ return platform_driver_register(&asoc_cpu_dai_driver);
+}
+
+static void __exit msm_cpu_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_cpu_dai_driver);
+}
+
+module_init(msm_codec_dai_init);
+module_exit(msm_codec_dai_exit);
+module_init(msm_cpu_dai_init);
+module_exit(msm_cpu_dai_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM Codec/Cpu Dai driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-multi-ch-pcm-q6.c b/sound/soc/msm/msm-multi-ch-pcm-q6.c
new file mode 100644
index 0000000..734d34f
--- /dev/null
+++ b/sound/soc/msm/msm-multi-ch-pcm-q6.c
@@ -0,0 +1,804 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+
+#include "msm-pcm-q6.h"
+#include "msm-pcm-routing.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+struct snd_msm_volume {
+ struct msm_audio *prtd;
+ unsigned volume;
+};
+static struct snd_msm_volume multi_ch_pcm_audio = {NULL, 0x2000};
+
+#define PLAYBACK_NUM_PERIODS 8
+#define PLAYBACK_MAX_PERIOD_SIZE 4032
+#define PLAYBACK_MIN_PERIOD_SIZE 256
+#define CAPTURE_NUM_PERIODS 16
+#define CAPTURE_PERIOD_SIZE 320
+
+static struct snd_pcm_hardware msm_pcm_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = CAPTURE_NUM_PERIODS * CAPTURE_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_PERIOD_SIZE,
+ .periods_min = CAPTURE_NUM_PERIODS,
+ .periods_max = CAPTURE_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 6,
+ .buffer_bytes_max = PLAYBACK_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE,
+ .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
+ .periods_min = PLAYBACK_NUM_PERIODS,
+ .periods_max = PLAYBACK_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static uint32_t in_frame_info[CAPTURE_NUM_PERIODS][2];
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ int i = 0;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ pr_debug("%s\n", __func__);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag)
+ break;
+ if (q6asm_is_cpu_buf_avail_nolock(IN,
+ prtd->audio_client,
+ &size, &idx)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ }
+ break;
+ }
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+ pr_debug("token = 0x%08x\n", token);
+ for (i = 0; i < 8; i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ in_frame_info[token][0] = payload[2];
+ in_frame_info[token][1] = payload[3];
+ prtd->pcm_irq_pos += in_frame_info[token][0];
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag
+ && q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx))
+ q6asm_read_nolock(prtd->audio_client);
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK) {
+ atomic_set(&prtd->start, 1);
+ break;
+ }
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ atomic_set(&prtd->start, 1);
+ break;
+ default:
+ break;
+ }
+ }
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_multi_ch_pcm(prtd->audio_client,
+ runtime->rate, runtime->channels);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ int i = 0;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return 0;
+
+ pr_debug("Samp_rate = %d\n", prtd->samp_rate);
+ pr_debug("Channel = %d\n", prtd->channel_mode);
+ ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client, prtd->samp_rate,
+ prtd->channel_mode);
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+ prtd->periods = runtime->periods;
+
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ int ret = 0;
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_err("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_pcm_hardware_playback;
+ ret = q6asm_open_write(prtd->audio_client,
+ FORMAT_MULTI_CHANNEL_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_hardware_capture;
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm in open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_integer failed\n");
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PLAYBACK_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+ PLAYBACK_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+
+ prtd->dsp_cnt = 0;
+ runtime->private_data = prtd;
+ pr_debug("substream->pcm->device = %d\n", substream->pcm->device);
+ pr_debug("soc_prtd->dai_link->be_id = %d\n", soc_prtd->dai_link->be_id);
+ multi_ch_pcm_audio.prtd = prtd;
+ ret = multi_ch_pcm_set_volume(multi_ch_pcm_audio.volume);
+ if (ret < 0)
+ pr_err("%s : Set Volume failed : %d", __func__, ret);
+
+ ret = q6asm_set_softpause(multi_ch_pcm_audio.prtd->audio_client,
+ &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(multi_ch_pcm_audio.prtd->audio_client,
+ &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ return 0;
+}
+
+int multi_ch_pcm_set_volume(unsigned volume)
+{
+ int rc = 0;
+ pr_err("multi_ch_pcm_set_volume\n");
+
+ if (multi_ch_pcm_audio.prtd && multi_ch_pcm_audio.prtd->audio_client) {
+ pr_err("%s q6asm_set_volume\n", __func__);
+ rc = q6asm_set_volume(multi_ch_pcm_audio.prtd->audio_client,
+ volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed"
+ " rc=%d\n", __func__, rc);
+ }
+ }
+ multi_ch_pcm_audio.volume = volume;
+ return rc;
+}
+
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_err("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_err("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ xfer = fbytes;
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+fail:
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ ret = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (ret < 0)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ multi_ch_pcm_audio.prtd = NULL;
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%08x\n",
+ (unsigned int) data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = in_frame_info[idx][1];
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ in_frame_info[idx][1] += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&in_frame_info[idx], 0,
+ sizeof(uint32_t) * 2);
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+
+ if (dir == OUT) {
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ } else {
+ /*
+ *TODO : Need to Add Async IO changes. All period
+ * size might not be supported.
+ */
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ (params_buffer_bytes(params) / params_periods(params)),
+ params_periods(params));
+ }
+
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ if (dir == OUT)
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ else
+ dma_buf->bytes = params_buffer_bytes(params);
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-multi-ch-pcm-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("Multi channel PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-mvs.c b/sound/soc/msm/msm-mvs.c
new file mode 100644
index 0000000..2e7114c
--- /dev/null
+++ b/sound/soc/msm/msm-mvs.c
@@ -0,0 +1,936 @@
+/* Copyright (c) 2010, Code Aurora Forum. All rights reserved.
+ *
+ * All source code in this file is licensed under the following license except
+ * where indicated.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <linux/kthread.h>
+#include <linux/uaccess.h>
+#include <linux/mutex.h>
+#include <linux/wakelock.h>
+#include <mach/msm_rpcrouter.h>
+#include <mach/debug_mm.h>
+#include "msm_audio_mvs.h"
+
+
+static struct audio_mvs_info_type audio_mvs_info;
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000),
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MVS_MAX_VOC_PKT_SIZE * MVS_MAX_Q_LEN,
+ .period_bytes_min = MVS_MAX_VOC_PKT_SIZE,
+ .period_bytes_max = MVS_MAX_VOC_PKT_SIZE,
+ .periods_min = MVS_MAX_Q_LEN,
+ .periods_max = MVS_MAX_Q_LEN,
+ .fifo_size = 0,
+};
+
+static void snd_pcm_mvs_timer(unsigned long data)
+{
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ MM_DBG("%s\n", __func__);
+ if (audio->playback_start) {
+ if (audio->ack_dl_count) {
+ audio->pcm_playback_irq_pos += audio->pcm_count;
+ audio->ack_dl_count--;
+ snd_pcm_period_elapsed(audio->playback_substream);
+ }
+ }
+
+ if (audio->capture_start) {
+ if (audio->ack_ul_count) {
+ audio->pcm_capture_irq_pos += audio->pcm_capture_count;
+ audio->ack_ul_count--;
+ snd_pcm_period_elapsed(audio->capture_substream);
+ }
+ }
+ audio->timer.expires += audio->expiry_delta;
+ add_timer(&audio->timer);
+}
+
+static int audio_mvs_setup_mvs(struct audio_mvs_info_type *audio)
+{
+ int rc = 0;
+ struct audio_mvs_enable_msg enable_msg;
+ MM_DBG("%s\n", __func__);
+
+ /* Enable MVS. */
+
+ memset(&enable_msg, 0, sizeof(enable_msg));
+ audio->rpc_status = RPC_STATUS_FAILURE;
+ enable_msg.enable_args.client_id = cpu_to_be32(MVS_CLIENT_ID_VOIP);
+ enable_msg.enable_args.mode = cpu_to_be32(MVS_MODE_LINEAR_PCM);
+ enable_msg.enable_args.ul_cb_func_id = (int) NULL;
+ enable_msg.enable_args.dl_cb_func_id = (int) NULL;
+ enable_msg.enable_args.context = cpu_to_be32(MVS_PKT_CONTEXT_ISR);
+
+ msm_rpc_setup_req(&enable_msg.rpc_hdr, MVS_PROG,
+ MVS_VERS, MVS_ENABLE_PROC);
+
+ rc = msm_rpc_write(audio->rpc_endpt,
+ &enable_msg, sizeof(enable_msg));
+
+ if (rc >= 0) {
+ MM_DBG("RPC write for enable done\n");
+
+ rc = wait_event_timeout(audio->wait,
+ (audio->rpc_status !=
+ RPC_STATUS_FAILURE), 1 * HZ);
+
+ if (rc > 0) {
+ MM_DBG("Wait event for enable succeeded\n");
+
+ mutex_lock(&audio->lock);
+ audio->mvs_mode = MVS_MODE_LINEAR_PCM;
+ audio->frame_mode = MVS_FRAME_MODE_PCM_DL;
+ audio->pcm_frame = 0;
+ mutex_unlock(&audio->lock);
+ rc = 0;
+
+ } else
+ MM_ERR("Wait event for enable failed %d\n", rc);
+ } else
+ MM_ERR("RPC write for enable failed %d\n", rc);
+ return rc;
+}
+
+static void audio_mvs_rpc_reply(struct msm_rpc_endpoint *endpoint,
+ uint32_t xid)
+{
+ int rc = 0;
+ struct rpc_reply_hdr reply_hdr;
+ MM_DBG("%s\n", __func__);
+
+ memset(&reply_hdr, 0, sizeof(reply_hdr));
+ reply_hdr.xid = cpu_to_be32(xid);
+ reply_hdr.type = cpu_to_be32(RPC_TYPE_REPLY);
+ reply_hdr.reply_stat = cpu_to_be32(RPCMSG_REPLYSTAT_ACCEPTED);
+ reply_hdr.data.acc_hdr.accept_stat =
+ cpu_to_be32(RPC_ACCEPTSTAT_SUCCESS);
+ reply_hdr.data.acc_hdr.verf_flavor = 0;
+ reply_hdr.data.acc_hdr.verf_length = 0;
+
+ rc = msm_rpc_write(endpoint, &reply_hdr, sizeof(reply_hdr));
+
+ if (rc < 0)
+ MM_ERR("RPC write for response failed %d\n", rc);
+}
+
+static void audio_mvs_process_rpc_request(uint32_t procedure, uint32_t xid,
+ void *data, uint32_t length,
+ struct audio_mvs_info_type *audio)
+{
+
+ int rc = 0;
+ uint32_t index;
+ MM_DBG("%s\n", __func__);
+ switch (procedure) {
+ case MVS_EVENT_CB_TYPE_PROC:{
+ struct audio_mvs_cb_func_args *args = data;
+ uint32_t event_type = be32_to_cpu(args->event);
+ uint32_t cmd_status =
+ be32_to_cpu(args->
+ event_data.mvs_ev_command_type.cmd_status);
+ uint32_t mode_status =
+ be32_to_cpu(args->
+ event_data.mvs_ev_mode_type.mode_status);
+ audio_mvs_rpc_reply(audio->rpc_endpt, xid);
+ if (be32_to_cpu(args->valid_ptr)) {
+ if (event_type == AUDIO_MVS_COMMAND) {
+ if (cmd_status == AUDIO_MVS_CMD_SUCCESS)
+ audio->rpc_status = RPC_STATUS_SUCCESS;
+ wake_up(&audio->wait);
+ } else if (event_type == AUDIO_MVS_MODE) {
+ if (mode_status != AUDIO_MVS_MODE_NOT_AVAIL) {
+ audio->rpc_status =
+ RPC_STATUS_SUCCESS;
+ }
+ audio->prepare_ack++;
+ wake_up(&audio->wait);
+ wake_up(&audio->prepare_wait);
+ } else {
+ /*nothing to do */
+ }
+ } else
+ MM_ERR("ALSA: CB event pointer not valid\n");
+ break;
+ }
+ case MVS_PACKET_UL_FN_TYPE_PROC:{
+ uint32_t *cb_data = data;
+ uint32_t pkt_len ;
+ struct audio_mvs_ul_reply ul_reply;
+ MM_DBG("MVS_PACKET_UL_FN_TYPE_PROC\n");
+
+ memset(&ul_reply, 0, sizeof(ul_reply));
+ cb_data++;
+ pkt_len = be32_to_cpu(*cb_data);
+ cb_data++;
+ if (audio->capture_enable) {
+ audio_mvs_info.ack_ul_count++;
+ mutex_lock(&audio->out_lock);
+ index = audio->out_write % MVS_MAX_Q_LEN;
+ memcpy(audio->out[index].voc_pkt, cb_data,
+ pkt_len);
+ audio->out[index].len = pkt_len;
+ audio->out_write++;
+ mutex_unlock(&audio->out_lock);
+ }
+ MM_DBG(" audio->out_read = %d audio->out write = %d\n",
+ audio->out_read, audio->out_write);
+ ul_reply.reply_hdr.xid = cpu_to_be32(xid);
+ ul_reply.reply_hdr.type = cpu_to_be32(RPC_TYPE_REPLY);
+ ul_reply.reply_hdr.reply_stat =
+ cpu_to_be32(RPCMSG_REPLYSTAT_ACCEPTED);
+ ul_reply.reply_hdr.data.acc_hdr.accept_stat =
+ cpu_to_be32(RPC_ACCEPTSTAT_SUCCESS);
+ ul_reply.reply_hdr.data.acc_hdr.verf_flavor = 0;
+ ul_reply.reply_hdr.data.acc_hdr.verf_length = 0;
+ ul_reply.valid_pkt_status_ptr = cpu_to_be32(0x00000001);
+ ul_reply.pkt_status = cpu_to_be32(0x00000000);
+ rc = msm_rpc_write(audio->rpc_endpt, &ul_reply,
+ sizeof(ul_reply));
+ wake_up(&audio->out_wait);
+ if (rc < 0)
+ MM_ERR("RPC write for UL response failed %d\n",
+ rc);
+ break;
+ }
+ case MVS_PACKET_DL_FN_TYPE_PROC:{
+ struct audio_mvs_dl_reply dl_reply;
+ MM_DBG("MVS_PACKET_DL_FN_TYPE_PROC\n");
+ memset(&dl_reply, 0, sizeof(dl_reply));
+ dl_reply.reply_hdr.xid = cpu_to_be32(xid);
+ dl_reply.reply_hdr.type = cpu_to_be32(RPC_TYPE_REPLY);
+ dl_reply.reply_hdr.reply_stat =
+ cpu_to_be32(RPCMSG_REPLYSTAT_ACCEPTED);
+ dl_reply.reply_hdr.data.acc_hdr.accept_stat =
+ cpu_to_be32(RPC_ACCEPTSTAT_SUCCESS);
+ dl_reply.reply_hdr.data.acc_hdr.verf_flavor = 0;
+ dl_reply.reply_hdr.data.acc_hdr.verf_length = 0;
+ mutex_lock(&audio->in_lock);
+ if (audio->in_read < audio->in_write
+ && audio->dl_play) {
+ index = audio->in_read % MVS_MAX_Q_LEN;
+ memcpy(&dl_reply.voc_pkt,
+ audio->in[index].voc_pkt,
+ audio->in[index].len);
+ audio->in_read++;
+ audio_mvs_info.ack_dl_count++;
+ dl_reply.pkt_status =
+ cpu_to_be32(AUDIO_MVS_PKT_NORMAL);
+ wake_up(&audio->in_wait);
+ } else {
+ dl_reply.pkt_status =
+ cpu_to_be32(AUDIO_MVS_PKT_SLOW);
+ }
+ mutex_unlock(&audio->in_lock);
+ MM_DBG(" audio->in_read = %d audio->in write = %d\n",
+ audio->in_read, audio->in_write);
+ dl_reply.valid_frame_info_ptr = cpu_to_be32(0x00000001);
+ dl_reply.frame_mode = cpu_to_be32(audio->frame_mode);
+ dl_reply.frame_mode_again =
+ cpu_to_be32(audio->frame_mode);
+ dl_reply.frame_info_hdr.frame_mode =
+ cpu_to_be32(audio->frame_mode);
+ dl_reply.frame_info_hdr.mvs_mode =
+ cpu_to_be32(audio->mvs_mode);
+ dl_reply.frame_info_hdr.buf_free_cnt = 0;
+ dl_reply.pcm_frame = cpu_to_be32(audio->pcm_frame);
+ dl_reply.pcm_mode = cpu_to_be32(audio->pcm_mode);
+ dl_reply.valid_pkt_status_ptr = cpu_to_be32(0x00000001);
+ rc = msm_rpc_write(audio->rpc_endpt, &dl_reply,
+ sizeof(dl_reply));
+ if (rc < 0)
+ MM_ERR("RPC write for DL response failed %d\n",
+ rc);
+ break;
+ }
+ default:
+ MM_ERR("Unknown CB type %d\n", procedure);
+ }
+}
+
+static int audio_mvs_thread(void *data)
+{
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ struct rpc_request_hdr *rpc_hdr = NULL;
+ struct rpc_reply_hdr *rpc_reply = NULL;
+ uint32_t reply_status = 0;
+ uint32_t rpc_type;
+ int rpc_hdr_len;
+ MM_DBG("%s\n", __func__);
+
+ while (!kthread_should_stop()) {
+ rpc_hdr_len =
+ msm_rpc_read(audio->rpc_endpt, (void **)&rpc_hdr, -1, -1);
+ if (rpc_hdr_len < 0) {
+ MM_ERR("RPC read failed %d\n", rpc_hdr_len);
+ break;
+ } else if (rpc_hdr_len < RPC_COMMON_HDR_SZ)
+ continue;
+ else {
+ rpc_type = be32_to_cpu(rpc_hdr->type);
+ if (rpc_type == RPC_TYPE_REPLY) {
+ if (rpc_hdr_len < RPC_REPLY_HDR_SZ)
+ continue;
+ rpc_reply = (void *)rpc_hdr;
+ reply_status = be32_to_cpu(rpc_reply->
+ reply_stat);
+ if (reply_status != RPCMSG_REPLYSTAT_ACCEPTED) {
+ /* If the command is not accepted,
+ * there will be no response callback.
+ * Wake the caller and report error. */
+ audio->rpc_status = RPC_STATUS_REJECT;
+ wake_up(&audio->wait);
+ MM_ERR("RPC reply status denied\n");
+ }
+ } else if (rpc_type == RPC_TYPE_REQUEST) {
+ if (rpc_hdr_len < RPC_REQUEST_HDR_SZ)
+ continue;
+ MM_DBG("ALSA: kthread call procedure\n");
+ audio_mvs_process_rpc_request(
+ be32_to_cpu(rpc_hdr->procedure),
+ be32_to_cpu(rpc_hdr->xid),
+ (void *)(rpc_hdr + 1),
+ (rpc_hdr_len - sizeof(*rpc_hdr)),
+ audio);
+ } else
+ MM_ERR("Unexpected RPC type %d\n", rpc_type);
+ }
+ kfree(rpc_hdr);
+ rpc_hdr = NULL;
+ }
+ return 0;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ MM_DBG("%s\n", __func__);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio->playback_start = 1;
+ else
+ audio->capture_start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio->playback_start = 0;
+ else
+ audio->capture_start = 0;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+
+ MM_DBG("%s\n", __func__);
+ mutex_lock(&audio->lock);
+ if (audio->state < AUDIO_MVS_OPENED) {
+ audio->rpc_endpt =
+ msm_rpc_connect_compatible(MVS_PROG,
+ MVS_VERS,
+ MSM_RPC_UNINTERRUPTIBLE);
+ audio->state = AUDIO_MVS_OPENED;
+ }
+
+ if (IS_ERR(audio->rpc_endpt)) {
+ MM_ERR("ALSA MVS RPC connect failed with version 0x%x\n",
+ MVS_VERS);
+ ret = PTR_ERR(audio->rpc_endpt);
+ audio->rpc_endpt = NULL;
+ goto err;
+ } else {
+ MM_DBG("ALSA MVS RPC connect succeeded\n");
+ if (audio->playback_substream == NULL ||
+ audio->capture_substream == NULL) {
+ if (substream->stream ==
+ SNDRV_PCM_STREAM_PLAYBACK) {
+ audio->playback_substream =
+ substream;
+ runtime->hw = msm_pcm_hardware;
+ } else if (substream->stream ==
+ SNDRV_PCM_STREAM_CAPTURE) {
+ audio->capture_substream =
+ substream;
+ runtime->hw = msm_pcm_hardware;
+ }
+ } else {
+ ret = -EPERM;
+ goto err;
+ }
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ MM_ERR("snd_pcm_hw_constraint_integer failed\n");
+ if (!audio->instance) {
+ msm_rpc_close(audio->rpc_endpt);
+ audio->rpc_endpt = NULL;
+ }
+ goto err;
+ }
+ audio->instance++;
+ }
+err:
+ mutex_unlock(&audio->lock);
+ return ret;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int rc = 0;
+ int count = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ uint32_t index;
+ MM_DBG("%s\n", __func__);
+ if (audio->dl_play == 1) {
+ rc = wait_event_interruptible_timeout(audio->in_wait,
+ (audio->in_write - audio->in_read <= 3),
+ 100 * HZ);
+ if (!rc) {
+ MM_ERR("MVS: write time out\n");
+ return -ETIMEDOUT;
+ } else if (rc < 0) {
+ MM_ERR("MVS: write was interrupted\n");
+ return -ERESTARTSYS;
+ }
+ }
+ mutex_lock(&audio->in_lock);
+ if (audio->state == AUDIO_MVS_ENABLED) {
+ index = audio->in_write % MVS_MAX_Q_LEN;
+ count = frames_to_bytes(runtime, frames);
+ if (count <= MVS_MAX_VOC_PKT_SIZE) {
+ rc = copy_from_user(audio->in[index].voc_pkt, buf,
+ count);
+ } else
+ rc = -ENOMEM;
+ if (!rc) {
+ audio->in[index].len = count;
+ audio->in_write++;
+ rc = count;
+ if (audio->in_write >= 3)
+ audio->dl_play = 1;
+ } else {
+ MM_ERR("Copy from user returned %d\n", rc);
+ rc = -EFAULT;
+ }
+
+ } else {
+ MM_ERR("Write performed in invalid state %d\n",
+ audio->state);
+ rc = -EINVAL;
+ }
+ mutex_unlock(&audio->in_lock);
+ return rc;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff,
+ void __user *buf, snd_pcm_uframes_t frames)
+{
+ int rc = 0;
+ int count = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ uint32_t index = 0;
+
+ MM_DBG("%s\n", __func__);
+
+ /* Ensure the driver has been enabled. */
+ if (audio->state != AUDIO_MVS_ENABLED) {
+ MM_ERR("Read performed in invalid state %d\n", audio->state);
+ return -EPERM;
+ }
+ rc = wait_event_interruptible_timeout(audio->out_wait,
+ (audio->out_read < audio->out_write ||
+ audio->state == AUDIO_MVS_CLOSING ||
+ audio->state == AUDIO_MVS_CLOSED),
+ 100 * HZ);
+ if (!rc) {
+ MM_ERR("MVS: No UL data available\n");
+ return -ETIMEDOUT;
+ } else if (rc < 0) {
+ MM_ERR("MVS: Read was interrupted\n");
+ return -ERESTARTSYS;
+ }
+
+ mutex_lock(&audio->out_lock);
+ if (audio->state == AUDIO_MVS_CLOSING
+ || audio->state == AUDIO_MVS_CLOSED) {
+ rc = -EBUSY;
+ } else {
+ count = frames_to_bytes(runtime, frames);
+ index = audio->out_read % MVS_MAX_Q_LEN;
+ if (audio->out[index].len <= count) {
+ rc = copy_to_user(buf,
+ audio->out[index].voc_pkt,
+ audio->out[index].len);
+ if (rc == 0) {
+ rc = audio->out[index].len;
+ audio->out_read++;
+ } else {
+ MM_ERR("Copy to user %d\n", rc);
+ rc = -EFAULT;
+ }
+ } else
+ rc = -ENOMEM;
+ }
+ mutex_unlock(&audio->out_lock);
+ return rc;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ MM_DBG("%s\n", __func__);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ struct audio_mvs_release_msg release_msg;
+ MM_DBG("%s\n", __func__);
+ memset(&release_msg, 0, sizeof(release_msg));
+ mutex_lock(&audio->lock);
+
+ audio->instance--;
+ wake_up(&audio->out_wait);
+
+ if (!audio->instance) {
+ if (audio->state == AUDIO_MVS_ENABLED) {
+ audio->state = AUDIO_MVS_CLOSING;
+ /* Release MVS. */
+ release_msg.client_id = cpu_to_be32(MVS_CLIENT_ID_VOIP);
+ msm_rpc_setup_req(&release_msg.rpc_hdr, audio->rpc_prog,
+ audio->rpc_ver,
+ MVS_RELEASE_PROC);
+ audio->rpc_status = RPC_STATUS_FAILURE;
+ rc = msm_rpc_write(audio->rpc_endpt, &release_msg,
+ sizeof(release_msg));
+ if (rc >= 0) {
+ MM_DBG("RPC write for release done\n");
+ rc = wait_event_timeout(audio->wait,
+ (audio->rpc_status !=
+ RPC_STATUS_FAILURE), 1 * HZ);
+ if (rc != 0) {
+ MM_DBG
+ ("Wait event for release succeeded\n");
+ rc = 0;
+ kthread_stop(audio->task);
+ audio->prepare_ack = 0;
+ audio->task = NULL;
+ del_timer_sync(&audio->timer);
+ } else {
+ MM_ERR
+ ("Wait event for release failed %d\n",
+ rc);
+ }
+ } else {
+ MM_ERR("RPC write for release failed %d\n", rc);
+ }
+ }
+ audio->state = AUDIO_MVS_CLOSED;
+ msm_rpc_close(audio->rpc_endpt);
+ audio->rpc_endpt = NULL;
+ }
+
+ mutex_unlock(&audio->lock);
+
+ wake_unlock(&audio->suspend_lock);
+ wake_unlock(&audio->idle_lock);
+ /* Release the IO buffers. */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mutex_lock(&audio->in_lock);
+ audio->in_write = 0;
+ audio->in_read = 0;
+ audio->playback_enable = 0;
+ audio->dl_play = 0;
+ audio->ack_dl_count = 0;
+ memset(audio->in[0].voc_pkt, 0,
+ MVS_MAX_VOC_PKT_SIZE * MVS_MAX_Q_LEN);
+ audio->in->len = 0;
+ audio->playback_substream = NULL;
+ mutex_unlock(&audio->in_lock);
+ } else {
+ mutex_lock(&audio->out_lock);
+ audio->out_write = 0;
+ audio->out_read = 0;
+ audio->capture_enable = 0;
+ audio->ack_ul_count = 0;
+ memset(audio->out[0].voc_pkt, 0,
+ MVS_MAX_VOC_PKT_SIZE * MVS_MAX_Q_LEN);
+ audio->out->len = 0;
+ audio->capture_substream = NULL;
+ mutex_unlock(&audio->out_lock);
+ }
+ return rc;
+}
+
+static int msm_mvs_pcm_setup(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct audio_mvs_acquire_msg acquire_msg;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+ memset(&acquire_msg, 0, sizeof(acquire_msg));
+
+ /*Create an Kthread */
+ MM_DBG("ALSA MVS thread creating\n");
+ if (!IS_ERR(audio->rpc_endpt)) {
+ audio->task =
+ kthread_run(audio_mvs_thread, audio,
+ "audio_alsa_mvs_thread");
+ if (!IS_ERR(audio->task)) {
+ MM_DBG("ALSA MVS thread create succeeded\n");
+ audio->rpc_prog = MVS_PROG;
+ audio->rpc_ver = MVS_VERS;
+ /* Acquire MVS. */
+ acquire_msg.acquire_args.client_id =
+ cpu_to_be32(MVS_CLIENT_ID_VOIP);
+ acquire_msg.acquire_args.cb_func_id =
+ cpu_to_be32(MVS_CB_FUNC_ID);
+ msm_rpc_setup_req(&acquire_msg.rpc_hdr,
+ audio->rpc_prog,
+ audio->rpc_ver,
+ MVS_ACQUIRE_PROC);
+ audio->rpc_status = RPC_STATUS_FAILURE;
+ rc = msm_rpc_write(audio->rpc_endpt,
+ &acquire_msg, sizeof(acquire_msg));
+ if (rc >= 0) {
+ MM_DBG("RPC write for acquire done\n");
+
+ rc = wait_event_timeout(audio->wait,
+ (audio->rpc_status !=
+ RPC_STATUS_FAILURE),
+ 1 * HZ);
+ if (rc != 0) {
+ audio->state =
+ AUDIO_MVS_ACQUIRE;
+ rc = 0;
+ MM_DBG
+ ("MVS driver in acquire state\n");
+ } else {
+ MM_ERR
+ ("acquire Wait event failed %d\n",
+ rc);
+ rc = -EBUSY;
+ }
+ } else {
+ MM_ERR("RPC write for acquire failed %d\n",
+ rc);
+ rc = -EBUSY;
+ }
+ } else {
+ MM_ERR("ALSA MVS thread create failed\n");
+ rc = PTR_ERR(audio->task);
+ audio->task = NULL;
+ msm_rpc_close(audio->rpc_endpt);
+ audio->rpc_endpt = NULL;
+ }
+ } else {
+ MM_ERR("RPC connect is not setup with version 0x%x\n",
+ MVS_VERS);
+ rc = PTR_ERR(audio->rpc_endpt);
+ audio->rpc_endpt = NULL;
+ }
+ /*mvs mode setup */
+ if (audio->state == AUDIO_MVS_ACQUIRE)
+ rc = audio_mvs_setup_mvs(audio);
+ else
+ rc = -EBUSY;
+ return rc;
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct audio_mvs_info_type *prtd = &audio_mvs_info;
+ MM_DBG("%s\n", __func__);
+ prtd->pcm_playback_irq_pos = 0;
+ prtd->pcm_playback_buf_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->playback_enable = 1;
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct audio_mvs_info_type *prtd = &audio_mvs_info;
+ prtd->pcm_capture_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_capture_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_capture_irq_pos = 0;
+ prtd->pcm_capture_buf_pos = 0;
+ prtd->capture_enable = 1;
+ return 0;
+}
+
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_mvs_info_type *prtd = &audio_mvs_info;
+ unsigned long expiry = 0;
+ MM_DBG("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+
+ mutex_lock(&prtd->prepare_lock);
+ if (prtd->state == AUDIO_MVS_ENABLED)
+ goto enabled;
+ else if (prtd->state == AUDIO_MVS_PREPARING)
+ goto prepairing;
+ else if (prtd->state == AUDIO_MVS_OPENED) {
+ prtd->state = AUDIO_MVS_PREPARING;
+ rc = msm_mvs_pcm_setup(substream);
+ }
+ if (!rc) {
+ expiry = ((unsigned long)((prtd->pcm_count * 1000)
+ /(runtime->rate * runtime->channels * 2)));
+ expiry -= (expiry % 10);
+ prtd->timer.expires = jiffies + (msecs_to_jiffies(expiry));
+ prtd->expiry_delta = (msecs_to_jiffies(expiry));
+ if (prtd->expiry_delta <= 2)
+ prtd->expiry_delta = 1;
+ setup_timer(&prtd->timer, snd_pcm_mvs_timer,
+ (unsigned long)prtd);
+ prtd->ack_ul_count = 0;
+ prtd->ack_dl_count = 0;
+ add_timer(&prtd->timer);
+
+ } else {
+ MM_ERR("ALSA MVS setup is not done");
+ rc = -EPERM;
+ prtd->state = AUDIO_MVS_OPENED;
+ goto err;
+ }
+
+prepairing:
+ rc = wait_event_interruptible(prtd->prepare_wait,
+ (prtd->prepare_ack == 2));
+ if (rc < 0) {
+ MM_ERR("Wait event for prepare faild rc %d", rc);
+ rc = -EINTR;
+ prtd->state = AUDIO_MVS_OPENED;
+ goto err;
+ } else
+ MM_DBG("Wait event for prepare succeeded\n");
+
+ prtd->state = AUDIO_MVS_ENABLED;
+enabled:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ rc = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ rc = msm_pcm_capture_prepare(substream);
+err:
+ mutex_unlock(&prtd->prepare_lock);
+ return rc;
+}
+
+int msm_mvs_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ MM_DBG("%s\n", __func__);
+ if (substream->pcm->device & 1) {
+ runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED;
+ runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED;
+ }
+ return 0;
+}
+
+static snd_pcm_uframes_t
+msm_pcm_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+
+ if (audio->pcm_playback_irq_pos >= audio->pcm_size)
+ audio->pcm_playback_irq_pos = 0;
+ return bytes_to_frames(runtime, (audio->pcm_playback_irq_pos));
+}
+
+static snd_pcm_uframes_t
+msm_pcm_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_mvs_info_type *audio = &audio_mvs_info;
+
+ if (audio->pcm_capture_irq_pos >= audio->pcm_capture_size)
+ audio->pcm_capture_irq_pos = 0;
+ return bytes_to_frames(runtime, (audio->pcm_capture_irq_pos));
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t ret = 0;
+ MM_DBG("%s\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_pointer(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_pointer(substream);
+ return ret;
+}
+
+static struct snd_pcm_ops msm_mvs_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_mvs_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+
+};
+
+static int msm_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ int i, ret, offset = 0;
+ struct snd_pcm *pcm = rtd->pcm;
+
+ audio_mvs_info.mem_chunk = kmalloc(
+ 2 * MVS_MAX_VOC_PKT_SIZE * MVS_MAX_Q_LEN, GFP_KERNEL);
+ if (audio_mvs_info.mem_chunk != NULL) {
+ audio_mvs_info.in_read = 0;
+ audio_mvs_info.in_write = 0;
+ audio_mvs_info.out_read = 0;
+ audio_mvs_info.out_write = 0;
+ for (i = 0; i < MVS_MAX_Q_LEN; i++) {
+ audio_mvs_info.in[i].voc_pkt =
+ audio_mvs_info.mem_chunk + offset;
+ offset = offset + MVS_MAX_VOC_PKT_SIZE;
+ }
+ for (i = 0; i < MVS_MAX_Q_LEN; i++) {
+ audio_mvs_info.out[i].voc_pkt =
+ audio_mvs_info.mem_chunk + offset;
+ offset = offset + MVS_MAX_VOC_PKT_SIZE;
+ }
+ audio_mvs_info.playback_substream = NULL;
+ audio_mvs_info.capture_substream = NULL;
+ } else {
+ MM_ERR("MSM MVS kmalloc failed\n");
+ return -ENODEV;
+ }
+
+
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, 1);
+ if (ret)
+ return ret;
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+ if (ret)
+ return ret;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &msm_mvs_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &msm_mvs_pcm_ops);
+
+ return 0;
+}
+
+struct snd_soc_platform_driver msm_mvs_soc_platform = {
+ .ops = &msm_mvs_pcm_ops,
+ .pcm_new = msm_pcm_new,
+};
+EXPORT_SYMBOL(msm_mvs_soc_platform);
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_mvs_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-mvs-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_mvs_soc_platform_init(void)
+{
+ memset(&audio_mvs_info, 0, sizeof(audio_mvs_info));
+ mutex_init(&audio_mvs_info.lock);
+ mutex_init(&audio_mvs_info.prepare_lock);
+ mutex_init(&audio_mvs_info.in_lock);
+ mutex_init(&audio_mvs_info.out_lock);
+ init_waitqueue_head(&audio_mvs_info.wait);
+ init_waitqueue_head(&audio_mvs_info.prepare_wait);
+ init_waitqueue_head(&audio_mvs_info.out_wait);
+ init_waitqueue_head(&audio_mvs_info.in_wait);
+ wake_lock_init(&audio_mvs_info.suspend_lock, WAKE_LOCK_SUSPEND,
+ "audio_mvs_suspend");
+ wake_lock_init(&audio_mvs_info.idle_lock, WAKE_LOCK_IDLE,
+ "audio_mvs_idle");
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_mvs_soc_platform_init);
+
+static void __exit msm_mvs_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_mvs_soc_platform_exit);
+
+MODULE_DESCRIPTION("MVS PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-afe.c b/sound/soc/msm/msm-pcm-afe.c
new file mode 100644
index 0000000..482cbee
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-afe.c
@@ -0,0 +1,621 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License version 2 and
+* only version 2 as published by the Free Software Foundation.
+*
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*/
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6adm.h>
+#include <asm/dma.h>
+#include <linux/memory_alloc.h>
+#include <mach/msm_subsystem_map.h>
+#include "msm-pcm-afe.h"
+
+#define MIN_PERIOD_SIZE (128 * 2)
+#define MAX_PERIOD_SIZE (128 * 2 * 2 * 6)
+static struct snd_pcm_hardware msm_afe_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_PERIOD_SIZE * 32,
+ .period_bytes_min = MIN_PERIOD_SIZE,
+ .period_bytes_max = MAX_PERIOD_SIZE,
+ .periods_min = 32,
+ .periods_max = 384,
+ .fifo_size = 0,
+};
+static enum hrtimer_restart afe_hrtimer_callback(struct hrtimer *hrt);
+static enum hrtimer_restart afe_hrtimer_rec_callback(struct hrtimer *hrt);
+
+static enum hrtimer_restart afe_hrtimer_callback(struct hrtimer *hrt)
+{
+ struct pcm_afe_info *prtd =
+ container_of(hrt, struct pcm_afe_info, hrt);
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (prtd->start) {
+ pr_debug("sending frame to DSP: poll_time: %d\n",
+ prtd->poll_time);
+ if (prtd->dsp_cnt == runtime->periods)
+ prtd->dsp_cnt = 0;
+ afe_rt_proxy_port_write(
+ (prtd->dma_addr +
+ (prtd->dsp_cnt *
+ snd_pcm_lib_period_bytes(prtd->substream))),
+ snd_pcm_lib_period_bytes(prtd->substream));
+ prtd->dsp_cnt++;
+ hrtimer_forward_now(hrt, ns_to_ktime(prtd->poll_time
+ * 1000));
+
+ return HRTIMER_RESTART;
+ } else
+ return HRTIMER_NORESTART;
+}
+static enum hrtimer_restart afe_hrtimer_rec_callback(struct hrtimer *hrt)
+{
+ struct pcm_afe_info *prtd =
+ container_of(hrt, struct pcm_afe_info, hrt);
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ if (prtd->start) {
+ if (prtd->dsp_cnt == runtime->periods)
+ prtd->dsp_cnt = 0;
+ afe_rt_proxy_port_read(
+ (prtd->dma_addr + (prtd->dsp_cnt
+ * snd_pcm_lib_period_bytes(prtd->substream))),
+ snd_pcm_lib_period_bytes(prtd->substream));
+ prtd->dsp_cnt++;
+ pr_debug("sending frame rec to DSP: poll_time: %d\n",
+ prtd->poll_time);
+ hrtimer_forward_now(hrt, ns_to_ktime(prtd->poll_time
+ * 1000));
+
+ return HRTIMER_RESTART;
+ } else
+ return HRTIMER_NORESTART;
+}
+static void pcm_afe_process_tx_pkt(uint32_t opcode,
+ uint32_t token, uint32_t *payload,
+ void *priv)
+{
+ struct pcm_afe_info *prtd = priv;
+ unsigned long dsp_flags;
+ struct snd_pcm_substream *substream = NULL;
+ struct snd_pcm_runtime *runtime = NULL;
+ uint16_t event;
+
+ if (prtd == NULL)
+ return;
+ substream = prtd->substream;
+ runtime = substream->runtime;
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&prtd->dsp_lock, dsp_flags);
+ switch (opcode) {
+ case AFE_EVENT_RT_PROXY_PORT_STATUS: {
+ event = (uint16_t)((0xFFFF0000 & payload[0]) >> 0x10);
+ switch (event) {
+ case AFE_EVENT_RTPORT_START: {
+ prtd->dsp_cnt = 0;
+ prtd->poll_time = ((unsigned long)((
+ snd_pcm_lib_period_bytes
+ (prtd->substream) *
+ 1000 * 1000)/
+ (runtime->rate *
+ runtime->channels * 2)));
+ pr_debug("prtd->poll_time: %d",
+ prtd->poll_time);
+ hrtimer_start(&prtd->hrt,
+ ns_to_ktime(0),
+ HRTIMER_MODE_REL);
+ break;
+ }
+ case AFE_EVENT_RTPORT_STOP:
+ pr_debug("%s: event!=0\n", __func__);
+ prtd->start = 0;
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ break;
+ case AFE_EVENT_RTPORT_LOW_WM:
+ pr_debug("%s: Underrun\n", __func__);
+ break;
+ case AFE_EVENT_RTPORT_HI_WM:
+ pr_debug("%s: Overrun\n", __func__);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case AFE_SERVICE_CMD_RTPORT_WR:
+ pr_debug("write done\n");
+ prtd->pcm_irq_pos += snd_pcm_lib_period_bytes
+ (prtd->substream);
+ snd_pcm_period_elapsed(prtd->substream);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ break;
+ }
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+}
+
+static void pcm_afe_process_rx_pkt(uint32_t opcode,
+ uint32_t token, uint32_t *payload,
+ void *priv)
+{
+ struct pcm_afe_info *prtd = priv;
+ unsigned long dsp_flags;
+ struct snd_pcm_substream *substream = NULL;
+ struct snd_pcm_runtime *runtime = NULL;
+ uint16_t event;
+
+ if (prtd == NULL)
+ return;
+ substream = prtd->substream;
+ runtime = substream->runtime;
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&prtd->dsp_lock, dsp_flags);
+ switch (opcode) {
+ case AFE_EVENT_RT_PROXY_PORT_STATUS: {
+ event = (uint16_t)((0xFFFF0000 & payload[0]) >> 0x10);
+ switch (event) {
+ case AFE_EVENT_RTPORT_START: {
+ prtd->dsp_cnt = 0;
+ prtd->poll_time = ((unsigned long)((
+ snd_pcm_lib_period_bytes(prtd->substream)
+ * 1000 * 1000)/(runtime->rate
+ * runtime->channels * 2)));
+ hrtimer_start(&prtd->hrt,
+ ns_to_ktime(0),
+ HRTIMER_MODE_REL);
+ pr_debug("prtd->poll_time : %d", prtd->poll_time);
+ break;
+ }
+ case AFE_EVENT_RTPORT_STOP:
+ pr_debug("%s: event!=0\n", __func__);
+ prtd->start = 0;
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ break;
+ case AFE_EVENT_RTPORT_LOW_WM:
+ pr_debug("%s: Underrun\n", __func__);
+ break;
+ case AFE_EVENT_RTPORT_HI_WM:
+ pr_debug("%s: Overrun\n", __func__);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case AFE_SERVICE_CMD_RTPORT_RD:
+ pr_debug("Read done\n");
+ prtd->pcm_irq_pos += snd_pcm_lib_period_bytes
+ (prtd->substream);
+ snd_pcm_period_elapsed(prtd->substream);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ break;
+ }
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+}
+
+static int msm_afe_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ int ret = 0;
+
+ pr_debug("%s: sample_rate=%d\n", __func__, runtime->rate);
+
+ pr_debug("%s: dai->id =%x\n", __func__, dai->id);
+ ret = afe_register_get_events(dai->id,
+ pcm_afe_process_tx_pkt, prtd);
+ if (ret < 0) {
+ pr_err("afe-pcm:register for events failed\n");
+ return ret;
+ }
+ pr_debug("%s:success\n", __func__);
+ prtd->prepared++;
+ return ret;
+}
+
+static int msm_afe_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ pr_debug("%s: dai->id =%x\n", __func__, dai->id);
+ ret = afe_register_get_events(dai->id,
+ pcm_afe_process_rx_pkt, prtd);
+ if (ret < 0) {
+ pr_err("afe-pcm:register for events failed\n");
+ return ret;
+ }
+ pr_debug("%s:success\n", __func__);
+ prtd->prepared++;
+ return 0;
+}
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 16000, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static int msm_afe_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = NULL;
+ int ret = 0;
+
+ prtd = kzalloc(sizeof(struct pcm_afe_info), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ } else
+ pr_debug("prtd %x\n", (unsigned int)prtd);
+
+ mutex_init(&prtd->lock);
+ spin_lock_init(&prtd->dsp_lock);
+ prtd->dsp_cnt = 0;
+
+ mutex_lock(&prtd->lock);
+
+ runtime->hw = msm_afe_hardware;
+ prtd->substream = substream;
+ runtime->private_data = prtd;
+ mutex_unlock(&prtd->lock);
+ hrtimer_init(&prtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->hrt.function = afe_hrtimer_callback;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ prtd->hrt.function = afe_hrtimer_rec_callback;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_integer failed\n");
+
+ return 0;
+}
+
+static int msm_afe_close(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct snd_dma_buffer *dma_buf;
+ struct snd_pcm_runtime *runtime;
+ struct pcm_afe_info *prtd;
+ struct snd_soc_pcm_runtime *rtd = NULL;
+ struct snd_soc_dai *dai = NULL;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ if (substream == NULL) {
+ pr_err("substream is NULL\n");
+ return -EINVAL;
+ }
+ rtd = substream->private_data;
+ dai = rtd->cpu_dai;
+ runtime = substream->runtime;
+ prtd = runtime->private_data;
+
+ mutex_lock(&prtd->lock);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = afe_unregister_get_events(dai->id);
+ if (ret < 0)
+ pr_err("AFE unregister for events failed\n");
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = afe_unregister_get_events(dai->id);
+ if (ret < 0)
+ pr_err("AFE unregister for events failed\n");
+ }
+ hrtimer_cancel(&prtd->hrt);
+
+ rc = afe_cmd_memory_unmap(runtime->dma_addr);
+ if (rc < 0)
+ pr_err("AFE memory unmap failed\n");
+
+ pr_debug("release all buffer\n");
+ dma_buf = &substream->dma_buffer;
+ if (dma_buf == NULL) {
+ pr_debug("dma_buf is NULL\n");
+ goto done;
+ }
+
+ if (dma_buf->area) {
+ if (msm_subsystem_unmap_buffer(prtd->mem_buffer) < 0) {
+ pr_err("%s: unmap buffer failed\n", __func__);
+ prtd->mem_buffer = NULL;
+ dma_buf->area = NULL;
+ }
+ }
+
+ if (dma_buf->addr)
+ free_contiguous_memory_by_paddr(dma_buf->addr);
+done:
+ pr_debug("%s: dai->id =%x\n", __func__, dai->id);
+ mutex_unlock(&prtd->lock);
+ prtd->prepared--;
+ kfree(prtd);
+ return 0;
+}
+static int msm_afe_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ prtd->pcm_irq_pos = 0;
+ if (prtd->prepared)
+ return 0;
+ mutex_lock(&prtd->lock);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_afe_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_afe_capture_prepare(substream);
+ mutex_unlock(&prtd->lock);
+ return ret;
+}
+static int msm_afe_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ int result = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+ return result;
+}
+static int msm_afe_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__);
+ prtd->start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("%s: SNDRV_PCM_TRIGGER_STOP\n", __func__);
+ prtd->start = 0;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+static int msm_afe_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ int rc;
+ unsigned int flags = 0;
+
+ pr_debug("%s:\n", __func__);
+
+ mutex_lock(&prtd->lock);
+
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+
+ dma_buf->addr = allocate_contiguous_ebi_nomap(
+ runtime->hw.buffer_bytes_max, SZ_4K);
+ if (!dma_buf->addr) {
+ pr_err("%s:MSM AFE physical memory allocation failed\n",
+ __func__);
+ mutex_unlock(&prtd->lock);
+ return -ENOMEM;
+ }
+
+ flags = MSM_SUBSYSTEM_MAP_KADDR | MSM_SUBSYSTEM_MAP_CACHED;
+
+ prtd->mem_buffer = msm_subsystem_map_buffer(dma_buf->addr,
+ runtime->hw.buffer_bytes_max, flags,
+ NULL, 0);
+ if (IS_ERR((void *) prtd->mem_buffer)) {
+ pr_err("%s: map_buffer failed error = %ld\n", __func__,
+ PTR_ERR((void *)prtd->mem_buffer));
+ free_contiguous_memory_by_paddr(dma_buf->addr);
+ mutex_unlock(&prtd->lock);
+ return -ENOMEM;
+ }
+
+ dma_buf->area = prtd->mem_buffer->vaddr;
+
+ pr_debug("%s: dma_buf->area: 0x%p, dma_buf->addr: 0x%x", __func__,
+ (unsigned int *) dma_buf->area, dma_buf->addr);
+
+ if (!dma_buf->area) {
+ pr_err("%s: Invalid Virtual address\n", __func__);
+ if (prtd->mem_buffer) {
+ msm_subsystem_unmap_buffer(prtd->mem_buffer);
+ prtd->mem_buffer = NULL;
+ dma_buf->area = NULL;
+ }
+ free_contiguous_memory_by_paddr(dma_buf->addr);
+ mutex_unlock(&prtd->lock);
+ return -ENOMEM;
+ }
+
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ memset(dma_buf->area, 0, runtime->hw.buffer_bytes_max);
+ prtd->dma_addr = (u32) dma_buf->addr;
+
+ mutex_unlock(&prtd->lock);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ rc = afe_cmd_memory_map(dma_buf->addr, dma_buf->bytes);
+ if (rc < 0)
+ pr_err("fail to map memory to DSP\n");
+
+ return rc;
+}
+static snd_pcm_uframes_t msm_afe_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= snd_pcm_lib_buffer_bytes(substream))
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static struct snd_pcm_ops msm_afe_ops = {
+ .open = msm_afe_open,
+ .hw_params = msm_afe_hw_params,
+ .trigger = msm_afe_trigger,
+ .close = msm_afe_close,
+ .prepare = msm_afe_prepare,
+ .mmap = msm_afe_mmap,
+ .pointer = msm_afe_pointer,
+};
+
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static int msm_afe_afe_probe(struct snd_soc_platform *platform)
+{
+ pr_debug("%s\n", __func__);
+ return 0;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_afe_ops,
+ .pcm_new = msm_asoc_pcm_new,
+ .probe = msm_afe_afe_probe,
+};
+
+static __devinit int msm_afe_probe(struct platform_device *pdev)
+{
+ pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_afe_remove(struct platform_device *pdev)
+{
+ pr_debug("%s\n", __func__);
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_afe_driver = {
+ .driver = {
+ .name = "msm-pcm-afe",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_afe_probe,
+ .remove = __devexit_p(msm_afe_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ pr_debug("%s\n", __func__);
+ return platform_driver_register(&msm_afe_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ pr_debug("%s\n", __func__);
+ platform_driver_unregister(&msm_afe_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("AFE PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-afe.h b/sound/soc/msm/msm-pcm-afe.h
new file mode 100644
index 0000000..38026d5
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-afe.h
@@ -0,0 +1,46 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef _MSM_PCM_AFE_H
+#define _MSM_PCM_AFE_H
+#include <sound/apr_audio.h>
+#include <sound/q6afe.h>
+
+
+struct pcm_afe_info {
+ unsigned long dma_addr;
+ struct snd_pcm_substream *substream;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ struct mutex lock;
+ spinlock_t dsp_lock;
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint8_t start;
+ uint32_t dsp_cnt;
+ uint32_t buf_phys;
+ int32_t mmap_flag;
+ int prepared;
+ struct hrtimer hrt;
+ int poll_time;
+ struct msm_mapped_buffer *mem_buffer;
+};
+
+
+#define MSM_EXT(xname, fp_info, fp_get, fp_put, addr) \
+ {.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .name = xname, \
+ .info = fp_info,\
+ .get = fp_get, .put = fp_put, \
+ .private_value = addr, \
+ }
+
+#endif /*_MSM_PCM_AFE_H*/
diff --git a/sound/soc/msm/msm-pcm-hostless.c b/sound/soc/msm/msm-pcm-hostless.c
new file mode 100644
index 0000000..c61511d
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-hostless.c
@@ -0,0 +1,61 @@
+/* Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+
+static struct snd_pcm_ops msm_pcm_hostless_ops = {};
+
+static struct snd_soc_platform_driver msm_soc_hostless_platform = {
+ .ops = &msm_pcm_hostless_ops,
+};
+
+static __devinit int msm_pcm_hostless_probe(struct platform_device *pdev)
+{
+ pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_hostless_platform);
+}
+
+static int msm_pcm_hostless_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_hostless_driver = {
+ .driver = {
+ .name = "msm-pcm-hostless",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_hostless_probe,
+ .remove = __devexit_p(msm_pcm_hostless_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ return platform_driver_register(&msm_pcm_hostless_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_hostless_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("Hostless platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-lpa.c b/sound/soc/msm/msm-pcm-lpa.c
new file mode 100644
index 0000000..c65a7d2
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-lpa.c
@@ -0,0 +1,607 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <sound/compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+#include <sound/timer.h>
+
+#include "msm-pcm-q6.h"
+#include "msm-pcm-routing.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct msm_audio *prtd;
+ unsigned volume;
+};
+static struct snd_msm lpa_audio;
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 1024 * 1024,
+/* TODO: Check on the lowest period size we can support */
+ .period_bytes_min = 128 * 1024,
+ .period_bytes_max = 256 * 1024,
+ .periods_min = 4,
+ .periods_max = 8,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_aio_write_param param;
+ struct audio_buffer *buf = NULL;
+ unsigned long flag = 0;
+ int i = 0;
+
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&the_locks.event_lock, flag);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE: {
+ uint32_t *ptrmem = (uint32_t *)¶m;
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ else
+ if (substream->timer_running)
+ snd_timer_interrupt(substream->timer, 1);
+
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start)) {
+ atomic_set(&prtd->pending_buffer, 1);
+ break;
+ } else
+ atomic_set(&prtd->pending_buffer, 0);
+ if (runtime->status->hw_ptr >= runtime->control->appl_ptr)
+ break;
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+
+ buf = prtd->audio_client->port[IN].buf;
+ param.paddr = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
+ i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1) & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ break;
+ }
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN: {
+ if (!atomic_read(&prtd->pending_buffer))
+ break;
+ if (runtime->status->hw_ptr >=
+ runtime->control->appl_ptr)
+ break;
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__, prtd->pcm_count);
+ buf = prtd->audio_client->port[IN].buf;
+ param.paddr = (unsigned long)buf[prtd->out_head].phys;
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[prtd->out_head].phys;
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1)
+ & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ }
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ pr_debug("ASM_STREAM_CMD_FLUSH\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+ spin_unlock_irqrestore(&the_locks.event_lock, flag);
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ prtd->out_head = 0;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_pcm(prtd->audio_client, runtime->rate,
+ runtime->channels);
+ if (ret < 0)
+ pr_debug("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+ return 0;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->pcm_irq_pos = 0;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("SNDRV_PCM_TRIGGER_START\n");
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->start, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ runtime->hw = msm_pcm_hardware;
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_debug("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return -EPERM;
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ atomic_set(&prtd->pending_buffer, 1);
+ runtime->private_data = prtd;
+ lpa_audio.prtd = prtd;
+ lpa_set_volume(lpa_audio.volume);
+ ret = q6asm_set_softpause(lpa_audio.prtd->audio_client, &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(lpa_audio.prtd->audio_client, &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ return 0;
+}
+
+int lpa_set_volume(unsigned volume)
+{
+ int rc = 0;
+ if (lpa_audio.prtd && lpa_audio.prtd->audio_client) {
+ rc = q6asm_set_volume(lpa_audio.prtd->audio_client, volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed"
+ " rc=%d\n", __func__, rc);
+ }
+ }
+ lpa_audio.volume = volume;
+ return rc;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int rc = 0;
+
+ /*
+ If routing is still enabled, we need to issue EOS to
+ the DSP
+ To issue EOS to dsp, we need to be run state otherwise
+ EOS is not honored.
+ */
+ if (msm_routing_check_backend_enabled(soc_prtd->dai_link->be_id)) {
+ rc = q6asm_run(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->pending_buffer, 0);
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ pr_debug("%s\n", __func__);
+ rc = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (rc < 0)
+ pr_err("EOS cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ }
+
+ dir = IN;
+ atomic_set(&prtd->pending_buffer, 0);
+ lpa_audio.prtd = NULL;
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ pr_debug("%s\n", __func__);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ pr_debug("%s\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ return ret;
+}
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s: pcm_irq_pos = %d\n", __func__, prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ return -EPERM;
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed \
+ rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ if (buf == NULL || buf[0].data == NULL)
+ return -ENOMEM;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static int msm_pcm_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int rc = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ uint64_t timestamp;
+ uint64_t temp;
+
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP: {
+ struct snd_compr_tstamp tstamp;
+ pr_debug("SNDRV_COMPRESS_TSTAMP\n");
+
+ memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
+ timestamp = q6asm_get_session_time(prtd->audio_client);
+ if (timestamp < 0) {
+ pr_err("%s: Get Session Time return value =%lld\n",
+ __func__, timestamp);
+ return -EAGAIN;
+ }
+ temp = (timestamp * 2 * runtime->channels);
+ temp = temp * (runtime->rate/1000);
+ temp = div_u64(temp, 1000);
+ tstamp.sampling_rate = runtime->rate;
+ tstamp.timestamp = timestamp;
+ pr_debug("%s: bytes_consumed:"
+ "timestamp = %lld,\n",__func__,
+ tstamp.timestamp);
+ if (copy_to_user((void *) arg, &tstamp,
+ sizeof(struct snd_compr_tstamp)))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_PCM_IOCTL1_RESET:
+ prtd->cmd_ack = 0;
+ rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ if (rc < 0)
+ pr_err("%s: flush cmd failed rc=%d\n", __func__, rc);
+ rc = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (rc < 0)
+ pr_err("Flush cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ break;
+ default:
+ break;
+ }
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = msm_pcm_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n",
+ __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-lpa",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ spin_lock_init(&the_locks.event_lock);
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-q6.c b/sound/soc/msm/msm-pcm-q6.c
new file mode 100644
index 0000000..39ce436
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-q6.c
@@ -0,0 +1,740 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+
+#include "msm-pcm-q6.h"
+#include "msm-pcm-routing.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+#define PLAYBACK_NUM_PERIODS 8
+#define PLAYBACK_PERIOD_SIZE 2048
+#define CAPTURE_NUM_PERIODS 16
+#define CAPTURE_PERIOD_SIZE 320
+
+static struct snd_pcm_hardware msm_pcm_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 4,
+ .buffer_bytes_max = CAPTURE_NUM_PERIODS * CAPTURE_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_PERIOD_SIZE,
+ .periods_min = CAPTURE_NUM_PERIODS,
+ .periods_max = CAPTURE_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = PLAYBACK_NUM_PERIODS * PLAYBACK_PERIOD_SIZE,
+ .period_bytes_min = PLAYBACK_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_PERIOD_SIZE,
+ .periods_min = PLAYBACK_NUM_PERIODS,
+ .periods_max = PLAYBACK_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static uint32_t in_frame_info[CAPTURE_NUM_PERIODS][2];
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ int i = 0;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ pr_debug("%s\n", __func__);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag)
+ break;
+ if (q6asm_is_cpu_buf_avail_nolock(IN,
+ prtd->audio_client,
+ &size, &idx)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ }
+ break;
+ }
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+ pr_debug("token = 0x%08x\n", token);
+ for (i = 0; i < 8; i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ in_frame_info[token][0] = payload[2];
+ in_frame_info[token][1] = payload[3];
+ prtd->pcm_irq_pos += in_frame_info[token][0];
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag
+ && q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx))
+ q6asm_read_nolock(prtd->audio_client);
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK) {
+ atomic_set(&prtd->start, 1);
+ break;
+ }
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ atomic_set(&prtd->start, 1);
+ break;
+ default:
+ break;
+ }
+ }
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_pcm(prtd->audio_client, runtime->rate,
+ runtime->channels);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ int i = 0;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return 0;
+
+ pr_debug("Samp_rate = %d\n", prtd->samp_rate);
+ pr_debug("Channel = %d\n", prtd->channel_mode);
+ if (prtd->channel_mode > 2) {
+ ret = q6asm_enc_cfg_blk_multi_ch_pcm(prtd->audio_client,
+ prtd->samp_rate, prtd->channel_mode);
+ } else {
+ ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client,
+ prtd->samp_rate, prtd->channel_mode);
+ }
+
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+ prtd->periods = runtime->periods;
+
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_pcm_hardware_playback;
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_hardware_capture;
+ }
+
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_err("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_err("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ xfer = fbytes;
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+fail:
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ ret = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (ret < 0)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%08x\n",
+ (unsigned int) data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = in_frame_info[idx][1];
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ in_frame_info[idx][1] += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&in_frame_info[idx], 0,
+ sizeof(uint32_t) * 2);
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+ int format = FORMAT_LINEAR_PCM;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+
+ /*capture path*/
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (params_channels(params) > 2)
+ format = FORMAT_MULTI_CHANNEL_LINEAR_PCM;
+ pr_debug("%s format = :0x%x\n", __func__, format);
+
+ ret = q6asm_open_read(prtd->audio_client, format);
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_read failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed \
+ rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+ if (buf == NULL || buf[0].data == NULL)
+ return -ENOMEM;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-q6.h b/sound/soc/msm/msm-pcm-q6.h
new file mode 100644
index 0000000..e5551ea
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-q6.h
@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2009,2011 Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+#include <sound/apr_audio.h>
+#include <sound/q6asm.h>
+
+
+/* Support unconventional sample rates 12000, 24000 as well */
+#define USE_RATE \
+ (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+
+extern int copy_count;
+
+struct buffer {
+ void *data;
+ unsigned size;
+ unsigned used;
+ unsigned addr;
+};
+
+struct buffer_rec {
+ void *data;
+ unsigned int size;
+ unsigned int read;
+ unsigned int addr;
+};
+
+struct audio_locks {
+ spinlock_t event_lock;
+ wait_queue_head_t read_wait;
+ wait_queue_head_t write_wait;
+ wait_queue_head_t eos_wait;
+ wait_queue_head_t enable_wait;
+};
+
+struct msm_audio {
+ struct snd_pcm_substream *substream;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ uint16_t source; /* Encoding source bit mask */
+
+ struct audio_client *audio_client;
+
+ uint16_t session_id;
+
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint32_t dsp_cnt;
+
+ int abort; /* set when error, like sample rate mismatch */
+
+ int enabled;
+ int close_ack;
+ int cmd_ack;
+ atomic_t start;
+ atomic_t out_count;
+ atomic_t in_count;
+ atomic_t out_needed;
+ int out_head;
+ int periods;
+ int mmap_flag;
+ atomic_t pending_buffer;
+};
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/msm-pcm-routing.c b/sound/soc/msm/msm-pcm-routing.c
new file mode 100644
index 0000000..7a269ca
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-routing.c
@@ -0,0 +1,2409 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/bitops.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6adm.h>
+#include <sound/q6asm.h>
+#include <sound/q6afe.h>
+#include <sound/tlv.h>
+#include "msm-pcm-routing.h"
+#include "qdsp6/q6voice.h"
+
+struct msm_pcm_routing_bdai_data {
+ u16 port_id; /* AFE port ID */
+ u8 active; /* track if this backend is enabled */
+ unsigned long fe_sessions; /* Front-end sessions */
+ unsigned long port_sessions; /* track Tx BE ports -> Rx BE */
+ unsigned int sample_rate;
+ unsigned int channel;
+};
+
+#define INVALID_SESSION -1
+#define SESSION_TYPE_RX 0
+#define SESSION_TYPE_TX 1
+
+static struct mutex routing_lock;
+
+static int fm_switch_enable;
+static int fm_pcmrx_switch_enable;
+
+#define INT_RX_VOL_MAX_STEPS 0x2000
+#define INT_RX_VOL_GAIN 0x2000
+
+static int msm_route_fm_vol_control;
+static const DECLARE_TLV_DB_LINEAR(fm_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS);
+
+static int msm_route_lpa_vol_control;
+static const DECLARE_TLV_DB_LINEAR(lpa_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS);
+
+static int msm_route_multimedia2_vol_control;
+static const DECLARE_TLV_DB_LINEAR(multimedia2_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS);
+
+static int msm_route_compressed_vol_control;
+static const DECLARE_TLV_DB_LINEAR(compressed_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS);
+
+
+
+/* Equal to Frontend after last of the MULTIMEDIA SESSIONS */
+#define MAX_EQ_SESSIONS MSM_FRONTEND_DAI_CS_VOICE
+
+enum {
+ EQ_BAND1 = 0,
+ EQ_BAND2,
+ EQ_BAND3,
+ EQ_BAND4,
+ EQ_BAND5,
+ EQ_BAND6,
+ EQ_BAND7,
+ EQ_BAND8,
+ EQ_BAND9,
+ EQ_BAND10,
+ EQ_BAND11,
+ EQ_BAND12,
+ EQ_BAND_MAX,
+};
+
+struct msm_audio_eq_band {
+ uint16_t band_idx; /* The band index, 0 .. 11 */
+ uint32_t filter_type; /* Filter band type */
+ uint32_t center_freq_hz; /* Filter band center frequency */
+ uint32_t filter_gain; /* Filter band initial gain (dB) */
+ /* Range is +12 dB to -12 dB with 1dB increments. */
+ uint32_t q_factor;
+} __packed;
+
+struct msm_audio_eq_stream_config {
+ uint32_t enable; /* Number of consequtive bands specified */
+ uint32_t num_bands;
+ struct msm_audio_eq_band eq_bands[EQ_BAND_MAX];
+} __packed;
+
+struct msm_audio_eq_stream_config eq_data[MAX_EQ_SESSIONS];
+
+static void msm_send_eq_values(int eq_idx);
+/* This array is indexed by back-end DAI ID defined in msm-pcm-routing.h
+ * If new back-end is defined, add new back-end DAI ID at the end of enum
+ */
+
+union srs_trumedia_params_u {
+ struct srs_trumedia_params srs_params;
+ unsigned short int raw_params[1];
+};
+static union srs_trumedia_params_u msm_srs_trumedia_params[2];
+static int srs_port_id = -1;
+
+static void srs_send_params(int port_id, unsigned int techs,
+ int param_block_idx) {
+ pr_debug("SRS %s: called, port_id = %d, techs flags = %u,"
+ " paramblockidx %d", __func__, port_id, techs,
+ param_block_idx);
+ /* force all if techs is set to 1 */
+ if (techs == 1)
+ techs = 0xFFFFFFFF;
+
+ if (techs & (1 << SRS_ID_WOWHD))
+ srs_trumedia_open(port_id, SRS_ID_WOWHD,
+ (void *)&msm_srs_trumedia_params[param_block_idx].srs_params.wowhd);
+ if (techs & (1 << SRS_ID_CSHP))
+ srs_trumedia_open(port_id, SRS_ID_CSHP,
+ (void *)&msm_srs_trumedia_params[param_block_idx].srs_params.cshp);
+ if (techs & (1 << SRS_ID_HPF))
+ srs_trumedia_open(port_id, SRS_ID_HPF,
+ (void *)&msm_srs_trumedia_params[param_block_idx].srs_params.hpf);
+ if (techs & (1 << SRS_ID_PEQ))
+ srs_trumedia_open(port_id, SRS_ID_PEQ,
+ (void *)&msm_srs_trumedia_params[param_block_idx].srs_params.peq);
+ if (techs & (1 << SRS_ID_HL))
+ srs_trumedia_open(port_id, SRS_ID_HL,
+ (void *)&msm_srs_trumedia_params[param_block_idx].srs_params.hl);
+ if (techs & (1 << SRS_ID_GLOBAL))
+ srs_trumedia_open(port_id, SRS_ID_GLOBAL,
+ (void *)&msm_srs_trumedia_params[param_block_idx].srs_params.global);
+}
+
+static struct msm_pcm_routing_bdai_data msm_bedais[MSM_BACKEND_DAI_MAX] = {
+ { PRIMARY_I2S_RX, 0, 0, 0, 0, 0},
+ { PRIMARY_I2S_TX, 0, 0, 0, 0, 0},
+ { SLIMBUS_0_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_0_TX, 0, 0, 0, 0, 0},
+ { HDMI_RX, 0, 0, 0, 0, 0},
+ { INT_BT_SCO_RX, 0, 0, 0, 0, 0},
+ { INT_BT_SCO_TX, 0, 0, 0, 0, 0},
+ { INT_FM_RX, 0, 0, 0, 0, 0},
+ { INT_FM_TX, 0, 0, 0, 0, 0},
+ { RT_PROXY_PORT_001_RX, 0, 0, 0, 0, 0},
+ { RT_PROXY_PORT_001_TX, 0, 0, 0, 0, 0},
+ { PCM_RX, 0, 0, 0, 0, 0},
+ { PCM_TX, 0, 0, 0, 0, 0},
+ { VOICE_PLAYBACK_TX, 0, 0, 0, 0, 0},
+ { VOICE_RECORD_RX, 0, 0, 0, 0, 0},
+ { VOICE_RECORD_TX, 0, 0, 0, 0, 0},
+ { MI2S_RX, 0, 0, 0, 0, 0},
+ { MI2S_TX, 0, 0, 0, 0},
+ { SECONDARY_I2S_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_1_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_1_TX, 0, 0, 0, 0, 0},
+ { SLIMBUS_4_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_4_TX, 0, 0, 0, 0, 0},
+ { SLIMBUS_3_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_EXTPROC_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_EXTPROC_RX, 0, 0, 0, 0, 0},
+ { SLIMBUS_EXTPROC_RX, 0, 0, 0, 0, 0},
+};
+
+
+/* Track ASM playback & capture sessions of DAI */
+static int fe_dai_map[MSM_FRONTEND_DAI_MM_SIZE][2] = {
+ /* MULTIMEDIA1 */
+ {INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA2 */
+ {INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA3 */
+ {INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA4 */
+ {INVALID_SESSION, INVALID_SESSION},
+};
+
+static uint8_t is_be_dai_extproc(int be_dai)
+{
+ if (be_dai == MSM_BACKEND_DAI_EXTPROC_RX ||
+ be_dai == MSM_BACKEND_DAI_EXTPROC_TX ||
+ be_dai == MSM_BACKEND_DAI_EXTPROC_EC_TX)
+ return 1;
+ else
+ return 0;
+}
+
+static void msm_pcm_routing_build_matrix(int fedai_id, int dspst_id,
+ int path_type)
+{
+ int i, port_type;
+ struct route_payload payload;
+
+ payload.num_copps = 0;
+ port_type = (path_type == ADM_PATH_PLAYBACK ?
+ MSM_AFE_PORT_TYPE_RX : MSM_AFE_PORT_TYPE_TX);
+
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if (!is_be_dai_extproc(i) &&
+ (afe_get_port_type(msm_bedais[i].port_id) == port_type) &&
+ (msm_bedais[i].active) &&
+ (test_bit(fedai_id, &msm_bedais[i].fe_sessions)))
+ payload.copp_ids[payload.num_copps++] =
+ msm_bedais[i].port_id;
+ }
+
+ if (payload.num_copps)
+ adm_matrix_map(dspst_id, path_type,
+ payload.num_copps, payload.copp_ids, 0);
+}
+
+void msm_pcm_routing_reg_psthr_stream(int fedai_id, int dspst_id,
+ int stream_type)
+{
+ int i, session_type, path_type, port_type;
+ u32 mode = 0;
+
+ if (fedai_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID\n", __func__);
+ return;
+ }
+
+ if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) {
+ session_type = SESSION_TYPE_RX;
+ path_type = ADM_PATH_PLAYBACK;
+ port_type = MSM_AFE_PORT_TYPE_RX;
+ } else {
+ session_type = SESSION_TYPE_TX;
+ path_type = ADM_PATH_LIVE_REC;
+ port_type = MSM_AFE_PORT_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ fe_dai_map[fedai_id][session_type] = dspst_id;
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if (!is_be_dai_extproc(i) &&
+ (afe_get_port_type(msm_bedais[i].port_id) == port_type) &&
+ (msm_bedais[i].active) &&
+ (test_bit(fedai_id, &msm_bedais[i].fe_sessions))) {
+ mode = afe_get_port_type(msm_bedais[i].port_id);
+ adm_connect_afe_port(mode, dspst_id,
+ msm_bedais[i].port_id);
+ break;
+ }
+ }
+ mutex_unlock(&routing_lock);
+}
+
+void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id, int stream_type)
+{
+ int i, session_type, path_type, port_type;
+ struct route_payload payload;
+ u32 channels;
+
+ if (fedai_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID %d\n", __func__, fedai_id);
+ return;
+ }
+
+ if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) {
+ session_type = SESSION_TYPE_RX;
+ path_type = ADM_PATH_PLAYBACK;
+ port_type = MSM_AFE_PORT_TYPE_RX;
+ } else {
+ session_type = SESSION_TYPE_TX;
+ path_type = ADM_PATH_LIVE_REC;
+ port_type = MSM_AFE_PORT_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ payload.num_copps = 0; /* only RX needs to use payload */
+ fe_dai_map[fedai_id][session_type] = dspst_id;
+ /* re-enable EQ if active */
+ if (eq_data[fedai_id].enable)
+ msm_send_eq_values(fedai_id);
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if (!is_be_dai_extproc(i) &&
+ (afe_get_port_type(msm_bedais[i].port_id) == port_type) &&
+ (msm_bedais[i].active) &&
+ (test_bit(fedai_id, &msm_bedais[i].fe_sessions))) {
+
+ channels = msm_bedais[i].channel;
+
+ if ((stream_type == SNDRV_PCM_STREAM_PLAYBACK) &&
+ (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[i].port_id,
+ path_type,
+ msm_bedais[i].sample_rate,
+ msm_bedais[i].channel,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(msm_bedais[i].port_id,
+ path_type,
+ msm_bedais[i].sample_rate,
+ msm_bedais[i].channel,
+ DEFAULT_COPP_TOPOLOGY);
+
+ payload.copp_ids[payload.num_copps++] =
+ msm_bedais[i].port_id;
+ }
+ }
+ if (payload.num_copps)
+ adm_matrix_map(dspst_id, path_type,
+ payload.num_copps, payload.copp_ids, 0);
+
+ mutex_unlock(&routing_lock);
+}
+
+void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type)
+{
+ int i, port_type, session_type;
+
+ if (fedai_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID\n", __func__);
+ return;
+ }
+
+ if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) {
+ port_type = MSM_AFE_PORT_TYPE_RX;
+ session_type = SESSION_TYPE_RX;
+ } else {
+ port_type = MSM_AFE_PORT_TYPE_TX;
+ session_type = SESSION_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if (!is_be_dai_extproc(i) &&
+ (afe_get_port_type(msm_bedais[i].port_id) == port_type) &&
+ (msm_bedais[i].active) &&
+ (test_bit(fedai_id, &msm_bedais[i].fe_sessions)))
+ adm_close(msm_bedais[i].port_id);
+ }
+
+ fe_dai_map[fedai_id][session_type] = INVALID_SESSION;
+
+ mutex_unlock(&routing_lock);
+}
+
+/* Check if FE/BE route is set */
+static bool msm_pcm_routing_route_is_set(u16 be_id, u16 fe_id)
+{
+ bool rc = false;
+
+ if (fe_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* recheck FE ID in the mixer control defined in this file */
+ pr_err("%s: bad MM ID\n", __func__);
+ return rc;
+ }
+
+ if (test_bit(fe_id, &msm_bedais[be_id].fe_sessions))
+ rc = true;
+
+ return rc;
+}
+
+static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
+{
+ int session_type, path_type;
+ u32 channels;
+
+ pr_debug("%s: reg %x val %x set %x\n", __func__, reg, val, set);
+
+ if (val > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* recheck FE ID in the mixer control defined in this file */
+ pr_err("%s: bad MM ID\n", __func__);
+ return;
+ }
+
+ if (afe_get_port_type(msm_bedais[reg].port_id) ==
+ MSM_AFE_PORT_TYPE_RX) {
+ session_type = SESSION_TYPE_RX;
+ path_type = ADM_PATH_PLAYBACK;
+ } else {
+ session_type = SESSION_TYPE_TX;
+ path_type = ADM_PATH_LIVE_REC;
+ }
+
+ mutex_lock(&routing_lock);
+
+ if (set) {
+ if (!test_bit(val, &msm_bedais[reg].fe_sessions) &&
+ (msm_bedais[reg].port_id == VOICE_PLAYBACK_TX))
+ voc_start_playback(set);
+
+ set_bit(val, &msm_bedais[reg].fe_sessions);
+ if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
+ INVALID_SESSION) {
+
+ channels = msm_bedais[reg].channel;
+
+ if ((session_type == SESSION_TYPE_RX) && (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[reg].port_id,
+ path_type,
+ msm_bedais[reg].sample_rate,
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(msm_bedais[reg].port_id,
+ path_type,
+ msm_bedais[reg].sample_rate, channels,
+ DEFAULT_COPP_TOPOLOGY);
+
+ msm_pcm_routing_build_matrix(val,
+ fe_dai_map[val][session_type], path_type);
+ }
+ } else {
+ if (test_bit(val, &msm_bedais[reg].fe_sessions) &&
+ (msm_bedais[reg].port_id == VOICE_PLAYBACK_TX))
+ voc_start_playback(set);
+ clear_bit(val, &msm_bedais[reg].fe_sessions);
+ if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
+ INVALID_SESSION) {
+ adm_close(msm_bedais[reg].port_id);
+ msm_pcm_routing_build_matrix(val,
+ fe_dai_map[val][session_type], path_type);
+ }
+ }
+ if ((msm_bedais[reg].port_id == VOICE_RECORD_RX)
+ || (msm_bedais[reg].port_id == VOICE_RECORD_TX))
+ voc_start_record(msm_bedais[reg].port_id, set);
+
+ mutex_unlock(&routing_lock);
+}
+
+static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+
+ if (ucontrol->value.integer.value[0] &&
+ msm_pcm_routing_route_is_set(mc->reg, mc->shift) == false) {
+ msm_pcm_routing_process_audio(mc->reg, mc->shift, 1);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else if (!ucontrol->value.integer.value[0] &&
+ msm_pcm_routing_route_is_set(mc->reg, mc->shift) == true) {
+ msm_pcm_routing_process_audio(mc->reg, mc->shift, 0);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+
+ return 1;
+}
+
+static void msm_pcm_routing_process_voice(u16 reg, u16 val, int set)
+{
+ u16 session_id = 0;
+
+ pr_debug("%s: reg %x val %x set %x\n", __func__, reg, val, set);
+
+ if (val == MSM_FRONTEND_DAI_CS_VOICE)
+ session_id = voc_get_session_id(VOICE_SESSION_NAME);
+ else if (val == MSM_FRONTEND_DAI_VOLTE)
+ session_id = voc_get_session_id(VOLTE_SESSION_NAME);
+ else
+ session_id = voc_get_session_id(VOIP_SESSION_NAME);
+
+ pr_debug("%s: FE DAI 0x%x session_id 0x%x\n",
+ __func__, val, session_id);
+
+ mutex_lock(&routing_lock);
+
+ if (set)
+ set_bit(val, &msm_bedais[reg].fe_sessions);
+ else
+ clear_bit(val, &msm_bedais[reg].fe_sessions);
+
+ mutex_unlock(&routing_lock);
+
+ if (afe_get_port_type(msm_bedais[reg].port_id) ==
+ MSM_AFE_PORT_TYPE_RX) {
+ voc_set_route_flag(session_id, RX_PATH, set);
+ if (set) {
+ voc_set_rxtx_port(session_id,
+ msm_bedais[reg].port_id, DEV_RX);
+
+ if (voc_get_route_flag(session_id, RX_PATH) &&
+ voc_get_route_flag(session_id, TX_PATH))
+ voc_enable_cvp(session_id);
+ } else {
+ voc_disable_cvp(session_id);
+ }
+ } else {
+ voc_set_route_flag(session_id, TX_PATH, set);
+ if (set) {
+ voc_set_rxtx_port(session_id,
+ msm_bedais[reg].port_id, DEV_TX);
+ if (voc_get_route_flag(session_id, RX_PATH) &&
+ voc_get_route_flag(session_id, TX_PATH))
+ voc_enable_cvp(session_id);
+ } else {
+ voc_disable_cvp(session_id);
+ }
+ }
+}
+
+static int msm_routing_get_voice_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ mutex_lock(&routing_lock);
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ mutex_unlock(&routing_lock);
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_voice_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (ucontrol->value.integer.value[0]) {
+ msm_pcm_routing_process_voice(mc->reg, mc->shift, 1);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else {
+ msm_pcm_routing_process_voice(mc->reg, mc->shift, 0);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+
+ return 1;
+}
+
+static int msm_routing_get_voice_stub_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ mutex_lock(&routing_lock);
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ mutex_unlock(&routing_lock);
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_voice_stub_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (ucontrol->value.integer.value[0]) {
+ mutex_lock(&routing_lock);
+ set_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions);
+ mutex_unlock(&routing_lock);
+
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else {
+ mutex_lock(&routing_lock);
+ clear_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions);
+ mutex_unlock(&routing_lock);
+
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 1;
+}
+
+static int msm_routing_get_switch_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = fm_switch_enable;
+ pr_debug("%s: FM Switch enable %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int msm_routing_put_switch_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+
+ pr_debug("%s: FM Switch enable %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+ if (ucontrol->value.integer.value[0])
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ else
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ fm_switch_enable = ucontrol->value.integer.value[0];
+ return 1;
+}
+
+static int msm_routing_get_fm_pcmrx_switch_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = fm_pcmrx_switch_enable;
+ pr_debug("%s: FM Switch enable %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int msm_routing_put_fm_pcmrx_switch_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+
+ pr_debug("%s: FM Switch enable %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+ if (ucontrol->value.integer.value[0])
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ else
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ fm_pcmrx_switch_enable = ucontrol->value.integer.value[0];
+ return 1;
+}
+
+static int msm_routing_get_port_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].port_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_port_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg,
+ mc->shift, ucontrol->value.integer.value[0]);
+
+ if (ucontrol->value.integer.value[0]) {
+ afe_loopback(1, msm_bedais[mc->reg].port_id,
+ msm_bedais[mc->shift].port_id);
+ set_bit(mc->shift,
+ &msm_bedais[mc->reg].port_sessions);
+ } else {
+ afe_loopback(0, msm_bedais[mc->reg].port_id,
+ msm_bedais[mc->shift].port_id);
+ clear_bit(mc->shift,
+ &msm_bedais[mc->reg].port_sessions);
+ }
+
+ return 1;
+}
+
+static int msm_routing_get_fm_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm_route_fm_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_fm_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ afe_loopback_gain(INT_FM_TX , ucontrol->value.integer.value[0]);
+
+ msm_route_fm_vol_control = ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int msm_routing_get_lpa_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm_route_lpa_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_lpa_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (!lpa_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_lpa_vol_control =
+ ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int msm_routing_get_multimedia2_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ ucontrol->value.integer.value[0] = msm_route_multimedia2_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_multimedia2_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ if (!multi_ch_pcm_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_multimedia2_vol_control =
+ ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int msm_routing_get_compressed_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ ucontrol->value.integer.value[0] = msm_route_compressed_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_compressed_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (!compressed_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_compressed_vol_control =
+ ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int msm_routing_get_srs_trumedia_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_routing_set_srs_trumedia_control_(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int techs = 0;
+ unsigned short offset, value, max, index;
+
+ max = sizeof(msm_srs_trumedia_params) >> 1;
+ index = (unsigned short)((ucontrol->value.integer.value[0] &
+ SRS_PARAM_INDEX_MASK) >> 31);
+ if (SRS_CMD_UPLOAD ==
+ (ucontrol->value.integer.value[0] & SRS_CMD_UPLOAD)) {
+ techs = ucontrol->value.integer.value[0] & 0xFF;
+ pr_debug("SRS %s: send params request, flags = %u",
+ __func__, techs);
+ if (srs_port_id >= 0 && techs)
+ srs_send_params(srs_port_id, techs, index);
+ return 0;
+ }
+ offset = (unsigned short)((ucontrol->value.integer.value[0] &
+ SRS_PARAM_OFFSET_MASK) >> 16);
+ value = (unsigned short)(ucontrol->value.integer.value[0] &
+ SRS_PARAM_VALUE_MASK);
+ if (offset < max) {
+ msm_srs_trumedia_params[index].raw_params[offset] = value;
+ pr_debug("SRS %s: index set... (max %d, requested %d,"
+ " val %d, paramblockidx %d)", __func__, max, offset,
+ value, index);
+ } else {
+ pr_err("SRS %s: index out of bounds! (max %d, requested %d)",
+ __func__, max, offset);
+ }
+ if (offset == 4) {
+ int i;
+ for (i = 0; i < max; i++) {
+ if (i == 0) {
+ pr_debug("SRS %s: global block start",
+ __func__);
+ }
+ if (i ==
+ (sizeof(struct srs_trumedia_params_GLOBAL) >> 1)) {
+ break;
+ pr_debug("SRS %s: wowhd block start at"
+ " offset %d word offset %d", __func__,
+ i, i>>1);
+ }
+ pr_debug("SRS %s: param_index %d index %d val %d",
+ __func__, index, i,
+ msm_srs_trumedia_params[index].raw_params[i]);
+ }
+ }
+ return 0;
+}
+
+static int msm_routing_set_srs_trumedia_control(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol) {
+ int ret;
+
+ pr_debug("SRS control normal called");
+ mutex_lock(&routing_lock);
+ srs_port_id = SLIMBUS_0_RX;
+ ret = msm_routing_set_srs_trumedia_control_(kcontrol, ucontrol);
+ mutex_unlock(&routing_lock);
+ return ret;
+}
+
+static int msm_routing_set_srs_trumedia_control_I2S(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol) {
+ int ret;
+
+ pr_debug("SRS control I2S called");
+ mutex_lock(&routing_lock);
+ srs_port_id = PRIMARY_I2S_RX;
+ ret = msm_routing_set_srs_trumedia_control_(kcontrol, ucontrol);
+ mutex_unlock(&routing_lock);
+ return ret;
+}
+
+static int msm_routing_set_srs_trumedia_control_HDMI(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol) {
+ int ret;
+
+ pr_debug("SRS control HDMI called");
+ mutex_lock(&routing_lock);
+ srs_port_id = HDMI_RX;
+ ret = msm_routing_set_srs_trumedia_control_(kcontrol, ucontrol);
+ mutex_unlock(&routing_lock);
+ return ret;
+}
+
+static void msm_send_eq_values(int eq_idx)
+{
+ int result;
+ struct audio_client *ac =
+ q6asm_get_audio_client(fe_dai_map[eq_idx][SESSION_TYPE_RX]);
+
+ if (ac == NULL) {
+ pr_err("%s: Could not get audio client for session: %d\n",
+ __func__, fe_dai_map[eq_idx][SESSION_TYPE_RX]);
+ goto done;
+ }
+
+ result = q6asm_equalizer(ac, &eq_data[eq_idx]);
+
+ if (result < 0)
+ pr_err("%s: Call to ASM equalizer failed, returned = %d\n",
+ __func__, result);
+done:
+ return;
+}
+
+static int msm_routing_get_eq_enable_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+
+ ucontrol->value.integer.value[0] = eq_data[eq_idx].enable;
+
+ pr_debug("%s: EQ #%d enable %d\n", __func__,
+ eq_idx, eq_data[eq_idx].enable);
+ return 0;
+}
+
+static int msm_routing_put_eq_enable_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int value = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: EQ #%d enable %d\n", __func__,
+ eq_idx, value);
+ eq_data[eq_idx].enable = value;
+
+ msm_send_eq_values(eq_idx);
+ return 0;
+}
+
+static int msm_routing_get_eq_band_count_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+
+ ucontrol->value.integer.value[0] = eq_data[eq_idx].num_bands;
+
+ pr_debug("%s: EQ #%d bands %d\n", __func__,
+ eq_idx, eq_data[eq_idx].num_bands);
+ return eq_data[eq_idx].num_bands;
+}
+
+static int msm_routing_put_eq_band_count_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int value = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: EQ #%d bands %d\n", __func__,
+ eq_idx, value);
+ eq_data[eq_idx].num_bands = value;
+ return 0;
+}
+
+static int msm_routing_get_eq_band_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ ucontrol->value.integer.value[0] =
+ eq_data[eq_idx].eq_bands[band_idx].band_idx;
+ ucontrol->value.integer.value[1] =
+ eq_data[eq_idx].eq_bands[band_idx].filter_type;
+ ucontrol->value.integer.value[2] =
+ eq_data[eq_idx].eq_bands[band_idx].center_freq_hz;
+ ucontrol->value.integer.value[3] =
+ eq_data[eq_idx].eq_bands[band_idx].filter_gain;
+ ucontrol->value.integer.value[4] =
+ eq_data[eq_idx].eq_bands[band_idx].q_factor;
+
+ pr_debug("%s: band_idx = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].band_idx);
+ pr_debug("%s: filter_type = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].filter_type);
+ pr_debug("%s: center_freq_hz = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].center_freq_hz);
+ pr_debug("%s: filter_gain = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].filter_gain);
+ pr_debug("%s: q_factor = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].q_factor);
+ return 0;
+}
+
+static int msm_routing_put_eq_band_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ eq_data[eq_idx].eq_bands[band_idx].band_idx =
+ ucontrol->value.integer.value[0];
+ eq_data[eq_idx].eq_bands[band_idx].filter_type =
+ ucontrol->value.integer.value[1];
+ eq_data[eq_idx].eq_bands[band_idx].center_freq_hz =
+ ucontrol->value.integer.value[2];
+ eq_data[eq_idx].eq_bands[band_idx].filter_gain =
+ ucontrol->value.integer.value[3];
+ eq_data[eq_idx].eq_bands[band_idx].q_factor =
+ ucontrol->value.integer.value[4];
+ return 0;
+}
+
+static const struct snd_kcontrol_new pri_i2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_PRI_I2S_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_SEC_I2S_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_SLIMBUS_0_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_MI2S_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+ /* incall music delivery mixer */
+static const struct snd_kcontrol_new incall_music_delivery_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_4_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_SLIMBUS_4_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_SLIMBUS_4_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SLIM_4_TX", MSM_BACKEND_DAI_SLIMBUS_4_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul2_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new sec_i2s_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_SLIMBUS_0_RX ,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_SLIMBUS_0_RX ,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new bt_sco_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_INT_BT_SCO_RX ,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_INT_BT_SCO_RX ,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new mi2s_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new afe_pcm_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new aux_pcm_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new hdmi_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("VoLTE", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new stub_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_EXTPROC_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_1_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_3_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new tx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX_Voice", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("MI2S_TX_Voice", MSM_BACKEND_DAI_MI2S_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX_Voice", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX_Voice",
+ MSM_BACKEND_DAI_INT_BT_SCO_TX, MSM_FRONTEND_DAI_CS_VOICE, 1, 0,
+ msm_routing_get_voice_mixer, msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX_Voice", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_TX_Voice", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new tx_volte_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX_VoLTE", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX_VoLTE", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX_VoLTE",
+ MSM_BACKEND_DAI_INT_BT_SCO_TX, MSM_FRONTEND_DAI_VOLTE, 1, 0,
+ msm_routing_get_voice_mixer, msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX_VoLTE", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_TX_VoLTE", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_VOLTE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new tx_voip_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX_Voip", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("MI2S_TX_Voip", MSM_BACKEND_DAI_MI2S_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX_Voip", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX_Voip", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX_Voip", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_TX_Voip", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new tx_voice_stub_mixer_controls[] = {
+ SOC_SINGLE_EXT("STUB_TX_HL", MSM_BACKEND_DAI_EXTPROC_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("SLIM_1_TX", MSM_BACKEND_DAI_SLIMBUS_1_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("STUB_1_TX_HL", MSM_BACKEND_DAI_EXTPROC_EC_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_MI2S_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new sbus_0_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_INT_FM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_SLIMBUS_0_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_MI2S_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new auxpcm_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_BACKEND_DAI_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_BACKEND_DAI_SLIMBUS_0_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new sbus_1_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new sbus_3_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_RX", MSM_BACKEND_DAI_SLIMBUS_3_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_RX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_SLIMBUS_3_RX,
+ MSM_BACKEND_DAI_MI2S_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+static const struct snd_kcontrol_new bt_sco_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("SLIM_1_TX", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new afe_pcm_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_BACKEND_DAI_INT_FM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+
+static const struct snd_kcontrol_new hdmi_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_BACKEND_DAI_MI2S_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new sec_i2s_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("MI2S_TX", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_BACKEND_DAI_MI2S_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new mi2s_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("SLIM_1_TX", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new fm_switch_mixer_controls =
+ SOC_SINGLE_EXT("Switch", SND_SOC_NOPM,
+ 0, 1, 0, msm_routing_get_switch_mixer,
+ msm_routing_put_switch_mixer);
+
+static const struct snd_kcontrol_new pcm_rx_switch_mixer_controls =
+ SOC_SINGLE_EXT("Switch", SND_SOC_NOPM,
+ 0, 1, 0, msm_routing_get_fm_pcmrx_switch_mixer,
+ msm_routing_put_fm_pcmrx_switch_mixer);
+
+static const struct snd_kcontrol_new int_fm_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("Internal FM RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_fm_vol_mixer,
+ msm_routing_set_fm_vol_mixer, fm_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new lpa_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("LPA RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_lpa_vol_mixer,
+ msm_routing_set_lpa_vol_mixer, lpa_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new multimedia2_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("HIFI2 RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_multimedia2_vol_mixer,
+ msm_routing_set_multimedia2_vol_mixer, multimedia2_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new compressed_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("COMPRESSED RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_compressed_vol_mixer,
+ msm_routing_set_compressed_vol_mixer, compressed_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new lpa_SRS_trumedia_controls[] = {
+ {.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SRS TruMedia",
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_soc_info_volsw, \
+ .get = msm_routing_get_srs_trumedia_control,
+ .put = msm_routing_set_srs_trumedia_control,
+ .private_value = ((unsigned long)&(struct soc_mixer_control)
+ {.reg = SND_SOC_NOPM,
+ .shift = 0,
+ .rshift = 0,
+ .max = 0xFFFFFFFF,
+ .platform_max = 0xFFFFFFFF,
+ .invert = 0
+ })
+ }
+};
+
+static const struct snd_kcontrol_new lpa_SRS_trumedia_controls_HDMI[] = {
+ {.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SRS TruMedia HDMI",
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_soc_info_volsw, \
+ .get = msm_routing_get_srs_trumedia_control,
+ .put = msm_routing_set_srs_trumedia_control_HDMI,
+ .private_value = ((unsigned long)&(struct soc_mixer_control)
+ {.reg = SND_SOC_NOPM,
+ .shift = 0,
+ .rshift = 0,
+ .max = 0xFFFFFFFF,
+ .platform_max = 0xFFFFFFFF,
+ .invert = 0
+ })
+ }
+};
+
+static const struct snd_kcontrol_new lpa_SRS_trumedia_controls_I2S[] = {
+ {.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "SRS TruMedia I2S",
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_soc_info_volsw, \
+ .get = msm_routing_get_srs_trumedia_control,
+ .put = msm_routing_set_srs_trumedia_control_I2S,
+ .private_value = ((unsigned long)&(struct soc_mixer_control)
+ {.reg = SND_SOC_NOPM,
+ .shift = 0,
+ .rshift = 0,
+ .max = 0xFFFFFFFF,
+ .platform_max = 0xFFFFFFFF,
+ .invert = 0
+ })
+ }
+};
+
+static const struct snd_kcontrol_new eq_enable_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1 EQ Enable", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_eq_enable_mixer,
+ msm_routing_put_eq_enable_mixer),
+ SOC_SINGLE_EXT("MultiMedia2 EQ Enable", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_eq_enable_mixer,
+ msm_routing_put_eq_enable_mixer),
+ SOC_SINGLE_EXT("MultiMedia3 EQ Enable", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_eq_enable_mixer,
+ msm_routing_put_eq_enable_mixer),
+};
+
+static const struct snd_kcontrol_new eq_band_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1 EQ Band Count", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 11, 0,
+ msm_routing_get_eq_band_count_audio_mixer,
+ msm_routing_put_eq_band_count_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2 EQ Band Count", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 11, 0,
+ msm_routing_get_eq_band_count_audio_mixer,
+ msm_routing_put_eq_band_count_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3 EQ Band Count", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 11, 0,
+ msm_routing_get_eq_band_count_audio_mixer,
+ msm_routing_put_eq_band_count_audio_mixer),
+};
+
+static const struct snd_kcontrol_new eq_coeff_mixer_controls[] = {
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band1", EQ_BAND1,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band2", EQ_BAND2,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band3", EQ_BAND3,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band4", EQ_BAND4,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band5", EQ_BAND5,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band6", EQ_BAND6,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band7", EQ_BAND7,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band8", EQ_BAND8,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band9", EQ_BAND9,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band10", EQ_BAND10,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band11", EQ_BAND11,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band12", EQ_BAND12,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band1", EQ_BAND1,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band2", EQ_BAND2,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band3", EQ_BAND3,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band4", EQ_BAND4,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band5", EQ_BAND5,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band6", EQ_BAND6,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band7", EQ_BAND7,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band8", EQ_BAND8,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band9", EQ_BAND9,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band10", EQ_BAND10,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band11", EQ_BAND11,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band12", EQ_BAND12,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band1", EQ_BAND1,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band2", EQ_BAND2,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band3", EQ_BAND3,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band4", EQ_BAND4,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band5", EQ_BAND5,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band6", EQ_BAND6,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band7", EQ_BAND7,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band8", EQ_BAND8,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band9", EQ_BAND9,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band10", EQ_BAND10,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band11", EQ_BAND11,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band12", EQ_BAND12,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+};
+
+static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
+ /* Frontend AIF */
+ /* Widget name equals to Front-End DAI name<Need confirmation>,
+ * Stream name must contains substring of front-end dai name
+ */
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("VOIP_DL", "VoIP Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("VoLTE_DL", "VoLTE Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("VoLTE_UL", "VoLTE Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("VOIP_UL", "VoIP Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIM0_DL_HL", "SLIMBUS0_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIM0_UL_HL", "SLIMBUS0_HOSTLESS Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("INTFM_DL_HL", "INT_FM_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("INTFM_UL_HL", "INT_FM_HOSTLESS Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("HDMI_DL_HL", "HDMI_HOSTLESS Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SEC_I2S_DL_HL", "SEC_I2S_RX_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AUXPCM_DL_HL", "AUXPCM_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AUXPCM_UL_HL", "AUXPCM_HOSTLESS Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MI2S_UL_HL", "MI2S_TX_HOSTLESS Capture",
+ 0, 0, 0, 0),
+
+ /* Backend AIF */
+ /* Stream name equals to backend dai link stream name
+ */
+ SND_SOC_DAPM_AIF_OUT("PRI_I2S_RX", "Primary I2S Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SEC_I2S_RX", "Secondary I2S Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("HDMI", "HDMI Playback", 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_OUT("MI2S_RX", "MI2S Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("PRI_I2S_TX", "Primary I2S Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MI2S_TX", "MI2S Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("INT_BT_SCO_RX", "Internal BT-SCO Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INT_BT_SCO_TX", "Internal BT-SCO Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("INT_FM_RX", "Internal FM Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INT_FM_TX", "Internal FM Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("PCM_RX", "AFE Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("PCM_TX", "AFE Capture",
+ 0, 0, 0 , 0),
+ /* incall */
+ SND_SOC_DAPM_AIF_OUT("VOICE_PLAYBACK_TX", "Voice Farend Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INCALL_RECORD_TX", "Voice Uplink Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("INCALL_RECORD_RX", "Voice Downlink Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_4_TX", "Slimbus4 Capture",
+ 0, 0, 0, 0),
+
+ SND_SOC_DAPM_AIF_OUT("AUX_PCM_RX", "AUX PCM Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AUX_PCM_TX", "AUX PCM Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("VOICE_STUB_DL", "VOICE_STUB Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("VOICE_STUB_UL", "VOICE_STUB Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("STUB_RX", "Stub Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("STUB_TX", "Stub Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("STUB_1_TX", "Stub1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0),
+
+ /* Switch Definitions */
+ SND_SOC_DAPM_SWITCH("SLIMBUS_DL_HL", SND_SOC_NOPM, 0, 0,
+ &fm_switch_mixer_controls),
+ SND_SOC_DAPM_SWITCH("PCM_RX_DL_HL", SND_SOC_NOPM, 0, 0,
+ &pcm_rx_switch_mixer_controls),
+ /* Mixer definitions */
+ SND_SOC_DAPM_MIXER("PRI_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ pri_i2s_rx_mixer_controls, ARRAY_SIZE(pri_i2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ sec_i2s_rx_mixer_controls, ARRAY_SIZE(sec_i2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_rx_mixer_controls, ARRAY_SIZE(slimbus_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
+ hdmi_mixer_controls, ARRAY_SIZE(hdmi_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ mi2s_rx_mixer_controls, ARRAY_SIZE(mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul2_mixer_controls, ARRAY_SIZE(mmul2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)),
+ /* incall */
+ SND_SOC_DAPM_MIXER("Incall_Music Audio Mixer", SND_SOC_NOPM, 0, 0,
+ incall_music_delivery_mixer_controls,
+ ARRAY_SIZE(incall_music_delivery_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_4_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_4_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_4_rx_mixer_controls)),
+ /* Voice Mixer */
+ SND_SOC_DAPM_MIXER("PRI_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0, pri_rx_voice_mixer_controls,
+ ARRAY_SIZE(pri_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ sec_i2s_rx_voice_mixer_controls,
+ ARRAY_SIZE(sec_i2s_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIM_0_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ slimbus_rx_voice_mixer_controls,
+ ARRAY_SIZE(slimbus_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_BT_SCO_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ bt_sco_rx_voice_mixer_controls,
+ ARRAY_SIZE(bt_sco_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AFE_PCM_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ afe_pcm_rx_voice_mixer_controls,
+ ARRAY_SIZE(afe_pcm_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AUX_PCM_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ aux_pcm_rx_voice_mixer_controls,
+ ARRAY_SIZE(aux_pcm_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("HDMI_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ hdmi_rx_voice_mixer_controls,
+ ARRAY_SIZE(hdmi_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MI2S_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ mi2s_rx_voice_mixer_controls,
+ ARRAY_SIZE(mi2s_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Voice_Tx Mixer",
+ SND_SOC_NOPM, 0, 0, tx_voice_mixer_controls,
+ ARRAY_SIZE(tx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Voip_Tx Mixer",
+ SND_SOC_NOPM, 0, 0, tx_voip_mixer_controls,
+ ARRAY_SIZE(tx_voip_mixer_controls)),
+ SND_SOC_DAPM_MIXER("VoLTE_Tx Mixer",
+ SND_SOC_NOPM, 0, 0, tx_volte_mixer_controls,
+ ARRAY_SIZE(tx_volte_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_BT_SCO_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ int_bt_sco_rx_mixer_controls, ARRAY_SIZE(int_bt_sco_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_FM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ int_fm_rx_mixer_controls, ARRAY_SIZE(int_fm_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AFE_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ afe_pcm_rx_mixer_controls, ARRAY_SIZE(afe_pcm_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Voice Stub Tx Mixer", SND_SOC_NOPM, 0, 0,
+ tx_voice_stub_mixer_controls, ARRAY_SIZE(tx_voice_stub_mixer_controls)),
+ SND_SOC_DAPM_MIXER("STUB_RX Mixer", SND_SOC_NOPM, 0, 0,
+ stub_rx_mixer_controls, ARRAY_SIZE(stub_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_1_RX Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_1_rx_mixer_controls, ARRAY_SIZE(slimbus_1_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_3_RX_Voice Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_3_rx_mixer_controls, ARRAY_SIZE(slimbus_3_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, sbus_0_rx_port_mixer_controls,
+ ARRAY_SIZE(sbus_0_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AUXPCM_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, auxpcm_rx_port_mixer_controls,
+ ARRAY_SIZE(auxpcm_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_1_RX Port Mixer", SND_SOC_NOPM, 0, 0,
+ sbus_1_rx_port_mixer_controls,
+ ARRAY_SIZE(sbus_1_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_BT_SCO_RX Port Mixer", SND_SOC_NOPM, 0, 0,
+ bt_sco_rx_port_mixer_controls,
+ ARRAY_SIZE(bt_sco_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AFE_PCM_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, afe_pcm_rx_port_mixer_controls,
+ ARRAY_SIZE(afe_pcm_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("HDMI_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, hdmi_rx_port_mixer_controls,
+ ARRAY_SIZE(hdmi_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_I2S_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, sec_i2s_rx_port_mixer_controls,
+ ARRAY_SIZE(sec_i2s_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_3_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, sbus_3_rx_port_mixer_controls,
+ ARRAY_SIZE(sbus_3_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MI2S_RX Port Mixer", SND_SOC_NOPM, 0, 0,
+ mi2s_rx_port_mixer_controls, ARRAY_SIZE(mi2s_rx_port_mixer_controls)),
+
+ /* Virtual Pins to force backends ON atm */
+ SND_SOC_DAPM_OUTPUT("BE_OUT"),
+ SND_SOC_DAPM_INPUT("BE_IN"),
+
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"PRI_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"PRI_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"PRI_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"PRI_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PRI_I2S_RX", NULL, "PRI_RX Audio Mixer"},
+
+ {"SEC_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SEC_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SEC_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SEC_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SEC_I2S_RX", NULL, "SEC_RX Audio Mixer"},
+
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"},
+
+ {"HDMI Mixer", "MultiMedia1", "MM_DL1"},
+ {"HDMI Mixer", "MultiMedia2", "MM_DL2"},
+ {"HDMI Mixer", "MultiMedia3", "MM_DL3"},
+ {"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+ {"HDMI", NULL, "HDMI Mixer"},
+
+ /* incall */
+ {"Incall_Music Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"Incall_Music Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"VOICE_PLAYBACK_TX", NULL, "Incall_Music Audio Mixer"},
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_4_RX", NULL, "SLIMBUS_4_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},
+ {"MultiMedia1 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},
+ {"MultiMedia1 Mixer", "SLIM_4_TX", "SLIMBUS_4_TX"},
+ {"MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"MI2S_RX", NULL, "MI2S_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"},
+ {"MultiMedia1 Mixer", "MI2S_TX", "MI2S_TX"},
+ {"MultiMedia2 Mixer", "MI2S_TX", "MI2S_TX"},
+ {"MultiMedia1 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+ {"MultiMedia1 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Audio Mixer"},
+
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"INT_FM_RX", NULL, "INTERNAL_FM_RX Audio Mixer"},
+
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PCM_RX", NULL, "AFE_PCM_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+
+ {"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"},
+ {"MM_UL1", NULL, "MultiMedia1 Mixer"},
+ {"MultiMedia2 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+ {"MM_UL2", NULL, "MultiMedia2 Mixer"},
+
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"AUX_PCM_RX", NULL, "AUX_PCM_RX Audio Mixer"},
+
+ {"PRI_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"PRI_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"PRI_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"PRI_I2S_RX", NULL, "PRI_RX_Voice Mixer"},
+
+ {"SEC_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"SEC_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"SEC_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"SEC_I2S_RX", NULL, "SEC_RX_Voice Mixer"},
+
+ {"SLIM_0_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"SLIM_0_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"SLIM_0_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"SLIMBUS_0_RX", NULL, "SLIM_0_RX_Voice Mixer"},
+
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX_Voice Mixer"},
+
+ {"AFE_PCM_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"AFE_PCM_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"AFE_PCM_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"PCM_RX", NULL, "AFE_PCM_RX_Voice Mixer"},
+
+ {"AUX_PCM_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"AUX_PCM_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"AUX_PCM_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"AUX_PCM_RX", NULL, "AUX_PCM_RX_Voice Mixer"},
+
+ {"HDMI_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"HDMI_RX_Voice Mixer", "VoLTE", "VoLTE_DL"},
+ {"HDMI_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"HDMI", NULL, "HDMI_RX_Voice Mixer"},
+ {"HDMI", NULL, "HDMI_DL_HL"},
+
+ {"Voice_Tx Mixer", "PRI_TX_Voice", "PRI_I2S_TX"},
+ {"Voice_Tx Mixer", "MI2S_TX_Voice", "MI2S_TX"},
+ {"Voice_Tx Mixer", "SLIM_0_TX_Voice", "SLIMBUS_0_TX"},
+ {"Voice_Tx Mixer", "INTERNAL_BT_SCO_TX_Voice", "INT_BT_SCO_TX"},
+ {"Voice_Tx Mixer", "AFE_PCM_TX_Voice", "PCM_TX"},
+ {"Voice_Tx Mixer", "AUX_PCM_TX_Voice", "AUX_PCM_TX"},
+ {"CS-VOICE_UL1", NULL, "Voice_Tx Mixer"},
+ {"VoLTE_Tx Mixer", "PRI_TX_VoLTE", "PRI_I2S_TX"},
+ {"VoLTE_Tx Mixer", "SLIM_0_TX_VoLTE", "SLIMBUS_0_TX"},
+ {"VoLTE_Tx Mixer", "INTERNAL_BT_SCO_TX_VoLTE", "INT_BT_SCO_TX"},
+ {"VoLTE_Tx Mixer", "AFE_PCM_TX_VoLTE", "PCM_TX"},
+ {"VoLTE_Tx Mixer", "AUX_PCM_TX_VoLTE", "AUX_PCM_TX"},
+ {"VoLTE_UL", NULL, "VoLTE_Tx Mixer"},
+ {"Voip_Tx Mixer", "PRI_TX_Voip", "PRI_I2S_TX"},
+ {"Voip_Tx Mixer", "MI2S_TX_Voip", "MI2S_TX"},
+ {"Voip_Tx Mixer", "SLIM_0_TX_Voip", "SLIMBUS_0_TX"},
+ {"Voip_Tx Mixer", "INTERNAL_BT_SCO_TX_Voip", "INT_BT_SCO_TX"},
+ {"Voip_Tx Mixer", "AFE_PCM_TX_Voip", "PCM_TX"},
+ {"Voip_Tx Mixer", "AUX_PCM_TX_Voip", "AUX_PCM_TX"},
+
+ {"VOIP_UL", NULL, "Voip_Tx Mixer"},
+ {"SLIMBUS_DL_HL", "Switch", "SLIM0_DL_HL"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_DL_HL"},
+ {"SLIM0_UL_HL", NULL, "SLIMBUS_0_TX"},
+ {"INT_FM_RX", NULL, "INTFM_DL_HL"},
+ {"INTFM_UL_HL", NULL, "INT_FM_TX"},
+ {"AUX_PCM_RX", NULL, "AUXPCM_DL_HL"},
+ {"AUXPCM_UL_HL", NULL, "AUX_PCM_TX"},
+ {"PCM_RX_DL_HL", "Switch", "SLIM0_DL_HL"},
+ {"PCM_RX", NULL, "PCM_RX_DL_HL"},
+ {"MI2S_UL_HL", NULL, "MI2S_TX"},
+ {"SEC_I2S_RX", NULL, "SEC_I2S_DL_HL"},
+ {"SLIMBUS_0_RX Port Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+ {"SLIMBUS_0_RX Port Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+ {"SLIMBUS_0_RX Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+ {"SLIMBUS_0_RX Port Mixer", "MI2S_TX", "MI2S_TX"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Port Mixer"},
+ {"AFE_PCM_RX Port Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+ {"PCM_RX", NULL, "AFE_PCM_RX Port Mixer"},
+
+ {"AUXPCM_RX Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+ {"AUXPCM_RX Port Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+ {"AUX_PCM_RX", NULL, "AUXPCM_RX Port Mixer"},
+
+ {"Voice Stub Tx Mixer", "STUB_TX_HL", "STUB_TX"},
+ {"Voice Stub Tx Mixer", "SLIM_1_TX", "SLIMBUS_1_TX"},
+ {"Voice Stub Tx Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"Voice Stub Tx Mixer", "STUB_1_TX_HL", "STUB_1_TX"},
+ {"Voice Stub Tx Mixer", "MI2S_TX", "MI2S_TX"},
+ {"VOICE_STUB_UL", NULL, "Voice Stub Tx Mixer"},
+
+ {"STUB_RX Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"STUB_RX", NULL, "STUB_RX Mixer"},
+ {"SLIMBUS_1_RX Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Mixer"},
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"MI2S_RX_Voice Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"MI2S_RX", NULL, "MI2S_RX_Voice Mixer"},
+
+ {"SLIMBUS_3_RX_Voice Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"SLIMBUS_3_RX", NULL, "SLIMBUS_3_RX_Voice Mixer"},
+
+ {"SLIMBUS_1_RX Port Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Port Mixer"},
+ {"INTERNAL_BT_SCO_RX Port Mixer", "SLIM_1_TX", "SLIMBUS_1_TX"},
+ {"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Port Mixer"},
+ {"SLIMBUS_3_RX Port Mixer", "INTERNAL_BT_SCO_RX", "INT_BT_SCO_RX"},
+ {"SLIMBUS_3_RX Port Mixer", "MI2S_TX", "MI2S_TX"},
+ {"SLIMBUS_3_RX", NULL, "SLIMBUS_3_RX Port Mixer"},
+
+
+ {"HDMI_RX Port Mixer", "MI2S_TX", "MI2S_TX"},
+ {"HDMI", NULL, "HDMI_RX Port Mixer"},
+
+ {"SEC_I2S_RX Port Mixer", "MI2S_TX", "MI2S_TX"},
+ {"SEC_I2S_RX", NULL, "SEC_I2S_RX Port Mixer"},
+
+ {"MI2S_RX Port Mixer", "SLIM_1_TX", "SLIMBUS_1_TX"},
+ {"MI2S_RX", NULL, "MI2S_RX Port Mixer"},
+ /* Backend Enablement */
+
+ {"BE_OUT", NULL, "PRI_I2S_RX"},
+ {"BE_OUT", NULL, "SEC_I2S_RX"},
+ {"BE_OUT", NULL, "SLIMBUS_0_RX"},
+ {"BE_OUT", NULL, "HDMI"},
+ {"BE_OUT", NULL, "MI2S_RX"},
+ {"PRI_I2S_TX", NULL, "BE_IN"},
+ {"MI2S_TX", NULL, "BE_IN"},
+ {"SLIMBUS_0_TX", NULL, "BE_IN" },
+ {"BE_OUT", NULL, "INT_BT_SCO_RX"},
+ {"INT_BT_SCO_TX", NULL, "BE_IN"},
+ {"BE_OUT", NULL, "INT_FM_RX"},
+ {"INT_FM_TX", NULL, "BE_IN"},
+ {"BE_OUT", NULL, "PCM_RX"},
+ {"PCM_TX", NULL, "BE_IN"},
+ {"BE_OUT", NULL, "SLIMBUS_3_RX"},
+};
+
+static int msm_pcm_routing_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->dai_link->be_id;
+
+ if (be_id >= MSM_BACKEND_DAI_MAX) {
+ pr_err("%s: unexpected be_id %d\n", __func__, be_id);
+ return -EINVAL;
+ }
+
+ mutex_lock(&routing_lock);
+ msm_bedais[be_id].sample_rate = params_rate(params);
+ msm_bedais[be_id].channel = params_channels(params);
+ mutex_unlock(&routing_lock);
+ return 0;
+}
+
+static int msm_pcm_routing_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->dai_link->be_id;
+ int i, session_type;
+ struct msm_pcm_routing_bdai_data *bedai;
+
+ if (be_id >= MSM_BACKEND_DAI_MAX) {
+ pr_err("%s: unexpected be_id %d\n", __func__, be_id);
+ return -EINVAL;
+ }
+
+ bedai = &msm_bedais[be_id];
+ session_type = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ 0 : 1);
+
+ mutex_lock(&routing_lock);
+
+ for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
+ if (fe_dai_map[i][session_type] != INVALID_SESSION) {
+ adm_close(bedai->port_id);
+ srs_port_id = -1;
+ }
+ }
+
+ bedai->active = 0;
+ bedai->sample_rate = 0;
+ bedai->channel = 0;
+ mutex_unlock(&routing_lock);
+
+ return 0;
+}
+
+static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->dai_link->be_id;
+ int i, path_type, session_type;
+ struct msm_pcm_routing_bdai_data *bedai;
+ u32 channels;
+
+ if (be_id >= MSM_BACKEND_DAI_MAX) {
+ pr_err("%s: unexpected be_id %d\n", __func__, be_id);
+ return -EINVAL;
+ }
+
+ bedai = &msm_bedais[be_id];
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ path_type = ADM_PATH_PLAYBACK;
+ session_type = SESSION_TYPE_RX;
+ } else {
+ path_type = ADM_PATH_LIVE_REC;
+ session_type = SESSION_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ if (bedai->active == 1)
+ goto done; /* Ignore prepare if back-end already active */
+
+ /* AFE port is not active at this point. However, still
+ * go ahead setting active flag under the notion that
+ * QDSP6 is able to handle ADM starting before AFE port
+ * is started.
+ */
+ bedai->active = 1;
+ for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
+ if (fe_dai_map[i][session_type] != INVALID_SESSION) {
+
+ channels = bedai->channel;
+ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ && (channels > 2))
+ adm_multi_ch_copp_open(bedai->port_id,
+ path_type,
+ bedai->sample_rate,
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(bedai->port_id,
+ path_type,
+ bedai->sample_rate,
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+
+ msm_pcm_routing_build_matrix(i,
+ fe_dai_map[i][session_type], path_type);
+ srs_port_id = bedai->port_id;
+ srs_send_params(srs_port_id, 1, 0);
+ }
+ }
+
+done:
+ mutex_unlock(&routing_lock);
+
+ return 0;
+}
+
+static struct snd_pcm_ops msm_routing_pcm_ops = {
+ .hw_params = msm_pcm_routing_hw_params,
+ .close = msm_pcm_routing_close,
+ .prepare = msm_pcm_routing_prepare,
+};
+
+static unsigned int msm_routing_read(struct snd_soc_platform *platform,
+ unsigned int reg)
+{
+ dev_dbg(platform->dev, "reg %x\n", reg);
+ return 0;
+}
+
+/* Not used but frame seems to require it */
+static int msm_routing_write(struct snd_soc_platform *platform,
+ unsigned int reg, unsigned int val)
+{
+ dev_dbg(platform->dev, "reg %x val %x\n", reg, val);
+ return 0;
+}
+
+/* Not used but frame seems to require it */
+static int msm_routing_probe(struct snd_soc_platform *platform)
+{
+ snd_soc_dapm_new_controls(&platform->dapm, msm_qdsp6_widgets,
+ ARRAY_SIZE(msm_qdsp6_widgets));
+ snd_soc_dapm_add_routes(&platform->dapm, intercon,
+ ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(&platform->dapm);
+
+ snd_soc_add_platform_controls(platform,
+ int_fm_vol_mixer_controls,
+ ARRAY_SIZE(int_fm_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ lpa_vol_mixer_controls,
+ ARRAY_SIZE(lpa_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ eq_enable_mixer_controls,
+ ARRAY_SIZE(eq_enable_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ eq_band_mixer_controls,
+ ARRAY_SIZE(eq_band_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ eq_coeff_mixer_controls,
+ ARRAY_SIZE(eq_coeff_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ multimedia2_vol_mixer_controls,
+ ARRAY_SIZE(multimedia2_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ compressed_vol_mixer_controls,
+ ARRAY_SIZE(compressed_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ lpa_SRS_trumedia_controls,
+ ARRAY_SIZE(lpa_SRS_trumedia_controls));
+
+ snd_soc_add_platform_controls(platform,
+ lpa_SRS_trumedia_controls_HDMI,
+ ARRAY_SIZE(lpa_SRS_trumedia_controls_HDMI));
+
+ snd_soc_add_platform_controls(platform,
+ lpa_SRS_trumedia_controls_I2S,
+ ARRAY_SIZE(lpa_SRS_trumedia_controls_I2S));
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver msm_soc_routing_platform = {
+ .ops = &msm_routing_pcm_ops,
+ .probe = msm_routing_probe,
+ .read = msm_routing_read,
+ .write = msm_routing_write,
+};
+
+static __devinit int msm_routing_pcm_probe(struct platform_device *pdev)
+{
+ dev_dbg(&pdev->dev, "dev name %s\n", dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_routing_platform);
+}
+
+static int msm_routing_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_routing_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-routing",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_routing_pcm_probe,
+ .remove = __devexit_p(msm_routing_pcm_remove),
+};
+
+int msm_routing_check_backend_enabled(int fedai_id)
+{
+ int i;
+ if (fedai_id >= MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID\n", __func__);
+ return 0;
+ }
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if (test_bit(fedai_id, &msm_bedais[i].fe_sessions))
+ return msm_bedais[i].active;
+ }
+ return 0;
+}
+
+static int __init msm_soc_routing_platform_init(void)
+{
+ mutex_init(&routing_lock);
+ return platform_driver_register(&msm_routing_pcm_driver);
+}
+module_init(msm_soc_routing_platform_init);
+
+static void __exit msm_soc_routing_platform_exit(void)
+{
+ platform_driver_unregister(&msm_routing_pcm_driver);
+}
+module_exit(msm_soc_routing_platform_exit);
+
+MODULE_DESCRIPTION("MSM routing platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-routing.h b/sound/soc/msm/msm-pcm-routing.h
new file mode 100644
index 0000000..ff073ce
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-routing.h
@@ -0,0 +1,118 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef _MSM_PCM_ROUTING_H
+#define _MSM_PCM_ROUTING_H
+#include <sound/apr_audio.h>
+
+#define LPASS_BE_PRI_I2S_RX "PRIMARY_I2S_RX"
+#define LPASS_BE_PRI_I2S_TX "PRIMARY_I2S_TX"
+#define LPASS_BE_SLIMBUS_0_RX "SLIMBUS_0_RX"
+#define LPASS_BE_SLIMBUS_0_TX "SLIMBUS_0_TX"
+#define LPASS_BE_HDMI "HDMI"
+#define LPASS_BE_INT_BT_SCO_RX "INT_BT_SCO_RX"
+#define LPASS_BE_INT_BT_SCO_TX "INT_BT_SCO_TX"
+#define LPASS_BE_INT_FM_RX "INT_FM_RX"
+#define LPASS_BE_INT_FM_TX "INT_FM_TX"
+#define LPASS_BE_AFE_PCM_RX "RT_PROXY_DAI_001_RX"
+#define LPASS_BE_AFE_PCM_TX "RT_PROXY_DAI_002_TX"
+#define LPASS_BE_AUXPCM_RX "AUX_PCM_RX"
+#define LPASS_BE_AUXPCM_TX "AUX_PCM_TX"
+#define LPASS_BE_VOICE_PLAYBACK_TX "VOICE_PLAYBACK_TX"
+#define LPASS_BE_INCALL_RECORD_RX "INCALL_RECORD_TX"
+#define LPASS_BE_INCALL_RECORD_TX "INCALL_RECORD_RX"
+#define LPASS_BE_SEC_I2S_RX "SECONDARY_I2S_RX"
+
+#define LPASS_BE_MI2S_RX "MI2S_RX"
+#define LPASS_BE_MI2S_TX "MI2S_TX"
+#define LPASS_BE_STUB_RX "STUB_RX"
+#define LPASS_BE_STUB_TX "STUB_TX"
+#define LPASS_BE_SLIMBUS_1_RX "SLIMBUS_1_RX"
+#define LPASS_BE_SLIMBUS_1_TX "SLIMBUS_1_TX"
+#define LPASS_BE_STUB_1_TX "STUB_1_TX"
+#define LPASS_BE_SLIMBUS_3_RX "SLIMBUS_3_RX"
+#define LPASS_BE_SLIMBUS_4_RX "SLIMBUS_4_RX"
+#define LPASS_BE_SLIMBUS_4_TX "SLIMBUS_4_TX"
+
+/* For multimedia front-ends, asm session is allocated dynamically.
+ * Hence, asm session/multimedia front-end mapping has to be maintained.
+ * Due to this reason, additional multimedia front-end must be placed before
+ * non-multimedia front-ends.
+ */
+
+enum {
+ MSM_FRONTEND_DAI_MULTIMEDIA1 = 0,
+ MSM_FRONTEND_DAI_MULTIMEDIA2,
+ MSM_FRONTEND_DAI_MULTIMEDIA3,
+ MSM_FRONTEND_DAI_MULTIMEDIA4,
+ MSM_FRONTEND_DAI_CS_VOICE,
+ MSM_FRONTEND_DAI_VOIP,
+ MSM_FRONTEND_DAI_AFE_RX,
+ MSM_FRONTEND_DAI_AFE_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB,
+ MSM_FRONTEND_DAI_VOLTE,
+ MSM_FRONTEND_DAI_MAX,
+};
+
+#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA4 + 1)
+#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA4
+
+enum {
+ MSM_BACKEND_DAI_PRI_I2S_RX = 0,
+ MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_BACKEND_DAI_HDMI_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ MSM_BACKEND_DAI_MI2S_RX,
+ MSM_BACKEND_DAI_MI2S_TX,
+ MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_TX,
+ MSM_BACKEND_DAI_SLIMBUS_4_RX,
+ MSM_BACKEND_DAI_SLIMBUS_4_TX,
+ MSM_BACKEND_DAI_SLIMBUS_3_RX,
+ MSM_BACKEND_DAI_EXTPROC_RX,
+ MSM_BACKEND_DAI_EXTPROC_TX,
+ MSM_BACKEND_DAI_EXTPROC_EC_TX,
+ MSM_BACKEND_DAI_MAX,
+};
+
+/* dai_id: front-end ID,
+ * dspst_id: DSP audio stream ID
+ * stream_type: playback or capture
+ */
+void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id,
+ int stream_type);
+void msm_pcm_routing_reg_psthr_stream(int fedai_id, int dspst_id,
+ int stream_type);
+
+void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type);
+
+int lpa_set_volume(unsigned volume);
+
+int msm_routing_check_backend_enabled(int fedai_id);
+
+int multi_ch_pcm_set_volume(unsigned volume);
+
+int compressed_set_volume(unsigned volume);
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/msm-pcm-voice.c b/sound/soc/msm/msm-pcm-voice.c
new file mode 100644
index 0000000..7bdb4f0
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-voice.c
@@ -0,0 +1,558 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+
+#include "msm-pcm-voice.h"
+#include "qdsp6/q6voice.h"
+
+static struct msm_voice voice_info[VOICE_SESSION_INDEX_MAX];
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+
+ .info = (SNDRV_PCM_INFO_INTERLEAVED|
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 1,
+ .channels_max = 1,
+
+ .buffer_bytes_max = 4096 * 2,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 4096,
+ .periods_min = 2,
+ .periods_max = 2,
+
+ .fifo_size = 0,
+};
+static int is_volte(struct msm_voice *pvolte)
+{
+ if (pvolte == &voice_info[VOLTE_SESSION_INDEX])
+ return true;
+ else
+ return false;
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+
+ if (!prtd->playback_start)
+ prtd->playback_start = 1;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+
+ if (!prtd->capture_start)
+ prtd->capture_start = 1;
+
+ return 0;
+}
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *voice;
+
+ if (!strncmp("VoLTE", substream->pcm->id, 5)) {
+ voice = &voice_info[VOLTE_SESSION_INDEX];
+ pr_debug("%s: Open VoLTE Substream Id=%s\n",
+ __func__, substream->pcm->id);
+ } else {
+ voice = &voice_info[VOICE_SESSION_INDEX];
+ pr_debug("%s: Open VOICE Substream Id=%s\n",
+ __func__, substream->pcm->id);
+ }
+ mutex_lock(&voice->lock);
+
+ runtime->hw = msm_pcm_hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ voice->playback_substream = substream;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ voice->capture_substream = substream;
+
+ voice->instance++;
+ pr_debug("%s: Instance = %d, Stream ID = %s\n",
+ __func__ , voice->instance, substream->pcm->id);
+ runtime->private_data = voice;
+
+ mutex_unlock(&voice->lock);
+
+ return 0;
+}
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+
+ if (prtd->playback_start)
+ prtd->playback_start = 0;
+
+ prtd->playback_substream = NULL;
+
+ return 0;
+}
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+
+ if (prtd->capture_start)
+ prtd->capture_start = 0;
+ prtd->capture_substream = NULL;
+
+ return 0;
+}
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+ uint16_t session_id = 0;
+ int ret = 0;
+
+ mutex_lock(&prtd->lock);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+
+ prtd->instance--;
+ if (!prtd->playback_start && !prtd->capture_start) {
+ pr_debug("end voice call\n");
+ if (is_volte(prtd))
+ session_id = voc_get_session_id(VOLTE_SESSION_NAME);
+ else
+ session_id = voc_get_session_id(VOICE_SESSION_NAME);
+ voc_end_voice_call(session_id);
+ }
+ mutex_unlock(&prtd->lock);
+
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+ uint16_t session_id = 0;
+
+ mutex_lock(&prtd->lock);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+
+ if (prtd->playback_start && prtd->capture_start) {
+ if (is_volte(prtd))
+ session_id = voc_get_session_id(VOLTE_SESSION_NAME);
+ else
+ session_id = voc_get_session_id(VOICE_SESSION_NAME);
+ voc_start_voice_call(session_id);
+ }
+ mutex_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+
+ pr_debug("%s: Voice\n", __func__);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_voice *prtd = runtime->private_data;
+ uint16_t session_id = 0;
+
+ pr_debug("%s: cmd = %d\n", __func__, cmd);
+ if (is_volte(prtd))
+ session_id = voc_get_session_id(VOLTE_SESSION_NAME);
+ else
+ session_id = voc_get_session_id(VOICE_SESSION_NAME);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("Start & Stop Voice call not handled in Trigger.\n");
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: resume call session_id = %d\n", __func__,
+ session_id);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ if (prtd->playback_start && prtd->capture_start)
+ voc_resume_voice_call(session_id);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("%s: pause call session_id=%d\n",
+ __func__, session_id);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (prtd->playback_start)
+ prtd->playback_start = 0;
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (prtd->capture_start)
+ prtd->capture_start = 0;
+ }
+ voc_standby_voice_call(session_id);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+
+static int msm_voice_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_voice_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int volume = ucontrol->value.integer.value[0];
+ pr_debug("%s: volume: %d\n", __func__, volume);
+ voc_set_rx_vol_index(voc_get_session_id(VOICE_SESSION_NAME),
+ RX_PATH, volume);
+ return 0;
+}
+
+static int msm_volte_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_volte_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int volume = ucontrol->value.integer.value[0];
+ pr_debug("%s: volume: %d\n", __func__, volume);
+ voc_set_rx_vol_index(voc_get_session_id(VOLTE_SESSION_NAME),
+ RX_PATH, volume);
+ return 0;
+}
+
+static int msm_voice_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_voice_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int mute = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: mute=%d\n", __func__, mute);
+
+ voc_set_tx_mute(voc_get_session_id(VOICE_SESSION_NAME), TX_PATH, mute);
+
+ return 0;
+}
+
+static int msm_volte_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_volte_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int mute = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: mute=%d\n", __func__, mute);
+
+ voc_set_tx_mute(voc_get_session_id(VOLTE_SESSION_NAME), TX_PATH, mute);
+
+ return 0;
+}
+
+static int msm_voice_rx_device_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voc_get_rx_device_mute(voc_get_session_id(VOICE_SESSION_NAME));
+ return 0;
+}
+
+static int msm_voice_rx_device_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int mute = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: mute=%d\n", __func__, mute);
+
+ voc_set_rx_device_mute(voc_get_session_id(VOICE_SESSION_NAME), mute);
+
+ return 0;
+}
+
+static int msm_volte_rx_device_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voc_get_rx_device_mute(voc_get_session_id(VOLTE_SESSION_NAME));
+ return 0;
+}
+
+static int msm_volte_rx_device_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int mute = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: mute=%d\n", __func__, mute);
+
+ voc_set_rx_device_mute(voc_get_session_id(VOLTE_SESSION_NAME), mute);
+
+ return 0;
+}
+
+static const char const *tty_mode[] = {"OFF", "HCO", "VCO", "FULL"};
+static const struct soc_enum msm_tty_mode_enum[] = {
+ SOC_ENUM_SINGLE_EXT(4, tty_mode),
+};
+
+static int msm_voice_tty_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voc_get_tty_mode(voc_get_session_id(VOICE_SESSION_NAME));
+ return 0;
+}
+
+static int msm_voice_tty_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int tty_mode = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: tty_mode=%d\n", __func__, tty_mode);
+
+ voc_set_tty_mode(voc_get_session_id(VOICE_SESSION_NAME), tty_mode);
+
+ return 0;
+}
+static int msm_voice_widevoice_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int wv_enable = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: wv enable=%d\n", __func__, wv_enable);
+
+ voc_set_widevoice_enable(voc_get_session_id(VOICE_SESSION_NAME),
+ wv_enable);
+
+ return 0;
+}
+
+static int msm_voice_widevoice_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voc_get_widevoice_enable(voc_get_session_id(VOICE_SESSION_NAME));
+ return 0;
+}
+
+
+static int msm_voice_slowtalk_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int st_enable = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: st enable=%d\n", __func__, st_enable);
+
+ voc_set_pp_enable(voc_get_session_id(VOICE_SESSION_NAME),
+ MODULE_ID_VOICE_MODULE_ST, st_enable);
+
+ return 0;
+}
+
+static int msm_voice_slowtalk_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voc_get_pp_enable(voc_get_session_id(VOICE_SESSION_NAME),
+ MODULE_ID_VOICE_MODULE_ST);
+ return 0;
+}
+
+static int msm_voice_fens_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int fens_enable = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: fens enable=%d\n", __func__, fens_enable);
+
+ voc_set_pp_enable(voc_get_session_id(VOICE_SESSION_NAME),
+ MODULE_ID_VOICE_MODULE_FENS, fens_enable);
+
+ return 0;
+}
+
+static int msm_voice_fens_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voc_get_pp_enable(voc_get_session_id(VOICE_SESSION_NAME),
+ MODULE_ID_VOICE_MODULE_FENS);
+ return 0;
+}
+
+static struct snd_kcontrol_new msm_voice_controls[] = {
+ SOC_SINGLE_EXT("Voice Rx Device Mute", SND_SOC_NOPM, 0, 1, 0,
+ msm_voice_rx_device_mute_get,
+ msm_voice_rx_device_mute_put),
+ SOC_SINGLE_EXT("Voice Tx Mute", SND_SOC_NOPM, 0, 1, 0,
+ msm_voice_mute_get, msm_voice_mute_put),
+ SOC_SINGLE_EXT("Voice Rx Volume", SND_SOC_NOPM, 0, 5, 0,
+ msm_voice_volume_get, msm_voice_volume_put),
+ SOC_ENUM_EXT("TTY Mode", msm_tty_mode_enum[0], msm_voice_tty_mode_get,
+ msm_voice_tty_mode_put),
+ SOC_SINGLE_EXT("Widevoice Enable", SND_SOC_NOPM, 0, 1, 0,
+ msm_voice_widevoice_get, msm_voice_widevoice_put),
+ SOC_SINGLE_EXT("Slowtalk Enable", SND_SOC_NOPM, 0, 1, 0,
+ msm_voice_slowtalk_get, msm_voice_slowtalk_put),
+ SOC_SINGLE_EXT("FENS Enable", SND_SOC_NOPM, 0, 1, 0,
+ msm_voice_fens_get, msm_voice_fens_put),
+ SOC_SINGLE_EXT("VoLTE Rx Device Mute", SND_SOC_NOPM, 0, 1, 0,
+ msm_volte_rx_device_mute_get,
+ msm_volte_rx_device_mute_put),
+ SOC_SINGLE_EXT("VoLTE Tx Mute", SND_SOC_NOPM, 0, 1, 0,
+ msm_volte_mute_get, msm_volte_mute_put),
+ SOC_SINGLE_EXT("VoLTE Rx Volume", SND_SOC_NOPM, 0, 5, 0,
+ msm_volte_volume_get, msm_volte_volume_put),
+};
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+};
+
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static int msm_pcm_voice_probe(struct snd_soc_platform *platform)
+{
+ snd_soc_add_platform_controls(platform, msm_voice_controls,
+ ARRAY_SIZE(msm_voice_controls));
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+ .probe = msm_pcm_voice_probe,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-voice",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ memset(&voice_info, 0, sizeof(voice_info));
+ mutex_init(&voice_info[VOICE_SESSION_INDEX].lock);
+ mutex_init(&voice_info[VOLTE_SESSION_INDEX].lock);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("Voice PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm-voice.h b/sound/soc/msm/msm-pcm-voice.h
new file mode 100644
index 0000000..aa00577
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-voice.h
@@ -0,0 +1,37 @@
+/* Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef _MSM_PCM_VOICE_H
+#define _MSM_PCM_VOICE_H
+#include <sound/apr_audio.h>
+
+enum {
+ VOICE_SESSION_INDEX,
+ VOLTE_SESSION_INDEX,
+ VOICE_SESSION_INDEX_MAX,
+};
+
+struct msm_voice {
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ int instance;
+
+ struct mutex lock;
+
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+
+ int playback_start;
+ int capture_start;
+};
+
+#endif /*_MSM_PCM_VOICE_H*/
diff --git a/sound/soc/msm/msm-pcm-voip.c b/sound/soc/msm/msm-pcm-voip.c
new file mode 100644
index 0000000..570d71c
--- /dev/null
+++ b/sound/soc/msm/msm-pcm-voip.c
@@ -0,0 +1,1169 @@
+/* Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+
+#include "msm-pcm-q6.h"
+#include "msm-pcm-routing.h"
+#include "qdsp6/q6voice.h"
+
+#define VOIP_MAX_Q_LEN 10
+#define VOIP_MAX_VOC_PKT_SIZE 640
+#define VOIP_MIN_VOC_PKT_SIZE 320
+
+/* Length of the DSP frame info header added to the voc packet. */
+#define DSP_FRAME_HDR_LEN 1
+
+#define MODE_IS127 0x2
+#define MODE_4GV_NB 0x3
+#define MODE_4GV_WB 0x4
+#define MODE_AMR 0x5
+#define MODE_AMR_WB 0xD
+#define MODE_PCM 0xC
+
+enum format {
+ FORMAT_S16_LE = 2,
+ FORMAT_SPECIAL = 31,
+};
+
+
+enum amr_rate_type {
+ AMR_RATE_4750, /* AMR 4.75 kbps */
+ AMR_RATE_5150, /* AMR 5.15 kbps */
+ AMR_RATE_5900, /* AMR 5.90 kbps */
+ AMR_RATE_6700, /* AMR 6.70 kbps */
+ AMR_RATE_7400, /* AMR 7.40 kbps */
+ AMR_RATE_7950, /* AMR 7.95 kbps */
+ AMR_RATE_10200, /* AMR 10.20 kbps */
+ AMR_RATE_12200, /* AMR 12.20 kbps */
+ AMR_RATE_6600, /* AMR-WB 6.60 kbps */
+ AMR_RATE_8850, /* AMR-WB 8.85 kbps */
+ AMR_RATE_12650, /* AMR-WB 12.65 kbps */
+ AMR_RATE_14250, /* AMR-WB 14.25 kbps */
+ AMR_RATE_15850, /* AMR-WB 15.85 kbps */
+ AMR_RATE_18250, /* AMR-WB 18.25 kbps */
+ AMR_RATE_19850, /* AMR-WB 19.85 kbps */
+ AMR_RATE_23050, /* AMR-WB 23.05 kbps */
+ AMR_RATE_23850, /* AMR-WB 23.85 kbps */
+ AMR_RATE_UNDEF
+};
+
+enum voip_state {
+ VOIP_STOPPED,
+ VOIP_STARTED,
+};
+
+struct voip_frame {
+ union {
+ uint32_t frame_type;
+ uint32_t packet_rate;
+ } header;
+ uint32_t len;
+ uint8_t voc_pkt[VOIP_MAX_VOC_PKT_SIZE];
+};
+
+struct voip_buf_node {
+ struct list_head list;
+ struct voip_frame frame;
+};
+
+struct voip_drv_info {
+ enum voip_state state;
+
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ struct list_head in_queue;
+ struct list_head free_in_queue;
+
+ struct list_head out_queue;
+ struct list_head free_out_queue;
+
+ wait_queue_head_t out_wait;
+ wait_queue_head_t in_wait;
+
+ struct mutex lock;
+ struct mutex in_lock;
+ struct mutex out_lock;
+
+ spinlock_t dsp_lock;
+
+ uint32_t mode;
+ uint32_t rate_type;
+ uint32_t rate;
+ uint32_t dtx_mode;
+
+ uint8_t capture_start;
+ uint8_t playback_start;
+
+ uint8_t playback_instance;
+ uint8_t capture_instance;
+
+ unsigned int play_samp_rate;
+ unsigned int cap_samp_rate;
+
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_playback_irq_pos; /* IRQ position */
+ unsigned int pcm_playback_buf_pos; /* position in buffer */
+
+ unsigned int pcm_capture_size;
+ unsigned int pcm_capture_count;
+ unsigned int pcm_capture_irq_pos; /* IRQ position */
+ unsigned int pcm_capture_buf_pos; /* position in buffer */
+};
+
+static int voip_get_media_type(uint32_t mode,
+ unsigned int samp_rate);
+static int voip_get_rate_type(uint32_t mode,
+ uint32_t rate,
+ uint32_t *rate_type);
+static int msm_voip_mode_rate_config_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+static int msm_voip_mode_rate_config_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
+static struct voip_drv_info voip_info;
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_SPECIAL,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = sizeof(struct voip_buf_node) * VOIP_MAX_Q_LEN,
+ .period_bytes_min = VOIP_MIN_VOC_PKT_SIZE,
+ .period_bytes_max = VOIP_MAX_VOC_PKT_SIZE,
+ .periods_min = VOIP_MAX_Q_LEN,
+ .periods_max = VOIP_MAX_Q_LEN,
+ .fifo_size = 0,
+};
+
+
+static int msm_voip_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int mute = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: mute=%d\n", __func__, mute);
+
+ voc_set_tx_mute(voc_get_session_id(VOIP_SESSION_NAME), TX_PATH, mute);
+
+ return 0;
+}
+
+static int msm_voip_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_voip_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int volume = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: volume: %d\n", __func__, volume);
+
+ voc_set_rx_vol_index(voc_get_session_id(VOIP_SESSION_NAME),
+ RX_PATH,
+ volume);
+ return 0;
+}
+static int msm_voip_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_voip_dtx_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ mutex_lock(&voip_info.lock);
+
+ voip_info.dtx_mode = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: dtx: %d\n", __func__, voip_info.dtx_mode);
+
+ mutex_unlock(&voip_info.lock);
+
+ return 0;
+}
+static int msm_voip_dtx_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ mutex_lock(&voip_info.lock);
+
+ ucontrol->value.integer.value[0] = voip_info.dtx_mode;
+
+ mutex_unlock(&voip_info.lock);
+
+ return 0;
+}
+
+static struct snd_kcontrol_new msm_voip_controls[] = {
+ SOC_SINGLE_EXT("Voip Tx Mute", SND_SOC_NOPM, 0, 1, 0,
+ msm_voip_mute_get, msm_voip_mute_put),
+ SOC_SINGLE_EXT("Voip Rx Volume", SND_SOC_NOPM, 0, 5, 0,
+ msm_voip_volume_get, msm_voip_volume_put),
+ SOC_SINGLE_MULTI_EXT("Voip Mode Rate Config", SND_SOC_NOPM, 0, 23850,
+ 0, 2, msm_voip_mode_rate_config_get,
+ msm_voip_mode_rate_config_put),
+ SOC_SINGLE_EXT("Voip Dtx Mode", SND_SOC_NOPM, 0, 1, 0,
+ msm_voip_dtx_mode_get, msm_voip_dtx_mode_put),
+};
+
+static int msm_pcm_voip_probe(struct snd_soc_platform *platform)
+{
+ snd_soc_add_platform_controls(platform, msm_voip_controls,
+ ARRAY_SIZE(msm_voip_controls));
+
+ return 0;
+}
+
+/* sample rate supported */
+static unsigned int supported_sample_rates[] = {8000, 16000};
+
+/* capture path */
+static void voip_process_ul_pkt(uint8_t *voc_pkt,
+ uint32_t pkt_len,
+ void *private_data)
+{
+ struct voip_buf_node *buf_node = NULL;
+ struct voip_drv_info *prtd = private_data;
+ unsigned long dsp_flags;
+
+ if (prtd->capture_substream == NULL)
+ return;
+
+ /* Copy up-link packet into out_queue. */
+ spin_lock_irqsave(&prtd->dsp_lock, dsp_flags);
+
+ /* discarding UL packets till start is received */
+ if (!list_empty(&prtd->free_out_queue) && prtd->capture_start) {
+ buf_node = list_first_entry(&prtd->free_out_queue,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ switch (prtd->mode) {
+ case MODE_AMR_WB:
+ case MODE_AMR: {
+ /* Remove the DSP frame info header. Header format:
+ * Bits 0-3: Frame rate
+ * Bits 4-7: Frame type
+ */
+ buf_node->frame.header.frame_type =
+ ((*voc_pkt) & 0xF0) >> 4;
+ voc_pkt = voc_pkt + DSP_FRAME_HDR_LEN;
+ buf_node->frame.len = pkt_len - DSP_FRAME_HDR_LEN;
+ memcpy(&buf_node->frame.voc_pkt[0],
+ voc_pkt,
+ buf_node->frame.len);
+ list_add_tail(&buf_node->list, &prtd->out_queue);
+ break;
+ }
+ case MODE_IS127:
+ case MODE_4GV_NB:
+ case MODE_4GV_WB: {
+ /* Remove the DSP frame info header.
+ * Header format:
+ * Bits 0-3: frame rate
+ */
+ buf_node->frame.header.packet_rate = (*voc_pkt) & 0x0F;
+ voc_pkt = voc_pkt + DSP_FRAME_HDR_LEN;
+ buf_node->frame.len = pkt_len - DSP_FRAME_HDR_LEN;
+
+ memcpy(&buf_node->frame.voc_pkt[0],
+ voc_pkt,
+ buf_node->frame.len);
+
+ list_add_tail(&buf_node->list, &prtd->out_queue);
+ break;
+ }
+ default: {
+ buf_node->frame.len = pkt_len;
+ memcpy(&buf_node->frame.voc_pkt[0],
+ voc_pkt,
+ buf_node->frame.len);
+ list_add_tail(&buf_node->list, &prtd->out_queue);
+ }
+ }
+ pr_debug("ul_pkt: pkt_len =%d, frame.len=%d\n", pkt_len,
+ buf_node->frame.len);
+ prtd->pcm_capture_irq_pos += prtd->pcm_capture_count;
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+ snd_pcm_period_elapsed(prtd->capture_substream);
+ } else {
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+ pr_err("UL data dropped\n");
+ }
+
+ wake_up(&prtd->out_wait);
+}
+
+/* playback path */
+static void voip_process_dl_pkt(uint8_t *voc_pkt,
+ uint32_t *pkt_len,
+ void *private_data)
+{
+ struct voip_buf_node *buf_node = NULL;
+ struct voip_drv_info *prtd = private_data;
+ unsigned long dsp_flags;
+
+
+ if (prtd->playback_substream == NULL)
+ return;
+
+ spin_lock_irqsave(&prtd->dsp_lock, dsp_flags);
+
+ if (!list_empty(&prtd->in_queue) && prtd->playback_start) {
+ buf_node = list_first_entry(&prtd->in_queue,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ switch (prtd->mode) {
+ case MODE_AMR:
+ case MODE_AMR_WB: {
+ /* Add the DSP frame info header. Header format:
+ * Bits 0-3: Frame rate
+ * Bits 4-7: Frame type
+ */
+ *voc_pkt = ((buf_node->frame.header.frame_type &
+ 0x0F) << 4) | (prtd->rate_type & 0x0F);
+ voc_pkt = voc_pkt + DSP_FRAME_HDR_LEN;
+ *pkt_len = buf_node->frame.len + DSP_FRAME_HDR_LEN;
+ memcpy(voc_pkt,
+ &buf_node->frame.voc_pkt[0],
+ buf_node->frame.len);
+ list_add_tail(&buf_node->list, &prtd->free_in_queue);
+ break;
+ }
+ case MODE_IS127:
+ case MODE_4GV_NB:
+ case MODE_4GV_WB: {
+ /* Add the DSP frame info header. Header format:
+ * Bits 0-3 : Frame rate
+ */
+ *voc_pkt = buf_node->frame.header.packet_rate & 0x0F;
+ voc_pkt = voc_pkt + DSP_FRAME_HDR_LEN;
+ *pkt_len = buf_node->frame.len + DSP_FRAME_HDR_LEN;
+
+ memcpy(voc_pkt,
+ &buf_node->frame.voc_pkt[0],
+ buf_node->frame.len);
+
+ list_add_tail(&buf_node->list, &prtd->free_in_queue);
+ break;
+ }
+ default: {
+ *pkt_len = buf_node->frame.len;
+
+ memcpy(voc_pkt,
+ &buf_node->frame.voc_pkt[0],
+ buf_node->frame.len);
+
+ list_add_tail(&buf_node->list, &prtd->free_in_queue);
+ }
+ }
+ pr_debug("dl_pkt: pkt_len=%d, frame_len=%d\n", *pkt_len,
+ buf_node->frame.len);
+ prtd->pcm_playback_irq_pos += prtd->pcm_count;
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+ snd_pcm_period_elapsed(prtd->playback_substream);
+ } else {
+ *pkt_len = 0;
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+ pr_err("DL data not available\n");
+ }
+ wake_up(&prtd->in_wait);
+}
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+
+ prtd->play_samp_rate = runtime->rate;
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_playback_irq_pos = 0;
+ prtd->pcm_playback_buf_pos = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+ int ret = 0;
+
+ prtd->cap_samp_rate = runtime->rate;
+ prtd->pcm_capture_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_capture_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_capture_irq_pos = 0;
+ prtd->pcm_capture_buf_pos = 0;
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ pr_debug("%s: Trigger start\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ prtd->capture_start = 1;
+ else
+ prtd->playback_start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->playback_start = 0;
+ else
+ prtd->capture_start = 0;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = &voip_info;
+ int ret = 0;
+
+ pr_debug("%s, VoIP\n", __func__);
+ mutex_lock(&prtd->lock);
+
+ runtime->hw = msm_pcm_hardware;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_list failed\n");
+
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ pr_debug("snd_pcm_hw_constraint_integer failed\n");
+ goto err;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->playback_substream = substream;
+ prtd->playback_instance++;
+ } else {
+ prtd->capture_substream = substream;
+ prtd->capture_instance++;
+ }
+ runtime->private_data = prtd;
+err:
+ mutex_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ struct voip_buf_node *buf_node = NULL;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+
+ int count = frames_to_bytes(runtime, frames);
+ pr_debug("%s: count = %d, frames=%d\n", __func__, count, (int)frames);
+
+ ret = wait_event_interruptible_timeout(prtd->in_wait,
+ (!list_empty(&prtd->free_in_queue) ||
+ prtd->state == VOIP_STOPPED),
+ 1 * HZ);
+ if (ret > 0) {
+ mutex_lock(&prtd->in_lock);
+ if (count <= VOIP_MAX_VOC_PKT_SIZE) {
+ buf_node =
+ list_first_entry(&prtd->free_in_queue,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ if (prtd->mode == MODE_PCM) {
+ ret = copy_from_user(&buf_node->frame.voc_pkt,
+ buf, count);
+ buf_node->frame.len = count;
+ } else
+ ret = copy_from_user(&buf_node->frame,
+ buf, count);
+ list_add_tail(&buf_node->list, &prtd->in_queue);
+ } else {
+ pr_err("%s: Write cnt %d is > VOIP_MAX_VOC_PKT_SIZE\n",
+ __func__, count);
+ ret = -ENOMEM;
+ }
+
+ mutex_unlock(&prtd->in_lock);
+ } else if (ret == 0) {
+ pr_err("%s: No free DL buffs\n", __func__);
+ ret = -ETIMEDOUT;
+ } else {
+ pr_err("%s: playback copy was interrupted\n", __func__);
+ }
+
+ return ret;
+}
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int count = 0;
+ struct voip_buf_node *buf_node = NULL;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+
+ count = frames_to_bytes(runtime, frames);
+
+ pr_debug("%s: count = %d\n", __func__, count);
+
+ ret = wait_event_interruptible_timeout(prtd->out_wait,
+ (!list_empty(&prtd->out_queue) ||
+ prtd->state == VOIP_STOPPED),
+ 1 * HZ);
+
+ if (ret > 0) {
+ mutex_lock(&prtd->out_lock);
+
+ if (count <= VOIP_MAX_VOC_PKT_SIZE) {
+ buf_node = list_first_entry(&prtd->out_queue,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ if (prtd->mode == MODE_PCM)
+ ret = copy_to_user(buf,
+ &buf_node->frame.voc_pkt,
+ count);
+ else
+ ret = copy_to_user(buf,
+ &buf_node->frame,
+ count);
+ if (ret) {
+ pr_err("%s: Copy to user retuned %d\n",
+ __func__, ret);
+ ret = -EFAULT;
+ }
+ list_add_tail(&buf_node->list,
+ &prtd->free_out_queue);
+ } else {
+ pr_err("%s: Read count %d > VOIP_MAX_VOC_PKT_SIZE\n",
+ __func__, count);
+ ret = -ENOMEM;
+ }
+
+ mutex_unlock(&prtd->out_lock);
+
+ } else if (ret == 0) {
+ pr_err("%s: No UL data available\n", __func__);
+ ret = -ETIMEDOUT;
+ } else {
+ pr_err("%s: Read was interrupted\n", __func__);
+ ret = -ERESTARTSYS;
+ }
+ return ret;
+}
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct list_head *ptr = NULL;
+ struct list_head *next = NULL;
+ struct voip_buf_node *buf_node = NULL;
+ struct snd_dma_buffer *p_dma_buf, *c_dma_buf;
+ struct snd_pcm_substream *p_substream, *c_substream;
+ struct snd_pcm_runtime *runtime;
+ struct voip_drv_info *prtd;
+
+ if (substream == NULL) {
+ pr_err("substream is NULL\n");
+ return -EINVAL;
+ }
+ runtime = substream->runtime;
+ prtd = runtime->private_data;
+
+ wake_up(&prtd->out_wait);
+
+ mutex_lock(&prtd->lock);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->playback_instance--;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ prtd->capture_instance--;
+
+ if (!prtd->playback_instance && !prtd->capture_instance) {
+ if (prtd->state == VOIP_STARTED) {
+ prtd->state = VOIP_STOPPED;
+ voc_end_voice_call(
+ voc_get_session_id(VOIP_SESSION_NAME));
+ voc_register_mvs_cb(NULL, NULL, prtd);
+ }
+ /* release all buffer */
+ /* release in_queue and free_in_queue */
+ pr_debug("release all buffer\n");
+ p_substream = prtd->playback_substream;
+ if (p_substream == NULL) {
+ pr_debug("p_substream is NULL\n");
+ goto capt;
+ }
+ p_dma_buf = &p_substream->dma_buffer;
+ if (p_dma_buf == NULL) {
+ pr_debug("p_dma_buf is NULL\n");
+ goto capt;
+ }
+ if (p_dma_buf->area != NULL) {
+ mutex_lock(&prtd->in_lock);
+ list_for_each_safe(ptr, next, &prtd->in_queue) {
+ buf_node = list_entry(ptr,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ }
+ list_for_each_safe(ptr, next, &prtd->free_in_queue) {
+ buf_node = list_entry(ptr,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ }
+ dma_free_coherent(p_substream->pcm->card->dev,
+ runtime->hw.buffer_bytes_max, p_dma_buf->area,
+ p_dma_buf->addr);
+ p_dma_buf->area = NULL;
+ mutex_unlock(&prtd->in_lock);
+ }
+ /* release out_queue and free_out_queue */
+capt: c_substream = prtd->capture_substream;
+ if (c_substream == NULL) {
+ pr_debug("c_substream is NULL\n");
+ goto done;
+ }
+ c_dma_buf = &c_substream->dma_buffer;
+ if (c_substream == NULL) {
+ pr_debug("c_dma_buf is NULL.\n");
+ goto done;
+ }
+ if (c_dma_buf->area != NULL) {
+ mutex_lock(&prtd->out_lock);
+ list_for_each_safe(ptr, next, &prtd->out_queue) {
+ buf_node = list_entry(ptr,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ }
+ list_for_each_safe(ptr, next, &prtd->free_out_queue) {
+ buf_node = list_entry(ptr,
+ struct voip_buf_node, list);
+ list_del(&buf_node->list);
+ }
+ dma_free_coherent(c_substream->pcm->card->dev,
+ runtime->hw.buffer_bytes_max, c_dma_buf->area,
+ c_dma_buf->addr);
+ c_dma_buf->area = NULL;
+ mutex_unlock(&prtd->out_lock);
+ }
+done:
+ prtd->capture_substream = NULL;
+ prtd->playback_substream = NULL;
+ }
+ mutex_unlock(&prtd->lock);
+
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+ uint32_t media_type = 0;
+ uint32_t rate_type = 0;
+
+ mutex_lock(&prtd->lock);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+
+ if ((runtime->format != FORMAT_SPECIAL) &&
+ ((prtd->mode == MODE_AMR) || (prtd->mode == MODE_AMR_WB) ||
+ (prtd->mode == MODE_IS127) || (prtd->mode == MODE_4GV_NB) ||
+ (prtd->mode == MODE_4GV_WB))) {
+ pr_err("mode:%d and format:%u are not mached\n",
+ prtd->mode, (uint32_t)runtime->format);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if ((runtime->format != FORMAT_S16_LE) &&
+ (prtd->mode == MODE_PCM)) {
+ pr_err("mode:%d and format:%u are not mached\n",
+ prtd->mode, (uint32_t)runtime->format);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (prtd->playback_instance && prtd->capture_instance
+ && (prtd->state != VOIP_STARTED)) {
+
+ ret = voip_get_rate_type(prtd->mode,
+ prtd->rate,
+ &rate_type);
+ if (ret < 0) {
+ pr_err("fail at getting rate_type\n");
+ ret = -EINVAL;
+ goto done;
+ }
+ prtd->rate_type = rate_type;
+ media_type = voip_get_media_type(prtd->mode,
+ prtd->play_samp_rate);
+ if (media_type < 0) {
+ pr_err("fail at getting media_type\n");
+ ret = -EINVAL;
+ goto done;
+ }
+ pr_debug(" media_type=%d, rate_type=%d\n", media_type,
+ rate_type);
+ if ((prtd->play_samp_rate == 8000) &&
+ (prtd->cap_samp_rate == 8000))
+ voc_config_vocoder(media_type, rate_type,
+ VSS_NETWORK_ID_VOIP_NB,
+ voip_info.dtx_mode);
+ else if ((prtd->play_samp_rate == 16000) &&
+ (prtd->cap_samp_rate == 16000))
+ voc_config_vocoder(media_type, rate_type,
+ VSS_NETWORK_ID_VOIP_WB,
+ voip_info.dtx_mode);
+ else {
+ pr_debug("%s: Invalid rate playback %d, capture %d\n",
+ __func__, prtd->play_samp_rate,
+ prtd->cap_samp_rate);
+ goto done;
+ }
+ voc_register_mvs_cb(voip_process_ul_pkt,
+ voip_process_dl_pkt, prtd);
+ voc_start_voice_call(voc_get_session_id(VOIP_SESSION_NAME));
+
+ prtd->state = VOIP_STARTED;
+ }
+done:
+ mutex_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t
+msm_pcm_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ if (prtd->pcm_playback_irq_pos >= prtd->pcm_size)
+ prtd->pcm_playback_irq_pos = 0;
+ return bytes_to_frames(runtime, (prtd->pcm_playback_irq_pos));
+}
+
+static snd_pcm_uframes_t
+msm_pcm_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct voip_drv_info *prtd = runtime->private_data;
+
+ if (prtd->pcm_capture_irq_pos >= prtd->pcm_capture_size)
+ prtd->pcm_capture_irq_pos = 0;
+ return bytes_to_frames(runtime, (prtd->pcm_capture_irq_pos));
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t ret = 0;
+ pr_debug("%s\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_pointer(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_pointer(substream);
+ return ret;
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ pr_debug("%s\n", __func__);
+ dma_mmap_coherent(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return 0;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct voip_buf_node *buf_node = NULL;
+ int i = 0, offset = 0;
+
+ pr_debug("%s: voip\n", __func__);
+
+ mutex_lock(&voip_info.lock);
+
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+
+ dma_buf->area = dma_alloc_coherent(substream->pcm->card->dev,
+ runtime->hw.buffer_bytes_max,
+ &dma_buf->addr, GFP_KERNEL);
+ if (!dma_buf->area) {
+ pr_err("%s:MSM VOIP dma_alloc failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ memset(dma_buf->area, 0, runtime->hw.buffer_bytes_max);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < VOIP_MAX_Q_LEN; i++) {
+ buf_node = (void *)dma_buf->area + offset;
+
+ mutex_lock(&voip_info.in_lock);
+ list_add_tail(&buf_node->list,
+ &voip_info.free_in_queue);
+ mutex_unlock(&voip_info.in_lock);
+ offset = offset + sizeof(struct voip_buf_node);
+ }
+ } else {
+ for (i = 0; i < VOIP_MAX_Q_LEN; i++) {
+ buf_node = (void *) dma_buf->area + offset;
+ mutex_lock(&voip_info.out_lock);
+ list_add_tail(&buf_node->list,
+ &voip_info.free_out_queue);
+ mutex_unlock(&voip_info.out_lock);
+ offset = offset + sizeof(struct voip_buf_node);
+ }
+ }
+
+ mutex_unlock(&voip_info.lock);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int msm_voip_mode_rate_config_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ mutex_lock(&voip_info.lock);
+
+ ucontrol->value.integer.value[0] = voip_info.mode;
+ ucontrol->value.integer.value[1] = voip_info.rate;
+
+ mutex_unlock(&voip_info.lock);
+
+ return 0;
+}
+
+static int msm_voip_mode_rate_config_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ mutex_lock(&voip_info.lock);
+
+ voip_info.mode = ucontrol->value.integer.value[0];
+ voip_info.rate = ucontrol->value.integer.value[1];
+
+ pr_debug("%s: mode=%d,rate=%d\n", __func__, voip_info.mode,
+ voip_info.rate);
+
+ mutex_unlock(&voip_info.lock);
+
+ return 0;
+}
+
+static int voip_get_rate_type(uint32_t mode, uint32_t rate,
+ uint32_t *rate_type)
+{
+ int ret = 0;
+
+ switch (mode) {
+ case MODE_AMR: {
+ switch (rate) {
+ case 4750:
+ *rate_type = AMR_RATE_4750;
+ break;
+ case 5150:
+ *rate_type = AMR_RATE_5150;
+ break;
+ case 5900:
+ *rate_type = AMR_RATE_5900;
+ break;
+ case 6700:
+ *rate_type = AMR_RATE_6700;
+ break;
+ case 7400:
+ *rate_type = AMR_RATE_7400;
+ break;
+ case 7950:
+ *rate_type = AMR_RATE_7950;
+ break;
+ case 10200:
+ *rate_type = AMR_RATE_10200;
+ break;
+ case 12200:
+ *rate_type = AMR_RATE_12200;
+ break;
+ default:
+ pr_err("wrong rate for AMR NB.\n");
+ ret = -EINVAL;
+ break;
+ }
+ break;
+ }
+ case MODE_AMR_WB: {
+ switch (rate) {
+ case 6600:
+ *rate_type = AMR_RATE_6600 - AMR_RATE_6600;
+ break;
+ case 8850:
+ *rate_type = AMR_RATE_8850 - AMR_RATE_6600;
+ break;
+ case 12650:
+ *rate_type = AMR_RATE_12650 - AMR_RATE_6600;
+ break;
+ case 14250:
+ *rate_type = AMR_RATE_14250 - AMR_RATE_6600;
+ break;
+ case 15850:
+ *rate_type = AMR_RATE_15850 - AMR_RATE_6600;
+ break;
+ case 18250:
+ *rate_type = AMR_RATE_18250 - AMR_RATE_6600;
+ break;
+ case 19850:
+ *rate_type = AMR_RATE_19850 - AMR_RATE_6600;
+ break;
+ case 23050:
+ *rate_type = AMR_RATE_23050 - AMR_RATE_6600;
+ break;
+ case 23850:
+ *rate_type = AMR_RATE_23850 - AMR_RATE_6600;
+ break;
+ default:
+ pr_err("wrong rate for AMR_WB.\n");
+ ret = -EINVAL;
+ break;
+ }
+ break;
+ }
+ case MODE_PCM: {
+ *rate_type = 0;
+ break;
+ }
+ case MODE_IS127:
+ case MODE_4GV_NB:
+ case MODE_4GV_WB: {
+ switch (rate) {
+ case VOC_0_RATE:
+ case VOC_8_RATE:
+ case VOC_4_RATE:
+ case VOC_2_RATE:
+ case VOC_1_RATE:
+ *rate_type = rate;
+ break;
+ default:
+ pr_err("wrong rate for IS127/4GV_NB/WB.\n");
+ ret = -EINVAL;
+ break;
+ }
+ break;
+ }
+ default:
+ pr_err("wrong mode type.\n");
+ ret = -EINVAL;
+ }
+ pr_debug("%s, mode=%d, rate=%u, rate_type=%d\n",
+ __func__, mode, rate, *rate_type);
+ return ret;
+}
+
+static int voip_get_media_type(uint32_t mode,
+ unsigned int samp_rate)
+{
+ uint32_t media_type;
+
+ pr_debug("%s: mode=%d, samp_rate=%d\n", __func__,
+ mode, samp_rate);
+ switch (mode) {
+ case MODE_AMR:
+ media_type = VSS_MEDIA_ID_AMR_NB_MODEM;
+ break;
+ case MODE_AMR_WB:
+ media_type = VSS_MEDIA_ID_AMR_WB_MODEM;
+ break;
+ case MODE_PCM:
+ if (samp_rate == 8000)
+ media_type = VSS_MEDIA_ID_PCM_NB;
+ else
+ media_type = VSS_MEDIA_ID_PCM_WB;
+ break;
+ case MODE_IS127: /* EVRC-A */
+ media_type = VSS_MEDIA_ID_EVRC_MODEM;
+ break;
+ case MODE_4GV_NB: /* EVRC-B */
+ media_type = VSS_MEDIA_ID_4GV_NB_MODEM;
+ break;
+ case MODE_4GV_WB: /* EVRC-WB */
+ media_type = VSS_MEDIA_ID_4GV_WB_MODEM;
+ break;
+ default:
+ pr_debug(" input mode is not supported\n");
+ media_type = -EINVAL;
+ }
+
+ pr_debug("%s: media_type is 0x%x\n", __func__, media_type);
+
+ return media_type;
+}
+
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ pr_debug("msm_asoc_pcm_new\n");
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+ .probe = msm_pcm_voip_probe,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-voip-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ memset(&voip_info, 0, sizeof(voip_info));
+ voip_info.mode = MODE_PCM;
+ mutex_init(&voip_info.lock);
+ mutex_init(&voip_info.in_lock);
+ mutex_init(&voip_info.out_lock);
+
+ spin_lock_init(&voip_info.dsp_lock);
+
+ init_waitqueue_head(&voip_info.out_wait);
+ init_waitqueue_head(&voip_info.in_wait);
+
+ INIT_LIST_HEAD(&voip_info.in_queue);
+ INIT_LIST_HEAD(&voip_info.free_in_queue);
+ INIT_LIST_HEAD(&voip_info.out_queue);
+ INIT_LIST_HEAD(&voip_info.free_out_queue);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm-pcm.c b/sound/soc/msm/msm-pcm.c
new file mode 100644
index 0000000..ea31985
--- /dev/null
+++ b/sound/soc/msm/msm-pcm.c
@@ -0,0 +1,646 @@
+/* sound/soc/msm/msm-pcm.c
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2009, 2012 Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+
+#include "msm-pcm.h"
+
+#define MAX_DATA_SIZE 496
+#define AUDPP_ALSA_DECODER (-1)
+
+#define DB_TABLE_INDEX (50)
+
+#define audio_send_queue_recbs(prtd, cmd, len) \
+ msm_adsp_write(prtd->audrec, QDSP_uPAudRecBitStreamQueue, cmd, len)
+#define audio_send_queue_rec(prtd, cmd, len) \
+ msm_adsp_write(prtd->audrec, QDSP_uPAudRecCmdQueue, cmd, len)
+
+int intcnt;
+
+struct audio_frame {
+ uint16_t count_low;
+ uint16_t count_high;
+ uint16_t bytes;
+ uint16_t unknown;
+ unsigned char samples[];
+} __attribute__ ((packed));
+
+/* Table contains dB to raw value mapping */
+static const unsigned decoder_db_table[] = {
+
+ 31 , /* -50 dB */
+ 35 , 39 , 44 , 50 , 56 ,
+ 63 , 70 , 79 , 89 , 99 ,
+ 112 , 125 , 141 , 158 , 177 ,
+ 199 , 223 , 251 , 281 , 316 ,
+ 354 , 398 , 446 , 501 , 562 ,
+ 630 , 707 , 794 , 891 , 999 ,
+ 1122 , 1258 , 1412 , 1584 , 1778 ,
+ 1995 , 2238 , 2511 , 2818 , 3162 ,
+ 3548 , 3981 , 4466 , 5011 , 5623 ,
+ 6309 , 7079 , 7943 , 8912 , 10000 ,
+ 11220 , 12589 , 14125 , 15848 , 17782 ,
+ 19952 , 22387 , 25118 , 28183 , 31622 ,
+ 35481 , 39810 , 44668 , 50118 , 56234 ,
+ 63095 , 70794 , 79432 , 89125 , 100000 ,
+ 112201 , 125892 , 141253 , 158489 , 177827 ,
+ 199526 , 223872 , 251188 , 281838 , 316227 ,
+ 354813 , 398107 , 446683 , 501187 , 562341 ,
+ 630957 , 707945 , 794328 , 891250 , 1000000 ,
+ 1122018 , 1258925 , 1412537 , 1584893 , 1778279 ,
+ 1995262 , 2238721 , 2511886 , 2818382 , 3162277 ,
+ 3548133 /* 51 dB */
+
+};
+
+static unsigned compute_db_raw(int db)
+{
+ unsigned reg_val = 0; /* Computed result for correspondent db */
+ /* Check if the given db is out of range */
+ if (db <= MIN_DB)
+ return 0;
+ else if (db > MAX_DB)
+ db = MAX_DB; /* If db is too high then set to max */
+ reg_val = decoder_db_table[DB_TABLE_INDEX+db];
+ return reg_val;
+}
+
+int msm_audio_volume_update(unsigned id,
+ int volume, int pan)
+{
+ unsigned vol_raw;
+
+ vol_raw = compute_db_raw(volume);
+ printk(KERN_INFO "volume: %8x vol_raw: %8x \n", volume, vol_raw);
+ return audpp_set_volume_and_pan(id, vol_raw, pan);
+}
+EXPORT_SYMBOL(msm_audio_volume_update);
+
+void alsa_dsp_event(void *data, unsigned id, uint16_t *msg)
+{
+ struct msm_audio *prtd = data;
+ struct buffer *frame;
+ unsigned long flag;
+
+ switch (id) {
+ case AUDPP_MSG_STATUS_MSG:
+ break;
+ case AUDPP_MSG_SPA_BANDS:
+ break;
+ case AUDPP_MSG_HOST_PCM_INTF_MSG:{
+ unsigned id = msg[2];
+ unsigned idx = msg[3] - 1;
+ if (id != AUDPP_MSG_HOSTPCM_ID_ARM_RX) {
+ printk(KERN_ERR "bogus id\n");
+ break;
+ }
+ if (idx > 1) {
+ printk(KERN_ERR "bogus buffer idx\n");
+ break;
+ }
+ /* Update with actual sent buffer size */
+ if (prtd->out[idx].used != BUF_INVALID_LEN)
+ prtd->pcm_irq_pos += prtd->out[idx].used;
+
+ if (prtd->pcm_irq_pos > prtd->pcm_size)
+ prtd->pcm_irq_pos = prtd->pcm_count;
+
+ if (prtd->ops->playback)
+ prtd->ops->playback(prtd);
+
+ if (prtd->mmap_flag)
+ break;
+
+ spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
+ if (prtd->running) {
+ prtd->out[idx].used = 0;
+ frame = prtd->out + prtd->out_tail;
+ if (frame->used) {
+ alsa_dsp_send_buffer(prtd,
+ prtd->out_tail,
+ frame->used);
+ prtd->out_tail ^= 1;
+ } else {
+ prtd->out_needed++;
+ }
+ wake_up(&the_locks.write_wait);
+ }
+ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ break;
+ }
+ case AUDPP_MSG_PCMDMAMISSED:
+ pr_info("alsa_dsp_event: PCMDMAMISSED %d\n", msg[0]);
+ prtd->eos_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case AUDPP_MSG_CFG_MSG:
+ if (msg[0] == AUDPP_MSG_ENA_ENA) {
+ prtd->out_needed = 0;
+ prtd->running = 1;
+ audio_dsp_out_enable(prtd, 1);
+ } else if (msg[0] == AUDPP_MSG_ENA_DIS) {
+ prtd->running = 0;
+ } else {
+ printk(KERN_ERR "alsa_dsp_event:CFG_MSG=%d\n", msg[0]);
+ }
+ break;
+ case EVENT_MSG_ID:
+ printk(KERN_INFO"alsa_dsp_event: arm9 event\n");
+ break;
+ default:
+ printk(KERN_ERR "alsa_dsp_event: UNKNOWN (%d)\n", id);
+ }
+}
+
+void alsa_audpre_dsp_event(void *data, unsigned id, size_t len,
+ void (*getevent) (void *ptr, size_t len))
+{
+ uint16_t msg[MAX_DATA_SIZE/2];
+
+ if (len > MAX_DATA_SIZE) {
+ printk(KERN_ERR"audpre: event too large(%d bytes)\n", len);
+ return;
+ }
+ getevent(msg, len);
+
+ switch (id) {
+ case AUDPREPROC_MSG_CMD_CFG_DONE_MSG:
+ break;
+ case AUDPREPROC_MSG_ERROR_MSG_ID:
+ printk(KERN_ERR "audpre: err_index %d\n", msg[0]);
+ break;
+ case EVENT_MSG_ID:
+ printk(KERN_INFO"audpre: arm9 event\n");
+ break;
+ default:
+ printk(KERN_ERR "audpre: unknown event %d\n", id);
+ }
+}
+
+void audrec_dsp_event(void *data, unsigned id, size_t len,
+ void (*getevent) (void *ptr, size_t len))
+{
+ struct msm_audio *prtd = data;
+ unsigned long flag;
+ uint16_t msg[MAX_DATA_SIZE/2];
+
+ if (len > MAX_DATA_SIZE) {
+ printk(KERN_ERR"audrec: event/msg too large(%d bytes)\n", len);
+ return;
+ }
+ getevent(msg, len);
+
+ switch (id) {
+ case AUDREC_MSG_CMD_CFG_DONE_MSG:
+ if (msg[0] & AUDREC_MSG_CFG_DONE_TYPE_0_UPDATE) {
+ if (msg[0] & AUDREC_MSG_CFG_DONE_TYPE_0_ENA)
+ audrec_encoder_config(prtd);
+ else
+ prtd->running = 0;
+ }
+ break;
+ case AUDREC_MSG_CMD_AREC_PARAM_CFG_DONE_MSG:{
+ prtd->running = 1;
+ break;
+ }
+ case AUDREC_MSG_FATAL_ERR_MSG:
+ printk(KERN_ERR "audrec: ERROR %x\n", msg[0]);
+ break;
+ case AUDREC_MSG_PACKET_READY_MSG:
+ alsa_get_dsp_frames(prtd);
+ ++intcnt;
+ if (prtd->channel_mode == 1) {
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+
+ if (prtd->ops->capture)
+ prtd->ops->capture(prtd);
+ } else if ((prtd->channel_mode == 0) && (intcnt % 2 == 0)) {
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+ if (prtd->ops->capture)
+ prtd->ops->capture(prtd);
+ }
+ break;
+ case EVENT_MSG_ID:
+ printk(KERN_INFO"audrec: arm9 event\n");
+ break;
+ default:
+ printk(KERN_ERR "audrec: unknown event %d\n", id);
+ }
+}
+
+struct msm_adsp_ops aud_pre_adsp_ops = {
+ .event = alsa_audpre_dsp_event,
+};
+
+struct msm_adsp_ops aud_rec_adsp_ops = {
+ .event = audrec_dsp_event,
+};
+
+int alsa_adsp_configure(struct msm_audio *prtd)
+{
+ int ret, i;
+
+ if (prtd->dir == SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->data = prtd->playback_substream->dma_buffer.area;
+ prtd->phys = prtd->playback_substream->dma_buffer.addr;
+ }
+ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE) {
+ prtd->data = prtd->capture_substream->dma_buffer.area;
+ prtd->phys = prtd->capture_substream->dma_buffer.addr;
+ }
+ if (!prtd->data) {
+ ret = -ENOMEM;
+ goto err1;
+ }
+
+ ret = audmgr_open(&prtd->audmgr);
+ if (ret)
+ goto err2;
+ if (prtd->dir == SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->out_buffer_size = PLAYBACK_DMASZ;
+ prtd->out_sample_rate = 44100;
+ prtd->out_channel_mode = AUDPP_CMD_PCM_INTF_STEREO_V;
+ prtd->out_weight = 100;
+
+ prtd->out[0].data = prtd->data + 0;
+ prtd->out[0].addr = prtd->phys + 0;
+ prtd->out[0].size = BUFSZ;
+ prtd->out[1].data = prtd->data + BUFSZ;
+ prtd->out[1].addr = prtd->phys + BUFSZ;
+ prtd->out[1].size = BUFSZ;
+ }
+ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE) {
+ prtd->samp_rate = RPC_AUD_DEF_SAMPLE_RATE_44100;
+ prtd->samp_rate_index = AUDREC_CMD_SAMP_RATE_INDX_44100;
+ prtd->channel_mode = AUDREC_CMD_STEREO_MODE_STEREO;
+ prtd->buffer_size = STEREO_DATA_SIZE;
+ prtd->type = AUDREC_CMD_TYPE_0_INDEX_WAV;
+ prtd->tx_agc_cfg.cmd_id = AUDPREPROC_CMD_CFG_AGC_PARAMS;
+ prtd->ns_cfg.cmd_id = AUDPREPROC_CMD_CFG_NS_PARAMS;
+ prtd->iir_cfg.cmd_id =
+ AUDPREPROC_CMD_CFG_IIR_TUNING_FILTER_PARAMS;
+
+ ret = msm_adsp_get("AUDPREPROCTASK",
+ &prtd->audpre, &aud_pre_adsp_ops, prtd);
+ if (ret)
+ goto err3;
+ ret = msm_adsp_get("AUDRECTASK",
+ &prtd->audrec, &aud_rec_adsp_ops, prtd);
+ if (ret) {
+ msm_adsp_put(prtd->audpre);
+ goto err3;
+ }
+ prtd->dsp_cnt = 0;
+ prtd->in_head = 0;
+ prtd->in_tail = 0;
+ prtd->in_count = 0;
+ for (i = 0; i < FRAME_NUM; i++) {
+ prtd->in[i].size = 0;
+ prtd->in[i].read = 0;
+ }
+ }
+
+ return 0;
+
+err3:
+ audmgr_close(&prtd->audmgr);
+
+err2:
+ prtd->data = NULL;
+err1:
+ return ret;
+}
+EXPORT_SYMBOL(alsa_adsp_configure);
+
+int alsa_audio_configure(struct msm_audio *prtd)
+{
+ struct audmgr_config cfg;
+ int rc;
+
+ if (prtd->enabled)
+ return 0;
+
+ /* refuse to start if we're not ready with first buffer */
+ if (!prtd->out[0].used)
+ return -EIO;
+
+ cfg.tx_rate = 0;
+ cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_48000;
+ cfg.def_method = RPC_AUD_DEF_METHOD_HOST_PCM;
+ cfg.codec = RPC_AUD_DEF_CODEC_PCM;
+ cfg.snd_method = RPC_SND_METHOD_MIDI;
+ rc = audmgr_enable(&prtd->audmgr, &cfg);
+ if (rc < 0)
+ return rc;
+
+ if (audpp_enable(AUDPP_ALSA_DECODER, alsa_dsp_event, prtd)) {
+ printk(KERN_ERR "audio: audpp_enable() failed\n");
+ audmgr_disable(&prtd->audmgr);
+ return -ENODEV;
+ }
+
+ prtd->enabled = 1;
+ return 0;
+}
+EXPORT_SYMBOL(alsa_audio_configure);
+
+ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf,
+ size_t count, loff_t *pos)
+{
+ unsigned long flag;
+ const char __user *start = buf;
+ struct buffer *frame;
+ size_t xfer;
+ int rc = 0;
+
+ mutex_lock(&the_locks.write_lock);
+ while (count > 0) {
+ frame = prtd->out + prtd->out_head;
+ rc = wait_event_interruptible(the_locks.write_wait,
+ (frame->used == 0)
+ || (prtd->stopped));
+ if (rc < 0)
+ break;
+ if (prtd->stopped) {
+ rc = -EBUSY;
+ break;
+ }
+ xfer = count > frame->size ? frame->size : count;
+ if (copy_from_user(frame->data, buf, xfer)) {
+ rc = -EFAULT;
+ break;
+ }
+ frame->used = xfer;
+ prtd->out_head ^= 1;
+ count -= xfer;
+ buf += xfer;
+
+ spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
+ frame = prtd->out + prtd->out_tail;
+ if (frame->used && prtd->out_needed) {
+ alsa_dsp_send_buffer(prtd, prtd->out_tail,
+ frame->used);
+ prtd->out_tail ^= 1;
+ prtd->out_needed--;
+ }
+ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ }
+ mutex_unlock(&the_locks.write_lock);
+ if (buf > start)
+ return buf - start;
+ return rc;
+}
+EXPORT_SYMBOL(alsa_send_buffer);
+
+int alsa_audio_disable(struct msm_audio *prtd)
+{
+ if (prtd->enabled) {
+ mutex_lock(&the_locks.lock);
+ prtd->enabled = 0;
+ audio_dsp_out_enable(prtd, 0);
+ wake_up(&the_locks.write_wait);
+ audpp_disable(AUDPP_ALSA_DECODER, prtd);
+ audmgr_disable(&prtd->audmgr);
+ prtd->out_needed = 0;
+ mutex_unlock(&the_locks.lock);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(alsa_audio_disable);
+
+int alsa_audrec_disable(struct msm_audio *prtd)
+{
+ if (prtd->enabled) {
+ mutex_lock(&the_locks.lock);
+ prtd->enabled = 0;
+ alsa_rec_dsp_enable(prtd, 0);
+ wake_up(&the_locks.read_wait);
+ msm_adsp_disable(prtd->audpre);
+ msm_adsp_disable(prtd->audrec);
+ audmgr_disable(&prtd->audmgr);
+ prtd->out_needed = 0;
+ prtd->opened = 0;
+ mutex_unlock(&the_locks.lock);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(alsa_audrec_disable);
+
+static int audio_dsp_read_buffer(struct msm_audio *prtd, uint32_t read_cnt)
+{
+ audrec_cmd_packet_ext_ptr cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDREC_CMD_PACKET_EXT_PTR;
+ /* Both WAV and AAC use AUDREC_CMD_TYPE_0 */
+ cmd.type = AUDREC_CMD_TYPE_0;
+ cmd.curr_rec_count_msw = read_cnt >> 16;
+ cmd.curr_rec_count_lsw = read_cnt;
+
+ return audio_send_queue_recbs(prtd, &cmd, sizeof(cmd));
+}
+
+int audrec_encoder_config(struct msm_audio *prtd)
+{
+ audrec_cmd_arec0param_cfg cmd;
+ uint16_t *data = (void *)prtd->data;
+ unsigned n;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDREC_CMD_AREC0PARAM_CFG;
+ cmd.ptr_to_extpkt_buffer_msw = prtd->phys >> 16;
+ cmd.ptr_to_extpkt_buffer_lsw = prtd->phys;
+ cmd.buf_len = FRAME_NUM; /* Both WAV and AAC use 8 frames */
+ cmd.samp_rate_index = prtd->samp_rate_index;
+ /* 0 for mono, 1 for stereo */
+ cmd.stereo_mode = prtd->channel_mode;
+ cmd.rec_quality = 0x1C00;
+
+ /* prepare buffer pointers:
+ * Mono: 1024 samples + 4 halfword header
+ * Stereo: 2048 samples + 4 halfword header
+ */
+
+ for (n = 0; n < FRAME_NUM; n++) {
+ prtd->in[n].data = data + 4;
+ data += (4 + (prtd->channel_mode ? 2048 : 1024));
+ }
+
+ return audio_send_queue_rec(prtd, &cmd, sizeof(cmd));
+}
+
+int audio_dsp_out_enable(struct msm_audio *prtd, int yes)
+{
+ audpp_cmd_pcm_intf cmd;
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDPP_CMD_PCM_INTF_2;
+ cmd.object_num = AUDPP_CMD_PCM_INTF_OBJECT_NUM;
+ cmd.config = AUDPP_CMD_PCM_INTF_CONFIG_CMD_V;
+ cmd.intf_type = AUDPP_CMD_PCM_INTF_RX_ENA_ARMTODSP_V;
+
+ if (yes) {
+ cmd.write_buf1LSW = prtd->out[0].addr;
+ cmd.write_buf1MSW = prtd->out[0].addr >> 16;
+ cmd.write_buf1_len = 0;
+ cmd.write_buf2LSW = prtd->out[1].addr;
+ cmd.write_buf2MSW = prtd->out[1].addr >> 16;
+ cmd.write_buf2_len = prtd->out[1].used;
+ cmd.arm_to_rx_flag = AUDPP_CMD_PCM_INTF_ENA_V;
+ cmd.weight_decoder_to_rx = prtd->out_weight;
+ cmd.weight_arm_to_rx = 1;
+ cmd.partition_number_arm_to_dsp = 0;
+ cmd.sample_rate = prtd->out_sample_rate;
+ cmd.channel_mode = prtd->out_channel_mode;
+ }
+ return audpp_send_queue2(&cmd, sizeof(cmd));
+}
+
+int alsa_buffer_read(struct msm_audio *prtd, void __user *buf,
+ size_t count, loff_t *pos)
+{
+ unsigned long flag;
+ void *data;
+ uint32_t index;
+ uint32_t size;
+ int rc = 0;
+
+ mutex_lock(&the_locks.read_lock);
+ while (count > 0) {
+ rc = wait_event_interruptible(the_locks.read_wait,
+ (prtd->in_count > 0)
+ || prtd->stopped);
+ if (rc < 0)
+ break;
+
+ if (prtd->stopped) {
+ rc = -EBUSY;
+ break;
+ }
+
+ index = prtd->in_tail;
+ data = (uint8_t *) prtd->in[index].data;
+ size = prtd->in[index].size;
+ if (count >= size) {
+ if (copy_to_user(buf, data, size)) {
+ rc = -EFAULT;
+ break;
+ }
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ if (index != prtd->in_tail) {
+ /* overrun: data is invalid, we need to retry */
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock,
+ flag);
+ continue;
+ }
+ prtd->in[index].size = 0;
+ prtd->in_tail = (prtd->in_tail + 1) & (FRAME_NUM - 1);
+ prtd->in_count--;
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+ count -= size;
+ buf += size;
+ } else {
+ break;
+ }
+ }
+ mutex_unlock(&the_locks.read_lock);
+ return rc;
+}
+EXPORT_SYMBOL(alsa_buffer_read);
+
+int alsa_dsp_send_buffer(struct msm_audio *prtd,
+ unsigned idx, unsigned len)
+{
+ audpp_cmd_pcm_intf_send_buffer cmd;
+ cmd.cmd_id = AUDPP_CMD_PCM_INTF_2;
+ cmd.host_pcm_object = AUDPP_CMD_PCM_INTF_OBJECT_NUM;
+ cmd.config = AUDPP_CMD_PCM_INTF_BUFFER_CMD_V;
+ cmd.intf_type = AUDPP_CMD_PCM_INTF_RX_ENA_ARMTODSP_V;
+ cmd.dsp_to_arm_buf_id = 0;
+ cmd.arm_to_dsp_buf_id = idx + 1;
+ cmd.arm_to_dsp_buf_len = len;
+ return audpp_send_queue2(&cmd, sizeof(cmd));
+}
+
+int alsa_rec_dsp_enable(struct msm_audio *prtd, int enable)
+{
+ audrec_cmd_cfg cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDREC_CMD_CFG;
+ cmd.type_0 = enable ? AUDREC_CMD_TYPE_0_ENA : AUDREC_CMD_TYPE_0_DIS;
+ cmd.type_0 |= (AUDREC_CMD_TYPE_0_UPDATE | prtd->type);
+ cmd.type_1 = 0;
+
+ return audio_send_queue_rec(prtd, &cmd, sizeof(cmd));
+}
+EXPORT_SYMBOL(alsa_rec_dsp_enable);
+
+void alsa_get_dsp_frames(struct msm_audio *prtd)
+{
+ struct audio_frame *frame;
+ uint32_t index = 0;
+ unsigned long flag;
+
+ if (prtd->type == AUDREC_CMD_TYPE_0_INDEX_WAV) {
+ index = prtd->in_head;
+
+ frame =
+ (void *)(((char *)prtd->in[index].data) - sizeof(*frame));
+
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ prtd->in[index].size = frame->bytes;
+
+ prtd->in_head = (prtd->in_head + 1) & (FRAME_NUM - 1);
+
+ /* If overflow, move the tail index foward. */
+ if (prtd->in_head == prtd->in_tail)
+ prtd->in_tail = (prtd->in_tail + 1) & (FRAME_NUM - 1);
+ else
+ prtd->in_count++;
+
+ audio_dsp_read_buffer(prtd, prtd->dsp_cnt++);
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+
+ wake_up(&the_locks.read_wait);
+ } else {
+ /* TODO AAC not supported yet. */
+ }
+}
+EXPORT_SYMBOL(alsa_get_dsp_frames);
diff --git a/sound/soc/msm/msm-pcm.h b/sound/soc/msm/msm-pcm.h
new file mode 100644
index 0000000..6e1325b
--- /dev/null
+++ b/sound/soc/msm/msm-pcm.h
@@ -0,0 +1,204 @@
+/* sound/soc/msm/msm-pcm.h
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2009, 2012 Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+
+
+#include <mach/qdsp5/qdsp5audppcmdi.h>
+#include <mach/qdsp5/qdsp5audppmsg.h>
+#include <mach/qdsp5/qdsp5audreccmdi.h>
+#include <mach/qdsp5/qdsp5audrecmsg.h>
+#include <mach/qdsp5/qdsp5audpreproccmdi.h>
+#include <mach/qdsp5/qdsp5audpreprocmsg.h>
+
+#include <../arch/arm/mach-msm/qdsp5/adsp.h>
+#include <../arch/arm/mach-msm/qdsp5/audmgr.h>
+
+
+#define FRAME_NUM (8)
+#define FRAME_SIZE (2052 * 2)
+#define MONO_DATA_SIZE (2048)
+#define STEREO_DATA_SIZE (MONO_DATA_SIZE * 2)
+#define CAPTURE_DMASZ (FRAME_SIZE * FRAME_NUM)
+
+#define BUFSZ (960 * 5)
+#define PLAYBACK_DMASZ (BUFSZ * 2)
+
+#define MSM_PLAYBACK_DEFAULT_VOLUME 0 /* 0dB */
+#define MSM_PLAYBACK_DEFAULT_PAN 0
+
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN 1
+#define USE_CHANNELS_MAX 2
+/* Support unconventional sample rates 12000, 24000 as well */
+#define USE_RATE \
+ (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+#define USE_RATE_MIN 8000
+#define USE_RATE_MAX 48000
+#define MAX_BUFFER_PLAYBACK_SIZE \
+ (4800*4)
+/* 2048 frames (Mono), 1024 frames (Stereo) */
+#define CAPTURE_SIZE 4096
+#define MAX_BUFFER_CAPTURE_SIZE (4096*4)
+#define MAX_PERIOD_SIZE BUFSZ
+#define USE_PERIODS_MAX 1024
+#define USE_PERIODS_MIN 1
+
+
+#define MAX_DB (16)
+#define MIN_DB (-50)
+#define PCMPLAYBACK_DECODERID 5
+
+/* 0xFFFFFFFF Indicates not to be used for audio data copy */
+#define BUF_INVALID_LEN 0xFFFFFFFF
+
+extern int copy_count;
+extern int intcnt;
+
+struct msm_volume {
+ bool update;
+ int volume; /* Volume parameter, in dB Scale */
+ int pan;
+};
+
+struct buffer {
+ void *data;
+ unsigned size;
+ unsigned used;
+ unsigned addr;
+};
+
+struct buffer_rec {
+ void *data;
+ unsigned int size;
+ unsigned int read;
+ unsigned int addr;
+};
+
+struct audio_locks {
+ struct mutex lock;
+ struct mutex write_lock;
+ struct mutex read_lock;
+ spinlock_t read_dsp_lock;
+ spinlock_t write_dsp_lock;
+ spinlock_t mixer_lock;
+ wait_queue_head_t read_wait;
+ wait_queue_head_t write_wait;
+ wait_queue_head_t eos_wait;
+};
+
+extern struct audio_locks the_locks;
+
+struct msm_audio_event_callbacks {
+ /* event is called from interrupt context when a message
+ * arrives from the DSP.
+ */
+ void (*playback)(void *);
+ void (*capture)(void *);
+};
+
+
+struct msm_audio {
+ struct buffer out[2];
+ struct buffer_rec in[8];
+
+ uint8_t out_head;
+ uint8_t out_tail;
+ uint8_t out_needed; /* number of buffers the dsp is waiting for */
+ atomic_t out_bytes;
+
+ /* configuration to use on next enable */
+ uint32_t out_sample_rate;
+ uint32_t out_channel_mode;
+ uint32_t out_weight;
+ uint32_t out_buffer_size;
+
+ struct audmgr audmgr;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ /* data allocated for various buffers */
+ char *data;
+ dma_addr_t phys;
+
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int pcm_buf_pos; /* position in buffer */
+
+ struct msm_adsp_module *audpre;
+ struct msm_adsp_module *audrec;
+
+ /* configuration to use on next enable */
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint32_t buffer_size; /* 2048 for mono, 4096 for stereo */
+ uint32_t type; /* 0 for PCM ,1 for AAC */
+ uint32_t dsp_cnt;
+ uint32_t in_head; /* next buffer dsp will write */
+ uint32_t in_tail; /* next buffer read() will read */
+ uint32_t in_count; /* number of buffers available to read() */
+
+ unsigned short samp_rate_index;
+
+ /* audpre settings */
+ audpreproc_cmd_cfg_agc_params tx_agc_cfg;
+ audpreproc_cmd_cfg_ns_params ns_cfg;
+ /* For different sample rate, the coeff might be different. *
+ * All the coeff should be passed from user space */
+ audpreproc_cmd_cfg_iir_tuning_filter_params iir_cfg;
+
+ struct msm_audio_event_callbacks *ops;
+
+ int dir;
+ int opened;
+ int enabled;
+ int running;
+ int stopped; /* set when stopped, cleared on flush */
+ int eos_ack;
+ int mmap_flag;
+ int period;
+};
+
+
+
+/* platform data */
+extern int alsa_dsp_send_buffer(struct msm_audio *prtd,
+ unsigned idx, unsigned len);
+extern int audio_dsp_out_enable(struct msm_audio *prtd, int yes);
+extern struct snd_soc_platform_driver msm_soc_platform;
+
+int audrec_encoder_config(struct msm_audio *prtd);
+extern void alsa_get_dsp_frames(struct msm_audio *prtd);
+extern int alsa_rec_dsp_enable(struct msm_audio *prtd, int enable);
+extern int alsa_audrec_disable(struct msm_audio *prtd);
+extern int alsa_audio_configure(struct msm_audio *prtd);
+extern int alsa_audio_disable(struct msm_audio *prtd);
+extern int alsa_adsp_configure(struct msm_audio *prtd);
+extern int alsa_buffer_read(struct msm_audio *prtd, void __user *buf,
+ size_t count, loff_t *pos);
+ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf,
+ size_t count, loff_t *pos);
+int msm_audio_volume_update(unsigned id,
+ int volume, int pan);
+extern struct audio_locks the_locks;
+extern struct msm_volume msm_vol_ctl;
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/msm-voip.c b/sound/soc/msm/msm-voip.c
new file mode 100644
index 0000000..082c840
--- /dev/null
+++ b/sound/soc/msm/msm-voip.c
@@ -0,0 +1,610 @@
+/* Copyright (c) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * All source code in this file is licensed under the following license except
+ * where indicated.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/mutex.h>
+#include <linux/uaccess.h>
+#include <linux/wakelock.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6asm.h>
+#include <sound/apr_audio.h>
+#include <mach/msm_rpcrouter.h>
+#include <mach/qdsp6v2/q6voice.h>
+#include <mach/qdsp6v2/audio_dev_ctl.h>
+#include "msm_audio_mvs.h"
+
+
+static struct audio_voip_info_type audio_voip_info;
+static void audio_mvs_process_ul_pkt(uint8_t *voc_pkt,
+ uint32_t pkt_len,
+ void *private_data);
+static void audio_mvs_process_dl_pkt(uint8_t *voc_pkt,
+ uint32_t *pkt_len,
+ void *private_data);
+
+struct msm_audio_mvs_frame {
+ uint32_t frame_type;
+ uint32_t len;
+ uint8_t voc_pkt[MVS_MAX_VOC_PKT_SIZE];
+};
+
+struct audio_mvs_buf_node {
+ struct list_head list;
+ struct msm_audio_mvs_frame frame;
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000),
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN,
+ .period_bytes_min = MVS_MAX_VOC_PKT_SIZE,
+ .period_bytes_max = MVS_MAX_VOC_PKT_SIZE,
+ .periods_min = VOIP_MAX_Q_LEN,
+ .periods_max = VOIP_MAX_Q_LEN,
+ .fifo_size = 0,
+};
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+
+ struct audio_voip_info_type *audio = &audio_voip_info;
+ pr_debug("%s\n", __func__);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio->playback_start = 1;
+ else
+ audio->capture_start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio->playback_start = 0;
+ else
+ audio->capture_start = 0;
+ break;
+ default:
+ break;
+ }
+ return 0;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct audio_voip_info_type *audio = &audio_voip_info;
+ struct audio_mvs_release_msg release_msg;
+
+ pr_debug("%s\n", __func__);
+ memset(&release_msg, 0, sizeof(release_msg));
+ mutex_lock(&audio->lock);
+
+ audio->instance--;
+ wake_up(&audio->out_wait);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ audio->playback_state = AUDIO_MVS_CLOSED;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ audio->capture_state = AUDIO_MVS_CLOSED;
+ if (!audio->instance) {
+ /* Release MVS. */
+ release_msg.client_id = cpu_to_be32(MVS_CLIENT_ID_VOIP);
+ /* Derigstering the callbacks with voice driver */
+ voice_register_mvs_cb(NULL, NULL, audio);
+ } else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ voice_register_mvs_cb(audio_mvs_process_ul_pkt,
+ NULL, audio);
+ } else {
+ voice_register_mvs_cb(NULL, audio_mvs_process_dl_pkt,
+ audio);
+ }
+
+ mutex_unlock(&audio->lock);
+
+ wake_unlock(&audio->suspend_lock);
+ wake_unlock(&audio->idle_lock);
+ /* Release the IO buffers. */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ audio->in_write = 0;
+ audio->in_read = 0;
+ memset(audio->in[0].voc_pkt, 0,
+ MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN);
+ audio->playback_substream = NULL;
+ } else {
+ audio->out_write = 0;
+ audio->out_read = 0;
+ memset(audio->out[0].voc_pkt, 0,
+ MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN);
+ audio->capture_substream = NULL;
+ }
+ return rc;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_voip_info_type *audio = &audio_voip_info;
+
+ pr_debug("%s\n", __func__);
+ mutex_lock(&audio->lock);
+
+ if (audio->playback_substream == NULL ||
+ audio->capture_substream == NULL) {
+ if (substream->stream ==
+ SNDRV_PCM_STREAM_PLAYBACK) {
+ audio->playback_substream = substream;
+ runtime->hw = msm_pcm_hardware;
+ audio_voip_info.in_read = 0;
+ audio_voip_info.in_write = 0;
+ if (audio->playback_state < AUDIO_MVS_OPENED)
+ audio->playback_state = AUDIO_MVS_OPENED;
+ } else if (substream->stream ==
+ SNDRV_PCM_STREAM_CAPTURE) {
+ audio->capture_substream = substream;
+ runtime->hw = msm_pcm_hardware;
+ audio_voip_info.out_read = 0;
+ audio_voip_info.out_write = 0;
+ if (audio->capture_state < AUDIO_MVS_OPENED)
+ audio->capture_state = AUDIO_MVS_OPENED;
+ }
+ } else {
+ ret = -EPERM;
+ goto err;
+ }
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0) {
+ pr_debug("%s:snd_pcm_hw_constraint_integer failed\n", __func__);
+ goto err;
+ }
+ audio->instance++;
+
+err:
+ mutex_unlock(&audio->lock);
+ return ret;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int rc = 0;
+ int count = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_voip_info_type *audio = &audio_voip_info;
+ uint32_t index;
+ pr_debug("%s\n", __func__);
+
+ rc = wait_event_timeout(audio->in_wait,
+ (audio->in_write - audio->in_read <= VOIP_MAX_Q_LEN-1),
+ 1 * HZ);
+ if (rc < 0) {
+ pr_debug("%s: write was interrupted\n", __func__);
+ return -ERESTARTSYS;
+ }
+
+ if (audio->playback_state == AUDIO_MVS_ENABLED) {
+ index = audio->in_write % VOIP_MAX_Q_LEN;
+ count = frames_to_bytes(runtime, frames);
+ if (count == MVS_MAX_VOC_PKT_SIZE) {
+ pr_debug("%s:write index = %d\n", __func__, index);
+ rc = copy_from_user(audio->in[index].voc_pkt, buf,
+ count);
+ if (!rc) {
+ audio->in[index].len = count;
+ audio->in_write++;
+ } else {
+ pr_debug("%s:Copy from user returned %d\n",
+ __func__, rc);
+ rc = -EFAULT;
+ }
+ } else
+ rc = -ENOMEM;
+
+ } else {
+ pr_debug("%s:Write performed in invalid state %d\n",
+ __func__, audio->playback_state);
+ rc = -EINVAL;
+ }
+ return rc;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff,
+ void __user *buf, snd_pcm_uframes_t frames)
+{
+ int rc = 0;
+ int count = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_voip_info_type *audio = &audio_voip_info;
+ uint32_t index = 0;
+
+ pr_debug("%s\n", __func__);
+
+ /* Ensure the driver has been enabled. */
+ if (audio->capture_state != AUDIO_MVS_ENABLED) {
+ pr_debug("%s:Read performed in invalid state %d\n",
+ __func__, audio->capture_state);
+ return -EPERM;
+ }
+ rc = wait_event_timeout(audio->out_wait,
+ ((audio->out_read < audio->out_write) ||
+ (audio->capture_state == AUDIO_MVS_CLOSING) ||
+ (audio->capture_state == AUDIO_MVS_CLOSED)),
+ 1 * HZ);
+
+ if (rc < 0) {
+ pr_debug("%s: Read was interrupted\n", __func__);
+ return -ERESTARTSYS;
+ }
+
+ if (audio->capture_state == AUDIO_MVS_CLOSING
+ || audio->capture_state == AUDIO_MVS_CLOSED) {
+ pr_debug("%s:EBUSY STATE\n", __func__);
+ rc = -EBUSY;
+ } else {
+ count = frames_to_bytes(runtime, frames);
+ index = audio->out_read % VOIP_MAX_Q_LEN;
+ pr_debug("%s:index=%d\n", __func__, index);
+ if (audio->out[index].len <= count) {
+ rc = copy_to_user(buf,
+ audio->out[index].voc_pkt,
+ audio->out[index].len);
+ if (rc) {
+ pr_debug("%s:Copy to user %d\n",
+ __func__, rc);
+ rc = -EFAULT;
+ } else
+ audio->out_read++;
+ } else {
+ pr_debug("%s:returning ENOMEM\n", __func__);
+ rc = -ENOMEM;
+ }
+ }
+ return rc;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ pr_debug("%s\n", __func__);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+/* Capture path */
+static void audio_mvs_process_ul_pkt(uint8_t *voc_pkt,
+ uint32_t pkt_len,
+ void *private_data)
+{
+ struct audio_voip_info_type *audio = private_data;
+ uint32_t index;
+ static int i;
+ pr_debug("%s\n", __func__);
+
+ if (audio->capture_substream == NULL)
+ return;
+ index = audio->out_write % VOIP_MAX_Q_LEN;
+ memcpy(audio->out[index].voc_pkt, voc_pkt, pkt_len);
+ audio->out[index].len = pkt_len;
+ audio->out_write++;
+ wake_up(&audio->out_wait);
+ i++;
+ if (audio->capture_start) {
+ audio->pcm_capture_irq_pos += audio->pcm_count;
+ if (!(i % 2))
+ snd_pcm_period_elapsed(audio->capture_substream);
+ }
+}
+
+/* Playback path */
+static void audio_mvs_process_dl_pkt(uint8_t *voc_pkt,
+ uint32_t *pkt_len,
+ void *private_data)
+{
+ struct audio_voip_info_type *audio = private_data;
+ uint32_t index;
+ static int i;
+ pr_debug("%s\n", __func__);
+
+ if (audio->playback_substream == NULL)
+ return;
+ if ((audio->in_write - audio->in_read >= 0)
+ && (audio->playback_start)) {
+ index = audio->in_read % VOIP_MAX_Q_LEN;
+ *pkt_len = audio->pcm_count;
+ memcpy(voc_pkt, audio->in[index].voc_pkt, *pkt_len);
+ audio->in_read++;
+ wake_up(&audio->in_wait);
+ i++;
+ audio->pcm_playback_irq_pos += audio->pcm_count;
+ if (!(i%2))
+ snd_pcm_period_elapsed(audio->playback_substream);
+ pr_debug("%s:read_index=%d\n", __func__, index);
+ }
+}
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct audio_voip_info_type *prtd = &audio_voip_info;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_playback_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ pr_debug("%s:prtd->pcm_playback_size:%d\n",
+ __func__, prtd->pcm_playback_size);
+ pr_debug("%s:prtd->pcm_count:%d\n", __func__, prtd->pcm_count);
+
+ mutex_lock(&prtd->prepare_lock);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (prtd->playback_state == AUDIO_MVS_ENABLED)
+ goto enabled;
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (prtd->capture_state == AUDIO_MVS_ENABLED)
+ goto enabled;
+ }
+
+ pr_debug("%s:Register cbs with voice driver check audio_mvs_driver\n",
+ __func__);
+ if (prtd->instance == 2) {
+ voice_register_mvs_cb(audio_mvs_process_ul_pkt,
+ audio_mvs_process_dl_pkt,
+ prtd);
+ } else {
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ voice_register_mvs_cb(NULL,
+ audio_mvs_process_dl_pkt,
+ prtd);
+ } else {
+ voice_register_mvs_cb(audio_mvs_process_ul_pkt,
+ NULL,
+ prtd);
+ }
+ }
+
+enabled:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->playback_state = AUDIO_MVS_ENABLED;
+ prtd->pcm_playback_irq_pos = 0;
+ prtd->pcm_playback_buf_pos = 0;
+ /* rate and channels are sent to audio driver */
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ prtd->capture_state = AUDIO_MVS_ENABLED;
+ prtd->pcm_capture_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_capture_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_capture_irq_pos = 0;
+ prtd->pcm_capture_buf_pos = 0;
+ }
+ mutex_unlock(&prtd->prepare_lock);
+ return rc;
+}
+
+static snd_pcm_uframes_t
+msm_pcm_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_voip_info_type *audio = &audio_voip_info;
+
+ if (audio->pcm_playback_irq_pos >= audio->pcm_playback_size)
+ audio->pcm_playback_irq_pos = 0;
+ return bytes_to_frames(runtime, (audio->pcm_playback_irq_pos));
+}
+
+static snd_pcm_uframes_t
+msm_pcm_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_voip_info_type *audio = &audio_voip_info;
+
+ if (audio->pcm_capture_irq_pos >= audio->pcm_capture_size)
+ audio->pcm_capture_irq_pos = 0;
+ return bytes_to_frames(runtime, (audio->pcm_capture_irq_pos));
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t ret = 0;
+ pr_debug("%s\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_pointer(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_pointer(substream);
+ return ret;
+}
+
+static struct snd_pcm_ops msm_mvs_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+
+};
+
+static int msm_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ int i, ret, offset = 0;
+ struct snd_pcm_substream *substream = NULL;
+ struct snd_dma_buffer *dma_buffer = NULL;
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, 1);
+ if (ret)
+ return ret;
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+ if (ret)
+ return ret;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &msm_mvs_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &msm_mvs_pcm_ops);
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (!substream)
+ return -ENOMEM;
+
+ dma_buffer = &substream->dma_buffer;
+ dma_buffer->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buffer->dev.dev = card->dev;
+ dma_buffer->private_data = NULL;
+ dma_buffer->area = dma_alloc_coherent(card->dev,
+ (MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN),
+ &dma_buffer->addr, GFP_KERNEL);
+ if (!dma_buffer->area) {
+ pr_err("%s:MSM VOIP dma_alloc failed\n", __func__);
+ return -ENOMEM;
+ }
+ dma_buffer->bytes = MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN;
+ memset(dma_buffer->area, 0, MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN);
+ audio_voip_info.in_read = 0;
+ audio_voip_info.in_write = 0;
+ audio_voip_info.out_read = 0;
+ audio_voip_info.out_write = 0;
+ for (i = 0; i < VOIP_MAX_Q_LEN; i++) {
+ audio_voip_info.in[i].voc_pkt =
+ dma_buffer->area + offset;
+ offset = offset + MVS_MAX_VOC_PKT_SIZE;
+ }
+ substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (!substream)
+ return -ENOMEM;
+
+ dma_buffer = &substream->dma_buffer;
+ dma_buffer->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buffer->dev.dev = card->dev;
+ dma_buffer->private_data = NULL;
+ dma_buffer->area = dma_alloc_coherent(card->dev,
+ (MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN),
+ &dma_buffer->addr, GFP_KERNEL);
+ if (!dma_buffer->area) {
+ pr_err("%s:MSM VOIP dma_alloc failed\n", __func__);
+ return -ENOMEM;
+ }
+ memset(dma_buffer->area, 0, MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN);
+ dma_buffer->bytes = MVS_MAX_VOC_PKT_SIZE * VOIP_MAX_Q_LEN;
+ for (i = 0; i < VOIP_MAX_Q_LEN; i++) {
+ audio_voip_info.out[i].voc_pkt =
+ dma_buffer->area + offset;
+ offset = offset + MVS_MAX_VOC_PKT_SIZE;
+ }
+ audio_voip_info.playback_substream = NULL;
+ audio_voip_info.capture_substream = NULL;
+
+ return 0;
+}
+
+static void msm_pcm_free_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+struct snd_soc_platform_driver msm_mvs_soc_platform = {
+ .ops = &msm_mvs_pcm_ops,
+ .pcm_new = msm_pcm_new,
+ .pcm_free = msm_pcm_free_buffers,
+};
+EXPORT_SYMBOL(msm_mvs_soc_platform);
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_mvs_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-mvs-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_mvs_soc_platform_init(void)
+{
+ memset(&audio_voip_info, 0, sizeof(audio_voip_info));
+ mutex_init(&audio_voip_info.lock);
+ mutex_init(&audio_voip_info.prepare_lock);
+ init_waitqueue_head(&audio_voip_info.out_wait);
+ init_waitqueue_head(&audio_voip_info.in_wait);
+ wake_lock_init(&audio_voip_info.suspend_lock, WAKE_LOCK_SUSPEND,
+ "audio_mvs_suspend");
+ wake_lock_init(&audio_voip_info.idle_lock, WAKE_LOCK_IDLE,
+ "audio_mvs_idle");
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_mvs_soc_platform_init);
+
+static void __exit msm_mvs_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_mvs_soc_platform_exit);
+
+MODULE_DESCRIPTION("MVS PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm7201.c b/sound/soc/msm/msm7201.c
new file mode 100644
index 0000000..2a73fd6
--- /dev/null
+++ b/sound/soc/msm/msm7201.c
@@ -0,0 +1,424 @@
+/* linux/sound/soc/msm/msm7201.c
+ *
+ * Copyright (c) 2008-2009, 2011, 2012 Code Aurora Forum. All rights reserved.
+ *
+ * All source code in this file is licensed under the following license except
+ * where indicated.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+
+#include "msm-pcm.h"
+#include <asm/mach-types.h>
+#include <mach/msm_rpcrouter.h>
+
+static struct msm_rpc_endpoint *snd_ep;
+static uint32_t snd_mute_ear_mute;
+static uint32_t snd_mute_mic_mute;
+
+struct msm_snd_rpc_ids {
+ unsigned long prog;
+ unsigned long vers;
+ unsigned long vers2;
+ unsigned long rpc_set_snd_device;
+ unsigned long rpc_set_device_vol;
+ int device;
+};
+
+static struct msm_snd_rpc_ids snd_rpc_ids;
+
+static struct platform_device *msm_audio_snd_device;
+
+static int snd_msm_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1; /* Volume Param, in dB */
+ uinfo->value.integer.min = MIN_DB;
+ uinfo->value.integer.max = MAX_DB;
+ return 0;
+}
+
+static int snd_msm_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ spin_lock_irq(&the_locks.mixer_lock);
+ ucontrol->value.integer.value[0] = msm_vol_ctl.volume;
+ spin_unlock_irq(&the_locks.mixer_lock);
+ return 0;
+}
+
+static int snd_msm_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int change;
+ int volume;
+
+ volume = ucontrol->value.integer.value[0];
+ spin_lock_irq(&the_locks.mixer_lock);
+ change = (msm_vol_ctl.volume != volume);
+ if (change) {
+ msm_vol_ctl.volume = volume;
+ msm_audio_volume_update(PCMPLAYBACK_DECODERID,
+ msm_vol_ctl.volume, msm_vol_ctl.pan);
+ }
+ spin_unlock_irq(&the_locks.mixer_lock);
+ return 0;
+}
+
+static int snd_msm_device_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* Device */
+
+ /*
+ * The number of devices supported is 26 (0 to 25)
+ */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 36;
+ return 0;
+}
+
+static int snd_msm_device_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = (uint32_t)snd_rpc_ids.device;
+ ucontrol->value.integer.value[1] = snd_mute_ear_mute;
+ ucontrol->value.integer.value[2] = snd_mute_mic_mute;
+ return 0;
+}
+
+int msm_snd_init_rpc_ids(void)
+{
+ snd_rpc_ids.prog = 0x30000002;
+ snd_rpc_ids.vers = 0x00020001;
+ snd_rpc_ids.vers2 = 0x00030001;
+ /*
+ * The magic number 2 corresponds to the rpc call
+ * index for snd_set_device
+ */
+ snd_rpc_ids.rpc_set_snd_device = 2;
+ snd_rpc_ids.rpc_set_device_vol = 3;
+ return 0;
+}
+
+int msm_snd_rpc_connect(void)
+{
+ if (snd_ep) {
+ printk(KERN_INFO "%s: snd_ep already connected\n", __func__);
+ return 0;
+ }
+
+ /* Initialize rpc ids */
+ if (msm_snd_init_rpc_ids()) {
+ printk(KERN_ERR "%s: snd rpc ids initialization failed\n"
+ , __func__);
+ return -ENODATA;
+ }
+
+ snd_ep = msm_rpc_connect_compatible(snd_rpc_ids.prog,
+ snd_rpc_ids.vers, 0);
+ if (IS_ERR(snd_ep)) {
+ printk(KERN_DEBUG "%s failed (compatible VERS = %ld) \
+ trying again with another API\n",
+ __func__, snd_rpc_ids.vers);
+ snd_ep =
+ msm_rpc_connect_compatible(snd_rpc_ids.prog,
+ snd_rpc_ids.vers2, 0);
+ }
+ if (IS_ERR(snd_ep)) {
+ printk(KERN_ERR "%s: failed (compatible VERS = %ld)\n",
+ __func__, snd_rpc_ids.vers2);
+ snd_ep = NULL;
+ return -EAGAIN;
+ }
+ return 0;
+}
+
+int msm_snd_rpc_close(void)
+{
+ int rc = 0;
+
+ if (IS_ERR(snd_ep)) {
+ printk(KERN_ERR "%s: snd handle unavailable, rc = %ld\n",
+ __func__, PTR_ERR(snd_ep));
+ return -EAGAIN;
+ }
+
+ rc = msm_rpc_close(snd_ep);
+ snd_ep = NULL;
+
+ if (rc < 0) {
+ printk(KERN_ERR "%s: close rpc failed! rc = %d\n",
+ __func__, rc);
+ return -EAGAIN;
+ } else
+ printk(KERN_INFO "rpc close success\n");
+
+ return rc;
+}
+
+static int snd_msm_device_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct snd_start_req {
+ struct rpc_request_hdr hdr;
+ uint32_t rpc_snd_device;
+ uint32_t snd_mute_ear_mute;
+ uint32_t snd_mute_mic_mute;
+ uint32_t callback_ptr;
+ uint32_t client_data;
+ } req;
+
+ snd_rpc_ids.device = (int)ucontrol->value.integer.value[0];
+
+ if (ucontrol->value.integer.value[1] > 1)
+ ucontrol->value.integer.value[1] = 1;
+ if (ucontrol->value.integer.value[2] > 1)
+ ucontrol->value.integer.value[2] = 1;
+
+ req.hdr.type = 0;
+ req.hdr.rpc_vers = 2;
+
+ req.rpc_snd_device = cpu_to_be32(snd_rpc_ids.device);
+ req.snd_mute_ear_mute =
+ cpu_to_be32((int)ucontrol->value.integer.value[1]);
+ req.snd_mute_mic_mute =
+ cpu_to_be32((int)ucontrol->value.integer.value[2]);
+ req.callback_ptr = -1;
+ req.client_data = cpu_to_be32(0);
+
+ req.hdr.prog = snd_rpc_ids.prog;
+ req.hdr.vers = snd_rpc_ids.vers;
+
+ rc = msm_rpc_call(snd_ep, snd_rpc_ids.rpc_set_snd_device ,
+ &req, sizeof(req), 5 * HZ);
+
+ if (rc < 0) {
+ printk(KERN_ERR "%s: snd rpc call failed! rc = %d\n",
+ __func__, rc);
+ } else {
+ printk(KERN_INFO "snd device connected\n");
+ snd_mute_ear_mute = ucontrol->value.integer.value[1];
+ snd_mute_mic_mute = ucontrol->value.integer.value[2];
+ printk(KERN_ERR "%s: snd_mute_ear_mute =%d, snd_mute_mic_mute = %d\n",
+ __func__, snd_mute_ear_mute, snd_mute_mic_mute);
+ }
+
+ return rc;
+}
+
+static int snd_msm_device_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2; /* Device/Volume */
+
+ /*
+ * The number of devices supported is 37 (0 to 36)
+ */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 36;
+ return 0;
+}
+
+static int snd_msm_device_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct snd_vol_req {
+ struct rpc_request_hdr hdr;
+ uint32_t device;
+ uint32_t method;
+ uint32_t volume;
+ uint32_t cb_func;
+ uint32_t client_data;
+ } req;
+
+ snd_rpc_ids.device = (int)ucontrol->value.integer.value[0];
+
+ if ((ucontrol->value.integer.value[1] < 0) ||
+ (ucontrol->value.integer.value[1] > 6)) {
+ pr_err("Device volume should be in range of 1 to 6\n");
+ return -EINVAL;
+ }
+ if ((ucontrol->value.integer.value[0] > 36) ||
+ (ucontrol->value.integer.value[0] < 0)) {
+ pr_err("Device range supported is 0 to 36\n");
+ return -EINVAL;
+ }
+
+ req.device = cpu_to_be32((int)ucontrol->value.integer.value[0]);
+ req.method = cpu_to_be32(0);
+ req.volume = cpu_to_be32((int)ucontrol->value.integer.value[1]);
+ req.cb_func = -1;
+ req.client_data = cpu_to_be32(0);
+
+ rc = msm_rpc_call(snd_ep, snd_rpc_ids.rpc_set_device_vol ,
+ &req, sizeof(req), 5 * HZ);
+
+ if (rc < 0) {
+ printk(KERN_ERR "%s: snd rpc call failed! rc = %d\n",
+ __func__, rc);
+ } else {
+ printk(KERN_ERR "%s: device [%d] volume set to [%d]\n",
+ __func__, (int)ucontrol->value.integer.value[0],
+ (int)ucontrol->value.integer.value[1]);
+ }
+
+ return rc;
+}
+
+/* Supported range -50dB to 18dB */
+static const DECLARE_TLV_DB_LINEAR(db_scale_linear, -5000, 1800);
+
+#define MSM_EXT(xname, xindex, fp_info, fp_get, fp_put, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .name = xname, .index = xindex, \
+ .info = fp_info,\
+ .get = fp_get, .put = fp_put, \
+ .private_value = addr, \
+}
+
+#define MSM_EXT_TLV(xname, xindex, fp_info, fp_get, fp_put, addr, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = (SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE), \
+ .name = xname, .index = xindex, \
+ .info = fp_info,\
+ .get = fp_get, .put = fp_put, .tlv.p = tlv_array, \
+ .private_value = addr, \
+}
+
+static struct snd_kcontrol_new snd_msm_controls[] = {
+ MSM_EXT_TLV("PCM Playback Volume", 0, snd_msm_volume_info, \
+ snd_msm_volume_get, snd_msm_volume_put, 0, db_scale_linear),
+ MSM_EXT("device", 0, snd_msm_device_info, snd_msm_device_get, \
+ snd_msm_device_put, 0),
+ MSM_EXT("Device Volume", 0, snd_msm_device_vol_info, NULL, \
+ snd_msm_device_vol_put, 0),
+};
+
+static int msm_new_mixer(struct snd_soc_codec *codec)
+{
+ unsigned int idx;
+ int err;
+
+ pr_err("msm_soc: ALSA MSM Mixer Setting\n");
+ strcpy(codec->card->snd_card->mixername, "MSM Mixer");
+ for (idx = 0; idx < ARRAY_SIZE(snd_msm_controls); idx++) {
+ err = snd_ctl_add(codec->card->snd_card,
+ snd_ctl_new1(&snd_msm_controls[idx], NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int msm_soc_dai_init(
+ struct snd_soc_pcm_runtime *rtd)
+{
+ int ret = 0;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ mutex_init(&the_locks.lock);
+ mutex_init(&the_locks.write_lock);
+ mutex_init(&the_locks.read_lock);
+ spin_lock_init(&the_locks.read_dsp_lock);
+ spin_lock_init(&the_locks.write_dsp_lock);
+ spin_lock_init(&the_locks.mixer_lock);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+ msm_vol_ctl.volume = MSM_PLAYBACK_DEFAULT_VOLUME;
+ msm_vol_ctl.pan = MSM_PLAYBACK_DEFAULT_PAN;
+
+ ret = msm_new_mixer(codec);
+ if (ret < 0) {
+ pr_err("msm_soc: ALSA MSM Mixer Fail\n");
+ }
+
+ return ret;
+}
+
+static struct snd_soc_dai_link msm_dai[] = {
+{
+ .name = "MSM Primary I2S",
+ .stream_name = "DSP 1",
+ .cpu_dai_name = "msm-cpu-dai.0",
+ .platform_name = "msm-dsp-audio.0",
+ .codec_name = "msm-codec-dai.0",
+ .codec_dai_name = "msm-codec-dai",
+ .init = &msm_soc_dai_init,
+},
+};
+
+static struct snd_soc_card snd_soc_card_msm = {
+ .name = "msm-audio",
+ .dai_link = msm_dai,
+ .num_links = ARRAY_SIZE(msm_dai),
+};
+
+static int __init msm_audio_init(void)
+{
+ int ret;
+
+ msm_audio_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!msm_audio_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(msm_audio_snd_device, &snd_soc_card_msm);
+ ret = platform_device_add(msm_audio_snd_device);
+ if (ret) {
+ platform_device_put(msm_audio_snd_device);
+ return ret;
+ }
+
+ ret = msm_snd_rpc_connect();
+ snd_mute_ear_mute = 0;
+ snd_mute_mic_mute = 0;
+
+ return ret;
+}
+
+static void __exit msm_audio_exit(void)
+{
+ msm_snd_rpc_close();
+ platform_device_unregister(msm_audio_snd_device);
+}
+
+module_init(msm_audio_init);
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("PCM module");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c
new file mode 100644
index 0000000..1f23a92
--- /dev/null
+++ b/sound/soc/msm/msm7k-pcm.c
@@ -0,0 +1,699 @@
+/* linux/sound/soc/msm/msm7k-pcm.c
+ *
+ * Copyright (c) 2008-2009, 2012 Code Aurora Forum. All rights reserved.
+ *
+ * All source code in this file is licensed under the following license except
+ * where indicated.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+
+#include "msm-pcm.h"
+
+#define SND_DRIVER "snd_msm"
+#define MAX_PCM_DEVICES SNDRV_CARDS
+#define MAX_PCM_SUBSTREAMS 1
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+int copy_count;
+
+struct audio_locks the_locks;
+EXPORT_SYMBOL(the_locks);
+struct msm_volume msm_vol_ctl;
+EXPORT_SYMBOL(msm_vol_ctl);
+
+
+static unsigned convert_dsp_samp_index(unsigned index)
+{
+ switch (index) {
+ case 48000:
+ return AUDREC_CMD_SAMP_RATE_INDX_48000;
+ case 44100:
+ return AUDREC_CMD_SAMP_RATE_INDX_44100;
+ case 32000:
+ return AUDREC_CMD_SAMP_RATE_INDX_32000;
+ case 24000:
+ return AUDREC_CMD_SAMP_RATE_INDX_24000;
+ case 22050:
+ return AUDREC_CMD_SAMP_RATE_INDX_22050;
+ case 16000:
+ return AUDREC_CMD_SAMP_RATE_INDX_16000;
+ case 12000:
+ return AUDREC_CMD_SAMP_RATE_INDX_12000;
+ case 11025:
+ return AUDREC_CMD_SAMP_RATE_INDX_11025;
+ case 8000:
+ return AUDREC_CMD_SAMP_RATE_INDX_8000;
+ default:
+ return AUDREC_CMD_SAMP_RATE_INDX_44100;
+ }
+}
+
+static unsigned convert_samp_rate(unsigned hz)
+{
+ switch (hz) {
+ case 48000:
+ return RPC_AUD_DEF_SAMPLE_RATE_48000;
+ case 44100:
+ return RPC_AUD_DEF_SAMPLE_RATE_44100;
+ case 32000:
+ return RPC_AUD_DEF_SAMPLE_RATE_32000;
+ case 24000:
+ return RPC_AUD_DEF_SAMPLE_RATE_24000;
+ case 22050:
+ return RPC_AUD_DEF_SAMPLE_RATE_22050;
+ case 16000:
+ return RPC_AUD_DEF_SAMPLE_RATE_16000;
+ case 12000:
+ return RPC_AUD_DEF_SAMPLE_RATE_12000;
+ case 11025:
+ return RPC_AUD_DEF_SAMPLE_RATE_11025;
+ case 8000:
+ return RPC_AUD_DEF_SAMPLE_RATE_8000;
+ default:
+ return RPC_AUD_DEF_SAMPLE_RATE_44100;
+ }
+}
+
+static struct snd_pcm_hardware msm_pcm_playback_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = USE_FORMATS,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .buffer_bytes_max = 4800 * 2,
+ .period_bytes_min = 4800,
+ .period_bytes_max = 4800,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_capture_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = USE_FORMATS,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .channels_min = USE_CHANNELS_MIN,
+ .channels_max = USE_CHANNELS_MAX,
+ .buffer_bytes_max = MAX_BUFFER_CAPTURE_SIZE,
+ .period_bytes_min = CAPTURE_SIZE,
+ .period_bytes_max = CAPTURE_SIZE,
+ .periods_min = USE_PERIODS_MIN,
+ .periods_max = USE_PERIODS_MAX,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void msm_pcm_enqueue_data(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ unsigned int period_size;
+
+ pr_debug("prtd->out_tail =%d mmap_flag=%d\n",
+ prtd->out_tail, prtd->mmap_flag);
+ period_size = snd_pcm_lib_period_bytes(substream);
+ alsa_dsp_send_buffer(prtd, prtd->out_tail, period_size);
+ prtd->out_tail ^= 1;
+ ++copy_count;
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+
+}
+
+static void playback_event_handler(void *data)
+{
+ struct msm_audio *prtd = data;
+ snd_pcm_period_elapsed(prtd->playback_substream);
+ if (prtd->mmap_flag) {
+ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE)
+ return;
+ if (!prtd->stopped)
+ msm_pcm_enqueue_data(prtd->playback_substream);
+ else
+ prtd->out_needed++;
+ }
+}
+
+static void capture_event_handler(void *data)
+{
+ struct msm_audio *prtd = data;
+ snd_pcm_period_elapsed(prtd->capture_substream);
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ prtd->pcm_buf_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->out_sample_rate = runtime->rate;
+ prtd->out_channel_mode = runtime->channels;
+
+ if (prtd->enabled | !(prtd->mmap_flag))
+ return 0;
+
+ prtd->data = substream->dma_buffer.area;
+ prtd->phys = substream->dma_buffer.addr;
+ prtd->out[0].data = prtd->data + 0;
+ prtd->out[0].addr = prtd->phys + 0;
+ prtd->out[0].size = BUFSZ;
+ prtd->out[1].data = prtd->data + BUFSZ;
+ prtd->out[1].addr = prtd->phys + BUFSZ;
+ prtd->out[1].size = BUFSZ;
+
+ prtd->out[0].used = prtd->pcm_count;
+ prtd->out[1].used = prtd->pcm_count;
+
+ mutex_lock(&the_locks.lock);
+ alsa_audio_configure(prtd);
+ mutex_unlock(&the_locks.lock);
+
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct audmgr_config cfg;
+ int rc;
+
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ prtd->pcm_buf_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = convert_samp_rate(runtime->rate);
+ prtd->samp_rate_index = convert_dsp_samp_index(runtime->rate);
+ prtd->channel_mode = (runtime->channels - 1);
+ prtd->buffer_size = prtd->channel_mode ? STEREO_DATA_SIZE : \
+ MONO_DATA_SIZE;
+
+ if (prtd->enabled == 1)
+ return 0;
+
+ prtd->type = AUDREC_CMD_TYPE_0_INDEX_WAV;
+
+ cfg.tx_rate = convert_samp_rate(runtime->rate);
+ cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_NONE;
+ cfg.def_method = RPC_AUD_DEF_METHOD_RECORD;
+ cfg.codec = RPC_AUD_DEF_CODEC_PCM;
+ cfg.snd_method = RPC_SND_METHOD_MIDI;
+
+ rc = audmgr_enable(&prtd->audmgr, &cfg);
+ if (rc < 0)
+ return rc;
+
+ if (msm_adsp_enable(prtd->audpre)) {
+ audmgr_disable(&prtd->audmgr);
+ return -ENODEV;
+ }
+ if (msm_adsp_enable(prtd->audrec)) {
+ msm_adsp_disable(prtd->audpre);
+ audmgr_disable(&prtd->audmgr);
+ return -ENODEV;
+ }
+ prtd->enabled = 1;
+ alsa_rec_dsp_enable(prtd, 1);
+
+ return 0;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ unsigned long flag = 0;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ || !prtd->mmap_flag)
+ break;
+ if (!prtd->out_needed) {
+ prtd->stopped = 0;
+ break;
+ }
+ spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
+ if (prtd->running == 1) {
+ if (prtd->stopped == 1) {
+ prtd->stopped = 0;
+ prtd->period = 0;
+ if (prtd->pcm_irq_pos == 0) {
+ prtd->out_tail = 0;
+ msm_pcm_enqueue_data(
+ prtd->playback_substream);
+ prtd->out_needed--;
+ } else {
+ prtd->out_tail = 1;
+ msm_pcm_enqueue_data(
+ prtd->playback_substream);
+ prtd->out_needed--;
+ }
+ if (prtd->out_needed) {
+ prtd->out_tail ^= 1;
+ msm_pcm_enqueue_data(
+ prtd->playback_substream);
+ prtd->out_needed--;
+ }
+ }
+ }
+ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ || !prtd->mmap_flag)
+ break;
+ prtd->stopped = 1;
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static snd_pcm_uframes_t
+msm_pcm_playback_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos == prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int rc = 0, rc1 = 0, rc2 = 0;
+ int fbytes = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+ int monofbytes = 0;
+ char *bufferp = NULL;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ monofbytes = fbytes / 2;
+ if (runtime->channels == 2) {
+ rc = alsa_buffer_read(prtd, buf, fbytes, NULL);
+ } else {
+ bufferp = buf;
+ rc1 = alsa_buffer_read(prtd, bufferp, monofbytes, NULL);
+ bufferp = buf + monofbytes ;
+ rc2 = alsa_buffer_read(prtd, bufferp, monofbytes, NULL);
+ rc = rc1 + rc2;
+ }
+ prtd->pcm_buf_pos += fbytes;
+ return rc;
+}
+
+static snd_pcm_uframes_t
+msm_pcm_capture_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ alsa_audrec_disable(prtd);
+ audmgr_close(&prtd->audmgr);
+ msm_adsp_put(prtd->audrec);
+ msm_adsp_put(prtd->audpre);
+ kfree(prtd);
+
+ return 0;
+}
+
+struct msm_audio_event_callbacks snd_msm_audio_ops = {
+ .playback = playback_event_handler,
+ .capture = capture_event_handler,
+};
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ return ret;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_pcm_playback_hardware;
+ prtd->dir = SNDRV_PCM_STREAM_PLAYBACK;
+ prtd->playback_substream = substream;
+ prtd->eos_ack = 0;
+ ret = msm_audio_volume_update(PCMPLAYBACK_DECODERID,
+ msm_vol_ctl.volume, msm_vol_ctl.pan);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_capture_hardware;
+ prtd->dir = SNDRV_PCM_STREAM_CAPTURE;
+ prtd->capture_substream = substream;
+ }
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ goto out;
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ goto out;
+
+ prtd->ops = &snd_msm_audio_ops;
+ prtd->out[0].used = BUF_INVALID_LEN;
+ prtd->out_head = 1; /* point to second buffer on startup */
+ runtime->private_data = prtd;
+
+ ret = alsa_adsp_configure(prtd);
+ if (ret)
+ goto out;
+ copy_count = 0;
+ return 0;
+
+ out:
+ kfree(prtd);
+ return ret;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int rc = 1;
+ int fbytes = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ rc = alsa_send_buffer(prtd, buf, fbytes, NULL);
+ ++copy_count;
+ if (copy_count == 1) {
+ mutex_lock(&the_locks.lock);
+ alsa_audio_configure(prtd);
+ mutex_unlock(&the_locks.lock);
+ }
+ return rc;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ int rc = 0;
+
+ pr_debug("%s()\n", __func__);
+
+ /* pcm dmamiss message is sent continously
+ * when decoder is starved so no race
+ * condition concern
+ */
+ if (prtd->enabled)
+ rc = wait_event_interruptible(the_locks.eos_wait,
+ prtd->eos_ack);
+
+ alsa_audio_disable(prtd);
+ audmgr_close(&prtd->audmgr);
+ kfree(prtd);
+
+ return 0;
+}
+
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_pointer(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_pointer(substream);
+ return ret;
+}
+
+int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (substream->pcm->device & 1) {
+ runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED;
+ runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED;
+ }
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+
+}
+
+int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ prtd->out_head = 0; /* point to First buffer on startup */
+ prtd->mmap_flag = 1;
+ runtime->dma_bytes = snd_pcm_lib_period_bytes(substream)*2;
+ dma_mmap_coherent(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size;
+ if (!stream)
+ size = PLAYBACK_DMASZ;
+ else
+ size = CAPTURE_DMASZ;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void msm_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static int msm_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, 1);
+ if (ret)
+ return ret;
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+ if (ret)
+ return ret;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &msm_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &msm_pcm_ops);
+
+ ret = pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+
+ ret = pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ msm_pcm_free_dma_buffers(pcm);
+ return ret;
+}
+
+struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_pcm_new,
+ .pcm_free = msm_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL(msm_soc_platform);
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-dsp-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm7kv2-dai.c b/sound/soc/msm/msm7kv2-dai.c
new file mode 100644
index 0000000..e8d51ac
--- /dev/null
+++ b/sound/soc/msm/msm7kv2-dai.c
@@ -0,0 +1,149 @@
+/* sound/soc/msm/msm-dai.c
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2010, Code Aurora Forum. All rights reserved.
+ *
+ * Derived from msm-pcm.c and msm7201.c.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <linux/slab.h>
+#include "msm7kv2-pcm.h"
+
+static struct snd_soc_dai_driver msm_pcm_codec_dais[] = {
+{
+ .name = "msm-codec-dai",
+ .playback = {
+ .channels_max = USE_CHANNELS_MAX,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_max = USE_CHANNELS_MAX,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+};
+static struct snd_soc_dai_driver msm_pcm_cpu_dais[] = {
+{
+ .name = "msm-cpu-dai",
+ .playback = {
+ .channels_max = USE_CHANNELS_MAX,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_max = USE_CHANNELS_MAX,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = USE_RATE_MAX,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_msm = {
+ .compress_type = SND_SOC_FLAT_COMPRESSION,
+};
+
+static __devinit int asoc_msm_codec_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_msm,
+ msm_pcm_codec_dais, ARRAY_SIZE(msm_pcm_codec_dais));
+}
+
+static int __devexit asoc_msm_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static __devinit int asoc_msm_cpu_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_dai(&pdev->dev, msm_pcm_cpu_dais);
+}
+
+static int __devexit asoc_msm_cpu_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_msm_codec_driver = {
+ .probe = asoc_msm_codec_probe,
+ .remove = __devexit_p(asoc_msm_codec_remove),
+ .driver = {
+ .name = "msm-codec-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static struct platform_driver asoc_msm_cpu_driver = {
+ .probe = asoc_msm_cpu_probe,
+ .remove = __devexit_p(asoc_msm_cpu_remove),
+ .driver = {
+ .name = "msm-cpu-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_codec_dai_init(void)
+{
+ return platform_driver_register(&asoc_msm_codec_driver);
+}
+
+static void __exit msm_codec_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_msm_codec_driver);
+}
+
+static int __init msm_cpu_dai_init(void)
+{
+ return platform_driver_register(&asoc_msm_cpu_driver);
+}
+
+static void __exit msm_cpu_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_msm_cpu_driver);
+}
+
+module_init(msm_codec_dai_init);
+module_exit(msm_codec_dai_exit);
+module_init(msm_cpu_dai_init);
+module_exit(msm_cpu_dai_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM Codec/Cpu Dai driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm7kv2-dsp.c b/sound/soc/msm/msm7kv2-dsp.c
new file mode 100644
index 0000000..50bf6fb
--- /dev/null
+++ b/sound/soc/msm/msm7kv2-dsp.c
@@ -0,0 +1,633 @@
+/* Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2010, Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <mach/qdsp5v2/audio_dev_ctl.h>
+#include <mach/debug_mm.h>
+
+#include "msm7kv2-pcm.h"
+
+/* Audrec Queue command sent macro's */
+#define audrec_send_bitstreamqueue(audio, cmd, len) \
+ msm_adsp_write(audio->audrec, ((audio->queue_id & 0xFFFF0000) >> 16),\
+ cmd, len)
+
+#define audrec_send_audrecqueue(audio, cmd, len) \
+ msm_adsp_write(audio->audrec, (audio->queue_id & 0x0000FFFF),\
+ cmd, len)
+
+static int alsa_dsp_read_buffer(struct msm_audio *audio,
+ uint32_t read_cnt);
+static void alsa_get_dsp_frames(struct msm_audio *prtd);
+static int alsa_in_param_config(struct msm_audio *audio);
+
+static int alsa_in_mem_config(struct msm_audio *audio);
+static int alsa_in_enc_config(struct msm_audio *audio, int enable);
+
+int intcnt;
+struct audio_frame {
+ uint16_t count_low;
+ uint16_t count_high;
+ uint16_t bytes;
+ uint16_t unknown;
+ unsigned char samples[];
+} __attribute__ ((packed));
+
+void alsa_dsp_event(void *data, unsigned id, uint16_t *msg)
+{
+ struct msm_audio *prtd = data;
+ struct buffer *frame;
+ unsigned long flag = 0;
+
+ MM_DBG("\n");
+ switch (id) {
+ case AUDPP_MSG_HOST_PCM_INTF_MSG: {
+ unsigned id = msg[3];
+ unsigned idx = msg[4] - 1;
+
+ MM_DBG("HOST_PCM id %d idx %d\n", id, idx);
+ if (id != AUDPP_MSG_HOSTPCM_ID_ARM_RX) {
+ MM_ERR("bogus id\n");
+ break;
+ }
+ if (idx > 1) {
+ MM_ERR("bogus buffer idx\n");
+ break;
+ }
+
+ /* Update with actual sent buffer size */
+ if (prtd->out[idx].used != BUF_INVALID_LEN)
+ prtd->pcm_irq_pos += prtd->out[idx].used;
+
+ if (prtd->pcm_irq_pos > prtd->pcm_size)
+ prtd->pcm_irq_pos = prtd->pcm_count;
+
+ if (prtd->ops->playback)
+ prtd->ops->playback(prtd);
+
+ if (prtd->mmap_flag)
+ break;
+
+ spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
+ if (prtd->running) {
+ prtd->out[idx].used = 0;
+ frame = prtd->out + prtd->out_tail;
+ if (frame->used) {
+ alsa_dsp_send_buffer(
+ prtd, prtd->out_tail, frame->used);
+ /* Reset eos_ack flag to avoid stale
+ * PCMDMAMISS been considered
+ */
+ prtd->eos_ack = 0;
+ prtd->out_tail ^= 1;
+ } else {
+ prtd->out_needed++;
+ }
+ wake_up(&the_locks.write_wait);
+ }
+ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ break;
+ }
+ case AUDPP_MSG_PCMDMAMISSED:
+ MM_INFO("PCMDMAMISSED %d\n", msg[0]);
+ prtd->eos_ack++;
+ MM_DBG("PCMDMAMISSED Count per Buffer %d\n", prtd->eos_ack);
+ wake_up(&the_locks.eos_wait);
+ break;
+ case AUDPP_MSG_CFG_MSG:
+ if (msg[0] == AUDPP_MSG_ENA_ENA) {
+ MM_DBG("CFG_MSG ENABLE\n");
+ prtd->out_needed = 0;
+ prtd->running = 1;
+ audpp_dsp_set_vol_pan(prtd->session_id, &prtd->vol_pan,
+ POPP);
+ audpp_route_stream(prtd->session_id,
+ msm_snddev_route_dec(prtd->session_id));
+ audio_dsp_out_enable(prtd, 1);
+ } else if (msg[0] == AUDPP_MSG_ENA_DIS) {
+ MM_DBG("CFG_MSG DISABLE\n");
+ prtd->running = 0;
+ } else {
+ MM_DBG("CFG_MSG %d?\n", msg[0]);
+ }
+ break;
+ default:
+ MM_DBG("UNKNOWN (%d)\n", id);
+ }
+}
+
+static void audpreproc_dsp_event(void *data, unsigned id, void *msg)
+{
+ struct msm_audio *prtd = data;
+
+ switch (id) {
+ case AUDPREPROC_ERROR_MSG: {
+ struct audpreproc_err_msg *err_msg = msg;
+
+ MM_ERR("ERROR_MSG: stream id %d err idx %d\n",
+ err_msg->stream_id, err_msg->aud_preproc_err_idx);
+ /* Error case */
+ break;
+ }
+ case AUDPREPROC_CMD_CFG_DONE_MSG: {
+ MM_DBG("CMD_CFG_DONE_MSG\n");
+ break;
+ }
+ case AUDPREPROC_CMD_ENC_CFG_DONE_MSG: {
+ struct audpreproc_cmd_enc_cfg_done_msg *enc_cfg_msg = msg;
+
+ MM_DBG("CMD_ENC_CFG_DONE_MSG: stream id %d enc type \
+ 0x%8x\n", enc_cfg_msg->stream_id,
+ enc_cfg_msg->rec_enc_type);
+ /* Encoder enable success */
+ if (enc_cfg_msg->rec_enc_type & ENCODE_ENABLE)
+ alsa_in_param_config(prtd);
+ else { /* Encoder disable success */
+ prtd->running = 0;
+ alsa_in_record_config(prtd, 0);
+ }
+ break;
+ }
+ case AUDPREPROC_CMD_ENC_PARAM_CFG_DONE_MSG: {
+ MM_DBG("CMD_ENC_PARAM_CFG_DONE_MSG\n");
+ alsa_in_mem_config(prtd);
+ break;
+ }
+ case AUDPREPROC_AFE_CMD_AUDIO_RECORD_CFG_DONE_MSG: {
+ MM_DBG("AFE_CMD_AUDIO_RECORD_CFG_DONE_MSG\n");
+ wake_up(&the_locks.enable_wait);
+ break;
+ }
+ default:
+ MM_DBG("Unknown Event id %d\n", id);
+ }
+}
+
+static void audrec_dsp_event(void *data, unsigned id, size_t len,
+ void (*getevent) (void *ptr, size_t len))
+{
+ struct msm_audio *prtd = data;
+ unsigned long flag = 0;
+
+ switch (id) {
+ case AUDREC_CMD_MEM_CFG_DONE_MSG: {
+ MM_DBG("AUDREC_CMD_MEM_CFG_DONE_MSG\n");
+ prtd->running = 1;
+ alsa_in_record_config(prtd, 1);
+ break;
+ }
+ case AUDREC_FATAL_ERR_MSG: {
+ struct audrec_fatal_err_msg fatal_err_msg;
+
+ getevent(&fatal_err_msg, AUDREC_FATAL_ERR_MSG_LEN);
+ MM_ERR("FATAL_ERR_MSG: err id %d\n",
+ fatal_err_msg.audrec_err_id);
+ /* Error stop the encoder */
+ prtd->stopped = 1;
+ wake_up(&the_locks.read_wait);
+ break;
+ }
+ case AUDREC_UP_PACKET_READY_MSG: {
+ struct audrec_up_pkt_ready_msg pkt_ready_msg;
+ MM_DBG("AUDREC_UP_PACKET_READY_MSG\n");
+
+ getevent(&pkt_ready_msg, AUDREC_UP_PACKET_READY_MSG_LEN);
+ MM_DBG("UP_PACKET_READY_MSG: write cnt lsw %d \
+ write cnt msw %d read cnt lsw %d read cnt msw %d \n",\
+ pkt_ready_msg.audrec_packet_write_cnt_lsw, \
+ pkt_ready_msg.audrec_packet_write_cnt_msw, \
+ pkt_ready_msg.audrec_up_prev_read_cnt_lsw, \
+ pkt_ready_msg.audrec_up_prev_read_cnt_msw);
+
+ alsa_get_dsp_frames(prtd);
+ ++intcnt;
+ if (prtd->channel_mode == 1) {
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+
+ if (prtd->ops->capture)
+ prtd->ops->capture(prtd);
+ } else if ((prtd->channel_mode == 0) && (intcnt % 2 == 0)) {
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+ if (prtd->ops->capture)
+ prtd->ops->capture(prtd);
+ }
+ break;
+ }
+ default:
+ MM_DBG("Unknown Event id %d\n", id);
+ }
+}
+
+struct msm_adsp_ops alsa_audrec_adsp_ops = {
+ .event = audrec_dsp_event,
+};
+
+int alsa_audio_configure(struct msm_audio *prtd)
+{
+ if (prtd->enabled)
+ return 0;
+
+ MM_DBG("\n");
+ if (prtd->dir == SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->out_weight = 100;
+ if (audpp_enable(-1, alsa_dsp_event, prtd)) {
+ MM_ERR("audpp_enable() failed\n");
+ return -ENODEV;
+ }
+ }
+ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE) {
+ if (audpreproc_enable(prtd->session_id,
+ &audpreproc_dsp_event, prtd)) {
+ MM_ERR("audpreproc_enable failed\n");
+ return -ENODEV;
+ }
+
+ if (msm_adsp_enable(prtd->audrec)) {
+ MM_ERR("msm_adsp_enable(audrec) enable failed\n");
+ audpreproc_disable(prtd->session_id, prtd);
+ return -ENODEV;
+ }
+ alsa_in_enc_config(prtd, 1);
+
+ }
+ prtd->enabled = 1;
+ return 0;
+}
+EXPORT_SYMBOL(alsa_audio_configure);
+
+ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf,
+ size_t count, loff_t *pos)
+{
+ unsigned long flag = 0;
+ const char __user *start = buf;
+ struct buffer *frame;
+ size_t xfer;
+ int ret = 0;
+
+ MM_DBG("\n");
+ mutex_lock(&the_locks.write_lock);
+ while (count > 0) {
+ frame = prtd->out + prtd->out_head;
+ ret = wait_event_interruptible(the_locks.write_wait,
+ (frame->used == 0)
+ || (prtd->stopped));
+ if (ret < 0)
+ break;
+ if (prtd->stopped) {
+ ret = -EBUSY;
+ break;
+ }
+ xfer = count > frame->size ? frame->size : count;
+ if (copy_from_user(frame->data, buf, xfer)) {
+ ret = -EFAULT;
+ break;
+ }
+ frame->used = xfer;
+ prtd->out_head ^= 1;
+ count -= xfer;
+ buf += xfer;
+
+ spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
+ frame = prtd->out + prtd->out_tail;
+ if (frame->used && prtd->out_needed) {
+ alsa_dsp_send_buffer(prtd, prtd->out_tail,
+ frame->used);
+ /* Reset eos_ack flag to avoid stale
+ * PCMDMAMISS been considered
+ */
+ prtd->eos_ack = 0;
+ prtd->out_tail ^= 1;
+ prtd->out_needed--;
+ }
+ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ }
+ mutex_unlock(&the_locks.write_lock);
+ if (buf > start)
+ return buf - start;
+ return ret;
+}
+EXPORT_SYMBOL(alsa_send_buffer);
+
+int alsa_audio_disable(struct msm_audio *prtd)
+{
+ if (prtd->enabled) {
+ MM_DBG("\n");
+ mutex_lock(&the_locks.lock);
+ prtd->enabled = 0;
+ audio_dsp_out_enable(prtd, 0);
+ wake_up(&the_locks.write_wait);
+ audpp_disable(-1, prtd);
+ prtd->out_needed = 0;
+ mutex_unlock(&the_locks.lock);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(alsa_audio_disable);
+
+int alsa_audrec_disable(struct msm_audio *prtd)
+{
+ if (prtd->enabled) {
+ prtd->enabled = 0;
+ alsa_in_enc_config(prtd, 0);
+ wake_up(&the_locks.read_wait);
+ msm_adsp_disable(prtd->audrec);
+ prtd->out_needed = 0;
+ audpreproc_disable(prtd->session_id, prtd);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(alsa_audrec_disable);
+
+static int alsa_in_enc_config(struct msm_audio *prtd, int enable)
+{
+ struct audpreproc_audrec_cmd_enc_cfg cmd;
+ int i;
+ unsigned short *ptrmem = (unsigned short *)&cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDPREPROC_AUDREC_CMD_ENC_CFG;
+ cmd.stream_id = prtd->session_id;
+
+ if (enable)
+ cmd.audrec_enc_type = prtd->type | ENCODE_ENABLE;
+ else
+ cmd.audrec_enc_type &= ~(ENCODE_ENABLE);
+ for (i = 0; i < sizeof(cmd)/2; i++, ++ptrmem)
+ MM_DBG("cmd[%d]=0x%04x\n", i, *ptrmem);
+
+ return audpreproc_send_audreccmdqueue(&cmd, sizeof(cmd));
+}
+
+static int alsa_in_param_config(struct msm_audio *prtd)
+{
+ struct audpreproc_audrec_cmd_parm_cfg_wav cmd;
+ int i;
+ unsigned short *ptrmem = (unsigned short *)&cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.common.cmd_id = AUDPREPROC_AUDREC_CMD_PARAM_CFG;
+ cmd.common.stream_id = prtd->session_id;
+
+ cmd.aud_rec_samplerate_idx = prtd->samp_rate;
+ cmd.aud_rec_stereo_mode = prtd->channel_mode;
+ for (i = 0; i < sizeof(cmd)/2; i++, ++ptrmem)
+ MM_DBG("cmd[%d]=0x%04x\n", i, *ptrmem);
+
+ return audpreproc_send_audreccmdqueue(&cmd, sizeof(cmd));
+}
+
+int alsa_in_record_config(struct msm_audio *prtd, int enable)
+{
+ struct audpreproc_afe_cmd_audio_record_cfg cmd;
+ int i;
+ unsigned short *ptrmem = (unsigned short *)&cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDPREPROC_AFE_CMD_AUDIO_RECORD_CFG;
+ cmd.stream_id = prtd->session_id;
+ if (enable)
+ cmd.destination_activity = AUDIO_RECORDING_TURN_ON;
+ else
+ cmd.destination_activity = AUDIO_RECORDING_TURN_OFF;
+ cmd.source_mix_mask = prtd->source;
+ if (prtd->session_id == 2) {
+ if ((cmd.source_mix_mask &
+ INTERNAL_CODEC_TX_SOURCE_MIX_MASK) ||
+ (cmd.source_mix_mask & AUX_CODEC_TX_SOURCE_MIX_MASK) ||
+ (cmd.source_mix_mask & VOICE_UL_SOURCE_MIX_MASK) ||
+ (cmd.source_mix_mask & VOICE_DL_SOURCE_MIX_MASK)) {
+ cmd.pipe_id = SOURCE_PIPE_1;
+ }
+ if (cmd.source_mix_mask &
+ AUDPP_A2DP_PIPE_SOURCE_MIX_MASK)
+ cmd.pipe_id |= SOURCE_PIPE_0;
+ }
+ for (i = 0; i < sizeof(cmd)/2; i++, ++ptrmem)
+ MM_DBG("cmd[%d]=0x%04x\n", i, *ptrmem);
+ return audpreproc_send_audreccmdqueue(&cmd, sizeof(cmd));
+}
+
+static int alsa_in_mem_config(struct msm_audio *prtd)
+{
+ struct audrec_cmd_arecmem_cfg cmd;
+ uint16_t *data = (void *) prtd->data;
+ int n;
+ int i;
+ unsigned short *ptrmem = (unsigned short *)&cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDREC_CMD_MEM_CFG_CMD;
+ cmd.audrec_up_pkt_intm_count = 1;
+ cmd.audrec_ext_pkt_start_addr_msw = prtd->phys >> 16;
+ cmd.audrec_ext_pkt_start_addr_lsw = prtd->phys;
+ cmd.audrec_ext_pkt_buf_number = FRAME_NUM;
+
+ /* prepare buffer pointers:
+ * Mono: 1024 samples + 4 halfword header
+ * Stereo: 2048 samples + 4 halfword header
+ */
+ for (n = 0; n < FRAME_NUM; n++) {
+ prtd->in[n].data = data + 4;
+ data += (4 + (prtd->channel_mode ? 2048 : 1024));
+ MM_DBG("0x%8x\n", (int)(prtd->in[n].data - 8));
+ }
+ for (i = 0; i < sizeof(cmd)/2; i++, ++ptrmem)
+ MM_DBG("cmd[%d]=0x%04x\n", i, *ptrmem);
+
+ return audrec_send_audrecqueue(prtd, &cmd, sizeof(cmd));
+}
+
+int audio_dsp_out_enable(struct msm_audio *prtd, int yes)
+{
+ struct audpp_cmd_pcm_intf cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDPP_CMD_PCM_INTF;
+ cmd.stream = AUDPP_CMD_POPP_STREAM;
+ cmd.stream_id = prtd->session_id;
+ cmd.config = AUDPP_CMD_PCM_INTF_CONFIG_CMD_V;
+ cmd.intf_type = AUDPP_CMD_PCM_INTF_RX_ENA_ARMTODSP_V;
+
+ if (yes) {
+ cmd.write_buf1LSW = prtd->out[0].addr;
+ cmd.write_buf1MSW = prtd->out[0].addr >> 16;
+ cmd.write_buf1_len = prtd->out[0].size;
+ cmd.write_buf2LSW = prtd->out[1].addr;
+ cmd.write_buf2MSW = prtd->out[1].addr >> 16;
+ if (prtd->out[1].used)
+ cmd.write_buf2_len = prtd->out[1].used;
+ else
+ cmd.write_buf2_len = prtd->out[1].size;
+ cmd.arm_to_rx_flag = AUDPP_CMD_PCM_INTF_ENA_V;
+ cmd.weight_decoder_to_rx = prtd->out_weight;
+ cmd.weight_arm_to_rx = 1;
+ cmd.partition_number_arm_to_dsp = 0;
+ cmd.sample_rate = prtd->out_sample_rate;
+ cmd.channel_mode = prtd->out_channel_mode;
+ }
+
+ return audpp_send_queue2(&cmd, sizeof(cmd));
+}
+
+int alsa_buffer_read(struct msm_audio *prtd, void __user *buf,
+ size_t count, loff_t *pos)
+{
+ unsigned long flag;
+ void *data;
+ uint32_t index;
+ uint32_t size;
+ int ret = 0;
+
+ mutex_lock(&the_locks.read_lock);
+ while (count > 0) {
+ ret = wait_event_interruptible(the_locks.read_wait,
+ (prtd->in_count > 0)
+ || prtd->stopped ||
+ prtd->abort);
+
+ if (ret < 0)
+ break;
+
+ if (prtd->stopped) {
+ ret = -EBUSY;
+ break;
+ }
+
+ if (prtd->abort) {
+ MM_DBG(" prtd->abort !\n");
+ ret = -EPERM; /* Not permitted due to abort */
+ break;
+ }
+
+ index = prtd->in_tail;
+ data = (uint8_t *) prtd->in[index].data;
+ size = prtd->in[index].size;
+ if (count >= size) {
+ if (copy_to_user(buf, data, size)) {
+ ret = -EFAULT;
+ break;
+ }
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ if (index != prtd->in_tail) {
+ /* overrun: data is invalid, we need to retry */
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock,
+ flag);
+ continue;
+ }
+ prtd->in[index].size = 0;
+ prtd->in_tail = (prtd->in_tail + 1) & (FRAME_NUM - 1);
+ prtd->in_count--;
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+ count -= size;
+ buf += size;
+ } else {
+ break;
+ }
+ }
+ mutex_unlock(&the_locks.read_lock);
+ return ret;
+}
+EXPORT_SYMBOL(alsa_buffer_read);
+
+int alsa_dsp_send_buffer(struct msm_audio *prtd,
+ unsigned idx, unsigned len)
+{
+ struct audpp_cmd_pcm_intf_send_buffer cmd;
+ int i;
+ unsigned short *ptrmem = (unsigned short *)&cmd;
+
+ cmd.cmd_id = AUDPP_CMD_PCM_INTF;
+ cmd.stream = AUDPP_CMD_POPP_STREAM;
+ cmd.stream_id = prtd->session_id;
+ cmd.config = AUDPP_CMD_PCM_INTF_BUFFER_CMD_V;
+ cmd.intf_type = AUDPP_CMD_PCM_INTF_RX_ENA_ARMTODSP_V;
+ cmd.dsp_to_arm_buf_id = 0;
+ cmd.arm_to_dsp_buf_id = idx + 1;
+ cmd.arm_to_dsp_buf_len = len;
+ for (i = 0; i < sizeof(cmd)/2; i++, ++ptrmem)
+ MM_DBG("cmd[%d]=0x%04x\n", i, *ptrmem);
+
+ return audpp_send_queue2(&cmd, sizeof(cmd));
+}
+
+static int alsa_dsp_read_buffer(struct msm_audio *audio, uint32_t read_cnt)
+{
+ struct up_audrec_packet_ext_ptr cmd;
+ int i;
+ unsigned short *ptrmem = (unsigned short *)&cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = UP_AUDREC_PACKET_EXT_PTR;
+ cmd.audrec_up_curr_read_count_msw = read_cnt >> 16;
+ cmd.audrec_up_curr_read_count_lsw = read_cnt;
+ for (i = 0; i < sizeof(cmd)/2; i++, ++ptrmem)
+ MM_DBG("cmd[%d]=0x%04x\n", i, *ptrmem);
+
+ return audrec_send_bitstreamqueue(audio, &cmd, sizeof(cmd));
+}
+
+static void alsa_get_dsp_frames(struct msm_audio *prtd)
+{
+ struct audio_frame *frame;
+ uint32_t index = 0;
+ unsigned long flag;
+
+ if (prtd->type == ENC_TYPE_WAV) {
+ index = prtd->in_head;
+
+ frame =
+ (void *)(((char *)prtd->in[index].data) - sizeof(*frame));
+
+ spin_lock_irqsave(&the_locks.read_dsp_lock, flag);
+ prtd->in[index].size = frame->bytes;
+ MM_DBG("frame = %08x\n", (unsigned int) frame);
+ MM_DBG("prtd->in[index].size = %08x\n",
+ (unsigned int) prtd->in[index].size);
+
+ prtd->in_head = (prtd->in_head + 1) & (FRAME_NUM - 1);
+
+ /* If overflow, move the tail index foward. */
+ if (prtd->in_head == prtd->in_tail)
+ prtd->in_tail = (prtd->in_tail + 1) & (FRAME_NUM - 1);
+ else
+ prtd->in_count++;
+
+ prtd->pcm_irq_pos += frame->bytes;
+ alsa_dsp_read_buffer(prtd, prtd->dsp_cnt++);
+ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag);
+
+ wake_up(&the_locks.read_wait);
+ }
+}
diff --git a/sound/soc/msm/msm7kv2-pcm.c b/sound/soc/msm/msm7kv2-pcm.c
new file mode 100644
index 0000000..c64a3ff
--- /dev/null
+++ b/sound/soc/msm/msm7kv2-pcm.c
@@ -0,0 +1,774 @@
+/* Copyright (c) 2008-2010, Code Aurora Forum. All rights reserved.
+ *
+ * All source code in this file is licensed under the following license except
+ * where indicated.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <linux/slab.h>
+#include "msm7kv2-pcm.h"
+#include <mach/qdsp5v2/audio_dev_ctl.h>
+#include <mach/debug_mm.h>
+
+#define HOSTPCM_STREAM_ID 5
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+int copy_count;
+
+static struct snd_pcm_hardware msm_pcm_playback_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = USE_FORMATS,
+ .rates = USE_RATE,
+ .rate_min = USE_RATE_MIN,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_BUFFER_PLAYBACK_SIZE,
+ .period_bytes_min = BUFSZ,
+ .period_bytes_max = BUFSZ,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_capture_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = USE_FORMATS,
+ .rates = USE_RATE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_BUFFER_CAPTURE_SIZE,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 4096,
+ .periods_min = 4,
+ .periods_max = 4,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+static void alsa_out_listener(u32 evt_id, union auddev_evt_data *evt_payload,
+ void *private_data)
+{
+ struct msm_audio *prtd = (struct msm_audio *) private_data;
+ MM_DBG("evt_id = 0x%8x\n", evt_id);
+ switch (evt_id) {
+ case AUDDEV_EVT_DEV_RDY:
+ MM_DBG("AUDDEV_EVT_DEV_RDY\n");
+ prtd->source |= (0x1 << evt_payload->routing_id);
+ if (prtd->running == 1 && prtd->enabled == 1)
+ audpp_route_stream(prtd->session_id, prtd->source);
+ break;
+ case AUDDEV_EVT_DEV_RLS:
+ MM_DBG("AUDDEV_EVT_DEV_RLS\n");
+ prtd->source &= ~(0x1 << evt_payload->routing_id);
+ if (prtd->running == 1 && prtd->enabled == 1)
+ audpp_route_stream(prtd->session_id, prtd->source);
+ break;
+ case AUDDEV_EVT_STREAM_VOL_CHG:
+ prtd->vol_pan.volume = evt_payload->session_vol;
+ MM_DBG("AUDDEV_EVT_STREAM_VOL_CHG, stream vol %d\n",
+ prtd->vol_pan.volume);
+ if (prtd->running)
+ audpp_set_volume_and_pan(prtd->session_id,
+ prtd->vol_pan.volume,
+ 0, POPP);
+ break;
+ default:
+ MM_DBG("Unknown Event\n");
+ break;
+ }
+}
+
+static void alsa_in_listener(u32 evt_id, union auddev_evt_data *evt_payload,
+ void *private_data)
+{
+ struct msm_audio *prtd = (struct msm_audio *) private_data;
+ MM_DBG("evt_id = 0x%8x\n", evt_id);
+
+ switch (evt_id) {
+ case AUDDEV_EVT_DEV_RDY: {
+ MM_DBG("AUDDEV_EVT_DEV_RDY\n");
+ prtd->source |= (0x1 << evt_payload->routing_id);
+
+ if ((prtd->running == 1) && (prtd->enabled == 1))
+ alsa_in_record_config(prtd, 1);
+
+ break;
+ }
+ case AUDDEV_EVT_DEV_RLS: {
+ MM_DBG("AUDDEV_EVT_DEV_RLS\n");
+ prtd->source &= ~(0x1 << evt_payload->routing_id);
+
+ if (!prtd->running || !prtd->enabled)
+ break;
+
+ /* Turn off as per source */
+ if (prtd->source)
+ alsa_in_record_config(prtd, 1);
+ else
+ /* Turn off all */
+ alsa_in_record_config(prtd, 0);
+
+ break;
+ }
+ case AUDDEV_EVT_FREQ_CHG: {
+ MM_DBG("Encoder Driver got sample rate change event\n");
+ MM_DBG("sample rate %d\n", evt_payload->freq_info.sample_rate);
+ MM_DBG("dev_type %d\n", evt_payload->freq_info.dev_type);
+ MM_DBG("acdb_dev_id %d\n", evt_payload->freq_info.acdb_dev_id);
+ if (prtd->running == 1) {
+ /* Stop Recording sample rate does not match
+ with device sample rate */
+ if (evt_payload->freq_info.sample_rate !=
+ prtd->samp_rate) {
+ alsa_in_record_config(prtd, 0);
+ prtd->abort = 1;
+ wake_up(&the_locks.read_wait);
+ }
+ }
+ break;
+ }
+ default:
+ MM_DBG("Unknown Event\n");
+ break;
+ }
+}
+
+static void msm_pcm_enqueue_data(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ unsigned int period_size;
+
+ MM_DBG("prtd->out_tail =%d mmap_flag=%d\n",
+ prtd->out_tail, prtd->mmap_flag);
+ period_size = snd_pcm_lib_period_bytes(substream);
+ alsa_dsp_send_buffer(prtd, prtd->out_tail, period_size);
+ prtd->out_tail ^= 1;
+ ++copy_count;
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+
+}
+
+static void event_handler(void *data)
+{
+ struct msm_audio *prtd = data;
+ MM_DBG("\n");
+ snd_pcm_period_elapsed(prtd->substream);
+ if (prtd->mmap_flag) {
+ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE)
+ return;
+ if (!prtd->stopped)
+ msm_pcm_enqueue_data(prtd->substream);
+ else
+ prtd->out_needed++;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ MM_DBG("\n");
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ prtd->pcm_buf_pos = 0;
+ if (prtd->enabled)
+ return 0;
+
+ MM_DBG("\n");
+ /* rate and channels are sent to audio driver */
+ prtd->out_sample_rate = runtime->rate;
+ prtd->out_channel_mode = runtime->channels;
+ prtd->data = prtd->substream->dma_buffer.area;
+ prtd->phys = prtd->substream->dma_buffer.addr;
+ prtd->out[0].data = prtd->data + 0;
+ prtd->out[0].addr = prtd->phys + 0;
+ prtd->out[0].size = BUFSZ;
+ prtd->out[1].data = prtd->data + BUFSZ;
+ prtd->out[1].addr = prtd->phys + BUFSZ;
+ prtd->out[1].size = BUFSZ;
+
+ if (prtd->enabled | !(prtd->mmap_flag))
+ return 0;
+
+ prtd->out[0].used = prtd->pcm_count;
+ prtd->out[1].used = prtd->pcm_count;
+
+ mutex_lock(&the_locks.lock);
+ alsa_audio_configure(prtd);
+ mutex_unlock(&the_locks.lock);
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ uint32_t freq;
+ MM_DBG("\n");
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ prtd->pcm_buf_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->type = ENC_TYPE_WAV;
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = (runtime->channels - 1);
+ prtd->buffer_size = prtd->channel_mode ? STEREO_DATA_SIZE : \
+ MONO_DATA_SIZE;
+
+ if (prtd->enabled)
+ return 0;
+
+ freq = prtd->samp_rate;
+
+ prtd->data = prtd->substream->dma_buffer.area;
+ prtd->phys = prtd->substream->dma_buffer.addr;
+ MM_DBG("prtd->data =%08x\n", (unsigned int)prtd->data);
+ MM_DBG("prtd->phys =%08x\n", (unsigned int)prtd->phys);
+
+ mutex_lock(&the_locks.lock);
+ ret = alsa_audio_configure(prtd);
+ mutex_unlock(&the_locks.lock);
+ if (ret)
+ return ret;
+ ret = wait_event_interruptible(the_locks.enable_wait,
+ prtd->running != 0);
+ MM_DBG("state prtd->running = %d ret = %d\n", prtd->running, ret);
+
+ if (prtd->running == 0)
+ ret = -ENODEV;
+ else
+ ret = 0;
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ unsigned long flag = 0;
+ int ret = 0;
+
+ MM_DBG("\n");
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ || !prtd->mmap_flag)
+ break;
+ if (!prtd->out_needed) {
+ prtd->stopped = 0;
+ break;
+ }
+ spin_lock_irqsave(&the_locks.write_dsp_lock, flag);
+ if (prtd->running == 1) {
+ if (prtd->stopped == 1) {
+ prtd->stopped = 0;
+ prtd->period = 0;
+ if (prtd->pcm_irq_pos == 0) {
+ prtd->out_tail = 0;
+ msm_pcm_enqueue_data(prtd->substream);
+ prtd->out_needed--;
+ } else {
+ prtd->out_tail = 1;
+ msm_pcm_enqueue_data(prtd->substream);
+ prtd->out_needed--;
+ }
+ if (prtd->out_needed) {
+ prtd->out_tail ^= 1;
+ msm_pcm_enqueue_data(prtd->substream);
+ prtd->out_needed--;
+ }
+ }
+ }
+ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ || !prtd->mmap_flag)
+ break;
+ prtd->stopped = 1;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+struct msm_audio_event_callbacks snd_msm_audio_ops = {
+ .playback = event_handler,
+ .capture = event_handler,
+};
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd;
+ int ret = 0;
+ int i = 0;
+ int session_attrb, sessionid;
+
+ MM_DBG("\n");
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ ret = -ENOMEM;
+ return ret;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (prtd->opened) {
+ kfree(prtd);
+ return -EBUSY;
+ }
+ runtime->hw = msm_pcm_playback_hardware;
+ prtd->dir = SNDRV_PCM_STREAM_PLAYBACK;
+ prtd->eos_ack = 0;
+ prtd->session_id = HOSTPCM_STREAM_ID;
+ prtd->device_events = AUDDEV_EVT_DEV_RDY |
+ AUDDEV_EVT_STREAM_VOL_CHG |
+ AUDDEV_EVT_DEV_RLS;
+ prtd->source = msm_snddev_route_dec(prtd->session_id);
+ MM_ERR("Register device event listener\n");
+ ret = auddev_register_evt_listner(prtd->device_events,
+ AUDDEV_CLNT_DEC, prtd->session_id,
+ alsa_out_listener, (void *) prtd);
+ if (ret) {
+ MM_ERR("failed to register device event listener\n");
+ goto evt_error;
+ }
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_capture_hardware;
+ prtd->dir = SNDRV_PCM_STREAM_CAPTURE;
+ session_attrb = ENC_TYPE_WAV;
+ sessionid = audpreproc_aenc_alloc(session_attrb,
+ &prtd->module_name, &prtd->queue_id);
+ if (sessionid < 0) {
+ MM_ERR("AUDREC not available\n");
+ kfree(prtd);
+ return -ENODEV;
+ }
+ prtd->session_id = sessionid;
+ MM_DBG("%s\n", prtd->module_name);
+ ret = msm_adsp_get(prtd->module_name, &prtd->audrec,
+ &alsa_audrec_adsp_ops, prtd);
+ if (ret < 0) {
+ audpreproc_aenc_free(prtd->session_id);
+ kfree(prtd);
+ return -ENODEV;
+ }
+
+ prtd->abort = 0;
+ prtd->device_events = AUDDEV_EVT_DEV_RDY | AUDDEV_EVT_DEV_RLS |
+ AUDDEV_EVT_FREQ_CHG;
+ prtd->source = msm_snddev_route_enc(prtd->session_id);
+ MM_ERR("Register device event listener\n");
+ ret = auddev_register_evt_listner(prtd->device_events,
+ AUDDEV_CLNT_ENC, prtd->session_id,
+ alsa_in_listener, (void *) prtd);
+ if (ret) {
+ MM_ERR("failed to register device event listener\n");
+ audpreproc_aenc_free(prtd->session_id);
+ msm_adsp_put(prtd->audrec);
+ goto evt_error;
+ }
+ }
+ prtd->substream = substream;
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ MM_ERR("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ MM_ERR("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->ops = &snd_msm_audio_ops;
+ prtd->out[0].used = BUF_INVALID_LEN;
+ prtd->out[1].used = 0;
+ prtd->out_head = 1; /* point to second buffer on startup */
+ prtd->out_tail = 0;
+ prtd->dsp_cnt = 0;
+ prtd->in_head = 0;
+ prtd->in_tail = 0;
+ prtd->in_count = 0;
+ prtd->out_needed = 0;
+ for (i = 0; i < FRAME_NUM; i++) {
+ prtd->in[i].size = 0;
+ prtd->in[i].read = 0;
+ }
+ prtd->vol_pan.volume = 0x2000;
+ prtd->vol_pan.pan = 0x0;
+ prtd->opened = 1;
+ runtime->private_data = prtd;
+
+ copy_count = 0;
+ return 0;
+evt_error:
+ kfree(prtd);
+ return ret;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ MM_DBG("%d\n", fbytes);
+ ret = alsa_send_buffer(prtd, buf, fbytes, NULL);
+ ++copy_count;
+ prtd->pcm_buf_pos += fbytes;
+ if (copy_count == 1) {
+ mutex_lock(&the_locks.lock);
+ ret = alsa_audio_configure(prtd);
+ mutex_unlock(&the_locks.lock);
+ }
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ int ret = 0;
+
+ MM_DBG("\n");
+ if ((!prtd->mmap_flag) && prtd->enabled) {
+ ret = wait_event_interruptible(the_locks.eos_wait,
+ (!(prtd->out[0].used) && !(prtd->out[1].used)));
+
+ if (ret < 0)
+ goto done;
+ }
+
+ /* PCM DMAMISS message is sent only once in
+ * hpcm interface. So, wait for buffer complete
+ * and teos flag.
+ */
+ if (prtd->enabled)
+ ret = wait_event_interruptible(the_locks.eos_wait,
+ prtd->eos_ack);
+
+done:
+ alsa_audio_disable(prtd);
+ auddev_unregister_evt_listner(AUDDEV_CLNT_DEC, prtd->session_id);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0, rc1 = 0, rc2 = 0;
+ int fbytes = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+ int monofbytes = 0;
+ char *bufferp = NULL;
+
+ if (prtd->abort)
+ return -EPERM;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ MM_DBG("%d\n", fbytes);
+ monofbytes = fbytes / 2;
+ if (runtime->channels == 2) {
+ ret = alsa_buffer_read(prtd, buf, fbytes, NULL);
+ } else {
+ bufferp = buf;
+ rc1 = alsa_buffer_read(prtd, bufferp, monofbytes, NULL);
+ bufferp = buf + monofbytes ;
+ rc2 = alsa_buffer_read(prtd, bufferp, monofbytes, NULL);
+ ret = rc1 + rc2;
+ }
+ prtd->pcm_buf_pos += fbytes;
+ MM_DBG("prtd->pcm_buf_pos =%d, prtd->mmap_flag =%d\n",
+ prtd->pcm_buf_pos, prtd->mmap_flag);
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+
+ MM_DBG("\n");
+ ret = msm_snddev_withdraw_freq(prtd->session_id,
+ SNDDEV_CAP_TX, AUDDEV_CLNT_ENC);
+ MM_DBG("msm_snddev_withdraw_freq\n");
+ auddev_unregister_evt_listner(AUDDEV_CLNT_ENC, prtd->session_id);
+ prtd->abort = 0;
+ wake_up(&the_locks.enable_wait);
+ alsa_audrec_disable(prtd);
+ audpreproc_aenc_free(prtd->session_id);
+ msm_adsp_put(prtd->audrec);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ MM_DBG("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ if (prtd->pcm_irq_pos == prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ prtd->out_head = 0; /* point to First buffer on startup */
+ prtd->mmap_flag = 1;
+ runtime->dma_bytes = snd_pcm_lib_period_bytes(substream)*2;
+ dma_mmap_coherent(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return 0;
+}
+
+int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int pcm_preallocate_buffer(struct snd_pcm *pcm,
+ int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size;
+ if (!stream)
+ size = PLAYBACK_DMASZ;
+ else
+ size = CAPTURE_DMASZ;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void msm_pcm_free_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static int msm_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret = 0;
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, 1);
+ if (ret)
+ return ret;
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+ if (ret)
+ return ret;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &msm_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &msm_pcm_ops);
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ ret = pcm_preallocate_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ ret = pcm_preallocate_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ msm_pcm_free_buffers(pcm);
+ return ret;
+}
+
+struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_pcm_new,
+ .pcm_free = msm_pcm_free_buffers,
+};
+EXPORT_SYMBOL(msm_soc_platform);
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-dsp-audio",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm7kv2-pcm.h b/sound/soc/msm/msm7kv2-pcm.h
new file mode 100644
index 0000000..fec7cf5
--- /dev/null
+++ b/sound/soc/msm/msm7kv2-pcm.h
@@ -0,0 +1,207 @@
+/*
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2010, Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+
+
+#include <mach/qdsp5v2/qdsp5audppcmdi.h>
+#include <mach/qdsp5v2/qdsp5audppmsg.h>
+#include <mach/qdsp5v2/qdsp5audplaycmdi.h>
+#include <mach/qdsp5v2/qdsp5audplaymsg.h>
+#include <mach/qdsp5v2/audpp.h>
+#include <mach/msm_adsp.h>
+#include <mach/qdsp5v2/qdsp5audreccmdi.h>
+#include <mach/qdsp5v2/qdsp5audrecmsg.h>
+#include <mach/qdsp5v2/audpreproc.h>
+
+
+#define FRAME_NUM (8)
+#define FRAME_SIZE (2052 * 2)
+#define MONO_DATA_SIZE (2048)
+#define STEREO_DATA_SIZE (MONO_DATA_SIZE * 2)
+#define CAPTURE_DMASZ (FRAME_SIZE * FRAME_NUM)
+
+#define BUFSZ (960 * 5)
+#define PLAYBACK_DMASZ (BUFSZ * 2)
+
+#define MSM_PLAYBACK_DEFAULT_VOLUME 0 /* 0dB */
+#define MSM_PLAYBACK_DEFAULT_PAN 0
+
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN 1
+#define USE_CHANNELS_MAX 2
+/* Support unconventional sample rates 12000, 24000 as well */
+#define USE_RATE \
+ (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+#define USE_RATE_MIN 8000
+#define USE_RATE_MAX 48000
+#define MAX_BUFFER_PLAYBACK_SIZE \
+ PLAYBACK_DMASZ
+/* 2048 frames (Mono), 1024 frames (Stereo) */
+#define CAPTURE_SIZE 4096
+#define MAX_BUFFER_CAPTURE_SIZE (4096*4)
+#define MAX_PERIOD_SIZE BUFSZ
+#define USE_PERIODS_MAX 1024
+#define USE_PERIODS_MIN 1
+
+
+#define MAX_DB (16)
+#define MIN_DB (-50)
+#define PCMPLAYBACK_DECODERID 5
+
+/* 0xFFFFFFFF Indicates not to be used for audio data copy */
+#define BUF_INVALID_LEN 0xFFFFFFFF
+#define EVENT_MSG_ID ((uint16_t)~0)
+
+#define AUDDEC_DEC_PCM 0
+/* Decoder status received from AUDPPTASK */
+#define AUDPP_DEC_STATUS_SLEEP 0
+#define AUDPP_DEC_STATUS_INIT 1
+#define AUDPP_DEC_STATUS_CFG 2
+#define AUDPP_DEC_STATUS_PLAY 3
+
+extern int copy_count;
+extern int intcnt;
+
+struct buffer {
+ void *data;
+ unsigned size;
+ unsigned used;
+ unsigned addr;
+};
+
+struct buffer_rec {
+ void *data;
+ unsigned int size;
+ unsigned int read;
+ unsigned int addr;
+};
+
+struct audio_locks {
+ struct mutex lock;
+ struct mutex write_lock;
+ struct mutex read_lock;
+ spinlock_t read_dsp_lock;
+ spinlock_t write_dsp_lock;
+ spinlock_t mixer_lock;
+ wait_queue_head_t read_wait;
+ wait_queue_head_t write_wait;
+ wait_queue_head_t wait;
+ wait_queue_head_t eos_wait;
+ wait_queue_head_t enable_wait;
+};
+
+extern struct audio_locks the_locks;
+
+struct msm_audio_event_callbacks {
+ /* event is called from interrupt context when a message
+ * arrives from the DSP.
+ */
+ void (*playback)(void *);
+ void (*capture)(void *);
+};
+
+
+struct msm_audio {
+ struct buffer out[2];
+ struct buffer_rec in[8];
+
+ uint8_t out_head;
+ uint8_t out_tail;
+ uint8_t out_needed; /* number of buffers the dsp is waiting for */
+ atomic_t out_bytes;
+
+ /* configuration to use on next enable */
+ uint32_t out_sample_rate;
+ uint32_t out_channel_mode;
+ uint32_t out_weight;
+ uint32_t out_buffer_size;
+
+ struct snd_pcm_substream *substream;
+
+ /* data allocated for various buffers */
+ char *data;
+ dma_addr_t phys;
+
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int pcm_buf_pos; /* position in buffer */
+ uint16_t source; /* Encoding source bit mask */
+
+ struct msm_adsp_module *audpre;
+ struct msm_adsp_module *audrec;
+ struct msm_adsp_module *audplay;
+ enum msm_aud_decoder_state dec_state; /* Represents decoder state */
+
+ uint16_t session_id;
+ uint32_t out_bits; /* bits per sample */
+ const char *module_name;
+ unsigned queue_id;
+
+ /* configuration to use on next enable */
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint32_t buffer_size; /* 2048 for mono, 4096 for stereo */
+ uint32_t type; /* 0 for PCM ,1 for AAC */
+ uint32_t dsp_cnt;
+ uint32_t in_head; /* next buffer dsp will write */
+ uint32_t in_tail; /* next buffer read() will read */
+ uint32_t in_count; /* number of buffers available to read() */
+
+ unsigned short samp_rate_index;
+ uint32_t device_events; /* device events interested in */
+ int abort; /* set when error, like sample rate mismatch */
+
+ /* audpre settings */
+ /* For different sample rate, the coeff might be different. *
+ * All the coeff should be passed from user space */
+
+ struct msm_audio_event_callbacks *ops;
+
+ int dir;
+ int opened;
+ int enabled;
+ int running;
+ int stopped; /* set when stopped, cleared on flush */
+ int eos_ack;
+ int mmap_flag;
+ int period;
+ struct audpp_cmd_cfg_object_params_volume vol_pan;
+};
+
+
+
+/* platform data */
+extern int alsa_dsp_send_buffer(struct msm_audio *prtd,
+ unsigned idx, unsigned len);
+extern int audio_dsp_out_enable(struct msm_audio *prtd, int yes);
+extern struct snd_soc_platform_driver msm_soc_platform;
+
+extern int audrec_encoder_config(struct msm_audio *prtd);
+extern int alsa_audrec_disable(struct msm_audio *prtd);
+extern int alsa_audio_configure(struct msm_audio *prtd);
+extern int alsa_audio_disable(struct msm_audio *prtd);
+extern int alsa_buffer_read(struct msm_audio *prtd, void __user *buf,
+ size_t count, loff_t *pos);
+ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf,
+ size_t count, loff_t *pos);
+extern struct msm_adsp_ops alsa_audrec_adsp_ops;
+extern int alsa_in_record_config(struct msm_audio *prtd, int enable);
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/msm7x30.c b/sound/soc/msm/msm7x30.c
new file mode 100644
index 0000000..94e37ca
--- /dev/null
+++ b/sound/soc/msm/msm7x30.c
@@ -0,0 +1,1004 @@
+/* Copyright (c) 2008-2011, Code Aurora Forum. All rights reserved.
+ *
+ * All source code in this file is licensed under the following license except
+ * where indicated.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio.h>
+
+#include "msm7kv2-pcm.h"
+#include <asm/mach-types.h>
+#include <mach/qdsp5v2/audio_dev_ctl.h>
+#include <mach/debug_mm.h>
+#include <mach/qdsp5v2/afe.h>
+
+static struct platform_device *msm_audio_snd_device;
+struct audio_locks the_locks;
+EXPORT_SYMBOL(the_locks);
+struct msm_volume msm_vol_ctl;
+EXPORT_SYMBOL(msm_vol_ctl);
+static struct snd_kcontrol_new snd_msm_controls[];
+
+char snddev_name[AUDIO_DEV_CTL_MAX_DEV][44];
+#define MSM_MAX_VOLUME 0x2000
+#define MSM_VOLUME_STEP ((MSM_MAX_VOLUME+17)/100) /* 17 added to avoid
+ more deviation */
+#define LOOPBACK_ENABLE 0x1
+#define LOOPBACK_DISABLE 0x0
+
+static int device_index; /* Count of Device controls */
+static int simple_control; /* Count of simple controls*/
+static int src_dev;
+static int dst_dev;
+static int loopback_status;
+
+
+static int msm_scontrol_count_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int msm_scontrol_count_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = simple_control;
+ return 0;
+}
+
+static int msm_v_call_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int msm_v_call_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_v_call_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int start = ucontrol->value.integer.value[0];
+ if (start)
+ broadcast_event(AUDDEV_EVT_START_VOICE, DEVICE_IGNORE,
+ SESSION_IGNORE);
+ else
+ broadcast_event(AUDDEV_EVT_END_VOICE, DEVICE_IGNORE,
+ SESSION_IGNORE);
+ return 0;
+}
+
+static int msm_v_mute_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 2;
+ return 0;
+}
+
+static int msm_v_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_v_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int dir = ucontrol->value.integer.value[0];
+ int mute = ucontrol->value.integer.value[1];
+ return msm_set_voice_mute(dir, mute);
+}
+
+static int msm_v_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2; /* Volume */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int msm_v_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_v_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int dir = ucontrol->value.integer.value[0];
+ int volume = ucontrol->value.integer.value[1];
+
+ return msm_set_voice_vol(dir, volume);
+}
+
+static int msm_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* Volume and 10-base multiply factor*/
+ uinfo->value.integer.min = 0;
+
+ /* limit the muliply factor to 4 decimal digit */
+ uinfo->value.integer.max = 1000000;
+ return 0;
+}
+static int msm_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int ret = 0;
+ int session_id = ucontrol->value.integer.value[0];
+ int volume = ucontrol->value.integer.value[1];
+ int factor = ucontrol->value.integer.value[2];
+ u32 session_mask = 0;
+
+
+ if (factor > 10000)
+ return -EINVAL;
+
+ if ((volume < 0) || (volume/factor > 100))
+ return -EINVAL;
+
+ volume = (MSM_VOLUME_STEP * volume);
+
+ /* Convert back to original decimal point by removing the 10-base factor
+ * and discard the fractional portion
+ */
+
+ volume = volume/factor;
+
+ if (volume > MSM_MAX_VOLUME)
+ volume = MSM_MAX_VOLUME;
+
+ /* Only Decoder volume control supported */
+ session_mask = (0x1 << (session_id) << (8 * ((int)AUDDEV_CLNT_DEC-1)));
+ msm_vol_ctl.volume = volume;
+ MM_DBG("session_id %d, volume %d", session_id, volume);
+ broadcast_event(AUDDEV_EVT_STREAM_VOL_CHG, DEVICE_IGNORE,
+ session_mask);
+
+ return ret;
+}
+
+static int msm_voice_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* Device */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_voice_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ uint32_t rx_dev_id;
+ uint32_t tx_dev_id;
+ struct msm_snddev_info *rx_dev_info;
+ struct msm_snddev_info *tx_dev_info;
+ int set = ucontrol->value.integer.value[2];
+ u32 session_mask;
+
+ if (!set)
+ return -EPERM;
+ /* Rx Device Routing */
+ rx_dev_id = ucontrol->value.integer.value[0];
+ rx_dev_info = audio_dev_ctrl_find_dev(rx_dev_id);
+
+ if (IS_ERR(rx_dev_info)) {
+ MM_ERR("pass invalid dev_id\n");
+ rc = PTR_ERR(rx_dev_info);
+ return rc;
+ }
+
+ if (!(rx_dev_info->capability & SNDDEV_CAP_RX)) {
+ MM_ERR("First Dev is supposed to be RX\n");
+ return -EFAULT;
+ }
+
+ MM_DBG("route cfg %d STREAM_VOICE_RX type\n",
+ rx_dev_id);
+
+ msm_set_voc_route(rx_dev_info, AUDIO_ROUTE_STREAM_VOICE_RX,
+ rx_dev_id);
+
+ session_mask = 0x1 << (8 * ((int)AUDDEV_CLNT_VOC-1));
+
+ broadcast_event(AUDDEV_EVT_DEV_CHG_VOICE, rx_dev_id, session_mask);
+
+
+ /* Tx Device Routing */
+ tx_dev_id = ucontrol->value.integer.value[1];
+ tx_dev_info = audio_dev_ctrl_find_dev(tx_dev_id);
+
+ if (IS_ERR(tx_dev_info)) {
+ MM_ERR("pass invalid dev_id\n");
+ rc = PTR_ERR(tx_dev_info);
+ return rc;
+ }
+
+ if (!(tx_dev_info->capability & SNDDEV_CAP_TX)) {
+ MM_ERR("Second Dev is supposed to be Tx\n");
+ return -EFAULT;
+ }
+
+ MM_DBG("route cfg %d %d type\n",
+ tx_dev_id, AUDIO_ROUTE_STREAM_VOICE_TX);
+
+ msm_set_voc_route(tx_dev_info, AUDIO_ROUTE_STREAM_VOICE_TX,
+ tx_dev_id);
+
+ broadcast_event(AUDDEV_EVT_DEV_CHG_VOICE, tx_dev_id, session_mask);
+
+ if (rx_dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RDY, rx_dev_id, session_mask);
+
+ if (tx_dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RDY, tx_dev_id, session_mask);
+
+ return rc;
+}
+
+static int msm_voice_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ /* TODO: query Device list */
+ return 0;
+}
+
+static int msm_device_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1; /* Device */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_device_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ int set = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+ struct msm_snddev_info *dst_dev_info;
+ struct msm_snddev_info *src_dev_info;
+ int tx_freq = 0;
+ int rx_freq = 0;
+ u32 set_freq = 0;
+
+ set = ucontrol->value.integer.value[0];
+ route_cfg.dev_id = ucontrol->id.numid - device_index;
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+ if (IS_ERR(dev_info)) {
+ MM_ERR("pass invalid dev_id\n");
+ rc = PTR_ERR(dev_info);
+ return rc;
+ }
+ MM_INFO("device %s set %d\n", dev_info->name, set);
+
+ if (set) {
+ if (!dev_info->opened) {
+ set_freq = dev_info->sample_rate;
+ if (!msm_device_is_voice(route_cfg.dev_id)) {
+ msm_get_voc_freq(&tx_freq, &rx_freq);
+ if (dev_info->capability & SNDDEV_CAP_TX)
+ set_freq = tx_freq;
+
+ if (set_freq == 0)
+ set_freq = dev_info->sample_rate;
+ } else
+ set_freq = dev_info->sample_rate;
+
+
+ MM_ERR("device freq =%d\n", set_freq);
+ rc = dev_info->dev_ops.set_freq(dev_info, set_freq);
+ if (rc < 0) {
+ MM_ERR("device freq failed!\n");
+ return rc;
+ }
+ dev_info->set_sample_rate = rc;
+ rc = 0;
+ rc = dev_info->dev_ops.open(dev_info);
+ if (rc < 0) {
+ MM_ERR("Enabling %s failed", dev_info->name);
+ return rc;
+ }
+ dev_info->opened = 1;
+ broadcast_event(AUDDEV_EVT_DEV_RDY, route_cfg.dev_id,
+ SESSION_IGNORE);
+ /* Event to notify client for device info */
+ broadcast_event(AUDDEV_EVT_DEVICE_INFO,
+ route_cfg.dev_id, SESSION_IGNORE);
+ if ((route_cfg.dev_id == src_dev) ||
+ (route_cfg.dev_id == dst_dev)) {
+ dst_dev_info = audio_dev_ctrl_find_dev(
+ dst_dev);
+ if (IS_ERR(dst_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(dst_dev_info);
+ return rc;
+ }
+ src_dev_info = audio_dev_ctrl_find_dev(
+ src_dev);
+ if (IS_ERR(src_dev_info)) {
+ pr_err("src_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(src_dev_info);
+ return rc;
+ }
+ if ((dst_dev_info->opened) &&
+ (src_dev_info->opened)) {
+ pr_debug("%d: Enable afe_loopback\n",
+ __LINE__);
+ afe_ext_loopback(LOOPBACK_ENABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ loopback_status = 1;
+ }
+ }
+ }
+ } else {
+ if (dev_info->opened) {
+ broadcast_event(AUDDEV_EVT_REL_PENDING,
+ route_cfg.dev_id,
+ SESSION_IGNORE);
+ rc = dev_info->dev_ops.close(dev_info);
+ if (rc < 0) {
+ MM_ERR("Snd device failed close!\n");
+ return rc;
+ } else {
+ dev_info->opened = 0;
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ SESSION_IGNORE);
+ }
+ if (loopback_status == 1) {
+ if ((route_cfg.dev_id == src_dev) ||
+ (route_cfg.dev_id == dst_dev)) {
+ dst_dev_info = audio_dev_ctrl_find_dev(
+ dst_dev);
+ if (IS_ERR(dst_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(dst_dev_info);
+ return rc;
+ }
+ src_dev_info = audio_dev_ctrl_find_dev(
+ src_dev);
+ if (IS_ERR(src_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(src_dev_info);
+ return rc;
+ }
+ pr_debug("%d: Disable afe_loopback\n",
+ __LINE__);
+ afe_ext_loopback(LOOPBACK_DISABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ loopback_status = 0;
+ }
+ }
+ }
+
+ }
+ return rc;
+}
+
+static int msm_device_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+
+ route_cfg.dev_id = ucontrol->id.numid - device_index;
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+
+ if (IS_ERR(dev_info)) {
+ MM_ERR("pass invalid dev_id\n");
+ rc = PTR_ERR(dev_info);
+ return rc;
+ }
+
+ ucontrol->value.integer.value[0] = dev_info->copp_id;
+ ucontrol->value.integer.value[1] = dev_info->capability;
+
+ return 0;
+}
+
+static int msm_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* Device */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_route_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ /* TODO: query Device list */
+ return 0;
+}
+
+static int msm_route_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ int enc_freq = 0;
+ int requested_freq = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+ int session_id = ucontrol->value.integer.value[0];
+ int set = ucontrol->value.integer.value[2];
+ u32 session_mask = 0;
+ route_cfg.dev_id = ucontrol->value.integer.value[1];
+
+ if (ucontrol->id.numid == 2)
+ route_cfg.stream_type = AUDIO_ROUTE_STREAM_PLAYBACK;
+ else
+ route_cfg.stream_type = AUDIO_ROUTE_STREAM_REC;
+
+ MM_DBG("route cfg %d %d type for popp %d set value %d\n",
+ route_cfg.dev_id, route_cfg.stream_type, session_id, set);
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+
+ if (IS_ERR(dev_info)) {
+ MM_ERR("pass invalid dev_id\n");
+ rc = PTR_ERR(dev_info);
+ return rc;
+ }
+ if (route_cfg.stream_type == AUDIO_ROUTE_STREAM_PLAYBACK) {
+ rc = msm_snddev_set_dec(session_id, dev_info->copp_id, set);
+ session_mask =
+ (0x1 << (session_id) << (8 * ((int)AUDDEV_CLNT_DEC-1)));
+ if (!set) {
+ if (dev_info->opened) {
+ broadcast_event(AUDDEV_EVT_REL_PENDING,
+ route_cfg.dev_id,
+ session_mask);
+
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ }
+ dev_info->sessions &= ~(session_mask);
+ } else {
+ dev_info->sessions = dev_info->sessions | session_mask;
+ if (dev_info->opened) {
+ broadcast_event(AUDDEV_EVT_DEV_RDY,
+ route_cfg.dev_id,
+ session_mask);
+ /* Event to notify client for device info */
+ broadcast_event(AUDDEV_EVT_DEVICE_INFO,
+ route_cfg.dev_id,
+ session_mask);
+ }
+ }
+ } else {
+ rc = msm_snddev_set_enc(session_id, dev_info->copp_id, set);
+ session_mask =
+ (0x1 << (session_id)) << (8 * ((int)AUDDEV_CLNT_ENC-1));
+ if (!set) {
+ if (dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ } else {
+ dev_info->sessions = dev_info->sessions | session_mask;
+ enc_freq = msm_snddev_get_enc_freq(session_id);
+ requested_freq = enc_freq;
+ if (enc_freq > 0) {
+ rc = msm_snddev_request_freq(&enc_freq,
+ session_id,
+ SNDDEV_CAP_TX,
+ AUDDEV_CLNT_ENC);
+ MM_DBG("sample rate configured %d"
+ "sample rate requested %d\n",
+ enc_freq, requested_freq);
+ if ((rc <= 0) || (enc_freq != requested_freq)) {
+ MM_DBG("msm_snddev_withdraw_freq\n");
+ rc = msm_snddev_withdraw_freq
+ (session_id,
+ SNDDEV_CAP_TX, AUDDEV_CLNT_ENC);
+ broadcast_event(AUDDEV_EVT_FREQ_CHG,
+ route_cfg.dev_id,
+ SESSION_IGNORE);
+ }
+ }
+ if (dev_info->opened) {
+ broadcast_event(AUDDEV_EVT_DEV_RDY,
+ route_cfg.dev_id,
+ session_mask);
+ /* Event to notify client for device info */
+ broadcast_event(AUDDEV_EVT_DEVICE_INFO,
+ route_cfg.dev_id,
+ session_mask);
+ }
+ }
+ }
+
+ if (rc < 0) {
+ MM_ERR("device could not be assigned!\n");
+ return -EFAULT;
+ }
+
+ return rc;
+}
+
+static int msm_device_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int msm_device_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct msm_snddev_info *dev_info;
+
+ int dev_id = ucontrol->value.integer.value[0];
+
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+ ucontrol->value.integer.value[0] = dev_info->dev_volume;
+
+ return 0;
+}
+
+static int msm_device_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = -EPERM;
+ struct msm_snddev_info *dev_info;
+
+ int dev_id = ucontrol->value.integer.value[0];
+ int volume = ucontrol->value.integer.value[1];
+
+ MM_DBG("dev_id = %d, volume = %d\n", dev_id, volume);
+
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+
+ if (IS_ERR(dev_info)) {
+ rc = PTR_ERR(dev_info);
+ MM_ERR("audio_dev_ctrl_find_dev failed. %ld\n",
+ PTR_ERR(dev_info));
+ return rc;
+ }
+
+ MM_DBG("dev_name = %s dev_id = %d, volume = %d\n",
+ dev_info->name, dev_id, volume);
+
+ if (dev_info->dev_ops.set_device_volume)
+ rc = dev_info->dev_ops.set_device_volume(dev_info, volume);
+ else {
+ MM_INFO("device %s does not support device volume "
+ "control.", dev_info->name);
+ return -EPERM;
+ }
+
+ return rc;
+}
+
+static int msm_reset_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0;
+ return 0;
+}
+
+static int msm_reset_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_reset_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ MM_DBG("Resetting all devices\n");
+ return msm_reset_all_device();
+}
+
+
+static int msm_dual_mic_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ /*Max value is decided based on MAX ENC sessions*/
+ uinfo->value.integer.max = MAX_AUDREC_SESSIONS - 1;
+ return 0;
+}
+
+static int msm_dual_mic_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int enc_session_id = ucontrol->value.integer.value[0];
+ ucontrol->value.integer.value[1] =
+ msm_get_dual_mic_config(enc_session_id);
+ MM_DBG("session id = %d, config = %ld\n", enc_session_id,
+ ucontrol->value.integer.value[1]);
+ return 0;
+}
+
+static int msm_dual_mic_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int enc_session_id = ucontrol->value.integer.value[0];
+ int dual_mic_config = ucontrol->value.integer.value[1];
+ MM_DBG("session id = %d, config = %d\n", enc_session_id,
+ dual_mic_config);
+ return msm_set_dual_mic_config(enc_session_id, dual_mic_config);
+}
+
+static int msm_device_mute_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_device_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return 0;
+}
+
+static int msm_device_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int dev_id = ucontrol->value.integer.value[0];
+ int mute = ucontrol->value.integer.value[1];
+ struct msm_snddev_info *dev_info;
+ int afe_dev_id = 0;
+ int volume = 0x4000;
+
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+ if (IS_ERR(dev_info)) {
+ MM_ERR("pass invalid dev_id %d\n", dev_id);
+ return PTR_ERR(dev_info);
+ }
+
+ if (dev_info->capability & SNDDEV_CAP_RX)
+ return -EPERM;
+
+ MM_DBG("Muting device id %d(%s)\n", dev_id, dev_info->name);
+
+ if (dev_info->copp_id == 0)
+ afe_dev_id = AFE_HW_PATH_CODEC_TX;
+ if (dev_info->copp_id == 1)
+ afe_dev_id = AFE_HW_PATH_AUXPCM_TX;
+ if (dev_info->copp_id == 2)
+ afe_dev_id = AFE_HW_PATH_MI2S_TX;
+ if (mute)
+ volume = 0;
+ afe_device_volume_ctrl(afe_dev_id, volume);
+ return 0;
+}
+
+static int msm_loopback_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_loopback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_loopback_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct msm_snddev_info *src_dev_info = NULL; /* TX device */
+ struct msm_snddev_info *dst_dev_info = NULL; /* RX device */
+ int dst_dev_id = ucontrol->value.integer.value[0];
+ int src_dev_id = ucontrol->value.integer.value[1];
+ int set = ucontrol->value.integer.value[2];
+
+ pr_debug("%s: set=%d\n", __func__, set);
+
+ dst_dev_info = audio_dev_ctrl_find_dev(dst_dev_id);
+ if (IS_ERR(dst_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(dst_dev_info);
+ return rc;
+ }
+ if (!(dst_dev_info->capability & SNDDEV_CAP_RX)) {
+ pr_err("Destination device %d is not RX device\n",
+ dst_dev_id);
+ return -EFAULT;
+ }
+
+ src_dev_info = audio_dev_ctrl_find_dev(src_dev_id);
+ if (IS_ERR(src_dev_info)) {
+ pr_err("src_dev:%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(src_dev_info);
+ return rc;
+ }
+ if (!(src_dev_info->capability & SNDDEV_CAP_TX)) {
+ pr_err("Source device %d is not TX device\n", src_dev_id);
+ return -EFAULT;
+ }
+
+ if (set) {
+ pr_debug("%s:%d:Enabling AFE_Loopback\n", __func__, __LINE__);
+ src_dev = src_dev_id;
+ dst_dev = dst_dev_id;
+ loopback_status = 1;
+ if ((dst_dev_info->opened) && (src_dev_info->opened))
+ afe_ext_loopback(LOOPBACK_ENABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ } else {
+ pr_debug("%s:%d:Disabling AFE_Loopback\n", __func__, __LINE__);
+ src_dev = DEVICE_IGNORE;
+ dst_dev = DEVICE_IGNORE;
+ loopback_status = 0;
+ afe_ext_loopback(LOOPBACK_DISABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ }
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_dev_controls[AUDIO_DEV_CTL_MAX_DEV];
+
+static int snd_dev_ctl_index(int idx)
+{
+ struct msm_snddev_info *dev_info;
+
+ dev_info = audio_dev_ctrl_find_dev(idx);
+ if (IS_ERR(dev_info)) {
+ MM_ERR("pass invalid dev_id\n");
+ return PTR_ERR(dev_info);
+ }
+ if (sizeof(dev_info->name) <= 44)
+ sprintf(&snddev_name[idx][0] , "%s", dev_info->name);
+
+ snd_dev_controls[idx].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ snd_dev_controls[idx].access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ snd_dev_controls[idx].name = &snddev_name[idx][0];
+ snd_dev_controls[idx].index = idx;
+ snd_dev_controls[idx].info = msm_device_info;
+ snd_dev_controls[idx].get = msm_device_get;
+ snd_dev_controls[idx].put = msm_device_put;
+ snd_dev_controls[idx].private_value = 0;
+ return 0;
+
+}
+
+#define MSM_EXT(xname, fp_info, fp_get, fp_put, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .name = xname, \
+ .info = fp_info,\
+ .get = fp_get, .put = fp_put, \
+ .private_value = addr, \
+}
+
+static struct snd_kcontrol_new snd_msm_controls[] = {
+ MSM_EXT("Count", msm_scontrol_count_info, msm_scontrol_count_get, \
+ NULL, 0),
+ MSM_EXT("Stream", msm_route_info, msm_route_get, \
+ msm_route_put, 0),
+ MSM_EXT("Record", msm_route_info, msm_route_get, \
+ msm_route_put, 0),
+ MSM_EXT("Voice", msm_voice_info, msm_voice_get, \
+ msm_voice_put, 0),
+ MSM_EXT("Volume", msm_volume_info, msm_volume_get, \
+ msm_volume_put, 0),
+ MSM_EXT("VoiceVolume", msm_v_volume_info, msm_v_volume_get, \
+ msm_v_volume_put, 0),
+ MSM_EXT("VoiceMute", msm_v_mute_info, msm_v_mute_get, \
+ msm_v_mute_put, 0),
+ MSM_EXT("Voice Call", msm_v_call_info, msm_v_call_get, \
+ msm_v_call_put, 0),
+ MSM_EXT("Device_Volume", msm_device_volume_info,
+ msm_device_volume_get, msm_device_volume_put, 0),
+ MSM_EXT("Reset", msm_reset_info,
+ msm_reset_get, msm_reset_put, 0),
+ MSM_EXT("DualMic Switch", msm_dual_mic_info,
+ msm_dual_mic_get, msm_dual_mic_put, 0),
+ MSM_EXT("Device_Mute", msm_device_mute_info,
+ msm_device_mute_get, msm_device_mute_put, 0),
+ MSM_EXT("Sound Device Loopback", msm_loopback_info,
+ msm_loopback_get, msm_loopback_put, 0),
+};
+
+static int msm_new_mixer(struct snd_soc_codec *codec)
+{
+ unsigned int idx;
+ int err;
+ int dev_cnt;
+
+ strcpy(codec->card->snd_card->mixername, "MSM Mixer");
+ for (idx = 0; idx < ARRAY_SIZE(snd_msm_controls); idx++) {
+ err = snd_ctl_add(codec->card->snd_card,
+ snd_ctl_new1(&snd_msm_controls[idx], NULL));
+ if (err < 0)
+ MM_ERR("ERR adding ctl\n");
+ }
+ dev_cnt = msm_snddev_devcount();
+
+ for (idx = 0; idx < dev_cnt; idx++) {
+ if (!snd_dev_ctl_index(idx)) {
+ err = snd_ctl_add(codec->card->snd_card,
+ snd_ctl_new1(&snd_dev_controls[idx], NULL));
+ if (err < 0)
+ MM_ERR("ERR adding ctl\n");
+ } else
+ return 0;
+ }
+ simple_control = ARRAY_SIZE(snd_msm_controls);
+ device_index = simple_control + 1;
+ return 0;
+}
+
+static int msm_soc_dai_init(
+ struct snd_soc_pcm_runtime *rtd)
+{
+ int ret = 0;
+ struct snd_soc_codec *codec = rtd->codec;
+ ret = msm_new_mixer(codec);
+ if (ret < 0)
+ MM_ERR("msm_soc: ALSA MSM Mixer Fail\n");
+
+ mutex_init(&the_locks.lock);
+ mutex_init(&the_locks.write_lock);
+ mutex_init(&the_locks.read_lock);
+ spin_lock_init(&the_locks.read_dsp_lock);
+ spin_lock_init(&the_locks.write_dsp_lock);
+ spin_lock_init(&the_locks.mixer_lock);
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+ src_dev = DEVICE_IGNORE;
+ dst_dev = DEVICE_IGNORE;
+
+ return ret;
+}
+
+static struct snd_soc_dai_link msm_dai[] = {
+{
+ .name = "MSM Primary I2S",
+ .stream_name = "DSP 1",
+ .cpu_dai_name = "msm-cpu-dai.0",
+ .platform_name = "msm-dsp-audio.0",
+ .codec_name = "msm-codec-dai.0",
+ .codec_dai_name = "msm-codec-dai",
+ .init = &msm_soc_dai_init,
+},
+#ifdef CONFIG_SND_MVS_SOC
+{
+ .name = "MSM Primary Voip",
+ .stream_name = "MVS",
+ .cpu_dai_name = "mvs-cpu-dai.0",
+ .platform_name = "msm-mvs-audio.0",
+ .codec_name = "mvs-codec-dai.0",
+ .codec_dai_name = "mvs-codec-dai",
+},
+#endif
+};
+
+static struct snd_soc_card snd_soc_card_msm = {
+ .name = "msm-audio",
+ .dai_link = msm_dai,
+ .num_links = ARRAY_SIZE(msm_dai),
+};
+
+static int __init msm_audio_init(void)
+{
+ int ret;
+
+ msm_audio_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!msm_audio_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(msm_audio_snd_device, &snd_soc_card_msm);
+ ret = platform_device_add(msm_audio_snd_device);
+ if (ret) {
+ platform_device_put(msm_audio_snd_device);
+ return ret;
+ }
+
+ return ret;
+}
+
+static void __exit msm_audio_exit(void)
+{
+ platform_device_unregister(msm_audio_snd_device);
+}
+
+module_init(msm_audio_init);
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("PCM module");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8660-apq-wm8903.c b/sound/soc/msm/msm8660-apq-wm8903.c
new file mode 100644
index 0000000..e697c3f
--- /dev/null
+++ b/sound/soc/msm/msm8660-apq-wm8903.c
@@ -0,0 +1,725 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/err.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pmic8058.h>
+#include <linux/mfd/pmic8901.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/delay.h>
+#include <mach/mpp.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc-dsp.h>
+#include <sound/pcm.h>
+#include <asm/mach-types.h>
+#include "msm-pcm-routing.h"
+#include "../codecs/wm8903.h"
+
+#define MSM_GPIO_CLASS_D0_EN 80
+#define MSM_GPIO_CLASS_D1_EN 81
+
+#define MSM_CDC_MIC_I2S_MCLK 108
+
+static int msm8660_spk_func;
+static int msm8660_headset_func;
+static int msm8660_headphone_func;
+
+static struct clk *mic_bit_clk;
+static struct clk *spkr_osr_clk;
+static struct clk *spkr_bit_clk;
+static struct clk *wm8903_mclk;
+
+static int rx_hw_param_status;
+static int tx_hw_param_status;
+/* Platform specific logic */
+
+enum {
+ GET_ERR,
+ SET_ERR,
+ ENABLE_ERR,
+ NONE
+};
+
+enum {
+ FUNC_OFF,
+ FUNC_ON,
+};
+
+static struct wm8903_vdd {
+ struct regulator *reg_id;
+ const char *name;
+ u32 voltage;
+} wm8903_vdds[] = {
+ { NULL, "8058_l16", 1800000 },
+ { NULL, "8058_l0", 1200000 },
+ { NULL, "8058_s3", 1800000 },
+};
+
+static void classd_amp_pwr(int enable)
+{
+ int rc;
+
+ pr_debug("%s, enable = %d\n", __func__, enable);
+ if (enable) {
+ /* currently external PA isn't used for LINEOUTL */
+ rc = gpio_request(MSM_GPIO_CLASS_D0_EN, "CLASSD0_EN");
+ if (rc) {
+ pr_err("%s: spkr PA gpio %d request failed\n",
+ __func__, MSM_GPIO_CLASS_D0_EN);
+ return;
+ }
+ gpio_direction_output(MSM_GPIO_CLASS_D0_EN, 1);
+ gpio_set_value_cansleep(MSM_GPIO_CLASS_D0_EN, 1);
+ rc = gpio_request(MSM_GPIO_CLASS_D1_EN, "CLASSD1_EN");
+ if (rc) {
+ pr_err("%s: spkr PA gpio %d request failed\n",
+ __func__, MSM_GPIO_CLASS_D1_EN);
+ return;
+ }
+ gpio_direction_output(MSM_GPIO_CLASS_D1_EN, 1);
+ gpio_set_value_cansleep(MSM_GPIO_CLASS_D1_EN, 1);
+ } else {
+ gpio_set_value_cansleep(MSM_GPIO_CLASS_D0_EN, 0);
+ gpio_free(MSM_GPIO_CLASS_D0_EN);
+
+ gpio_set_value_cansleep(MSM_GPIO_CLASS_D1_EN, 0);
+ gpio_free(MSM_GPIO_CLASS_D1_EN);
+ }
+}
+
+static void extern_poweramp_on(void)
+{
+ pr_debug("%s: enable stereo spkr amp\n", __func__);
+ classd_amp_pwr(1);
+}
+
+static void extern_poweramp_off(void)
+{
+ pr_debug("%s: disable stereo spkr amp\n", __func__);
+ classd_amp_pwr(0);
+}
+
+static int msm8660_wm8903_powerup(void)
+{
+ int rc = 0, index, stage = NONE;
+ struct wm8903_vdd *vdd = NULL;
+
+ for (index = 0; index < ARRAY_SIZE(wm8903_vdds); index++) {
+ vdd = &wm8903_vdds[index];
+ vdd->reg_id = regulator_get(NULL, vdd->name);
+ if (IS_ERR(vdd->reg_id)) {
+ pr_err("%s: Unable to get %s\n", __func__, vdd->name);
+ stage = GET_ERR;
+ rc = -ENODEV;
+ break;
+ }
+
+ rc = regulator_set_voltage(vdd->reg_id,
+ vdd->voltage, vdd->voltage);
+ if (rc) {
+ pr_err("%s: unable to set %s voltage to %dV\n",
+ __func__, vdd->name, vdd->voltage);
+ stage = SET_ERR;
+ break;
+ }
+
+ rc = regulator_enable(vdd->reg_id);
+ if (rc) {
+ pr_err("%s:failed to enable %s\n", __func__, vdd->name);
+ stage = ENABLE_ERR;
+ break;
+ }
+ }
+
+ if (index != ARRAY_SIZE(wm8903_vdds)) {
+ if (stage != GET_ERR) {
+ vdd = &wm8903_vdds[index];
+ regulator_put(vdd->reg_id);
+ vdd->reg_id = NULL;
+ }
+
+ while (index--) {
+ vdd = &wm8903_vdds[index];
+ regulator_disable(vdd->reg_id);
+ regulator_put(vdd->reg_id);
+ vdd->reg_id = NULL;
+ }
+ }
+
+ return rc;
+}
+
+static void msm8660_wm8903_powerdown(void)
+{
+ int index = ARRAY_SIZE(wm8903_vdds);
+ struct wm8903_vdd *vdd = NULL;
+
+ while (index--) {
+ vdd = &wm8903_vdds[index];
+ if (vdd->reg_id) {
+ regulator_disable(vdd->reg_id);
+ regulator_put(vdd->reg_id);
+ }
+ }
+}
+
+static int msm8660_wm8903_enable_mclk(int enable)
+{
+ int ret = 0;
+
+ if (enable) {
+ ret = gpio_request(MSM_CDC_MIC_I2S_MCLK, "I2S_Clock");
+ if (ret != 0) {
+ pr_err("%s: failed to request GPIO\n", __func__);
+ return ret;
+ }
+
+ wm8903_mclk = clk_get_sys(NULL, "i2s_mic_osr_clk");
+ if (IS_ERR(wm8903_mclk)) {
+ pr_err("Failed to get i2s_mic_osr_clk\n");
+ gpio_free(MSM_CDC_MIC_I2S_MCLK);
+ return IS_ERR(wm8903_mclk);
+ }
+ /* Master clock OSR 256 */
+ clk_set_rate(wm8903_mclk, 48000 * 256);
+ ret = clk_prepare_enable(wm8903_mclk);
+ if (ret != 0) {
+ pr_err("Unable to enable i2s_mic_osr_clk\n");
+ gpio_free(MSM_CDC_MIC_I2S_MCLK);
+ clk_put(wm8903_mclk);
+ return ret;
+ }
+ } else {
+ if (wm8903_mclk) {
+ clk_disable_unprepare(wm8903_mclk);
+ clk_put(wm8903_mclk);
+ gpio_free(MSM_CDC_MIC_I2S_MCLK);
+ wm8903_mclk = NULL;
+ }
+ }
+
+ return ret;
+}
+
+static int msm8660_wm8903_prepare(void)
+{
+ int ret = 0;
+
+ ret = msm8660_wm8903_powerup();
+ if (ret) {
+ pr_err("Unable to powerup wm8903\n");
+ return ret;
+ }
+
+ ret = msm8660_wm8903_enable_mclk(1);
+ if (ret) {
+ pr_err("Unable to enable mclk to wm8903\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static void msm8660_wm8903_unprepare(void)
+{
+ msm8660_wm8903_powerdown();
+ msm8660_wm8903_enable_mclk(0);
+}
+
+static int msm8660_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int rate = params_rate(params), ret = 0;
+
+ pr_debug("Enter %s rate = %d\n", __func__, rate);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (rx_hw_param_status)
+ return 0;
+ /* wm8903 run @ LRC*256 */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, rate * 256,
+ SND_SOC_CLOCK_IN);
+ snd_soc_dai_digital_mute(codec_dai, 0);
+ if (ret < 0) {
+ pr_err("can't set rx codec clk configuration\n");
+ return ret;
+ }
+ clk_set_rate(wm8903_mclk, rate * 256);
+ /* set as slave mode CPU */
+ clk_set_rate(spkr_bit_clk, 0);
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM);
+ rx_hw_param_status++;
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (tx_hw_param_status)
+ return 0;
+ clk_set_rate(wm8903_mclk, rate * 256);
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, rate * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ pr_err("can't set tx codec clk configuration\n");
+ return ret;
+ }
+ clk_set_rate(mic_bit_clk, 0);
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM);
+ tx_hw_param_status++;
+ }
+ return 0;
+}
+
+static int msm8660_i2s_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("Enter %s\n", __func__);
+ /* ON Dragonboard, I2S between wm8903 and CPU is shared by
+ * CODEC_SPEAKER and CODEC_MIC therefore CPU only can operate
+ * as input SLAVE mode.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* config WM8903 in Mater mode */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_I2S);
+ if (ret != 0) {
+ pr_err("codec_dai set_fmt error\n");
+ return ret;
+ }
+ /* config CPU in SLAVE mode */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM);
+ if (ret != 0) {
+ pr_err("cpu_dai set_fmt error\n");
+ return ret;
+ }
+ spkr_osr_clk = clk_get_sys(NULL, "i2s_spkr_osr_clk");
+ if (IS_ERR(spkr_osr_clk)) {
+ pr_err("Failed to get i2s_spkr_osr_clk\n");
+ return PTR_ERR(spkr_osr_clk);
+ }
+ clk_set_rate(spkr_osr_clk, 48000 * 256);
+ ret = clk_prepare_enable(spkr_osr_clk);
+ if (ret != 0) {
+ pr_err("Unable to enable i2s_spkr_osr_clk\n");
+ clk_put(spkr_osr_clk);
+ return ret;
+ }
+ spkr_bit_clk = clk_get_sys(NULL, "i2s_spkr_bit_clk");
+ if (IS_ERR(spkr_bit_clk)) {
+ pr_err("Failed to get i2s_spkr_bit_clk\n");
+ clk_disable_unprepare(spkr_osr_clk);
+ clk_put(spkr_osr_clk);
+ return PTR_ERR(spkr_bit_clk);
+ }
+ clk_set_rate(spkr_bit_clk, 0);
+ ret = clk_prepare_enable(spkr_bit_clk);
+ if (ret != 0) {
+ pr_err("Unable to enable i2s_spkr_bit_clk\n");
+ clk_disable_unprepare(spkr_osr_clk);
+ clk_put(spkr_osr_clk);
+ clk_put(spkr_bit_clk);
+ return ret;
+ }
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ /* config WM8903 in Mater mode */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_I2S);
+ if (ret != 0) {
+ pr_err("codec_dai set_fmt error\n");
+ return ret;
+ }
+ /* config CPU in SLAVE mode */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM);
+ if (ret != 0) {
+ pr_err("codec_dai set_fmt error\n");
+ return ret;
+ }
+
+ mic_bit_clk = clk_get_sys(NULL, "i2s_mic_bit_clk");
+ if (IS_ERR(mic_bit_clk)) {
+ pr_err("Failed to get i2s_mic_bit_clk\n");
+ return PTR_ERR(mic_bit_clk);
+ }
+ clk_set_rate(mic_bit_clk, 0);
+ ret = clk_prepare_enable(mic_bit_clk);
+ if (ret != 0) {
+ pr_err("Unable to enable i2s_mic_bit_clk\n");
+ clk_put(mic_bit_clk);
+ return ret;
+ }
+ }
+ return ret;
+}
+
+static void msm8660_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("Enter %s\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ tx_hw_param_status = 0;
+ rx_hw_param_status = 0;
+ if (spkr_bit_clk) {
+ clk_disable_unprepare(spkr_bit_clk);
+ clk_put(spkr_bit_clk);
+ spkr_bit_clk = NULL;
+ }
+ if (spkr_osr_clk) {
+ clk_disable_unprepare(spkr_osr_clk);
+ clk_put(spkr_osr_clk);
+ spkr_osr_clk = NULL;
+ }
+ if (mic_bit_clk) {
+ clk_disable_unprepare(mic_bit_clk);
+ clk_put(mic_bit_clk);
+ mic_bit_clk = NULL;
+ }
+ }
+}
+
+static void msm8660_ext_control(struct snd_soc_codec *codec)
+{
+ /* set the enpoints to their new connetion states */
+ if (msm8660_spk_func == FUNC_ON)
+ snd_soc_dapm_enable_pin(&codec->dapm, "Ext Spk");
+ else
+ snd_soc_dapm_disable_pin(&codec->dapm, "Ext Spk");
+
+ /* set the enpoints to their new connetion states */
+ if (msm8660_headset_func == FUNC_ON)
+ snd_soc_dapm_enable_pin(&codec->dapm, "Headset Jack");
+ else
+ snd_soc_dapm_disable_pin(&codec->dapm, "Headset Jack");
+
+ /* set the enpoints to their new connetion states */
+ if (msm8660_headphone_func == FUNC_ON)
+ snd_soc_dapm_enable_pin(&codec->dapm, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin(&codec->dapm, "Headphone Jack");
+
+ /* signal a DAPM event */
+ snd_soc_dapm_sync(&codec->dapm);
+}
+
+static int msm8660_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm8660_spk_func;
+ return 0;
+}
+
+static int msm8660_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm8660_spk_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm8660_spk_func = ucontrol->value.integer.value[0];
+ msm8660_ext_control(codec);
+ return 1;
+}
+
+static int msm8660_get_hs(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm8660_headset_func;
+ return 0;
+}
+
+static int msm8660_set_hs(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm8660_headset_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm8660_headset_func = ucontrol->value.integer.value[0];
+ msm8660_ext_control(codec);
+ return 1;
+}
+
+static int msm8660_get_hph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm8660_headphone_func;
+ return 0;
+}
+
+static int msm8660_set_hph(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm8660_headphone_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm8660_headphone_func = ucontrol->value.integer.value[0];
+ msm8660_ext_control(codec);
+ return 1;
+}
+
+static int msm8660_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
+ extern_poweramp_on();
+ else
+ extern_poweramp_off();
+ return 0;
+}
+
+static struct snd_soc_ops machine_ops = {
+ .startup = msm8660_i2s_startup,
+ .shutdown = msm8660_i2s_shutdown,
+ .hw_params = msm8660_i2s_hw_params,
+};
+
+static const struct snd_soc_dapm_widget msm8660_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", msm8660_spkramp_event),
+ SND_SOC_DAPM_MIC("Headset Jack", NULL),
+ SND_SOC_DAPM_MIC("Headphone Jack", NULL),
+ /* to fix a bug in wm8903.c, where audio doesn't function
+ * after suspend/resume
+ */
+ SND_SOC_DAPM_SUPPLY("CLK_SYS_ENA", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Match with wm8903 codec line out pin */
+ {"Ext Spk", NULL, "LINEOUTL"},
+ {"Ext Spk", NULL, "LINEOUTR"},
+ /* Headset connects to IN3L with Bias */
+ {"IN3L", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Jack"},
+ /* Headphone connects to IN3R with Bias */
+ {"IN3R", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headphone Jack"},
+ {"ADCL", NULL, "CLK_SYS_ENA"},
+ {"ADCR", NULL, "CLK_SYS_ENA"},
+ {"DACL", NULL, "CLK_SYS_ENA"},
+ {"DACR", NULL, "CLK_SYS_ENA"},
+};
+
+static const char *cmn_status[] = {"Off", "On"};
+static const struct soc_enum msm8660_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, cmn_status),
+};
+
+static const struct snd_kcontrol_new wm8903_msm8660_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", msm8660_enum[0], msm8660_get_spk,
+ msm8660_set_spk),
+ SOC_ENUM_EXT("Headset Function", msm8660_enum[0], msm8660_get_hs,
+ msm8660_set_hs),
+ SOC_ENUM_EXT("Headphone Function", msm8660_enum[0], msm8660_get_hph,
+ msm8660_set_hph),
+};
+
+static int msm8660_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int err;
+
+ snd_soc_dapm_disable_pin(&codec->dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin(&codec->dapm, "CLK_SYS_ENA");
+
+ err = snd_soc_add_controls(codec, wm8903_msm8660_controls,
+ ARRAY_SIZE(wm8903_msm8660_controls));
+ if (err < 0)
+ return err;
+
+ snd_soc_dapm_new_controls(&codec->dapm, msm8660_dapm_widgets,
+ ARRAY_SIZE(msm8660_dapm_widgets));
+
+ snd_soc_dapm_add_routes(&codec->dapm, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_sync(&codec->dapm);
+
+ return 0;
+}
+
+static int pri_i2s_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ rate->min = rate->max = 48000;
+ return 0;
+}
+/*
+ * LPA Needs only RX BE DAI links.
+ * Hence define seperate BE list for lpa
+ */
+static const char *lpa_mm_be[] = {
+ LPASS_BE_PRI_I2S_RX,
+};
+
+static struct snd_soc_dsp_link lpa_fe_media = {
+ .supported_be = lpa_mm_be,
+ .num_be = ARRAY_SIZE(lpa_mm_be),
+ .fe_playback_channels = 2,
+ .fe_capture_channels = 1,
+ .trigger = {
+ SND_SOC_DSP_TRIGGER_POST,
+ SND_SOC_DSP_TRIGGER_POST
+ },
+};
+
+static const char *mm1_be[] = {
+ LPASS_BE_PRI_I2S_RX,
+ LPASS_BE_PRI_I2S_TX,
+ LPASS_BE_HDMI,
+};
+
+static struct snd_soc_dsp_link fe_media = {
+ .supported_be = mm1_be,
+ .num_be = ARRAY_SIZE(mm1_be),
+ .fe_playback_channels = 2,
+ .fe_capture_channels = 1,
+ .trigger = {
+ SND_SOC_DSP_TRIGGER_POST, SND_SOC_DSP_TRIGGER_POST},
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm8660_dai[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MSM8660 Media",
+ .stream_name = "MultiMedia",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .dsp_link = &fe_media,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MSM8660 Media2",
+ .stream_name = "MultiMedia2",
+ .cpu_dai_name = "MultiMedia2",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .dsp_link = &fe_media,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+ /* Backend DAI Links */
+ {
+ .name = LPASS_BE_PRI_I2S_RX,
+ .stream_name = "Primary I2S Playback",
+ .cpu_dai_name = "msm-dai-q6.0",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "wm8903-codec.3-001a",
+ .codec_dai_name = "wm8903-hifi",
+ .no_pcm = 1,
+ .be_hw_params_fixup = pri_i2s_be_hw_params_fixup,
+ .ops = &machine_ops,
+ .init = &msm8660_audrx_init,
+ .be_id = MSM_BACKEND_DAI_PRI_I2S_RX
+ },
+ {
+ .name = LPASS_BE_PRI_I2S_TX,
+ .stream_name = "Primary I2S Capture",
+ .cpu_dai_name = "msm-dai-q6.1",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "wm8903-codec.3-001a",
+ .codec_dai_name = "wm8903-hifi",
+ .no_pcm = 1,
+ .ops = &machine_ops,
+ .be_hw_params_fixup = pri_i2s_be_hw_params_fixup,
+ .be_id = MSM_BACKEND_DAI_PRI_I2S_TX
+ },
+ /* LPA frontend DAI link*/
+ {
+ .name = "MSM8660 LPA",
+ .stream_name = "LPA",
+ .cpu_dai_name = "MultiMedia3",
+ .platform_name = "msm-pcm-lpa",
+ .dynamic = 1,
+ .dsp_link = &lpa_fe_media,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
+ },
+ /* HDMI backend DAI link */
+ {
+ .name = LPASS_BE_HDMI,
+ .stream_name = "HDMI Playback",
+ .cpu_dai_name = "msm-dai-q6.8",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_codec = 1,
+ .no_pcm = 1,
+ .be_hw_params_fixup = pri_i2s_be_hw_params_fixup,
+ .be_id = MSM_BACKEND_DAI_HDMI_RX
+ },
+};
+
+struct snd_soc_card snd_soc_card_msm8660 = {
+ .name = "msm8660-snd-card",
+ .dai_link = msm8660_dai,
+ .num_links = ARRAY_SIZE(msm8660_dai),
+};
+
+static struct platform_device *msm_snd_device;
+
+static int __init msm_audio_init(void)
+{
+ int ret = 0;
+
+ if (machine_is_msm8x60_dragon()) {
+ /* wm8903 audio codec needs to power up and mclk existing
+ before it's probed */
+ ret = msm8660_wm8903_prepare();
+ if (ret) {
+ pr_err("failed to prepare wm8903 audio codec\n");
+ return ret;
+ }
+
+ msm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ msm8660_wm8903_unprepare();
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(msm_snd_device, &snd_soc_card_msm8660);
+ ret = platform_device_add(msm_snd_device);
+ if (ret) {
+ platform_device_put(msm_snd_device);
+ msm8660_wm8903_unprepare();
+ return ret;
+ }
+ }
+ return ret;
+
+}
+module_init(msm_audio_init);
+
+static void __exit msm_audio_exit(void)
+{
+ msm8660_wm8903_unprepare();
+ platform_device_unregister(msm_snd_device);
+}
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC MSM8660");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8660.c b/sound/soc/msm/msm8660.c
new file mode 100644
index 0000000..4cbfd45
--- /dev/null
+++ b/sound/soc/msm/msm8660.c
@@ -0,0 +1,342 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/clk.h>
+#include <linux/err.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pmic8058.h>
+#include <linux/mfd/pmic8901.h>
+#include <linux/platform_device.h>
+#include <mach/board.h>
+#include <mach/mpp.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/dai.h>
+#include "msm8660-pcm.h"
+#include "../codecs/timpani.h"
+
+#define PM8058_GPIO_BASE NR_MSM_GPIOS
+#define PM8901_GPIO_BASE (PM8058_GPIO_BASE + \
+ PM8058_GPIOS + PM8058_MPPS)
+#define PM8901_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio + PM8901_GPIO_BASE)
+#define GPIO_EXPANDER_GPIO_BASE \
+ (PM8901_GPIO_BASE + PM8901_MPPS)
+
+static struct clk *rx_osr_clk;
+static struct clk *rx_bit_clk;
+static struct clk *tx_osr_clk;
+static struct clk *tx_bit_clk;
+
+static int rx_hw_param_status;
+static int tx_hw_param_status;
+/* Platform specific logic */
+
+static int timpani_rx_route_enable(void)
+{
+ int ret = 0;
+ pr_debug("%s\n", __func__);
+ ret = gpio_request(109, "I2S_Clock");
+ if (ret != 0) {
+ pr_err("%s: I2s clk gpio 109 request"
+ "failed\n", __func__);
+ return ret;
+ }
+ return ret;
+}
+
+static int timpani_rx_route_disable(void)
+{
+ int ret = 0;
+ pr_debug("%s\n", __func__);
+ gpio_free(109);
+ return ret;
+}
+
+
+#define GPIO_CLASS_D1_EN (GPIO_EXPANDER_GPIO_BASE + 0)
+#define PM8901_MPP_3 (2) /* PM8901 MPP starts from 0 */
+static void config_class_d1_gpio(int enable)
+{
+ int rc;
+
+ if (enable) {
+ rc = gpio_request(GPIO_CLASS_D1_EN, "CLASSD1_EN");
+ if (rc) {
+ pr_err("%s: spkr pamp gpio %d request"
+ "failed\n", __func__, GPIO_CLASS_D1_EN);
+ return;
+ }
+ gpio_direction_output(GPIO_CLASS_D1_EN, 1);
+ gpio_set_value_cansleep(GPIO_CLASS_D1_EN, 1);
+ } else {
+ gpio_set_value_cansleep(GPIO_CLASS_D1_EN, 0);
+ gpio_free(GPIO_CLASS_D1_EN);
+ }
+}
+
+static void config_class_d0_gpio(int enable)
+{
+ int rc;
+
+ if (enable) {
+ rc = pm8901_mpp_config_digital_out(PM8901_MPP_3,
+ PM8901_MPP_DIG_LEVEL_MSMIO, 1);
+
+ if (rc) {
+ pr_err("%s: CLASS_D0_EN failed\n", __func__);
+ return;
+ }
+
+ rc = gpio_request(PM8901_GPIO_PM_TO_SYS(PM8901_MPP_3),
+ "CLASSD0_EN");
+
+ if (rc) {
+ pr_err("%s: spkr pamp gpio pm8901 mpp3 request"
+ "failed\n", __func__);
+ pm8901_mpp_config_digital_out(PM8901_MPP_3,
+ PM8901_MPP_DIG_LEVEL_MSMIO, 0);
+ return;
+ }
+
+ gpio_direction_output(PM8901_GPIO_PM_TO_SYS(PM8901_MPP_3), 1);
+ gpio_set_value_cansleep(PM8901_GPIO_PM_TO_SYS(PM8901_MPP_3), 1);
+
+ } else {
+ pm8901_mpp_config_digital_out(PM8901_MPP_3,
+ PM8901_MPP_DIG_LEVEL_MSMIO, 0);
+ gpio_set_value_cansleep(PM8901_GPIO_PM_TO_SYS(PM8901_MPP_3), 0);
+ gpio_free(PM8901_GPIO_PM_TO_SYS(PM8901_MPP_3));
+ }
+}
+
+static void timpani_poweramp_on(void)
+{
+
+ pr_debug("%s: enable stereo spkr amp\n", __func__);
+ timpani_rx_route_enable();
+ config_class_d0_gpio(1);
+ config_class_d1_gpio(1);
+}
+
+static void timpani_poweramp_off(void)
+{
+
+ pr_debug("%s: disable stereo spkr amp\n", __func__);
+ timpani_rx_route_disable();
+ config_class_d0_gpio(0);
+ config_class_d1_gpio(0);
+}
+
+static int msm8660_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ int rate = params_rate(params);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (rx_hw_param_status)
+ return 0;
+ clk_set_rate(rx_osr_clk, rate * 256);
+ rx_hw_param_status++;
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (tx_hw_param_status)
+ return 0;
+ clk_set_rate(tx_osr_clk, rate * 256);
+ tx_hw_param_status++;
+ }
+ return 0;
+}
+
+static int msm8660_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ rx_osr_clk = clk_get(NULL, "i2s_spkr_osr_clk");
+ if (IS_ERR(rx_osr_clk)) {
+ pr_debug("Failed to get i2s_spkr_osr_clk\n");
+ return PTR_ERR(rx_osr_clk);
+ }
+ /* Master clock OSR 256 */
+ /* Initially set to Lowest sample rate Needed */
+ clk_set_rate(rx_osr_clk, 8000 * 256);
+ ret = clk_prepare_enable(rx_osr_clk);
+ if (ret != 0) {
+ pr_debug("Unable to enable i2s_spkr_osr_clk\n");
+ clk_put(rx_osr_clk);
+ return ret;
+ }
+ rx_bit_clk = clk_get(NULL, "i2s_spkr_bit_clk");
+ if (IS_ERR(rx_bit_clk)) {
+ pr_debug("Failed to get i2s_spkr_bit_clk\n");
+ clk_disable_unprepare(rx_osr_clk);
+ clk_put(rx_osr_clk);
+ return PTR_ERR(rx_bit_clk);
+ }
+ clk_set_rate(rx_bit_clk, 8);
+ ret = clk_prepare_enable(rx_bit_clk);
+ if (ret != 0) {
+ pr_debug("Unable to enable i2s_spkr_bit_clk\n");
+ clk_put(rx_bit_clk);
+ clk_disable_unprepare(rx_osr_clk);
+ clk_put(rx_osr_clk);
+ return ret;
+ }
+ timpani_poweramp_on();
+ msleep(30);
+ /* End of platform specific logic */
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ tx_osr_clk = clk_get(NULL, "i2s_mic_osr_clk");
+ if (IS_ERR(tx_osr_clk)) {
+ pr_debug("Failed to get i2s_mic_osr_clk\n");
+ return PTR_ERR(tx_osr_clk);
+ }
+ /* Master clock OSR 256 */
+ clk_set_rate(tx_osr_clk, 8000 * 256);
+ ret = clk_prepare_enable(tx_osr_clk);
+ if (ret != 0) {
+ pr_debug("Unable to enable i2s_mic_osr_clk\n");
+ clk_put(tx_osr_clk);
+ return ret;
+ }
+ tx_bit_clk = clk_get(NULL, "i2s_mic_bit_clk");
+ if (IS_ERR(tx_bit_clk)) {
+ pr_debug("Failed to get i2s_mic_bit_clk\n");
+ clk_disable_unprepare(tx_osr_clk);
+ clk_put(tx_osr_clk);
+ return PTR_ERR(tx_bit_clk);
+ }
+ clk_set_rate(tx_bit_clk, 8);
+ ret = clk_prepare_enable(tx_bit_clk);
+ if (ret != 0) {
+ pr_debug("Unable to enable i2s_mic_bit_clk\n");
+ clk_put(tx_bit_clk);
+ clk_disable_unprepare(tx_osr_clk);
+ clk_put(tx_osr_clk);
+ return ret;
+ }
+ msm_snddev_enable_dmic_power();
+ msleep(30);
+ }
+ return ret;
+}
+
+/*
+ * TODO: rx/tx_hw_param_status should be a counter in the below code
+ * when driver starts supporting mutisession else setting it to 0
+ * will stop audio in all sessions.
+ */
+static void msm8660_shutdown(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ rx_hw_param_status = 0;
+ timpani_poweramp_off();
+ msleep(30);
+ if (rx_bit_clk) {
+ clk_disable_unprepare(rx_bit_clk);
+ clk_put(rx_bit_clk);
+ rx_bit_clk = NULL;
+ }
+ if (rx_osr_clk) {
+ clk_disable_unprepare(rx_osr_clk);
+ clk_put(rx_osr_clk);
+ rx_osr_clk = NULL;
+ }
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ tx_hw_param_status = 0;
+ msm_snddev_disable_dmic_power();
+ msleep(30);
+ if (tx_bit_clk) {
+ clk_disable_unprepare(tx_bit_clk);
+ clk_put(tx_bit_clk);
+ tx_bit_clk = NULL;
+ }
+ if (tx_osr_clk) {
+ clk_disable_unprepare(tx_osr_clk);
+ clk_put(tx_osr_clk);
+ tx_osr_clk = NULL;
+ }
+ }
+}
+
+static struct snd_soc_ops machine_ops = {
+ .startup = msm8660_startup,
+ .shutdown = msm8660_shutdown,
+ .hw_params = msm8660_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm8660_dai[] = {
+ {
+ .name = "Audio Rx",
+ .stream_name = "Audio Rx",
+ .cpu_dai = &msm_cpu_dai[0],
+ .codec_dai = &timpani_codec_dai[0],
+ .ops = &machine_ops,
+ },
+ {
+ .name = "Audio Tx",
+ .stream_name = "Audio Tx",
+ .cpu_dai = &msm_cpu_dai[5],
+ .codec_dai = &timpani_codec_dai[1],
+ .ops = &machine_ops,
+ }
+};
+
+struct snd_soc_card snd_soc_card_msm8660 = {
+ .name = "msm8660-pcm-audio",
+ .dai_link = msm8660_dai,
+ .num_links = ARRAY_SIZE(msm8660_dai),
+ .platform = &msm8660_soc_platform,
+};
+
+/* msm_audio audio subsystem */
+static struct snd_soc_device msm_snd_devdata = {
+ .card = &snd_soc_card_msm8660,
+ .codec_dev = &soc_codec_dev_timpani,
+};
+
+static struct platform_device *msm_snd_device;
+
+
+static int __init msm_audio_init(void)
+{
+ int ret;
+
+ msm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(msm_snd_device, &msm_snd_devdata);
+
+ msm_snd_devdata.dev = &msm_snd_device->dev;
+ ret = platform_device_add(msm_snd_device);
+ if (ret) {
+ platform_device_put(msm_snd_device);
+ return ret;
+ }
+
+ return ret;
+}
+module_init(msm_audio_init);
+
+static void __exit msm_audio_exit(void)
+{
+ platform_device_unregister(msm_snd_device);
+}
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC MSM8660");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8930.c b/sound/soc/msm/msm8930.c
new file mode 100644
index 0000000..ad1278c
--- /dev/null
+++ b/sound/soc/msm/msm8930.c
@@ -0,0 +1,1146 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/spk.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <mach/socinfo.h>
+#include "msm-pcm-routing.h"
+#include "../codecs/wcd9304.h"
+
+/* 8930 machine driver */
+
+#define MSM8930_SPK_ON 1
+#define MSM8930_SPK_OFF 0
+
+#define BTSCO_RATE_8KHZ 8000
+#define BTSCO_RATE_16KHZ 16000
+
+#define SPK_AMP_POS 0x1
+#define SPK_AMP_NEG 0x2
+#define SPKR_BOOST_GPIO 15
+#define DEFAULT_PMIC_SPK_GAIN 0x0D
+#define SITAR_EXT_CLK_RATE 12288000
+
+#define SITAR_MBHC_DEF_BUTTONS 8
+#define SITAR_MBHC_DEF_RLOADS 5
+
+static int msm8930_spk_control;
+static int msm8930_slim_0_rx_ch = 1;
+static int msm8930_slim_0_tx_ch = 1;
+static int msm8930_pmic_spk_gain = DEFAULT_PMIC_SPK_GAIN;
+
+static int msm8930_ext_spk_pamp;
+static int msm8930_btsco_rate = BTSCO_RATE_8KHZ;
+static int msm8930_btsco_ch = 1;
+
+static struct clk *codec_clk;
+static int clk_users;
+
+static int msm8930_headset_gpios_configured;
+
+static struct snd_soc_jack hs_jack;
+static struct snd_soc_jack button_jack;
+
+static int msm8930_enable_codec_ext_clk(
+ struct snd_soc_codec *codec, int enable,
+ bool dapm);
+
+static struct sitar_mbhc_config mbhc_cfg = {
+ .headset_jack = &hs_jack,
+ .button_jack = &button_jack,
+ .read_fw_bin = false,
+ .calibration = NULL,
+ .micbias = SITAR_MICBIAS2,
+ .mclk_cb_fn = msm8930_enable_codec_ext_clk,
+ .mclk_rate = SITAR_EXT_CLK_RATE,
+ .gpio = 0,
+ .gpio_irq = 0,
+ .gpio_level_insert = 1,
+};
+
+
+static void msm8930_ext_control(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ pr_debug("%s: msm8930_spk_control = %d", __func__, msm8930_spk_control);
+ if (msm8930_spk_control == MSM8930_SPK_ON) {
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Left Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk left Neg");
+ } else {
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Left Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Left Neg");
+ }
+
+ snd_soc_dapm_sync(dapm);
+}
+
+static int msm8930_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8930_spk_control = %d", __func__, msm8930_spk_control);
+ ucontrol->value.integer.value[0] = msm8930_spk_control;
+ return 0;
+}
+static int msm8930_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm8930_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm8930_spk_control = ucontrol->value.integer.value[0];
+ msm8930_ext_control(codec);
+ return 1;
+}
+
+static void msm8960_ext_spk_power_amp_on(u32 spk)
+{
+ int ret = 0;
+
+ if (spk & (SPK_AMP_POS | SPK_AMP_NEG)) {
+ if ((msm8930_ext_spk_pamp & SPK_AMP_POS) &&
+ (msm8930_ext_spk_pamp & SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Bottom Speaker Ampl already "
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm8930_ext_spk_pamp |= spk;
+
+ if ((msm8930_ext_spk_pamp & SPK_AMP_POS) &&
+ (msm8930_ext_spk_pamp & SPK_AMP_NEG)) {
+
+ if (machine_is_msm8930_mtp()
+ || machine_is_msm8930_fluid()) {
+ pr_debug("%s: Configure Speaker Boost GPIO %u",
+ __func__, SPKR_BOOST_GPIO);
+ ret = gpio_request(SPKR_BOOST_GPIO,
+ "SPKR_BOOST_EN");
+ if (ret) {
+ pr_err("%s: Failed to configure speaker boost "
+ "gpio %u\n", __func__, SPKR_BOOST_GPIO);
+ return;
+ }
+
+ pr_debug("%s: Enable Speaker boost gpio %u\n",
+ __func__, SPKR_BOOST_GPIO);
+ gpio_direction_output(SPKR_BOOST_GPIO, 1);
+ }
+
+ pm8xxx_spk_enable(MSM8930_SPK_ON);
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Left Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm8960_ext_spk_power_amp_off(u32 spk)
+{
+ if (spk & (SPK_AMP_POS | SPK_AMP_NEG)) {
+ if (!msm8930_ext_spk_pamp)
+ return;
+ if (machine_is_msm8930_mtp()
+ || machine_is_msm8930_fluid()) {
+ pr_debug("%s: Free speaker boost gpio %u\n",
+ __func__, SPKR_BOOST_GPIO);
+ gpio_direction_output(SPKR_BOOST_GPIO, 0);
+ gpio_free(SPKR_BOOST_GPIO);
+ }
+
+ pm8xxx_spk_enable(MSM8930_SPK_OFF);
+ msm8930_ext_spk_pamp = 0;
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Left Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static int msm8930_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ if (!strncmp(w->name, "Ext Spk Left Pos", 17))
+ msm8960_ext_spk_power_amp_on(SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Left Neg", 17))
+ msm8960_ext_spk_power_amp_on(SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ } else {
+ if (!strncmp(w->name, "Ext Spk Left Pos", 17))
+ msm8960_ext_spk_power_amp_off(SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Left Neg", 17))
+ msm8960_ext_spk_power_amp_off(SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ }
+ return 0;
+}
+
+static int msm8930_enable_codec_ext_clk(
+ struct snd_soc_codec *codec, int enable,
+ bool dapm)
+{
+ pr_debug("%s: enable = %d\n", __func__, enable);
+ if (enable) {
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users != 1)
+ return 0;
+
+ if (codec_clk) {
+ clk_set_rate(codec_clk, SITAR_EXT_CLK_RATE);
+ clk_prepare_enable(codec_clk);
+ sitar_mclk_enable(codec, 1, dapm);
+ } else {
+ pr_err("%s: Error setting Sitar MCLK\n", __func__);
+ clk_users--;
+ return -EINVAL;
+ }
+ } else {
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users == 0)
+ return 0;
+ clk_users--;
+ if (!clk_users) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+ sitar_mclk_enable(codec, 0, dapm);
+ clk_disable_unprepare(codec_clk);
+ }
+ }
+ return 0;
+}
+
+static int msm8930_mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return msm8930_enable_codec_ext_clk(w->codec, 1, true);
+ case SND_SOC_DAPM_POST_PMD:
+ return msm8930_enable_codec_ext_clk(w->codec, 0, true);
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget msm8930_dapm_widgets[] = {
+
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ msm8930_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SPK("Ext Spk Left Pos", msm8930_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Left Neg", msm8930_spkramp_event),
+
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic2", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic3", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic4", NULL),
+
+};
+
+static const struct snd_soc_dapm_route common_audio_map[] = {
+
+ {"RX_BIAS", NULL, "MCLK"},
+ {"LDO_H", NULL, "MCLK"},
+
+ {"MIC BIAS1 Internal1", NULL, "MCLK"},
+ {"MIC BIAS2 Internal1", NULL, "MCLK"},
+
+ /* Speaker path */
+ {"Ext Spk Left Pos", NULL, "LINEOUT1"},
+ {"Ext Spk Left Neg", NULL, "LINEOUT2"},
+
+ /* Headset Mic */
+ {"AMIC2", NULL, "MIC BIAS2 Internal1"},
+ {"MIC BIAS2 Internal1", NULL, "Headset Mic"},
+
+ /* Microphone path */
+ {"AMIC1", NULL, "MIC BIAS2 Internal1"},
+ {"MIC BIAS2 Internal1", NULL, "ANCLeft Headset Mic"},
+
+ {"AMIC3", NULL, "MIC BIAS2 Internal1"},
+ {"MIC BIAS2 Internal1", NULL, "ANCRight Headset Mic"},
+
+ {"HEADPHONE", NULL, "LDO_H"},
+
+ /**
+ * The digital Mic routes are setup considering
+ * fluid as default device.
+ */
+
+ /**
+ * Digital Mic1. Front Bottom left Mic on Fluid and MTP.
+ * Digital Mic GM5 on CDP mainboard.
+ * Conncted to DMIC1 Input on Sitar codec.
+ */
+ {"DMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic1"},
+
+ /**
+ * Digital Mic2. Back top MIC on Fluid.
+ * Digital Mic GM6 on CDP mainboard.
+ * Conncted to DMIC2 Input on Sitar codec.
+ */
+ {"DMIC2", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic2"},
+ /**
+ * Digital Mic3. Back Bottom Digital Mic on Fluid.
+ * Digital Mic GM1 on CDP mainboard.
+ * Conncted to DMIC4 Input on Sitar codec.
+ */
+ {"DMIC3", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic3"},
+
+ /**
+ * Digital Mic4. Back top Digital Mic on Fluid.
+ * Digital Mic GM2 on CDP mainboard.
+ * Conncted to DMIC3 Input on Sitar codec.
+ */
+ {"DMIC4", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic4"},
+
+
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *slim0_rx_ch_text[] = {"One", "Two"};
+static const char *slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
+
+static const struct soc_enum msm8930_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
+ SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
+};
+
+static const char *btsco_rate_text[] = {"8000", "16000"};
+static const struct soc_enum msm8930_btsco_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, btsco_rate_text),
+};
+
+static int msm8930_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8930_slim_0_rx_ch = %d\n", __func__,
+ msm8930_slim_0_rx_ch);
+ ucontrol->value.integer.value[0] = msm8930_slim_0_rx_ch - 1;
+ return 0;
+}
+
+static int msm8930_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm8930_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm8930_slim_0_rx_ch = %d\n", __func__,
+ msm8930_slim_0_rx_ch);
+ return 1;
+}
+
+static int msm8930_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8930_slim_0_tx_ch = %d\n", __func__,
+ msm8930_slim_0_tx_ch);
+ ucontrol->value.integer.value[0] = msm8930_slim_0_tx_ch - 1;
+ return 0;
+}
+
+static int msm8930_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm8930_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm8930_slim_0_tx_ch = %d\n", __func__,
+ msm8930_slim_0_tx_ch);
+ return 1;
+}
+
+static int msm8930_btsco_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8930_btsco_rate = %d", __func__, msm8930_btsco_rate);
+ ucontrol->value.integer.value[0] = msm8930_btsco_rate;
+ return 0;
+}
+
+static int msm8930_btsco_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ msm8930_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ case 1:
+ msm8930_btsco_rate = BTSCO_RATE_16KHZ;
+ break;
+ default:
+ msm8930_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ }
+ pr_debug("%s: msm8930_btsco_rate = %d\n", __func__, msm8930_btsco_rate);
+ return 0;
+}
+
+static const char *pmic_spk_gain_text[] = {
+ "NEG_6_DB", "NEG_4_DB", "NEG_2_DB", "ZERO_DB", "POS_2_DB", "POS_4_DB",
+ "POS_6_DB", "POS_8_DB", "POS_10_DB", "POS_12_DB", "POS_14_DB",
+ "POS_16_DB", "POS_18_DB", "POS_20_DB", "POS_22_DB", "POS_24_DB"
+};
+
+static const struct soc_enum msm8960_pmic_spk_gain_enum[] = {
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(pmic_spk_gain_text),
+ pmic_spk_gain_text),
+};
+
+static int msm8930_pmic_gain_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8930_pmic_spk_gain = %d\n", __func__,
+ msm8930_pmic_spk_gain);
+ ucontrol->value.integer.value[0] = msm8930_pmic_spk_gain;
+ return 0;
+}
+
+static int msm8930_pmic_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int ret = 0;
+ msm8930_pmic_spk_gain = ucontrol->value.integer.value[0];
+ ret = pm8xxx_spk_gain(msm8930_pmic_spk_gain);
+ pr_debug("%s: msm8930_pmic_spk_gain = %d"
+ " ucontrol->value.integer.value[0] = %d\n", __func__,
+ msm8930_pmic_spk_gain,
+ (int) ucontrol->value.integer.value[0]);
+ return ret;
+}
+
+static const struct snd_kcontrol_new sitar_msm8930_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", msm8930_enum[0], msm8930_get_spk,
+ msm8930_set_spk),
+ SOC_ENUM_EXT("SLIM_0_RX Channels", msm8930_enum[1],
+ msm8930_slim_0_rx_ch_get, msm8930_slim_0_rx_ch_put),
+ SOC_ENUM_EXT("SLIM_0_TX Channels", msm8930_enum[2],
+ msm8930_slim_0_tx_ch_get, msm8930_slim_0_tx_ch_put),
+ SOC_ENUM_EXT("PMIC SPK Gain", msm8960_pmic_spk_gain_enum[0],
+ msm8930_pmic_gain_get, msm8930_pmic_gain_put),
+ SOC_ENUM_EXT("Internal BTSCO SampleRate", msm8930_btsco_enum[0],
+ msm8930_btsco_rate_get, msm8930_btsco_rate_put),
+};
+
+static void *def_sitar_mbhc_cal(void)
+{
+ void *sitar_cal;
+ struct sitar_mbhc_btn_detect_cfg *btn_cfg;
+ u16 *btn_low, *btn_high;
+ u8 *n_ready, *n_cic, *gain;
+
+ sitar_cal = kzalloc(SITAR_MBHC_CAL_SIZE(SITAR_MBHC_DEF_BUTTONS,
+ SITAR_MBHC_DEF_RLOADS),
+ GFP_KERNEL);
+ if (!sitar_cal) {
+ pr_err("%s: out of memory\n", __func__);
+ return NULL;
+ }
+
+#define S(X, Y) ((SITAR_MBHC_CAL_GENERAL_PTR(sitar_cal)->X) = (Y))
+ S(t_ldoh, 100);
+ S(t_bg_fast_settle, 100);
+ S(t_shutdown_plug_rem, 255);
+ S(mbhc_nsa, 4);
+ S(mbhc_navg, 4);
+#undef S
+#define S(X, Y) ((SITAR_MBHC_CAL_PLUG_DET_PTR(sitar_cal)->X) = (Y))
+ S(mic_current, SITAR_PID_MIC_5_UA);
+ S(hph_current, SITAR_PID_MIC_5_UA);
+ S(t_mic_pid, 100);
+ S(t_ins_complete, 250);
+ S(t_ins_retry, 200);
+#undef S
+#define S(X, Y) ((SITAR_MBHC_CAL_PLUG_TYPE_PTR(sitar_cal)->X) = (Y))
+ S(v_no_mic, 30);
+ S(v_hs_max, 1550);
+#undef S
+#define S(X, Y) ((SITAR_MBHC_CAL_BTN_DET_PTR(sitar_cal)->X) = (Y))
+ S(c[0], 62);
+ S(c[1], 124);
+ S(nc, 1);
+ S(n_meas, 3);
+ S(mbhc_nsc, 11);
+ S(n_btn_meas, 1);
+ S(n_btn_con, 2);
+ S(num_btn, SITAR_MBHC_DEF_BUTTONS);
+ S(v_btn_press_delta_sta, 100);
+ S(v_btn_press_delta_cic, 50);
+#undef S
+ btn_cfg = SITAR_MBHC_CAL_BTN_DET_PTR(sitar_cal);
+ btn_low = sitar_mbhc_cal_btn_det_mp(btn_cfg, SITAR_BTN_DET_V_BTN_LOW);
+ btn_high = sitar_mbhc_cal_btn_det_mp(btn_cfg, SITAR_BTN_DET_V_BTN_HIGH);
+ btn_low[0] = -50;
+ btn_high[0] = 10;
+ btn_low[1] = 11;
+ btn_high[1] = 38;
+ btn_low[2] = 39;
+ btn_high[2] = 64;
+ btn_low[3] = 65;
+ btn_high[3] = 91;
+ btn_low[4] = 92;
+ btn_high[4] = 115;
+ btn_low[5] = 116;
+ btn_high[5] = 141;
+ btn_low[6] = 142;
+ btn_high[6] = 163;
+ btn_low[7] = 164;
+ btn_high[7] = 250;
+ n_ready = sitar_mbhc_cal_btn_det_mp(btn_cfg, SITAR_BTN_DET_N_READY);
+ n_ready[0] = 48;
+ n_ready[1] = 38;
+ n_cic = sitar_mbhc_cal_btn_det_mp(btn_cfg, SITAR_BTN_DET_N_CIC);
+ n_cic[0] = 60;
+ n_cic[1] = 47;
+ gain = sitar_mbhc_cal_btn_det_mp(btn_cfg, SITAR_BTN_DET_GAIN);
+ gain[0] = 11;
+ gain[1] = 9;
+
+ return sitar_cal;
+}
+
+static int msm8930_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ unsigned int rx_ch_cnt = 0, tx_ch_cnt = 0;
+
+ pr_debug("%s: ch=%d\n", __func__,
+ msm8930_slim_0_rx_ch);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
+ msm8930_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai, 0, 0,
+ msm8930_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+ } else {
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(cpu_dai,
+ msm8930_slim_0_tx_ch, tx_ch, 0 , 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai,
+ msm8930_slim_0_tx_ch, tx_ch, 0, 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+
+ }
+end:
+ return ret;
+}
+
+static int msm8930_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ pr_debug("%s()\n", __func__);
+
+ snd_soc_dapm_new_controls(dapm, msm8930_dapm_widgets,
+ ARRAY_SIZE(msm8930_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, common_audio_map,
+ ARRAY_SIZE(common_audio_map));
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Left Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Left Neg");
+
+ snd_soc_dapm_sync(dapm);
+
+ err = snd_soc_jack_new(codec, "Headset Jack",
+ (SND_JACK_HEADSET | SND_JACK_OC_HPHL | SND_JACK_OC_HPHR),
+ &hs_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ err = snd_soc_jack_new(codec, "Button Jack",
+ SITAR_JACK_BUTTON_MASK, &button_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+ codec_clk = clk_get(cpu_dai->dev, "osr_clk");
+
+ mbhc_cfg.gpio = 37;
+ mbhc_cfg.gpio_irq = gpio_to_irq(mbhc_cfg.gpio);
+ sitar_hs_detect(codec, &mbhc_cfg);
+
+ /* Initialize default PMIC speaker gain */
+ pm8xxx_spk_gain(DEFAULT_PMIC_SPK_GAIN);
+
+ return 0;
+}
+
+static int msm8930_slim_0_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm8930_slim_0_rx_ch;
+
+ return 0;
+}
+
+static int msm8930_slim_0_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm8930_slim_0_tx_ch;
+
+ return 0;
+}
+
+static int msm8930_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+
+ return 0;
+}
+
+static int msm8930_hdmi_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ return 0;
+}
+
+static int msm8930_btsco_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = msm8930_btsco_rate;
+ channels->min = channels->max = msm8930_btsco_ch;
+
+ return 0;
+}
+
+static int msm8930_startup(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+ return 0;
+}
+
+static void msm8930_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+}
+
+static struct snd_soc_ops msm8930_be_ops = {
+ .startup = msm8930_startup,
+ .hw_params = msm8930_hw_params,
+ .shutdown = msm8930_shutdown,
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm8930_dai[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MSM8930 Media1",
+ .stream_name = "MultiMedia1",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MSM8930 Media2",
+ .stream_name = "MultiMedia2",
+ .cpu_dai_name = "MultiMedia2",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+ {
+ .name = "Circuit-Switch Voice",
+ .stream_name = "CS-Voice",
+ .cpu_dai_name = "CS-VOICE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .be_id = MSM_FRONTEND_DAI_CS_VOICE,
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = "MSM VoIP",
+ .stream_name = "VoIP",
+ .cpu_dai_name = "VoIP",
+ .platform_name = "msm-voip-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_VOIP,
+ },
+ {
+ .name = "MSM8930 LPA",
+ .stream_name = "LPA",
+ .cpu_dai_name = "MultiMedia3",
+ .platform_name = "msm-pcm-lpa",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
+ },
+ /* Hostless PMC purpose */
+ {
+ .name = "SLIMBUS_0 Hostless",
+ .stream_name = "SLIMBUS_0 Hostless",
+ .cpu_dai_name = "SLIMBUS0_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ /* .be_id = do not care */
+ },
+ {
+ .name = "INT_FM Hostless",
+ .stream_name = "INT_FM Hostless",
+ .cpu_dai_name = "INT_FM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ /* .be_id = do not care */
+ },
+ {
+ .name = "MSM AFE-PCM RX",
+ .stream_name = "AFE-PROXY RX",
+ .cpu_dai_name = "msm-dai-q6.241",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = "MSM AFE-PCM TX",
+ .stream_name = "AFE-PROXY TX",
+ .cpu_dai_name = "msm-dai-q6.240",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "MSM8930 Compr",
+ .stream_name = "COMPR",
+ .cpu_dai_name = "MultiMedia4",
+ .platform_name = "msm-compr-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA4,
+ },
+ {
+ .name = "AUXPCM Hostless",
+ .stream_name = "AUXPCM Hostless",
+ .cpu_dai_name = "AUXPCM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* HDMI Hostless */
+ {
+ .name = "HDMI_RX_HOSTLESS",
+ .stream_name = "HDMI_RX_HOSTLESS",
+ .cpu_dai_name = "HDMI_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* Backend DAI Links */
+ {
+ .name = LPASS_BE_SLIMBUS_0_RX,
+ .stream_name = "Slimbus Playback",
+ .cpu_dai_name = "msm-dai-q6.16384",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "sitar_codec",
+ .codec_dai_name = "sitar_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ .init = &msm8930_audrx_init,
+ .be_hw_params_fixup = msm8930_slim_0_rx_be_hw_params_fixup,
+ .ops = &msm8930_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_0_TX,
+ .stream_name = "Slimbus Capture",
+ .cpu_dai_name = "msm-dai-q6.16385",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "sitar_codec",
+ .codec_dai_name = "sitar_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ .be_hw_params_fixup = msm8930_slim_0_tx_be_hw_params_fixup,
+ .ops = &msm8930_be_ops,
+ },
+ /* Backend BT/FM DAI Links */
+ {
+ .name = LPASS_BE_INT_BT_SCO_RX,
+ .stream_name = "Internal BT-SCO Playback",
+ .cpu_dai_name = "msm-dai-q6.12288",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ .be_hw_params_fixup = msm8930_btsco_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_BT_SCO_TX,
+ .stream_name = "Internal BT-SCO Capture",
+ .cpu_dai_name = "msm-dai-q6.12289",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ .be_hw_params_fixup = msm8930_btsco_be_hw_params_fixup,
+ },
+ {
+ .name = LPASS_BE_INT_FM_RX,
+ .stream_name = "Internal FM Playback",
+ .cpu_dai_name = "msm-dai-q6.12292",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_RX,
+ .be_hw_params_fixup = msm8930_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_FM_TX,
+ .stream_name = "Internal FM Capture",
+ .cpu_dai_name = "msm-dai-q6.12293",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_TX,
+ .be_hw_params_fixup = msm8930_be_hw_params_fixup,
+ },
+ /* HDMI BACK END DAI Link */
+ {
+ .name = LPASS_BE_HDMI,
+ .stream_name = "HDMI Playback",
+ .cpu_dai_name = "msm-dai-q6-hdmi.8",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_HDMI_RX,
+ .be_hw_params_fixup = msm8930_hdmi_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ /* Backend AFE DAI Links */
+ {
+ .name = LPASS_BE_AFE_PCM_RX,
+ .stream_name = "AFE Playback",
+ .cpu_dai_name = "msm-dai-q6.224",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_RX,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_AFE_PCM_TX,
+ .stream_name = "AFE Capture",
+ .cpu_dai_name = "msm-dai-q6.225",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_TX,
+ },
+ /* Incall Music BACK END DAI Link */
+ {
+ .name = LPASS_BE_VOICE_PLAYBACK_TX,
+ .stream_name = "Voice Farend Playback",
+ .cpu_dai_name = "msm-dai-q6.32773",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ .be_hw_params_fixup = msm8930_be_hw_params_fixup,
+ },
+ /* Incall Record Uplink BACK END DAI Link */
+ {
+ .name = LPASS_BE_INCALL_RECORD_TX,
+ .stream_name = "Voice Uplink Capture",
+ .cpu_dai_name = "msm-dai-q6.32772",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ .be_hw_params_fixup = msm8930_be_hw_params_fixup,
+ },
+ /* Incall Record Downlink BACK END DAI Link */
+ {
+ .name = LPASS_BE_INCALL_RECORD_RX,
+ .stream_name = "Voice Downlink Capture",
+ .cpu_dai_name = "msm-dai-q6.32771",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ .be_hw_params_fixup = msm8930_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+};
+
+struct snd_soc_card snd_soc_card_msm8930 = {
+ .name = "msm8930-sitar-snd-card",
+ .dai_link = msm8930_dai,
+ .num_links = ARRAY_SIZE(msm8930_dai),
+ .controls = sitar_msm8930_controls,
+ .num_controls = ARRAY_SIZE(sitar_msm8930_controls),
+};
+
+static struct platform_device *msm8930_snd_device;
+
+static int msm8930_configure_headset_mic_gpios(void)
+{
+ int ret;
+ ret = gpio_request(80, "US_EURO_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio 80\n", __func__);
+ return ret;
+ }
+ ret = gpio_direction_output(80, 0);
+ if (ret) {
+ pr_err("%s: Unable to set direction\n", __func__);
+ gpio_free(80);
+ }
+ msm8930_headset_gpios_configured = 0;
+ return 0;
+}
+static void msm8930_free_headset_mic_gpios(void)
+{
+ if (msm8930_headset_gpios_configured)
+ gpio_free(80);
+}
+
+static int __init msm8930_audio_init(void)
+{
+ int ret;
+
+ if (!cpu_is_msm8930()) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return -ENODEV ;
+ }
+ mbhc_cfg.calibration = def_sitar_mbhc_cal();
+ if (!mbhc_cfg.calibration) {
+ pr_err("Calibration data allocation failed\n");
+ return -ENOMEM;
+ }
+
+ msm8930_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm8930_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(msm8930_snd_device, &snd_soc_card_msm8930);
+ ret = platform_device_add(msm8930_snd_device);
+ if (ret) {
+ platform_device_put(msm8930_snd_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+
+ if (msm8930_configure_headset_mic_gpios()) {
+ pr_err("%s Fail to configure headset mic gpios\n", __func__);
+ msm8930_headset_gpios_configured = 0;
+ } else
+ msm8930_headset_gpios_configured = 1;
+
+ return ret;
+
+}
+module_init(msm8930_audio_init);
+
+static void __exit msm8930_audio_exit(void)
+{
+ if (!cpu_is_msm8930()) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return ;
+ }
+ msm8930_free_headset_mic_gpios();
+ platform_device_unregister(msm8930_snd_device);
+ kfree(mbhc_cfg.calibration);
+}
+module_exit(msm8930_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC MSM8930");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8960.c b/sound/soc/msm/msm8960.c
new file mode 100644
index 0000000..bb5d817
--- /dev/null
+++ b/sound/soc/msm/msm8960.c
@@ -0,0 +1,1615 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <mach/socinfo.h>
+#include <linux/mfd/wcd9xxx/core.h>
+#include "msm-pcm-routing.h"
+#include "../codecs/wcd9310.h"
+
+/* 8960 machine driver */
+
+#define PM8921_GPIO_BASE NR_GPIO_IRQS
+#define PM8921_IRQ_BASE (NR_MSM_IRQS + NR_GPIO_IRQS)
+#define PM8921_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio - 1 + PM8921_GPIO_BASE)
+
+#define MSM8960_SPK_ON 1
+#define MSM8960_SPK_OFF 0
+
+#define msm8960_SLIM_0_RX_MAX_CHANNELS 2
+#define msm8960_SLIM_0_TX_MAX_CHANNELS 4
+
+#define SAMPLE_RATE_8KHZ 8000
+#define SAMPLE_RATE_16KHZ 16000
+
+#define BOTTOM_SPK_AMP_POS 0x1
+#define BOTTOM_SPK_AMP_NEG 0x2
+#define TOP_SPK_AMP_POS 0x4
+#define TOP_SPK_AMP_NEG 0x8
+
+#define GPIO_AUX_PCM_DOUT 63
+#define GPIO_AUX_PCM_DIN 64
+#define GPIO_AUX_PCM_SYNC 65
+#define GPIO_AUX_PCM_CLK 66
+
+#define TABLA_EXT_CLK_RATE 12288000
+
+#define TABLA_MBHC_DEF_BUTTONS 8
+#define TABLA_MBHC_DEF_RLOADS 5
+
+#define JACK_DETECT_GPIO 38
+#define JACK_DETECT_INT PM8921_GPIO_IRQ(PM8921_IRQ_BASE, JACK_DETECT_GPIO)
+#define JACK_US_EURO_SEL_GPIO 35
+
+static u32 top_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(18);
+static u32 bottom_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(19);
+static int msm8960_spk_control;
+static int msm8960_ext_bottom_spk_pamp;
+static int msm8960_ext_top_spk_pamp;
+static int msm8960_slim_0_rx_ch = 1;
+static int msm8960_slim_0_tx_ch = 1;
+
+static int msm8960_btsco_rate = SAMPLE_RATE_8KHZ;
+static int msm8960_btsco_ch = 1;
+
+static int msm8960_auxpcm_rate = SAMPLE_RATE_8KHZ;
+
+static struct clk *codec_clk;
+static int clk_users;
+
+static int msm8960_headset_gpios_configured;
+
+static struct snd_soc_jack hs_jack;
+static struct snd_soc_jack button_jack;
+
+static bool hs_detect_use_gpio;
+module_param(hs_detect_use_gpio, bool, 0444);
+MODULE_PARM_DESC(hs_detect_use_gpio, "Use GPIO for headset detection");
+
+static bool hs_detect_use_firmware;
+module_param(hs_detect_use_firmware, bool, 0444);
+MODULE_PARM_DESC(hs_detect_use_firmware, "Use firmware for headset detection");
+
+static int msm8960_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm);
+static bool msm8960_swap_gnd_mic(struct snd_soc_codec *codec);
+
+static struct tabla_mbhc_config mbhc_cfg = {
+ .headset_jack = &hs_jack,
+ .button_jack = &button_jack,
+ .read_fw_bin = false,
+ .calibration = NULL,
+ .micbias = TABLA_MICBIAS2,
+ .mclk_cb_fn = msm8960_enable_codec_ext_clk,
+ .mclk_rate = TABLA_EXT_CLK_RATE,
+ .gpio = 0,
+ .gpio_irq = 0,
+ .gpio_level_insert = 1,
+ .swap_gnd_mic = NULL,
+};
+
+static u32 us_euro_sel_gpio = PM8921_GPIO_PM_TO_SYS(JACK_US_EURO_SEL_GPIO);
+
+static struct mutex cdc_mclk_mutex;
+
+static void msm8960_enable_ext_spk_amp_gpio(u32 spk_amp_gpio)
+{
+ int ret = 0;
+
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .
+ function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ if (spk_amp_gpio == bottom_spk_pamp_gpio) {
+
+ ret = gpio_request(bottom_spk_pamp_gpio, "BOTTOM_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting BOTTOM SPK AMP GPIO %u\n",
+ __func__, bottom_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(bottom_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Bottom Spk Ampl"
+ " gpio %u\n", __func__, bottom_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Bottom spkr amp gpio\n", __func__);
+ gpio_direction_output(bottom_spk_pamp_gpio, 1);
+ }
+
+ } else if (spk_amp_gpio == top_spk_pamp_gpio) {
+
+ ret = gpio_request(top_spk_pamp_gpio, "TOP_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting GPIO %d\n", __func__,
+ top_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(top_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Top Spk Ampl"
+ " gpio %u\n", __func__, top_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Top spkr amp gpio\n", __func__);
+ gpio_direction_output(top_spk_pamp_gpio, 1);
+ }
+ } else {
+ pr_err("%s: ERROR : Invalid External Speaker Ampl GPIO."
+ " gpio = %u\n", __func__, spk_amp_gpio);
+ return;
+ }
+}
+
+static void msm8960_ext_spk_power_amp_on(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if ((msm8960_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm8960_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Bottom Speaker Ampl already "
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm8960_ext_bottom_spk_pamp |= spk;
+
+ if ((msm8960_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm8960_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ msm8960_enable_ext_spk_amp_gpio(bottom_spk_pamp_gpio);
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Bottom Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if ((msm8960_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm8960_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Top Speaker Ampl already"
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm8960_ext_top_spk_pamp |= spk;
+
+ if ((msm8960_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm8960_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ msm8960_enable_ext_spk_amp_gpio(top_spk_pamp_gpio);
+ pr_debug("%s: sleeping 4 ms after turning on "
+ " external Top Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm8960_ext_spk_power_amp_off(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if (!msm8960_ext_bottom_spk_pamp)
+ return;
+
+ gpio_direction_output(bottom_spk_pamp_gpio, 0);
+ gpio_free(bottom_spk_pamp_gpio);
+ msm8960_ext_bottom_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Bottom"
+ " Speaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if (!msm8960_ext_top_spk_pamp)
+ return;
+
+ gpio_direction_output(top_spk_pamp_gpio, 0);
+ gpio_free(top_spk_pamp_gpio);
+ msm8960_ext_top_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Top"
+ " Spkaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+ } else {
+
+ pr_err("%s: ERROR : Invalid Ext Spk Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm8960_ext_control(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ mutex_lock(&dapm->codec->mutex);
+
+ pr_debug("%s: msm8960_spk_control = %d", __func__, msm8960_spk_control);
+ if (msm8960_spk_control == MSM8960_SPK_ON) {
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+ } else {
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Neg");
+ }
+
+ snd_soc_dapm_sync(dapm);
+ mutex_unlock(&dapm->codec->mutex);
+}
+
+static int msm8960_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8960_spk_control = %d", __func__, msm8960_spk_control);
+ ucontrol->value.integer.value[0] = msm8960_spk_control;
+ return 0;
+}
+static int msm8960_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm8960_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm8960_spk_control = ucontrol->value.integer.value[0];
+ msm8960_ext_control(codec);
+ return 1;
+}
+static int msm8960_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm8960_ext_spk_power_amp_on(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm8960_ext_spk_power_amp_on(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm8960_ext_spk_power_amp_on(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm8960_ext_spk_power_amp_on(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+
+ } else {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm8960_ext_spk_power_amp_off(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm8960_ext_spk_power_amp_off(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm8960_ext_spk_power_amp_off(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm8960_ext_spk_power_amp_off(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ }
+ return 0;
+}
+
+static int msm8960_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm)
+{
+ int r = 0;
+ pr_debug("%s: enable = %d\n", __func__, enable);
+
+ mutex_lock(&cdc_mclk_mutex);
+ if (enable) {
+ clk_users++;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users == 1) {
+ if (codec_clk) {
+ clk_set_rate(codec_clk, TABLA_EXT_CLK_RATE);
+ clk_prepare_enable(codec_clk);
+ tabla_mclk_enable(codec, 1, dapm);
+ } else {
+ pr_err("%s: Error setting Tabla MCLK\n",
+ __func__);
+ clk_users--;
+ r = -EINVAL;
+ }
+ }
+ } else {
+ if (clk_users > 0) {
+ clk_users--;
+ pr_debug("%s: clk_users = %d\n", __func__, clk_users);
+ if (clk_users == 0) {
+ pr_debug("%s: disabling MCLK. clk_users = %d\n",
+ __func__, clk_users);
+ tabla_mclk_enable(codec, 0, dapm);
+ clk_disable_unprepare(codec_clk);
+ }
+ } else {
+ pr_err("%s: Error releasing Tabla MCLK\n", __func__);
+ r = -EINVAL;
+ }
+ }
+ mutex_unlock(&cdc_mclk_mutex);
+ return r;
+}
+
+static bool msm8960_swap_gnd_mic(struct snd_soc_codec *codec)
+{
+ int value = gpio_get_value_cansleep(us_euro_sel_gpio);
+ pr_debug("%s: US EURO select switch %d to %d\n", __func__, value,
+ !value);
+ gpio_set_value_cansleep(us_euro_sel_gpio, !value);
+ return true;
+}
+
+static int msm8960_mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ pr_debug("%s: event = %d\n", __func__, event);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return msm8960_enable_codec_ext_clk(w->codec, 1, true);
+ case SND_SOC_DAPM_POST_PMD:
+ return msm8960_enable_codec_ext_clk(w->codec, 0, true);
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget msm8960_dapm_widgets[] = {
+
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ msm8960_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Pos", msm8960_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Neg", msm8960_spkramp_event),
+
+ SND_SOC_DAPM_SPK("Ext Spk Top Pos", msm8960_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Top Neg", msm8960_spkramp_event),
+
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic2", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic3", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic4", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic5", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic6", NULL),
+
+};
+
+static const struct snd_soc_dapm_route common_audio_map[] = {
+
+ {"RX_BIAS", NULL, "MCLK"},
+ {"LDO_H", NULL, "MCLK"},
+
+ /* Speaker path */
+ {"Ext Spk Bottom Pos", NULL, "LINEOUT1"},
+ {"Ext Spk Bottom Neg", NULL, "LINEOUT3"},
+
+ {"Ext Spk Top Pos", NULL, "LINEOUT2"},
+ {"Ext Spk Top Neg", NULL, "LINEOUT4"},
+
+ /* Microphone path */
+ {"AMIC1", NULL, "MIC BIAS1 Internal1"},
+ {"MIC BIAS1 Internal1", NULL, "Handset Mic"},
+
+ {"AMIC2", NULL, "MIC BIAS2 External"},
+ {"MIC BIAS2 External", NULL, "Headset Mic"},
+
+ /**
+ * AMIC3 and AMIC4 inputs are connected to ANC microphones
+ * These mics are biased differently on CDP and FLUID
+ * routing entries below are based on bias arrangement
+ * on FLUID.
+ */
+ {"AMIC3", NULL, "MIC BIAS3 Internal1"},
+ {"MIC BIAS3 Internal1", NULL, "ANCRight Headset Mic"},
+
+ {"AMIC4", NULL, "MIC BIAS1 Internal2"},
+ {"MIC BIAS1 Internal2", NULL, "ANCLeft Headset Mic"},
+
+ {"HEADPHONE", NULL, "LDO_H"},
+
+ /**
+ * The digital Mic routes are setup considering
+ * fluid as default device.
+ */
+
+ /**
+ * Digital Mic1. Front Bottom left Digital Mic on Fluid and MTP.
+ * Digital Mic GM5 on CDP mainboard.
+ * Conncted to DMIC2 Input on Tabla codec.
+ */
+ {"DMIC2", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic1"},
+
+ /**
+ * Digital Mic2. Front Bottom right Digital Mic on Fluid and MTP.
+ * Digital Mic GM6 on CDP mainboard.
+ * Conncted to DMIC1 Input on Tabla codec.
+ */
+ {"DMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic2"},
+
+ /**
+ * Digital Mic3. Back Bottom Digital Mic on Fluid.
+ * Digital Mic GM1 on CDP mainboard.
+ * Conncted to DMIC4 Input on Tabla codec.
+ */
+ {"DMIC4", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic3"},
+
+ /**
+ * Digital Mic4. Back top Digital Mic on Fluid.
+ * Digital Mic GM2 on CDP mainboard.
+ * Conncted to DMIC3 Input on Tabla codec.
+ */
+ {"DMIC3", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic4"},
+
+ /**
+ * Digital Mic5. Front top Digital Mic on Fluid.
+ * Digital Mic GM3 on CDP mainboard.
+ * Conncted to DMIC5 Input on Tabla codec.
+ */
+ {"DMIC5", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic5"},
+
+ /* Tabla digital Mic6 - back bottom digital Mic on Liquid and
+ * bottom mic on CDP. FLUID/MTP do not have dmic6 installed.
+ */
+ {"DMIC6", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic6"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *slim0_rx_ch_text[] = {"One", "Two"};
+static const char *slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
+
+static const struct soc_enum msm8960_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
+ SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
+};
+
+static const char *btsco_rate_text[] = {"8000", "16000"};
+static const struct soc_enum msm8960_btsco_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, btsco_rate_text),
+};
+
+static const char *auxpcm_rate_text[] = {"rate_8000", "rate_16000"};
+static const struct soc_enum msm8960_auxpcm_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, auxpcm_rate_text),
+};
+
+static int msm8960_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8960_slim_0_rx_ch = %d\n", __func__,
+ msm8960_slim_0_rx_ch);
+ ucontrol->value.integer.value[0] = msm8960_slim_0_rx_ch - 1;
+ return 0;
+}
+
+static int msm8960_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm8960_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm8960_slim_0_rx_ch = %d\n", __func__,
+ msm8960_slim_0_rx_ch);
+ return 1;
+}
+
+static int msm8960_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8960_slim_0_tx_ch = %d\n", __func__,
+ msm8960_slim_0_tx_ch);
+ ucontrol->value.integer.value[0] = msm8960_slim_0_tx_ch - 1;
+ return 0;
+}
+
+static int msm8960_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm8960_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm8960_slim_0_tx_ch = %d\n", __func__,
+ msm8960_slim_0_tx_ch);
+ return 1;
+}
+
+static int msm8960_btsco_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8960_btsco_rate = %d", __func__, msm8960_btsco_rate);
+ ucontrol->value.integer.value[0] = msm8960_btsco_rate;
+ return 0;
+}
+
+static int msm8960_btsco_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ msm8960_btsco_rate = SAMPLE_RATE_8KHZ;
+ break;
+ case 1:
+ msm8960_btsco_rate = SAMPLE_RATE_16KHZ;
+ break;
+ default:
+ msm8960_btsco_rate = SAMPLE_RATE_8KHZ;
+ break;
+ }
+ pr_debug("%s: msm8960_btsco_rate = %d\n", __func__, msm8960_btsco_rate);
+ return 0;
+}
+
+static int msm8960_auxpcm_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm8960_auxpcm_rate = %d", __func__,
+ msm8960_auxpcm_rate);
+ ucontrol->value.integer.value[0] = msm8960_auxpcm_rate;
+ return 0;
+}
+
+static int msm8960_auxpcm_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ msm8960_auxpcm_rate = SAMPLE_RATE_8KHZ;
+ break;
+ case 1:
+ msm8960_auxpcm_rate = SAMPLE_RATE_16KHZ;
+ break;
+ default:
+ msm8960_auxpcm_rate = SAMPLE_RATE_8KHZ;
+ break;
+ }
+ pr_debug("%s: msm8960_auxpcm_rate = %d"
+ "ucontrol->value.integer.value[0] = %d\n", __func__,
+ msm8960_auxpcm_rate,
+ (int)ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static const struct snd_kcontrol_new tabla_msm8960_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", msm8960_enum[0], msm8960_get_spk,
+ msm8960_set_spk),
+ SOC_ENUM_EXT("SLIM_0_RX Channels", msm8960_enum[1],
+ msm8960_slim_0_rx_ch_get, msm8960_slim_0_rx_ch_put),
+ SOC_ENUM_EXT("SLIM_0_TX Channels", msm8960_enum[2],
+ msm8960_slim_0_tx_ch_get, msm8960_slim_0_tx_ch_put),
+ SOC_ENUM_EXT("Internal BTSCO SampleRate", msm8960_btsco_enum[0],
+ msm8960_btsco_rate_get, msm8960_btsco_rate_put),
+ SOC_ENUM_EXT("AUX PCM SampleRate", msm8960_auxpcm_enum[0],
+ msm8960_auxpcm_rate_get, msm8960_auxpcm_rate_put),
+};
+
+static void *def_tabla_mbhc_cal(void)
+{
+ void *tabla_cal;
+ struct tabla_mbhc_btn_detect_cfg *btn_cfg;
+ u16 *btn_low, *btn_high;
+ u8 *n_ready, *n_cic, *gain;
+
+ tabla_cal = kzalloc(TABLA_MBHC_CAL_SIZE(TABLA_MBHC_DEF_BUTTONS,
+ TABLA_MBHC_DEF_RLOADS),
+ GFP_KERNEL);
+ if (!tabla_cal) {
+ pr_err("%s: out of memory\n", __func__);
+ return NULL;
+ }
+
+#define S(X, Y) ((TABLA_MBHC_CAL_GENERAL_PTR(tabla_cal)->X) = (Y))
+ S(t_ldoh, 100);
+ S(t_bg_fast_settle, 100);
+ S(t_shutdown_plug_rem, 255);
+ S(mbhc_nsa, 4);
+ S(mbhc_navg, 4);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_DET_PTR(tabla_cal)->X) = (Y))
+ S(mic_current, TABLA_PID_MIC_5_UA);
+ S(hph_current, TABLA_PID_MIC_5_UA);
+ S(t_mic_pid, 100);
+ S(t_ins_complete, 250);
+ S(t_ins_retry, 200);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_PLUG_TYPE_PTR(tabla_cal)->X) = (Y))
+ S(v_no_mic, 30);
+ S(v_hs_max, 2400);
+#undef S
+#define S(X, Y) ((TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal)->X) = (Y))
+ S(c[0], 62);
+ S(c[1], 124);
+ S(nc, 1);
+ S(n_meas, 3);
+ S(mbhc_nsc, 11);
+ S(n_btn_meas, 1);
+ S(n_btn_con, 2);
+ S(num_btn, TABLA_MBHC_DEF_BUTTONS);
+ S(v_btn_press_delta_sta, 100);
+ S(v_btn_press_delta_cic, 50);
+#undef S
+ btn_cfg = TABLA_MBHC_CAL_BTN_DET_PTR(tabla_cal);
+ btn_low = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_LOW);
+ btn_high = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_V_BTN_HIGH);
+ btn_low[0] = -50;
+ btn_high[0] = 20;
+ btn_low[1] = 21;
+ btn_high[1] = 62;
+ btn_low[2] = 63;
+ btn_high[2] = 104;
+ btn_low[3] = 105;
+ btn_high[3] = 143;
+ btn_low[4] = 144;
+ btn_high[4] = 181;
+ btn_low[5] = 182;
+ btn_high[5] = 218;
+ btn_low[6] = 219;
+ btn_high[6] = 254;
+ btn_low[7] = 255;
+ btn_high[7] = 330;
+ n_ready = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_READY);
+ n_ready[0] = 80;
+ n_ready[1] = 68;
+ n_cic = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_N_CIC);
+ n_cic[0] = 60;
+ n_cic[1] = 47;
+ gain = tabla_mbhc_cal_btn_det_mp(btn_cfg, TABLA_BTN_DET_GAIN);
+ gain[0] = 11;
+ gain[1] = 9;
+
+ return tabla_cal;
+}
+
+static int msm8960_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int ret = 0;
+ unsigned int rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS];
+ unsigned int rx_ch_cnt = 0, tx_ch_cnt = 0;
+ unsigned int user_set_tx_ch = 0;
+
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ pr_debug("%s: rx_0_ch=%d\n", __func__, msm8960_slim_0_rx_ch);
+
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+
+ ret = snd_soc_dai_set_channel_map(cpu_dai, 0, 0,
+ msm8960_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai, 0, 0,
+ msm8960_slim_0_rx_ch, rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+ } else {
+
+ if (codec_dai->id == 2)
+ user_set_tx_ch = msm8960_slim_0_tx_ch;
+ else if (codec_dai->id == 4)
+ user_set_tx_ch = params_channels(params);
+
+ pr_debug("%s: %s_tx_dai_id_%d_ch=%d\n", __func__,
+ codec_dai->name, codec_dai->id, user_set_tx_ch);
+
+ ret = snd_soc_dai_get_channel_map(codec_dai,
+ &tx_ch_cnt, tx_ch, &rx_ch_cnt , rx_ch);
+ if (ret < 0) {
+ pr_err("%s: failed to get codec chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(cpu_dai,
+ user_set_tx_ch, tx_ch, 0 , 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set cpu chan map\n", __func__);
+ goto end;
+ }
+ ret = snd_soc_dai_set_channel_map(codec_dai,
+ user_set_tx_ch, tx_ch, 0, 0);
+ if (ret < 0) {
+ pr_err("%s: failed to set codec channel map\n",
+ __func__);
+ goto end;
+ }
+
+
+ }
+end:
+ return ret;
+}
+
+static int msm8960_audrx_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int err;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct pm_gpio jack_gpio_cfg = {
+ .direction = PM_GPIO_DIR_IN,
+ .pull = PM_GPIO_PULL_UP_1P5,
+ .function = PM_GPIO_FUNC_NORMAL,
+ .vin_sel = 2,
+ .inv_int_pol = 0,
+ };
+
+ pr_debug("%s(), dev_name%s\n", __func__, dev_name(cpu_dai->dev));
+
+ if (machine_is_msm8960_liquid()) {
+ top_spk_pamp_gpio = (PM8921_GPIO_PM_TO_SYS(19));
+ bottom_spk_pamp_gpio = (PM8921_GPIO_PM_TO_SYS(18));
+ }
+
+ snd_soc_dapm_new_controls(dapm, msm8960_dapm_widgets,
+ ARRAY_SIZE(msm8960_dapm_widgets));
+
+ snd_soc_dapm_add_routes(dapm, common_audio_map,
+ ARRAY_SIZE(common_audio_map));
+
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+
+ snd_soc_dapm_sync(dapm);
+
+ err = snd_soc_jack_new(codec, "Headset Jack",
+ (SND_JACK_HEADSET | SND_JACK_OC_HPHL |
+ SND_JACK_OC_HPHR | SND_JACK_UNSUPPORTED),
+ &hs_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ err = snd_soc_jack_new(codec, "Button Jack",
+ TABLA_JACK_BUTTON_MASK, &button_jack);
+ if (err) {
+ pr_err("failed to create new jack\n");
+ return err;
+ }
+
+ codec_clk = clk_get(cpu_dai->dev, "osr_clk");
+
+ if (machine_is_msm8960_cdp())
+ mbhc_cfg.swap_gnd_mic = msm8960_swap_gnd_mic;
+
+ if (hs_detect_use_gpio) {
+ mbhc_cfg.gpio = PM8921_GPIO_PM_TO_SYS(JACK_DETECT_GPIO);
+ mbhc_cfg.gpio_irq = JACK_DETECT_INT;
+ }
+
+ if (mbhc_cfg.gpio) {
+ err = pm8xxx_gpio_config(mbhc_cfg.gpio, &jack_gpio_cfg);
+ if (err) {
+ pr_err("%s: pm8xxx_gpio_config JACK_DETECT failed %d\n",
+ __func__, err);
+ return err;
+ }
+ }
+
+ mbhc_cfg.read_fw_bin = hs_detect_use_firmware;
+
+ err = tabla_hs_detect(codec, &mbhc_cfg);
+
+ return err;
+}
+
+static int msm8960_slim_0_rx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm8960_slim_0_rx_ch;
+
+ return 0;
+}
+
+static int msm8960_slim_0_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = msm8960_slim_0_tx_ch;
+
+ return 0;
+}
+
+static int msm8960_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ pr_debug("%s()\n", __func__);
+ rate->min = rate->max = 48000;
+
+ return 0;
+}
+
+static int msm8960_hdmi_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ pr_debug("%s channels->min %u channels->max %u ()\n", __func__,
+ channels->min, channels->max);
+
+ rate->min = rate->max = 48000;
+
+ return 0;
+}
+
+static int msm8960_btsco_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = msm8960_btsco_rate;
+ channels->min = channels->max = msm8960_btsco_ch;
+
+ return 0;
+}
+static int msm8960_auxpcm_be_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = msm8960_auxpcm_rate;
+ /* PCM only supports mono output */
+ channels->min = channels->max = 1;
+
+ return 0;
+}
+static int msm8960_aux_pcm_get_gpios(void)
+{
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ ret = gpio_request(GPIO_AUX_PCM_DOUT, "AUX PCM DOUT");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DOUT",
+ __func__, GPIO_AUX_PCM_DOUT);
+ goto fail_dout;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_DIN, "AUX PCM DIN");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DIN",
+ __func__, GPIO_AUX_PCM_DIN);
+ goto fail_din;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_SYNC, "AUX PCM SYNC");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM SYNC",
+ __func__, GPIO_AUX_PCM_SYNC);
+ goto fail_sync;
+ }
+ ret = gpio_request(GPIO_AUX_PCM_CLK, "AUX PCM CLK");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM CLK",
+ __func__, GPIO_AUX_PCM_CLK);
+ goto fail_clk;
+ }
+
+ return 0;
+
+fail_clk:
+ gpio_free(GPIO_AUX_PCM_SYNC);
+fail_sync:
+ gpio_free(GPIO_AUX_PCM_DIN);
+fail_din:
+ gpio_free(GPIO_AUX_PCM_DOUT);
+fail_dout:
+
+ return ret;
+}
+
+static int msm8960_aux_pcm_free_gpios(void)
+{
+ gpio_free(GPIO_AUX_PCM_DIN);
+ gpio_free(GPIO_AUX_PCM_DOUT);
+ gpio_free(GPIO_AUX_PCM_SYNC);
+ gpio_free(GPIO_AUX_PCM_CLK);
+
+ return 0;
+}
+static int msm8960_startup(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+ return 0;
+}
+
+static int msm8960_auxpcm_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ ret = msm8960_aux_pcm_get_gpios();
+ if (ret < 0) {
+ pr_err("%s: Aux PCM GPIO request failed\n", __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static void msm8960_auxpcm_shutdown(struct snd_pcm_substream *substream)
+{
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ msm8960_aux_pcm_free_gpios();
+}
+
+static void msm8960_shutdown(struct snd_pcm_substream *substream)
+{
+ pr_debug("%s(): substream = %s stream = %d\n", __func__,
+ substream->name, substream->stream);
+}
+
+static struct snd_soc_ops msm8960_be_ops = {
+ .startup = msm8960_startup,
+ .hw_params = msm8960_hw_params,
+ .shutdown = msm8960_shutdown,
+};
+
+static struct snd_soc_ops msm8960_auxpcm_be_ops = {
+ .startup = msm8960_auxpcm_startup,
+ .shutdown = msm8960_auxpcm_shutdown,
+};
+
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm8960_dai_common[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MSM8960 Media1",
+ .stream_name = "MultiMedia1",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MSM8960 Media2",
+ .stream_name = "MultiMedia2",
+ .cpu_dai_name = "MultiMedia2",
+ .platform_name = "msm-multi-ch-pcm-dsp",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA2,
+ },
+ {
+ .name = "Circuit-Switch Voice",
+ .stream_name = "CS-Voice",
+ .cpu_dai_name = "CS-VOICE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_CS_VOICE,
+ },
+ {
+ .name = "MSM VoIP",
+ .stream_name = "VoIP",
+ .cpu_dai_name = "VoIP",
+ .platform_name = "msm-voip-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_VOIP,
+ },
+ {
+ .name = "MSM8960 LPA",
+ .stream_name = "LPA",
+ .cpu_dai_name = "MultiMedia3",
+ .platform_name = "msm-pcm-lpa",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
+ },
+ /* Hostless PMC purpose */
+ {
+ .name = "SLIMBUS_0 Hostless",
+ .stream_name = "SLIMBUS_0 Hostless",
+ .cpu_dai_name = "SLIMBUS0_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "INT_FM Hostless",
+ .stream_name = "INT_FM Hostless",
+ .cpu_dai_name = "INT_FM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "MSM AFE-PCM RX",
+ .stream_name = "AFE-PROXY RX",
+ .cpu_dai_name = "msm-dai-q6.241",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = "MSM AFE-PCM TX",
+ .stream_name = "AFE-PROXY TX",
+ .cpu_dai_name = "msm-dai-q6.240",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .platform_name = "msm-pcm-afe",
+ .ignore_suspend = 1,
+ },
+ {
+ .name = "MSM8960 Compr",
+ .stream_name = "COMPR",
+ .cpu_dai_name = "MultiMedia4",
+ .platform_name = "msm-compr-dsp",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA4,
+ },
+ {
+ .name = "AUXPCM Hostless",
+ .stream_name = "AUXPCM Hostless",
+ .cpu_dai_name = "AUXPCM_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ /* HDMI Hostless */
+ {
+ .name = "HDMI_RX_HOSTLESS",
+ .stream_name = "HDMI_RX_HOSTLESS",
+ .cpu_dai_name = "HDMI_HOSTLESS",
+ .platform_name = "msm-pcm-hostless",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ },
+ {
+ .name = "VoLTE",
+ .stream_name = "VoLTE",
+ .cpu_dai_name = "VoLTE",
+ .platform_name = "msm-pcm-voice",
+ .dynamic = 1,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .be_id = MSM_FRONTEND_DAI_VOLTE,
+ },
+ /* Backend BT/FM DAI Links */
+ {
+ .name = LPASS_BE_INT_BT_SCO_RX,
+ .stream_name = "Internal BT-SCO Playback",
+ .cpu_dai_name = "msm-dai-q6.12288",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ .be_hw_params_fixup = msm8960_btsco_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_BT_SCO_TX,
+ .stream_name = "Internal BT-SCO Capture",
+ .cpu_dai_name = "msm-dai-q6.12289",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ .be_hw_params_fixup = msm8960_btsco_be_hw_params_fixup,
+ },
+ {
+ .name = LPASS_BE_INT_FM_RX,
+ .stream_name = "Internal FM Playback",
+ .cpu_dai_name = "msm-dai-q6.12292",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_RX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_INT_FM_TX,
+ .stream_name = "Internal FM Capture",
+ .cpu_dai_name = "msm-dai-q6.12293",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INT_FM_TX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ },
+ /* HDMI BACK END DAI Link */
+ {
+ .name = LPASS_BE_HDMI,
+ .stream_name = "HDMI Playback",
+ .cpu_dai_name = "msm-dai-q6-hdmi.8",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_HDMI_RX,
+ .be_hw_params_fixup = msm8960_hdmi_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ /* Backend AFE DAI Links */
+ {
+ .name = LPASS_BE_AFE_PCM_RX,
+ .stream_name = "AFE Playback",
+ .cpu_dai_name = "msm-dai-q6.224",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_RX,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_AFE_PCM_TX,
+ .stream_name = "AFE Capture",
+ .cpu_dai_name = "msm-dai-q6.225",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AFE_PCM_TX,
+ },
+ /* AUX PCM Backend DAI Links */
+ {
+ .name = LPASS_BE_AUXPCM_RX,
+ .stream_name = "AUX PCM Playback",
+ .cpu_dai_name = "msm-dai-q6.2",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_RX,
+ .be_hw_params_fixup = msm8960_auxpcm_be_params_fixup,
+ .ops = &msm8960_auxpcm_be_ops,
+ .ignore_pmdown_time = 1,
+ },
+ {
+ .name = LPASS_BE_AUXPCM_TX,
+ .stream_name = "AUX PCM Capture",
+ .cpu_dai_name = "msm-dai-q6.3",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_TX,
+ .be_hw_params_fixup = msm8960_auxpcm_be_params_fixup,
+ },
+ /* Incall Music BACK END DAI Link */
+ {
+ .name = LPASS_BE_VOICE_PLAYBACK_TX,
+ .stream_name = "Voice Farend Playback",
+ .cpu_dai_name = "msm-dai-q6.32773",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ },
+ /* Incall Record Uplink BACK END DAI Link */
+ {
+ .name = LPASS_BE_INCALL_RECORD_TX,
+ .stream_name = "Voice Uplink Capture",
+ .cpu_dai_name = "msm-dai-q6.32772",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ },
+ /* Incall Record Downlink BACK END DAI Link */
+ {
+ .name = LPASS_BE_INCALL_RECORD_RX,
+ .stream_name = "Voice Downlink Capture",
+ .cpu_dai_name = "msm-dai-q6.32771",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ .be_hw_params_fixup = msm8960_be_hw_params_fixup,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+};
+
+static struct snd_soc_dai_link msm8960_dai_delta_tabla1x[] = {
+ /* Backend DAI Links */
+ {
+ .name = LPASS_BE_SLIMBUS_0_RX,
+ .stream_name = "Slimbus Playback",
+ .cpu_dai_name = "msm-dai-q6.16384",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla1x_codec",
+ .codec_dai_name = "tabla_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ .init = &msm8960_audrx_init,
+ .be_hw_params_fixup = msm8960_slim_0_rx_be_hw_params_fixup,
+ .ops = &msm8960_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_0_TX,
+ .stream_name = "Slimbus Capture",
+ .cpu_dai_name = "msm-dai-q6.16385",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla1x_codec",
+ .codec_dai_name = "tabla_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ .be_hw_params_fixup = msm8960_slim_0_tx_be_hw_params_fixup,
+ .ops = &msm8960_be_ops,
+ },
+ {
+ .name = "SLIMBUS_2 Hostless",
+ .stream_name = "SLIMBUS_2 Hostless",
+ .cpu_dai_name = "msm-dai-q6.16389",
+ .platform_name = "msm-pcm-hostless",
+ .codec_name = "tabla1x_codec",
+ .codec_dai_name = "tabla_tx2",
+ .ignore_suspend = 1,
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ops = &msm8960_be_ops,
+ },
+};
+
+
+static struct snd_soc_dai_link msm8960_dai_delta_tabla2x[] = {
+ /* Backend DAI Links */
+ {
+ .name = LPASS_BE_SLIMBUS_0_RX,
+ .stream_name = "Slimbus Playback",
+ .cpu_dai_name = "msm-dai-q6.16384",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_rx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ .init = &msm8960_audrx_init,
+ .be_hw_params_fixup = msm8960_slim_0_rx_be_hw_params_fixup,
+ .ops = &msm8960_be_ops,
+ .ignore_pmdown_time = 1, /* this dainlink has playback support */
+ },
+ {
+ .name = LPASS_BE_SLIMBUS_0_TX,
+ .stream_name = "Slimbus Capture",
+ .cpu_dai_name = "msm-dai-q6.16385",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx1",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ .be_hw_params_fixup = msm8960_slim_0_tx_be_hw_params_fixup,
+ .ops = &msm8960_be_ops,
+ },
+ {
+ .name = "SLIMBUS_2 Hostless",
+ .stream_name = "SLIMBUS_2 Hostless",
+ .cpu_dai_name = "msm-dai-q6.16389",
+ .platform_name = "msm-pcm-hostless",
+ .codec_name = "tabla_codec",
+ .codec_dai_name = "tabla_tx2",
+ .ignore_suspend = 1,
+ .no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
+ .ops = &msm8960_be_ops,
+ },
+};
+
+static struct snd_soc_dai_link msm8960_tabla1x_dai[
+ ARRAY_SIZE(msm8960_dai_common) +
+ ARRAY_SIZE(msm8960_dai_delta_tabla1x)];
+
+
+static struct snd_soc_dai_link msm8960_dai[
+ ARRAY_SIZE(msm8960_dai_common) +
+ ARRAY_SIZE(msm8960_dai_delta_tabla2x)];
+
+static struct snd_soc_card snd_soc_tabla1x_card_msm8960 = {
+ .name = "msm8960-tabla1x-snd-card",
+ .dai_link = msm8960_tabla1x_dai,
+ .num_links = ARRAY_SIZE(msm8960_tabla1x_dai),
+ .controls = tabla_msm8960_controls,
+ .num_controls = ARRAY_SIZE(tabla_msm8960_controls),
+};
+
+static struct snd_soc_card snd_soc_card_msm8960 = {
+ .name = "msm8960-snd-card",
+ .dai_link = msm8960_dai,
+ .num_links = ARRAY_SIZE(msm8960_dai),
+ .controls = tabla_msm8960_controls,
+ .num_controls = ARRAY_SIZE(tabla_msm8960_controls),
+};
+
+static struct platform_device *msm8960_snd_device;
+static struct platform_device *msm8960_snd_tabla1x_device;
+
+static int msm8960_configure_headset_mic_gpios(void)
+{
+ int ret;
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ ret = gpio_request(PM8921_GPIO_PM_TO_SYS(23), "AV_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+
+ ret = pm8xxx_gpio_config(PM8921_GPIO_PM_TO_SYS(23), ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ PM8921_GPIO_PM_TO_SYS(23));
+ else
+ gpio_direction_output(PM8921_GPIO_PM_TO_SYS(23), 0);
+
+ ret = gpio_request(us_euro_sel_gpio, "US_EURO_SWITCH");
+ if (ret) {
+ pr_err("%s: Failed to request gpio %d\n", __func__,
+ us_euro_sel_gpio);
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ return ret;
+ }
+ ret = pm8xxx_gpio_config(us_euro_sel_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure gpio %d\n", __func__,
+ us_euro_sel_gpio);
+ else
+ gpio_direction_output(us_euro_sel_gpio, 0);
+
+ return 0;
+}
+static void msm8960_free_headset_mic_gpios(void)
+{
+ if (msm8960_headset_gpios_configured) {
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ gpio_free(us_euro_sel_gpio);
+ }
+}
+
+static int __init msm8960_audio_init(void)
+{
+ int ret;
+
+ if (!cpu_is_msm8960()) {
+ pr_debug("%s: Not the right machine type\n", __func__);
+ return -ENODEV ;
+ }
+
+ mbhc_cfg.calibration = def_tabla_mbhc_cal();
+ if (!mbhc_cfg.calibration) {
+ pr_err("Calibration data allocation failed\n");
+ return -ENOMEM;
+ }
+
+ msm8960_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm8960_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+
+ memcpy(msm8960_dai, msm8960_dai_common, sizeof(msm8960_dai_common));
+ memcpy(msm8960_dai + ARRAY_SIZE(msm8960_dai_common),
+ msm8960_dai_delta_tabla2x, sizeof(msm8960_dai_delta_tabla2x));
+
+ platform_set_drvdata(msm8960_snd_device, &snd_soc_card_msm8960);
+ ret = platform_device_add(msm8960_snd_device);
+ if (ret) {
+ platform_device_put(msm8960_snd_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+
+ msm8960_snd_tabla1x_device = platform_device_alloc("soc-audio", 1);
+ if (!msm8960_snd_tabla1x_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+
+ memcpy(msm8960_tabla1x_dai, msm8960_dai_common,
+ sizeof(msm8960_dai_common));
+ memcpy(msm8960_tabla1x_dai + ARRAY_SIZE(msm8960_dai_common),
+ msm8960_dai_delta_tabla1x, sizeof(msm8960_dai_delta_tabla1x));
+
+ platform_set_drvdata(msm8960_snd_tabla1x_device,
+ &snd_soc_tabla1x_card_msm8960);
+ ret = platform_device_add(msm8960_snd_tabla1x_device);
+ if (ret) {
+ platform_device_put(msm8960_snd_tabla1x_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+
+ if (msm8960_configure_headset_mic_gpios()) {
+ pr_err("%s Fail to configure headset mic gpios\n", __func__);
+ msm8960_headset_gpios_configured = 0;
+ } else
+ msm8960_headset_gpios_configured = 1;
+
+ mutex_init(&cdc_mclk_mutex);
+ return ret;
+
+}
+module_init(msm8960_audio_init);
+
+static void __exit msm8960_audio_exit(void)
+{
+ if (!cpu_is_msm8960()) {
+ pr_debug("%s: Not the right machine type\n", __func__);
+ return ;
+ }
+ msm8960_free_headset_mic_gpios();
+ platform_device_unregister(msm8960_snd_device);
+ platform_device_unregister(msm8960_snd_tabla1x_device);
+ kfree(mbhc_cfg.calibration);
+ mutex_destroy(&cdc_mclk_mutex);
+}
+module_exit(msm8960_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC MSM8960");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8974.c b/sound/soc/msm/msm8974.c
new file mode 100644
index 0000000..d47910b
--- /dev/null
+++ b/sound/soc/msm/msm8974.c
@@ -0,0 +1,752 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/mfd/pm8xxx/pm8921.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc-dsp.h>
+#include <sound/pcm.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <mach/socinfo.h>
+#include <qdsp6v2/msm-pcm-routing-v2.h>
+#include "../codecs/wcd9310.h"
+
+/* 8974 machine driver */
+
+#define PM8921_GPIO_BASE NR_GPIO_IRQS
+#define PM8921_GPIO_PM_TO_SYS(pm_gpio) (pm_gpio - 1 + PM8921_GPIO_BASE)
+
+#define MSM8974_SPK_ON 1
+#define MSM8974_SPK_OFF 0
+
+#define MSM_SLIM_0_RX_MAX_CHANNELS 2
+#define MSM_SLIM_0_TX_MAX_CHANNELS 4
+
+#define BTSCO_RATE_8KHZ 8000
+#define BTSCO_RATE_16KHZ 16000
+
+#define BOTTOM_SPK_AMP_POS 0x1
+#define BOTTOM_SPK_AMP_NEG 0x2
+#define TOP_SPK_AMP_POS 0x4
+#define TOP_SPK_AMP_NEG 0x8
+
+#define GPIO_AUX_PCM_DOUT 43
+#define GPIO_AUX_PCM_DIN 44
+#define GPIO_AUX_PCM_SYNC 45
+#define GPIO_AUX_PCM_CLK 46
+
+#define TABLA_EXT_CLK_RATE 12288000
+
+#define TABLA_MBHC_DEF_BUTTONS 8
+#define TABLA_MBHC_DEF_RLOADS 5
+
+/* Shared channel numbers for Slimbus ports that connect APQ to MDM. */
+enum {
+ SLIM_1_RX_1 = 145, /* BT-SCO and USB TX */
+ SLIM_1_TX_1 = 146, /* BT-SCO and USB RX */
+ SLIM_2_RX_1 = 147, /* HDMI RX */
+ SLIM_3_RX_1 = 148, /* In-call recording RX */
+ SLIM_3_RX_2 = 149, /* In-call recording RX */
+ SLIM_4_TX_1 = 150, /* In-call musid delivery TX */
+};
+
+static u32 top_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(18);
+static u32 bottom_spk_pamp_gpio = PM8921_GPIO_PM_TO_SYS(19);
+static int msm_spk_control;
+static int msm_ext_bottom_spk_pamp;
+static int msm_ext_top_spk_pamp;
+static int msm_slim_0_rx_ch = 1;
+static int msm_slim_0_tx_ch = 1;
+
+static int msm_btsco_rate = BTSCO_RATE_8KHZ;
+static int msm_headset_gpios_configured;
+
+static struct snd_soc_jack hs_jack;
+static struct snd_soc_jack button_jack;
+
+static int msm_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm);
+
+static struct tabla_mbhc_config mbhc_cfg = {
+ .headset_jack = &hs_jack,
+ .button_jack = &button_jack,
+ .read_fw_bin = false,
+ .calibration = NULL,
+ .micbias = TABLA_MICBIAS2,
+ .mclk_cb_fn = msm_enable_codec_ext_clk,
+ .mclk_rate = TABLA_EXT_CLK_RATE,
+ .gpio = 0, /* MBHC GPIO is not configured */
+ .gpio_irq = 0,
+ .gpio_level_insert = 1,
+};
+
+static void msm_enable_ext_spk_amp_gpio(u32 spk_amp_gpio)
+{
+ int ret = 0;
+
+ struct pm_gpio param = {
+ .direction = PM_GPIO_DIR_OUT,
+ .output_buffer = PM_GPIO_OUT_BUF_CMOS,
+ .output_value = 1,
+ .pull = PM_GPIO_PULL_NO,
+ .vin_sel = PM_GPIO_VIN_S4,
+ .out_strength = PM_GPIO_STRENGTH_MED,
+ .
+ function = PM_GPIO_FUNC_NORMAL,
+ };
+
+ if (spk_amp_gpio == bottom_spk_pamp_gpio) {
+
+ ret = gpio_request(bottom_spk_pamp_gpio, "BOTTOM_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting BOTTOM SPK AMP GPIO %u\n",
+ __func__, bottom_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(bottom_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Bottom Spk Ampl"
+ " gpio %u\n", __func__, bottom_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Bottom spkr amp gpio\n", __func__);
+ gpio_direction_output(bottom_spk_pamp_gpio, 1);
+ }
+
+ } else if (spk_amp_gpio == top_spk_pamp_gpio) {
+
+ ret = gpio_request(top_spk_pamp_gpio, "TOP_SPK_AMP");
+ if (ret) {
+ pr_err("%s: Error requesting GPIO %d\n", __func__,
+ top_spk_pamp_gpio);
+ return;
+ }
+ ret = pm8xxx_gpio_config(top_spk_pamp_gpio, ¶m);
+ if (ret)
+ pr_err("%s: Failed to configure Top Spk Ampl"
+ " gpio %u\n", __func__, top_spk_pamp_gpio);
+ else {
+ pr_debug("%s: enable Top spkr amp gpio\n", __func__);
+ gpio_direction_output(top_spk_pamp_gpio, 1);
+ }
+ } else {
+ pr_err("%s: ERROR : Invalid External Speaker Ampl GPIO."
+ " gpio = %u\n", __func__, spk_amp_gpio);
+ return;
+ }
+}
+
+static void msm_ext_spk_power_amp_on(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Bottom Speaker Ampl already "
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm_ext_bottom_spk_pamp |= spk;
+
+ if ((msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_POS) &&
+ (msm_ext_bottom_spk_pamp & BOTTOM_SPK_AMP_NEG)) {
+
+ msm_enable_ext_spk_amp_gpio(bottom_spk_pamp_gpio);
+ pr_debug("%s: slepping 4 ms after turning on external "
+ " Bottom Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ pr_debug("%s() External Top Speaker Ampl already"
+ "turned on. spk = 0x%08x\n", __func__, spk);
+ return;
+ }
+
+ msm_ext_top_spk_pamp |= spk;
+
+ if ((msm_ext_top_spk_pamp & TOP_SPK_AMP_POS) &&
+ (msm_ext_top_spk_pamp & TOP_SPK_AMP_NEG)) {
+
+ msm_enable_ext_spk_amp_gpio(top_spk_pamp_gpio);
+ pr_debug("%s: sleeping 4 ms after turning on "
+ " external Top Speaker Ampl\n", __func__);
+ usleep_range(4000, 4000);
+ }
+ } else {
+
+ pr_err("%s: ERROR : Invalid External Speaker Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm_ext_spk_power_amp_off(u32 spk)
+{
+ if (spk & (BOTTOM_SPK_AMP_POS | BOTTOM_SPK_AMP_NEG)) {
+
+ if (!msm_ext_bottom_spk_pamp)
+ return;
+
+ gpio_direction_output(bottom_spk_pamp_gpio, 0);
+ gpio_free(bottom_spk_pamp_gpio);
+ msm_ext_bottom_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Bottom"
+ " Speaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+
+ } else if (spk & (TOP_SPK_AMP_POS | TOP_SPK_AMP_NEG)) {
+
+ if (!msm_ext_top_spk_pamp)
+ return;
+
+ gpio_direction_output(top_spk_pamp_gpio, 0);
+ gpio_free(top_spk_pamp_gpio);
+ msm_ext_top_spk_pamp = 0;
+
+ pr_debug("%s: sleeping 4 ms after turning off external Top"
+ " Spkaker Ampl\n", __func__);
+
+ usleep_range(4000, 4000);
+ } else {
+
+ pr_err("%s: ERROR : Invalid Ext Spk Ampl. spk = 0x%08x\n",
+ __func__, spk);
+ return;
+ }
+}
+
+static void msm_ext_control(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
+ if (msm_spk_control == MSM8974_SPK_ON) {
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_enable_pin(dapm, "Ext Spk Top Neg");
+ } else {
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Bottom Neg");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Pos");
+ snd_soc_dapm_disable_pin(dapm, "Ext Spk Top Neg");
+ }
+
+ snd_soc_dapm_sync(dapm);
+}
+
+static int msm_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_spk_control = %d", __func__, msm_spk_control);
+ ucontrol->value.integer.value[0] = msm_spk_control;
+ return 0;
+}
+static int msm_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ pr_debug("%s()\n", __func__);
+ if (msm_spk_control == ucontrol->value.integer.value[0])
+ return 0;
+
+ msm_spk_control = ucontrol->value.integer.value[0];
+ msm_ext_control(codec);
+ return 1;
+}
+static int msm_spkramp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ pr_debug("%s() %x\n", __func__, SND_SOC_DAPM_EVENT_ON(event));
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm_ext_spk_power_amp_on(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm_ext_spk_power_amp_on(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm_ext_spk_power_amp_on(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+
+ } else {
+ if (!strncmp(w->name, "Ext Spk Bottom Pos", 18))
+ msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Bottom Neg", 18))
+ msm_ext_spk_power_amp_off(BOTTOM_SPK_AMP_NEG);
+ else if (!strncmp(w->name, "Ext Spk Top Pos", 15))
+ msm_ext_spk_power_amp_off(TOP_SPK_AMP_POS);
+ else if (!strncmp(w->name, "Ext Spk Top Neg", 15))
+ msm_ext_spk_power_amp_off(TOP_SPK_AMP_NEG);
+ else {
+ pr_err("%s() Invalid Speaker Widget = %s\n",
+ __func__, w->name);
+ return -EINVAL;
+ }
+ }
+ return 0;
+}
+
+static int msm_enable_codec_ext_clk(struct snd_soc_codec *codec, int enable,
+ bool dapm)
+{
+ return 0;
+}
+
+static int msm_mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget msm_dapm_widgets[] = {
+
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0,
+ msm_mclk_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Pos", msm_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Bottom Neg", msm_spkramp_event),
+
+ SND_SOC_DAPM_SPK("Ext Spk Top Pos", msm_spkramp_event),
+ SND_SOC_DAPM_SPK("Ext Spk Top Neg", msm_spkramp_event),
+
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("ANCRight Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("ANCLeft Headset Mic", NULL),
+
+ SND_SOC_DAPM_MIC("Digital Mic1", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic2", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic3", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic4", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic5", NULL),
+ SND_SOC_DAPM_MIC("Digital Mic6", NULL),
+
+};
+
+static const struct snd_soc_dapm_route common_audio_map[] = {
+
+ {"RX_BIAS", NULL, "MCLK"},
+ {"LDO_H", NULL, "MCLK"},
+
+ /* Speaker path */
+ {"Ext Spk Bottom Pos", NULL, "LINEOUT1"},
+ {"Ext Spk Bottom Neg", NULL, "LINEOUT3"},
+
+ {"Ext Spk Top Pos", NULL, "LINEOUT2"},
+ {"Ext Spk Top Neg", NULL, "LINEOUT4"},
+
+ /* Microphone path */
+ {"AMIC1", NULL, "MIC BIAS1 Internal1"},
+ {"MIC BIAS1 Internal1", NULL, "Handset Mic"},
+
+ {"AMIC2", NULL, "MIC BIAS2 External"},
+ {"MIC BIAS2 External", NULL, "Headset Mic"},
+
+ /**
+ * AMIC3 and AMIC4 inputs are connected to ANC microphones
+ * These mics are biased differently on CDP and FLUID
+ * routing entries below are based on bias arrangement
+ * on FLUID.
+ */
+ {"AMIC3", NULL, "MIC BIAS3 Internal1"},
+ {"MIC BIAS3 Internal1", NULL, "ANCRight Headset Mic"},
+
+ {"AMIC4", NULL, "MIC BIAS1 Internal2"},
+ {"MIC BIAS1 Internal2", NULL, "ANCLeft Headset Mic"},
+
+ {"HEADPHONE", NULL, "LDO_H"},
+
+ /**
+ * The digital Mic routes are setup considering
+ * fluid as default device.
+ */
+
+ /**
+ * Digital Mic1. Front Bottom left Digital Mic on Fluid and MTP.
+ * Digital Mic GM5 on CDP mainboard.
+ * Conncted to DMIC2 Input on Tabla codec.
+ */
+ {"DMIC2", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic1"},
+
+ /**
+ * Digital Mic2. Front Bottom right Digital Mic on Fluid and MTP.
+ * Digital Mic GM6 on CDP mainboard.
+ * Conncted to DMIC1 Input on Tabla codec.
+ */
+ {"DMIC1", NULL, "MIC BIAS1 External"},
+ {"MIC BIAS1 External", NULL, "Digital Mic2"},
+
+ /**
+ * Digital Mic3. Back Bottom Digital Mic on Fluid.
+ * Digital Mic GM1 on CDP mainboard.
+ * Conncted to DMIC4 Input on Tabla codec.
+ */
+ {"DMIC4", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic3"},
+
+ /**
+ * Digital Mic4. Back top Digital Mic on Fluid.
+ * Digital Mic GM2 on CDP mainboard.
+ * Conncted to DMIC3 Input on Tabla codec.
+ */
+ {"DMIC3", NULL, "MIC BIAS3 External"},
+ {"MIC BIAS3 External", NULL, "Digital Mic4"},
+
+ /**
+ * Digital Mic5. Front top Digital Mic on Fluid.
+ * Digital Mic GM3 on CDP mainboard.
+ * Conncted to DMIC5 Input on Tabla codec.
+ */
+ {"DMIC5", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic5"},
+
+ /* Tabla digital Mic6 - back bottom digital Mic on Liquid and
+ * bottom mic on CDP. FLUID/MTP do not have dmic6 installed.
+ */
+ {"DMIC6", NULL, "MIC BIAS4 External"},
+ {"MIC BIAS4 External", NULL, "Digital Mic6"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *slim0_rx_ch_text[] = {"One", "Two"};
+static const char *slim0_tx_ch_text[] = {"One", "Two", "Three", "Four"};
+
+static const struct soc_enum msm_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
+ SOC_ENUM_SINGLE_EXT(2, slim0_rx_ch_text),
+ SOC_ENUM_SINGLE_EXT(4, slim0_tx_ch_text),
+};
+
+static const char *btsco_rate_text[] = {"8000", "16000"};
+static const struct soc_enum msm_btsco_enum[] = {
+ SOC_ENUM_SINGLE_EXT(2, btsco_rate_text),
+};
+
+static int msm_slim_0_rx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
+ msm_slim_0_rx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_0_rx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_0_rx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_0_rx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_0_rx_ch = %d\n", __func__,
+ msm_slim_0_rx_ch);
+ return 1;
+}
+
+static int msm_slim_0_tx_ch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
+ msm_slim_0_tx_ch);
+ ucontrol->value.integer.value[0] = msm_slim_0_tx_ch - 1;
+ return 0;
+}
+
+static int msm_slim_0_tx_ch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ msm_slim_0_tx_ch = ucontrol->value.integer.value[0] + 1;
+
+ pr_debug("%s: msm_slim_0_tx_ch = %d\n", __func__,
+ msm_slim_0_tx_ch);
+ return 1;
+}
+
+static int msm_btsco_rate_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_debug("%s: msm_btsco_rate = %d", __func__,
+ msm_btsco_rate);
+ ucontrol->value.integer.value[0] = msm_btsco_rate;
+ return 0;
+}
+
+static int msm_btsco_rate_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ msm_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ case 1:
+ msm_btsco_rate = BTSCO_RATE_16KHZ;
+ break;
+ default:
+ msm_btsco_rate = BTSCO_RATE_8KHZ;
+ break;
+ }
+ pr_debug("%s: msm_btsco_rate = %d\n", __func__,
+ msm_btsco_rate);
+ return 0;
+}
+
+static const struct snd_kcontrol_new tabla_msm_controls[] = {
+ SOC_ENUM_EXT("Speaker Function", msm_enum[0], msm_get_spk,
+ msm_set_spk),
+ SOC_ENUM_EXT("SLIM_0_RX Channels", msm_enum[1],
+ msm_slim_0_rx_ch_get, msm_slim_0_rx_ch_put),
+ SOC_ENUM_EXT("SLIM_0_TX Channels", msm_enum[2],
+ msm_slim_0_tx_ch_get, msm_slim_0_tx_ch_put),
+};
+
+static const struct snd_kcontrol_new int_btsco_rate_mixer_controls[] = {
+ SOC_ENUM_EXT("Internal BTSCO SampleRate", msm_btsco_enum[0],
+ msm_btsco_rate_get, msm_btsco_rate_put),
+};
+
+static struct snd_soc_dsp_link lpa_fe_media = {
+ .playback = true,
+ .trigger = {
+ SND_SOC_DSP_TRIGGER_POST,
+ SND_SOC_DSP_TRIGGER_POST
+ },
+};
+static struct snd_soc_dsp_link fe_media = {
+ .playback = true,
+ .capture = true,
+ .trigger = {
+ SND_SOC_DSP_TRIGGER_POST,
+ SND_SOC_DSP_TRIGGER_POST
+ },
+};
+static int msm_auxpcm_be_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* PCM only supports mono output with 8khz sample rate */
+ rate->min = rate->max = 8000;
+ channels->min = channels->max = 1;
+
+ return 0;
+}
+static int msm_aux_pcm_get_gpios(void)
+{
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ ret = gpio_request(GPIO_AUX_PCM_DOUT, "AUX PCM DOUT");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DOUT",
+ __func__, GPIO_AUX_PCM_DOUT);
+ goto fail_dout;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_DIN, "AUX PCM DIN");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM DIN",
+ __func__, GPIO_AUX_PCM_DIN);
+ goto fail_din;
+ }
+
+ ret = gpio_request(GPIO_AUX_PCM_SYNC, "AUX PCM SYNC");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM SYNC",
+ __func__, GPIO_AUX_PCM_SYNC);
+ goto fail_sync;
+ }
+ ret = gpio_request(GPIO_AUX_PCM_CLK, "AUX PCM CLK");
+ if (ret < 0) {
+ pr_err("%s: Failed to request gpio(%d): AUX PCM CLK",
+ __func__, GPIO_AUX_PCM_CLK);
+ goto fail_clk;
+ }
+
+ return 0;
+
+fail_clk:
+ gpio_free(GPIO_AUX_PCM_SYNC);
+fail_sync:
+ gpio_free(GPIO_AUX_PCM_DIN);
+fail_din:
+ gpio_free(GPIO_AUX_PCM_DOUT);
+fail_dout:
+
+ return ret;
+}
+static int msm_aux_pcm_free_gpios(void)
+{
+ gpio_free(GPIO_AUX_PCM_DIN);
+ gpio_free(GPIO_AUX_PCM_DOUT);
+ gpio_free(GPIO_AUX_PCM_SYNC);
+ gpio_free(GPIO_AUX_PCM_CLK);
+
+ return 0;
+}
+
+static int msm_auxpcm_startup(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ ret = msm_aux_pcm_get_gpios();
+ if (ret < 0) {
+ pr_err("%s: Aux PCM GPIO request failed\n", __func__);
+ return -EINVAL;
+ }
+ return ret;
+}
+
+static void msm_auxpcm_shutdown(struct snd_pcm_substream *substream)
+{
+
+ pr_debug("%s(): substream = %s\n", __func__, substream->name);
+ msm_aux_pcm_free_gpios();
+}
+static struct snd_soc_ops msm_auxpcm_be_ops = {
+ .startup = msm_auxpcm_startup,
+ .shutdown = msm_auxpcm_shutdown,
+};
+/* Digital audio interface glue - connects codec <---> CPU */
+static struct snd_soc_dai_link msm_dai[] = {
+ /* FrontEnd DAI Links */
+ {
+ .name = "MSM8974 Media1",
+ .stream_name = "MultiMedia1",
+ .cpu_dai_name = "MultiMedia1",
+ .platform_name = "msm-pcm-dsp",
+ .dynamic = 1,
+ .dsp_link = &fe_media,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA1
+ },
+ {
+ .name = "MSM8974 LPA",
+ .stream_name = "LPA",
+ .cpu_dai_name = "MultiMedia3",
+ .platform_name = "msm-pcm-lpa",
+ .dynamic = 1,
+ .dsp_link = &lpa_fe_media,
+ .be_id = MSM_FRONTEND_DAI_MULTIMEDIA3,
+ },
+
+ /* AUX PCM Backend DAI Links */
+ {
+ .name = LPASS_BE_AUXPCM_RX,
+ .stream_name = "AUX PCM Playback",
+ .cpu_dai_name = "msm-dai-q6.4106",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-rx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_RX,
+ .be_hw_params_fixup = msm_auxpcm_be_params_fixup,
+ .ops = &msm_auxpcm_be_ops,
+ },
+ {
+ .name = LPASS_BE_AUXPCM_TX,
+ .stream_name = "AUX PCM Capture",
+ .cpu_dai_name = "msm-dai-q6.4107",
+ .platform_name = "msm-pcm-routing",
+ .codec_name = "msm-stub-codec.1",
+ .codec_dai_name = "msm-stub-tx",
+ .no_pcm = 1,
+ .be_id = MSM_BACKEND_DAI_AUXPCM_TX,
+ .be_hw_params_fixup = msm_auxpcm_be_params_fixup,
+ },
+
+};
+
+struct snd_soc_card snd_soc_card_msm = {
+ .name = "msm8974-taiko-snd-card",
+ .dai_link = msm_dai,
+ .num_links = ARRAY_SIZE(msm_dai),
+};
+
+static struct platform_device *msm_snd_device;
+
+static void msm_free_headset_mic_gpios(void)
+{
+ if (msm_headset_gpios_configured) {
+ gpio_free(PM8921_GPIO_PM_TO_SYS(23));
+ gpio_free(PM8921_GPIO_PM_TO_SYS(35));
+ }
+}
+
+static int __init msm_audio_init(void)
+{
+ int ret = 0;
+ if (!machine_is_copper_sim()) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return -ENODEV;
+ }
+ msm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!msm_snd_device) {
+ pr_err("Platform device allocation failed\n");
+ kfree(mbhc_cfg.calibration);
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(msm_snd_device, &snd_soc_card_msm);
+ ret = platform_device_add(msm_snd_device);
+ if (ret) {
+ platform_device_put(msm_snd_device);
+ kfree(mbhc_cfg.calibration);
+ return ret;
+ }
+ return ret;
+
+}
+module_init(msm_audio_init);
+
+static void __exit msm_audio_exit(void)
+{
+ if (!machine_is_copper_sim()) {
+ pr_err("%s: Not the right machine type\n", __func__);
+ return ;
+ }
+ msm_free_headset_mic_gpios();
+ platform_device_unregister(msm_snd_device);
+ kfree(mbhc_cfg.calibration);
+}
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("ALSA SoC msm");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8x60-dai.c b/sound/soc/msm/msm8x60-dai.c
new file mode 100644
index 0000000..8130f07
--- /dev/null
+++ b/sound/soc/msm/msm8x60-dai.c
@@ -0,0 +1,148 @@
+/* sound/soc/msm/msm-dai.c
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2010, Code Aurora Forum. All rights reserved.
+ *
+ * Derived from msm-pcm.c and msm7201.c.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include "msm8x60-pcm.h"
+
+static struct snd_soc_dai_driver msm_pcm_codec_dais[] = {
+{
+ .name = "msm-codec-dai",
+ .playback = {
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+};
+static struct snd_soc_dai_driver msm_pcm_cpu_dais[] = {
+{
+ .name = "msm-cpu-dai",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_msm = {
+ .compress_type = SND_SOC_FLAT_COMPRESSION,
+};
+
+static __devinit int asoc_msm_codec_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_msm,
+ msm_pcm_codec_dais, ARRAY_SIZE(msm_pcm_codec_dais));
+}
+
+static int __devexit asoc_msm_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static __devinit int asoc_msm_cpu_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_dai(&pdev->dev, msm_pcm_cpu_dais);
+}
+
+static int __devexit asoc_msm_cpu_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_msm_codec_driver = {
+ .probe = asoc_msm_codec_probe,
+ .remove = __devexit_p(asoc_msm_codec_remove),
+ .driver = {
+ .name = "msm-codec-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static struct platform_driver asoc_msm_cpu_driver = {
+ .probe = asoc_msm_cpu_probe,
+ .remove = __devexit_p(asoc_msm_cpu_remove),
+ .driver = {
+ .name = "msm-cpu-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_codec_dai_init(void)
+{
+ return platform_driver_register(&asoc_msm_codec_driver);
+}
+
+static void __exit msm_codec_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_msm_codec_driver);
+}
+
+static int __init msm_cpu_dai_init(void)
+{
+ return platform_driver_register(&asoc_msm_cpu_driver);
+}
+
+static void __exit msm_cpu_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_msm_cpu_driver);
+}
+
+module_init(msm_codec_dai_init);
+module_exit(msm_codec_dai_exit);
+module_init(msm_cpu_dai_init);
+module_exit(msm_cpu_dai_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM Codec/Cpu Dai driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8x60-pcm.c b/sound/soc/msm/msm8x60-pcm.c
new file mode 100644
index 0000000..6f5ad32
--- /dev/null
+++ b/sound/soc/msm/msm8x60-pcm.c
@@ -0,0 +1,806 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/delay.h>
+#include <linux/android_pmem.h>
+#include <mach/qdsp6v2/audio_dev_ctl.h>
+
+#include "msm8x60-pcm.h"
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 960 * 10,
+ .period_bytes_min = 960 * 5,
+ .period_bytes_max = 960 * 5,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+uint32_t in_frame_info[8][2];
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void alsa_out_listener(u32 evt_id, union auddev_evt_data *evt_payload,
+ void *private_data)
+{
+ int ret = 0;
+ struct msm_audio *prtd = (struct msm_audio *) private_data;
+ int dev_rate = 48000;
+ pr_debug("evt_id = 0x%8x\n", evt_id);
+ switch (evt_id) {
+ case AUDDEV_EVT_DEV_RDY:
+ pr_debug("AUDDEV_EVT_DEV_RDY\n");
+ prtd->copp_id = evt_payload->routing_id;
+ pr_debug("prtd->session_id = %d, copp_id= %d",
+ prtd->session_id, prtd->copp_id);
+ if (prtd->copp_id == PCM_RX)
+ dev_rate = 8000;
+
+ ret = msm_snddev_set_dec(prtd->session_id, prtd->copp_id, 1,
+ dev_rate, 1);
+ break;
+ case AUDDEV_EVT_DEV_RLS:
+ pr_debug("AUDDEV_EVT_DEV_RLS\n");
+ prtd->copp_id = evt_payload->routing_id;
+ pr_debug("prtd->session_id = %d, copp_id= %d",
+ prtd->session_id, prtd->copp_id);
+ if (prtd->copp_id == PCM_RX)
+ dev_rate = 8000;
+
+ ret = msm_snddev_set_dec(prtd->session_id, prtd->copp_id, 0,
+ dev_rate, 1);
+ break;
+ case AUDDEV_EVT_STREAM_VOL_CHG:
+ pr_debug("AUDDEV_EVT_STREAM_VOL_CHG\n");
+ break;
+ default:
+ pr_debug("Unknown Event\n");
+ break;
+ }
+}
+
+static void alsa_in_listener(u32 evt_id, union auddev_evt_data *evt_payload,
+ void *private_data)
+{
+ int ret = 0;
+ struct msm_audio *prtd = (struct msm_audio *) private_data;
+ int dev_rate = 48000;
+ pr_debug("evt_id = 0x%8x\n", evt_id);
+
+ switch (evt_id) {
+ case AUDDEV_EVT_DEV_RDY:
+ prtd->copp_id = evt_payload->routing_id;
+ if (prtd->copp_id == PCM_TX)
+ dev_rate = 8000;
+
+ ret = msm_snddev_set_enc(prtd->session_id, prtd->copp_id, 1,
+ dev_rate, 1);
+ break;
+ case AUDDEV_EVT_DEV_RLS:
+ prtd->copp_id = evt_payload->routing_id;
+ if (prtd->copp_id == PCM_TX)
+ dev_rate = 8000;
+
+ ret = msm_snddev_set_enc(prtd->session_id, prtd->copp_id, 0,
+ dev_rate, 1);
+ break;
+ default:
+ pr_debug("Unknown Event\n");
+ break;
+ }
+}
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ int i = 0;
+
+ pr_debug("%s\n", __func__);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag)
+ break;
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ break;
+ }
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+ pr_debug("token = 0x%08x\n", token);
+ for (i = 0; i < 8; i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ in_frame_info[token][0] = payload[2];
+ in_frame_info[token][1] = payload[3];
+ prtd->pcm_irq_pos += in_frame_info[token][0];
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag)
+ q6asm_read_nolock(prtd->audio_client);
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ if (!prtd->mmap_flag
+ && !atomic_read(&prtd->out_needed))
+ break;
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ break;
+ default:
+ break;
+ }
+ }
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+ int dev_rate = 48000;
+ int i = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_pcm(prtd->audio_client, runtime->rate,
+ runtime->channels);
+ if (ret < 0)
+ pr_debug("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+ atomic_set(&prtd->in_count, 0);
+ for (i = 0; i < MAX_COPP; i++) {
+ pr_debug("prtd->session_id = %d, copp_id= %d",
+ prtd->session_id, i);
+ if (session_route.playback_session[substream->number][i]
+ != DEVICE_IGNORE) {
+ pr_err("Device active\n");
+ if (i == PCM_RX)
+ dev_rate = 8000;
+ msm_snddev_set_dec(prtd->session_id,
+ i, 1, dev_rate, runtime->channels);
+ }
+ }
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ int i = 0;
+ int dev_rate = 48000;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return 0;
+
+ pr_debug("Samp_rate = %d\n", prtd->samp_rate);
+ pr_debug("Channel = %d\n", prtd->channel_mode);
+ ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client, prtd->samp_rate,
+ prtd->channel_mode);
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read_nolock(prtd->audio_client);
+ prtd->periods = runtime->periods;
+ for (i = 0; i < MAX_COPP; i++) {
+ pr_debug("prtd->session_id = %d, copp_id= %d",
+ prtd->session_id,
+ session_route.capture_session[prtd->session_id][i]);
+ if (session_route.capture_session[prtd->session_id][i]
+ != DEVICE_IGNORE) {
+ if (i == PCM_RX)
+ dev_rate = 8000;
+ msm_snddev_set_enc(prtd->session_id, i, 1, dev_rate, 1);
+ }
+ }
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("SNDRV_PCM_TRIGGER_START\n");
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->start, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ runtime->hw = msm_pcm_hardware;
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_debug("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm in open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* The session id returned by q6asm_open_read above is random and
+ * hence we cannot use the session id to route from user space.
+ * This results in need of a hardcoded session id for both playback
+ * and capture sessions. we can use the subdevice id to identify
+ * the session and use that for routing. Hence using
+ * substream->number as the session id for routing purpose. However
+ * DSP understands the session based on the allocated session id,
+ * hence using the variable prtd->session_id for all dsp commands.
+ */
+
+ prtd->session_id = prtd->audio_client->session;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->cmd_ack = 1;
+ prtd->device_events = AUDDEV_EVT_DEV_RDY |
+ AUDDEV_EVT_STREAM_VOL_CHG |
+ AUDDEV_EVT_DEV_RLS;
+ prtd->source = msm_snddev_route_dec(prtd->session_id);
+ pr_debug("Register device event listener for"
+ "SNDRV_PCM_STREAM_PLAYBACK session %d\n",
+ substream->number);
+ ret = auddev_register_evt_listner(prtd->device_events,
+ AUDDEV_CLNT_DEC, substream->number,
+ alsa_out_listener, (void *) prtd);
+ if (ret)
+ pr_debug("failed to register device event listener\n");
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ prtd->device_events = AUDDEV_EVT_DEV_RDY | AUDDEV_EVT_DEV_RLS |
+ AUDDEV_EVT_FREQ_CHG;
+ prtd->source = msm_snddev_route_enc(prtd->session_id);
+ pr_debug("Register device event listener for"
+ "SNDRV_PCM_STREAM_CAPTURE session %d\n",
+ substream->number);
+ ret = auddev_register_evt_listner(prtd->device_events,
+ AUDDEV_CLNT_ENC, substream->number,
+ alsa_in_listener, (void *) prtd);
+ if (ret)
+ pr_debug("failed to register device event listener\n");
+ }
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_debug("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ xfer = fbytes;
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write_nolock(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+fail:
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ ret = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (ret < 0)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ pr_debug("%s\n", __func__);
+ auddev_unregister_evt_listner(AUDDEV_CLNT_DEC,
+ substream->number);
+ pr_debug("%s\n", __func__);
+ msm_clear_session_id(prtd->session_id);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%08x\n",
+ (unsigned int) data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = in_frame_info[idx][1];
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ in_frame_info[idx][1] += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&in_frame_info[idx], 0,
+ sizeof(uint32_t) * 2);
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read_nolock(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ auddev_unregister_evt_listner(AUDDEV_CLNT_ENC,
+ substream->number);
+ msm_clear_session_id(prtd->session_id);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ pr_debug("%s: pcm_irq_pos = %d\n", __func__, prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+ dma_mmap_coherent(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return 0;
+}
+
+int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed \
+ rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret = 0;
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, 2);
+ if (ret)
+ return ret;
+ ret = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+ if (ret)
+ return ret;
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &msm_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &msm_pcm_ops);
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_pcm_new,
+};
+EXPORT_SYMBOL(msm_soc_platform);
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+ .driver = {
+ .name = "msm-dsp-audio",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm8x60-pcm.h b/sound/soc/msm/msm8x60-pcm.h
new file mode 100644
index 0000000..d7c9ad8
--- /dev/null
+++ b/sound/soc/msm/msm8x60-pcm.h
@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+#include <sound/apr_audio.h>
+#include <sound/q6asm.h>
+
+#define MAX_PLAYBACK_SESSIONS 2
+#define MAX_CAPTURE_SESSIONS 1
+#define MAX_COPP 12
+
+extern int copy_count;
+
+struct buffer {
+ void *data;
+ unsigned size;
+ unsigned used;
+ unsigned addr;
+};
+
+struct buffer_rec {
+ void *data;
+ unsigned int size;
+ unsigned int read;
+ unsigned int addr;
+};
+
+struct audio_locks {
+ wait_queue_head_t read_wait;
+ wait_queue_head_t write_wait;
+ wait_queue_head_t eos_wait;
+ wait_queue_head_t enable_wait;
+};
+
+extern struct audio_locks the_locks;
+
+struct msm_audio {
+ struct snd_pcm_substream *substream;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ uint16_t source; /* Encoding source bit mask */
+
+ struct audio_client *audio_client;
+
+ uint16_t session_id;
+ int copp_id;
+
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint32_t dsp_cnt;
+
+ uint32_t device_events; /* device events interested in */
+ int abort; /* set when error, like sample rate mismatch */
+
+ int enabled;
+ int close_ack;
+ int cmd_ack;
+ atomic_t start;
+ atomic_t out_count;
+ atomic_t in_count;
+ atomic_t out_needed;
+ int periods;
+ int mmap_flag;
+};
+
+struct pcm_session {
+ unsigned short playback_session[MAX_PLAYBACK_SESSIONS][MAX_COPP];
+ unsigned short capture_session[MAX_CAPTURE_SESSIONS][MAX_COPP];
+};
+
+/* platform data */
+extern struct snd_soc_platform_driver msm_soc_platform;
+extern struct pcm_session session_route;
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/msm8x60.c b/sound/soc/msm/msm8x60.c
new file mode 100644
index 0000000..48ce610
--- /dev/null
+++ b/sound/soc/msm/msm8x60.c
@@ -0,0 +1,1231 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6afe.h>
+#include <asm/dma.h>
+#include <asm/mach-types.h>
+#include <mach/qdsp6v2/audio_dev_ctl.h>
+#include <mach/qdsp6v2/q6voice.h>
+
+#define LOOPBACK_ENABLE 0x1
+#define LOOPBACK_DISABLE 0x0
+
+#include "msm8x60-pcm.h"
+
+static struct platform_device *msm_audio_snd_device;
+struct audio_locks the_locks;
+EXPORT_SYMBOL(the_locks);
+struct msm_volume msm_vol_ctl;
+EXPORT_SYMBOL(msm_vol_ctl);
+struct pcm_session session_route;
+EXPORT_SYMBOL(session_route);
+static struct snd_kcontrol_new snd_msm_controls[];
+
+char snddev_name[AUDIO_DEV_CTL_MAX_DEV][44];
+#define MSM_MAX_VOLUME 0x2000
+#define MSM_VOLUME_STEP ((MSM_MAX_VOLUME+17)/100) /* 17 added to avoid
+ more deviation */
+static int device_index; /* Count of Device controls */
+static int simple_control; /* Count of simple controls*/
+static int src_dev;
+static int dst_dev;
+static int loopback_status;
+
+static int msm_scontrol_count_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int msm_scontrol_count_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = simple_control;
+ return 0;
+}
+
+static int msm_v_call_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2; /* start, session_id */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = SESSION_ID_BASE + MAX_VOC_SESSIONS;
+ return 0;
+}
+
+static int msm_v_call_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ ucontrol->value.integer.value[1] = 0;
+ return 0;
+}
+
+static int msm_v_call_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int start = ucontrol->value.integer.value[0];
+ u32 session_id = ucontrol->value.integer.value[1];
+
+ if ((session_id != 0) &&
+ ((session_id < SESSION_ID_BASE) ||
+ (session_id >= SESSION_ID_BASE + MAX_VOC_SESSIONS))) {
+ pr_err("%s: Invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ if (start)
+ broadcast_event(AUDDEV_EVT_START_VOICE, DEVICE_IGNORE,
+ session_id);
+ else
+ broadcast_event(AUDDEV_EVT_END_VOICE, DEVICE_IGNORE,
+ session_id);
+ return 0;
+}
+
+static int msm_v_mute_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* dir, mute, session_id */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = SESSION_ID_BASE + MAX_VOC_SESSIONS;
+ return 0;
+}
+
+static int msm_v_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ ucontrol->value.integer.value[1] = 0;
+ ucontrol->value.integer.value[2] = 0;
+ return 0;
+}
+
+static int msm_v_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int dir = ucontrol->value.integer.value[0];
+ int mute = ucontrol->value.integer.value[1];
+ u32 session_id = ucontrol->value.integer.value[2];
+
+ if ((session_id != 0) &&
+ ((session_id < SESSION_ID_BASE) ||
+ (session_id >= SESSION_ID_BASE + MAX_VOC_SESSIONS))) {
+ pr_err("%s: Invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ return msm_set_voice_mute(dir, mute, session_id);
+}
+
+static int msm_v_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* dir, volume, session_id */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = SESSION_ID_BASE + MAX_VOC_SESSIONS;
+ return 0;
+}
+
+static int msm_v_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ ucontrol->value.integer.value[1] = 0;
+ ucontrol->value.integer.value[2] = 0;
+ return 0;
+}
+
+static int msm_v_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int dir = ucontrol->value.integer.value[0];
+ int volume = ucontrol->value.integer.value[1];
+ u32 session_id = ucontrol->value.integer.value[2];
+
+ if ((session_id != 0) &&
+ ((session_id < SESSION_ID_BASE) ||
+ (session_id >= SESSION_ID_BASE + MAX_VOC_SESSIONS))) {
+ pr_err("%s: Invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ return msm_set_voice_vol(dir, volume, session_id);
+}
+
+static int msm_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2; /* Volume */
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 16383;
+ return 0;
+}
+static int msm_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int ret = 0;
+ int session_id = ucontrol->value.integer.value[0];
+ int volume = ucontrol->value.integer.value[1];
+ int factor = ucontrol->value.integer.value[2];
+ u64 session_mask = 0;
+
+ if (factor > 10000)
+ return -EINVAL;
+
+ if ((volume < 0) || (volume/factor > 100))
+ return -EINVAL;
+
+ volume = (MSM_VOLUME_STEP * volume);
+
+ /* Convert back to original decimal point by removing the 10-base factor
+ * and discard the fractional portion
+ */
+
+ volume = volume/factor;
+
+ if (volume > MSM_MAX_VOLUME)
+ volume = MSM_MAX_VOLUME;
+
+ /* Only Decoder volume control supported */
+ session_mask = (((u64)0x1) << session_id) << (MAX_BIT_PER_CLIENT * \
+ ((int)AUDDEV_CLNT_DEC-1));
+ msm_vol_ctl.volume = volume;
+ pr_debug("%s:session_id %d, volume %d", __func__, session_id, volume);
+ broadcast_event(AUDDEV_EVT_STREAM_VOL_CHG, DEVICE_IGNORE,
+ session_mask);
+
+ return ret;
+}
+
+static int msm_voice_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* Device */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_voice_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ uint32_t rx_dev_id;
+ uint32_t tx_dev_id;
+ struct msm_snddev_info *rx_dev_info;
+ struct msm_snddev_info *tx_dev_info;
+ int set = ucontrol->value.integer.value[2];
+ u64 session_mask;
+
+ if (!set)
+ return -EPERM;
+ /* Rx Device Routing */
+ rx_dev_id = ucontrol->value.integer.value[0];
+ rx_dev_info = audio_dev_ctrl_find_dev(rx_dev_id);
+
+ if (IS_ERR(rx_dev_info)) {
+ pr_err("%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(rx_dev_info);
+ return rc;
+ }
+
+ if (!(rx_dev_info->capability & SNDDEV_CAP_RX)) {
+ pr_err("%s:First Dev is supposed to be RX\n", __func__);
+ return -EFAULT;
+ }
+
+ pr_debug("%s:route cfg %d STREAM_VOICE_RX type\n",
+ __func__, rx_dev_id);
+
+ msm_set_voc_route(rx_dev_info, AUDIO_ROUTE_STREAM_VOICE_RX,
+ rx_dev_id);
+
+ session_mask = ((u64)0x1) << (MAX_BIT_PER_CLIENT * \
+ ((int)AUDDEV_CLNT_VOC-1));
+
+ broadcast_event(AUDDEV_EVT_DEV_CHG_VOICE, rx_dev_id, session_mask);
+
+
+ /* Tx Device Routing */
+ tx_dev_id = ucontrol->value.integer.value[1];
+ tx_dev_info = audio_dev_ctrl_find_dev(tx_dev_id);
+
+ if (IS_ERR(tx_dev_info)) {
+ pr_err("%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(tx_dev_info);
+ return rc;
+ }
+
+ if (!(tx_dev_info->capability & SNDDEV_CAP_TX)) {
+ pr_err("%s:Second Dev is supposed to be Tx\n", __func__);
+ return -EFAULT;
+ }
+
+ pr_debug("%s:route cfg %d %d type\n",
+ __func__, tx_dev_id, AUDIO_ROUTE_STREAM_VOICE_TX);
+
+ msm_set_voc_route(tx_dev_info, AUDIO_ROUTE_STREAM_VOICE_TX,
+ tx_dev_id);
+
+ broadcast_event(AUDDEV_EVT_DEV_CHG_VOICE, tx_dev_id, session_mask);
+
+ if (rx_dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RDY, rx_dev_id, session_mask);
+
+ if (tx_dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RDY, tx_dev_id, session_mask);
+
+ return rc;
+}
+
+static int msm_voice_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ /* TODO: query Device list */
+ return 0;
+}
+
+static int msm_device_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1; /* Device */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_device_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ int set = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+ struct msm_snddev_info *dst_dev_info;
+ struct msm_snddev_info *src_dev_info;
+ int tx_freq = 0;
+ int rx_freq = 0;
+ u32 set_freq = 0;
+
+ set = ucontrol->value.integer.value[0];
+ route_cfg.dev_id = ucontrol->id.numid - device_index;
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+ if (IS_ERR(dev_info)) {
+ pr_err("%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(dev_info);
+ return rc;
+ }
+ pr_info("%s:device %s set %d\n", __func__, dev_info->name, set);
+
+ if (set) {
+ if (!dev_info->opened) {
+ set_freq = dev_info->sample_rate;
+ if (!msm_device_is_voice(route_cfg.dev_id)) {
+ msm_get_voc_freq(&tx_freq, &rx_freq);
+ if (dev_info->capability & SNDDEV_CAP_TX)
+ set_freq = tx_freq;
+
+ if (set_freq == 0)
+ set_freq = dev_info->sample_rate;
+ } else
+ set_freq = dev_info->sample_rate;
+
+
+ pr_err("%s:device freq =%d\n", __func__, set_freq);
+ rc = dev_info->dev_ops.set_freq(dev_info, set_freq);
+ if (rc < 0) {
+ pr_err("%s:device freq failed!\n", __func__);
+ return rc;
+ }
+ dev_info->set_sample_rate = rc;
+ rc = 0;
+ rc = dev_info->dev_ops.open(dev_info);
+ if (rc < 0) {
+ pr_err("%s:Enabling %s failed\n",
+ __func__, dev_info->name);
+ return rc;
+ }
+ dev_info->opened = 1;
+ broadcast_event(AUDDEV_EVT_DEV_RDY, route_cfg.dev_id,
+ SESSION_IGNORE);
+ if ((route_cfg.dev_id == src_dev) ||
+ (route_cfg.dev_id == dst_dev)) {
+ dst_dev_info = audio_dev_ctrl_find_dev(
+ dst_dev);
+ if (IS_ERR(dst_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(dst_dev_info);
+ return rc;
+ }
+ src_dev_info = audio_dev_ctrl_find_dev(
+ src_dev);
+ if (IS_ERR(src_dev_info)) {
+ pr_err("src_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(src_dev_info);
+ return rc;
+ }
+ if ((dst_dev_info->opened) &&
+ (src_dev_info->opened)) {
+ pr_debug("%d: Enable afe_loopback\n",
+ __LINE__);
+ afe_loopback(LOOPBACK_ENABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ loopback_status = 1;
+ }
+ }
+ }
+ } else {
+ if (dev_info->opened) {
+ broadcast_event(AUDDEV_EVT_REL_PENDING,
+ route_cfg.dev_id,
+ SESSION_IGNORE);
+ rc = dev_info->dev_ops.close(dev_info);
+ if (rc < 0) {
+ pr_err("%s:Snd device failed close!\n",
+ __func__);
+ return rc;
+ } else {
+ dev_info->opened = 0;
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ SESSION_IGNORE);
+ }
+ if (loopback_status == 1) {
+ if ((route_cfg.dev_id == src_dev) ||
+ (route_cfg.dev_id == dst_dev)) {
+ dst_dev_info = audio_dev_ctrl_find_dev(
+ dst_dev);
+ if (IS_ERR(dst_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(dst_dev_info);
+ return rc;
+ }
+ src_dev_info = audio_dev_ctrl_find_dev(
+ src_dev);
+ if (IS_ERR(src_dev_info)) {
+ pr_err("src_dev:%s:pass invalid"
+ "dev_id\n", __func__);
+ rc = PTR_ERR(src_dev_info);
+ return rc;
+ }
+ pr_debug("%d: Disable afe_loopback\n",
+ __LINE__);
+ afe_loopback(LOOPBACK_DISABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ loopback_status = 0;
+ }
+ }
+ }
+
+ }
+ return rc;
+}
+
+static int msm_device_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+
+ route_cfg.dev_id = ucontrol->id.numid - device_index;
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+
+ if (IS_ERR(dev_info)) {
+ pr_err("%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(dev_info);
+ return rc;
+ }
+
+ ucontrol->value.integer.value[0] = dev_info->copp_id;
+ ucontrol->value.integer.value[1] = dev_info->capability;
+
+ return 0;
+}
+
+static int msm_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3; /* Device */
+
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_route_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ /* TODO: query Device list */
+ return 0;
+}
+
+static int msm_route_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ int enc_freq = 0;
+ int requested_freq = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+ int session_id = ucontrol->value.integer.value[0];
+ int set = ucontrol->value.integer.value[2];
+ u64 session_mask = 0;
+ route_cfg.dev_id = ucontrol->value.integer.value[1];
+
+ if (ucontrol->id.numid == 2)
+ route_cfg.stream_type = AUDIO_ROUTE_STREAM_PLAYBACK;
+ else
+ route_cfg.stream_type = AUDIO_ROUTE_STREAM_REC;
+
+ pr_debug("%s:route cfg %d %d type for popp %d\n",
+ __func__, route_cfg.dev_id, route_cfg.stream_type, session_id);
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+
+ if (IS_ERR(dev_info)) {
+ pr_err("%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(dev_info);
+ return rc;
+ }
+ if (route_cfg.stream_type == AUDIO_ROUTE_STREAM_PLAYBACK) {
+ rc = msm_snddev_set_dec(session_id, dev_info->copp_id, set,
+ dev_info->sample_rate, dev_info->channel_mode);
+ session_mask =
+ (((u64)0x1) << session_id) << (MAX_BIT_PER_CLIENT * \
+ ((int)AUDDEV_CLNT_DEC-1));
+ if (!set) {
+ if (dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ } else {
+ dev_info->sessions = dev_info->sessions | session_mask;
+ if (dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RDY,
+ route_cfg.dev_id,
+ session_mask);
+ }
+ } else {
+
+ rc = msm_snddev_set_enc(session_id, dev_info->copp_id, set,
+ dev_info->sample_rate, dev_info->channel_mode);
+ session_mask =
+ (((u64)0x1) << session_id) << (MAX_BIT_PER_CLIENT * \
+ ((int)AUDDEV_CLNT_ENC-1));
+ if (!set) {
+ if (dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ } else {
+ dev_info->sessions = dev_info->sessions | session_mask;
+ enc_freq = msm_snddev_get_enc_freq(session_id);
+ requested_freq = enc_freq;
+ if (enc_freq > 0) {
+ rc = msm_snddev_request_freq(&enc_freq,
+ session_id,
+ SNDDEV_CAP_TX,
+ AUDDEV_CLNT_ENC);
+ pr_debug("%s:sample rate configured %d\
+ sample rate requested %d \n",
+ __func__, enc_freq, requested_freq);
+ if ((rc <= 0) || (enc_freq != requested_freq)) {
+ pr_debug("%s:msm_snddev_withdraw_freq\n",
+ __func__);
+ rc = msm_snddev_withdraw_freq
+ (session_id,
+ SNDDEV_CAP_TX, AUDDEV_CLNT_ENC);
+ broadcast_event(AUDDEV_EVT_FREQ_CHG,
+ route_cfg.dev_id,
+ SESSION_IGNORE);
+ }
+ }
+ if (dev_info->opened)
+ broadcast_event(AUDDEV_EVT_DEV_RDY,
+ route_cfg.dev_id,
+ session_mask);
+ }
+ }
+
+ if (rc < 0) {
+ pr_err("%s:device could not be assigned!\n", __func__);
+ return -EFAULT;
+ }
+
+ return rc;
+}
+
+static int msm_device_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ return 0;
+}
+
+static int msm_device_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct msm_snddev_info *dev_info;
+
+ int dev_id = ucontrol->value.integer.value[0];
+
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+ ucontrol->value.integer.value[0] = dev_info->dev_volume;
+
+ return 0;
+}
+
+static int msm_device_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = -EPERM;
+ struct msm_snddev_info *dev_info;
+
+ int dev_id = ucontrol->value.integer.value[0];
+ int volume = ucontrol->value.integer.value[1];
+
+ pr_debug("%s:dev_id = %d, volume = %d\n", __func__, dev_id, volume);
+
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+
+ if (IS_ERR(dev_info)) {
+ rc = PTR_ERR(dev_info);
+ pr_err("%s: audio_dev_ctrl_find_dev failed. %ld \n",
+ __func__, PTR_ERR(dev_info));
+ return rc;
+ }
+
+ pr_debug("%s:dev_name = %s dev_id = %d, volume = %d\n",
+ __func__, dev_info->name, dev_id, volume);
+
+ if (dev_info->dev_ops.set_device_volume)
+ rc = dev_info->dev_ops.set_device_volume(dev_info, volume);
+ else {
+ pr_info("%s : device %s does not support device volume "
+ "control.", __func__, dev_info->name);
+ return -EPERM;
+ }
+
+ return rc;
+}
+
+static int msm_reset_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0;
+ return 0;
+}
+
+static int msm_reset_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_reset_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ pr_err("%s:Resetting all devices\n", __func__);
+ return msm_reset_all_device();
+}
+
+static int msm_anc_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int msm_anc_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_anc_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = -EPERM;
+ struct msm_snddev_info *dev_info;
+
+ int dev_id = ucontrol->value.integer.value[0];
+ int enable = ucontrol->value.integer.value[1];
+
+ pr_debug("%s: dev_id = %d, enable = %d\n", __func__, dev_id, enable);
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+
+ if (IS_ERR(dev_info)) {
+ rc = PTR_ERR(dev_info);
+ pr_err("%s: audio_dev_ctrl_find_dev failed. %ld\n",
+ __func__, PTR_ERR(dev_info));
+ return rc;
+ }
+
+ if (dev_info->dev_ops.enable_anc) {
+ rc = dev_info->dev_ops.enable_anc(dev_info, enable);
+ } else {
+ pr_info("%s : device %s does not support anc control.",
+ __func__, dev_info->name);
+ return -EPERM;
+ }
+
+ return rc;
+}
+
+static int pcm_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ /*
+ * First parameter is session id ~ subdevice number
+ * Second parameter is device id.
+ */
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int pcm_route_get_rx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return 0;
+}
+
+static int pcm_route_get_tx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return 0;
+}
+
+static int pcm_route_put_rx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int session_id = 0;
+ int set = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+ u64 session_mask = 0;
+
+ /*
+ * session id is incremented by one and stored as session id 0
+ * is being used by dsp currently. whereas user space would use
+ * subdevice number as session id.
+ */
+ session_id = ucontrol->value.integer.value[0];
+ route_cfg.dev_id = ucontrol->value.integer.value[1];
+ set = ucontrol->value.integer.value[2];
+
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+ if (IS_ERR(dev_info)) {
+ pr_err("pass invalid dev_id %d\n", route_cfg.dev_id);
+ return PTR_ERR(dev_info);
+ }
+ if (!(dev_info->capability & SNDDEV_CAP_RX))
+ return -EINVAL;
+ session_mask =
+ (((u64)0x1) << session_id) << (MAX_BIT_PER_CLIENT * \
+ ((int)AUDDEV_CLNT_DEC-1));
+ if (!set) {
+ session_route.playback_session[session_id][dev_info->copp_id]
+ = DEVICE_IGNORE;
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ return 0;
+ }
+ pr_debug("%s:Routing playback session %d to %s\n",
+ __func__, (session_id),
+ dev_info->name);
+ session_route.playback_session[session_id][dev_info->copp_id] =
+ dev_info->copp_id;
+ if (dev_info->opened) {
+ dev_info->sessions = dev_info->sessions | session_mask;
+ broadcast_event(AUDDEV_EVT_DEV_RDY,
+ route_cfg.dev_id,
+ session_mask);
+ } else {
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ }
+ return 0;
+}
+
+static int pcm_route_put_tx(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int session_id = 0;
+ int set = 0;
+ struct msm_audio_route_config route_cfg;
+ struct msm_snddev_info *dev_info;
+ u64 session_mask = 0;
+
+ session_id = ucontrol->value.integer.value[0];
+ route_cfg.dev_id = ucontrol->value.integer.value[1];
+ set = ucontrol->value.integer.value[2];
+
+ dev_info = audio_dev_ctrl_find_dev(route_cfg.dev_id);
+ if (IS_ERR(dev_info)) {
+ pr_err("pass invalid dev_id %d\n", route_cfg.dev_id);
+ return PTR_ERR(dev_info);
+ }
+ pr_debug("%s:Routing capture session %d to %s\n", __func__,
+ session_id,
+ dev_info->name);
+ if (!(dev_info->capability & SNDDEV_CAP_TX))
+ return -EINVAL;
+ session_mask =
+ (((u64)0x1) << session_id) << (MAX_BIT_PER_CLIENT * \
+ ((int)AUDDEV_CLNT_ENC-1));
+ if (!set) {
+ session_route.capture_session[session_id][dev_info->copp_id]
+ = DEVICE_IGNORE;
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ return 0;
+ }
+
+ session_route.capture_session[session_id][dev_info->copp_id] =
+ dev_info->copp_id;
+ if (dev_info->opened) {
+ dev_info->sessions = dev_info->sessions | session_mask;
+ broadcast_event(AUDDEV_EVT_DEV_RDY,
+ route_cfg.dev_id,
+ session_mask);
+ } else {
+ broadcast_event(AUDDEV_EVT_DEV_RLS,
+ route_cfg.dev_id,
+ session_mask);
+ dev_info->sessions &= ~(session_mask);
+ }
+ return 0;
+}
+
+static int msm_loopback_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_loopback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = 0;
+ return 0;
+}
+
+static int msm_loopback_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct msm_snddev_info *src_dev_info = NULL; /* TX device */
+ struct msm_snddev_info *dst_dev_info = NULL; /* RX device */
+ int dst_dev_id = ucontrol->value.integer.value[0];
+ int src_dev_id = ucontrol->value.integer.value[1];
+ int set = ucontrol->value.integer.value[2];
+
+ pr_debug("%s: dst=%d :src=%d set=%d\n", __func__,
+ dst_dev_id, src_dev_id, set);
+
+ dst_dev_info = audio_dev_ctrl_find_dev(dst_dev_id);
+ if (IS_ERR(dst_dev_info)) {
+ pr_err("dst_dev:%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(dst_dev_info);
+ return rc;
+ }
+ if (!(dst_dev_info->capability & SNDDEV_CAP_RX)) {
+ pr_err("Destination device %d is not RX device\n",
+ dst_dev_id);
+ return -EFAULT;
+ }
+
+ src_dev_info = audio_dev_ctrl_find_dev(src_dev_id);
+ if (IS_ERR(src_dev_info)) {
+ pr_err("src_dev:%s:pass invalid dev_id\n", __func__);
+ rc = PTR_ERR(src_dev_info);
+ return rc;
+ }
+ if (!(src_dev_info->capability & SNDDEV_CAP_TX)) {
+ pr_err("Source device %d is not TX device\n", src_dev_id);
+ return -EFAULT;
+ }
+
+ if (set) {
+ pr_debug("%s:%d:Enabling AFE_Loopback\n", __func__, __LINE__);
+ src_dev = src_dev_id;
+ dst_dev = dst_dev_id;
+ loopback_status = 1;
+ if ((dst_dev_info->opened) && (src_dev_info->opened))
+ rc = afe_loopback(LOOPBACK_ENABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ } else {
+ pr_debug("%s:%d:Disabling AFE_Loopback\n", __func__, __LINE__);
+ src_dev = DEVICE_IGNORE;
+ dst_dev = DEVICE_IGNORE;
+ loopback_status = 0;
+ rc = afe_loopback(LOOPBACK_DISABLE,
+ dst_dev_info->copp_id,
+ src_dev_info->copp_id);
+ }
+ return rc;
+}
+static int msm_device_mute_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = msm_snddev_devcount();
+ return 0;
+}
+
+static int msm_device_mute_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return 0;
+}
+
+static int msm_device_mute_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int dev_id = ucontrol->value.integer.value[0];
+ int mute = ucontrol->value.integer.value[1];
+ struct msm_snddev_info *dev_info;
+ int rc = 0;
+ u16 gain = 0x2000;
+
+ dev_info = audio_dev_ctrl_find_dev(dev_id);
+ if (IS_ERR(dev_info)) {
+ rc = PTR_ERR(dev_info);
+ pr_err("%s: audio_dev_ctrl_find_dev failed. %ld\n",
+ __func__, PTR_ERR(dev_info));
+ return rc;
+ }
+ if (!(dev_info->capability & SNDDEV_CAP_TX)) {
+ rc = -EINVAL;
+ return rc;
+ }
+ if (mute)
+ gain = 0;
+
+ pr_debug("%s:dev_name = %s dev_id = %d, gain = %hX\n",
+ __func__, dev_info->name, dev_id, gain);
+ rc = afe_apply_gain(dev_info->copp_id, gain);
+ if (rc < 0) {
+ pr_err("%s : device %s not able to set device gain "
+ "control.", __func__, dev_info->name);
+ return rc;
+ }
+ pr_debug("Muting/Unmuting device id %d(%s)\n", dev_id, dev_info->name);
+
+ return rc;
+}
+
+static int msm_voc_session_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = SESSION_ID_BASE + MAX_VOC_SESSIONS;
+ return 0;
+}
+
+static int msm_voice_session_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] =
+ voice_get_session_id("Voice session");
+ return 0;
+}
+
+static int msm_voip_session_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = voice_get_session_id("VoIP session");
+ return 0;
+}
+
+static struct snd_kcontrol_new snd_dev_controls[AUDIO_DEV_CTL_MAX_DEV];
+
+static int snd_dev_ctl_index(int idx)
+{
+ struct msm_snddev_info *dev_info;
+
+ dev_info = audio_dev_ctrl_find_dev(idx);
+ if (IS_ERR(dev_info)) {
+ pr_err("%s:pass invalid dev_id\n", __func__);
+ return PTR_ERR(dev_info);
+ }
+ if (sizeof(dev_info->name) <= 44)
+ sprintf(&snddev_name[idx][0] , "%s", dev_info->name);
+
+ snd_dev_controls[idx].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ snd_dev_controls[idx].access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+ snd_dev_controls[idx].name = &snddev_name[idx][0];
+ snd_dev_controls[idx].index = idx;
+ snd_dev_controls[idx].info = msm_device_info;
+ snd_dev_controls[idx].get = msm_device_get;
+ snd_dev_controls[idx].put = msm_device_put;
+ snd_dev_controls[idx].private_value = 0;
+ return 0;
+
+}
+
+#define MSM_EXT(xname, fp_info, fp_get, fp_put, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .name = xname, \
+ .info = fp_info,\
+ .get = fp_get, .put = fp_put, \
+ .private_value = addr, \
+}
+
+/* If new controls are to be added which would be constant across the
+ * different targets, please add to the structure
+ * snd_msm_controls. Please do not add any controls to the structure
+ * snd_msm_secondary_controls defined below unless they are msm8x60
+ * specific.
+ */
+
+static struct snd_kcontrol_new snd_msm_controls[] = {
+ MSM_EXT("Count", msm_scontrol_count_info, msm_scontrol_count_get, \
+ NULL, 0),
+ MSM_EXT("Stream", msm_route_info, msm_route_get, \
+ msm_route_put, 0),
+ MSM_EXT("Record", msm_route_info, msm_route_get, \
+ msm_route_put, 0),
+ MSM_EXT("Voice", msm_voice_info, msm_voice_get, \
+ msm_voice_put, 0),
+ MSM_EXT("Volume", msm_volume_info, msm_volume_get, \
+ msm_volume_put, 0),
+ MSM_EXT("VoiceVolume", msm_v_volume_info, msm_v_volume_get, \
+ msm_v_volume_put, 0),
+ MSM_EXT("VoiceMute", msm_v_mute_info, msm_v_mute_get, \
+ msm_v_mute_put, 0),
+ MSM_EXT("Voice Call", msm_v_call_info, msm_v_call_get, \
+ msm_v_call_put, 0),
+ MSM_EXT("Device_Volume", msm_device_volume_info,
+ msm_device_volume_get, msm_device_volume_put, 0),
+ MSM_EXT("Reset", msm_reset_info,
+ msm_reset_get, msm_reset_put, 0),
+ MSM_EXT("ANC", msm_anc_info, msm_anc_get, msm_anc_put, 0),
+ MSM_EXT("Device_Mute", msm_device_mute_info,
+ msm_device_mute_get, msm_device_mute_put, 0),
+};
+
+static struct snd_kcontrol_new snd_msm_secondary_controls[] = {
+ MSM_EXT("PCM Playback Sink",
+ pcm_route_info, pcm_route_get_rx, pcm_route_put_rx, 0),
+ MSM_EXT("PCM Capture Source",
+ pcm_route_info, pcm_route_get_tx, pcm_route_put_tx, 0),
+ MSM_EXT("Sound Device Loopback", msm_loopback_info,
+ msm_loopback_get, msm_loopback_put, 0),
+ MSM_EXT("VoiceVolume Ext",
+ msm_v_volume_info, msm_v_volume_get, msm_v_volume_put, 0),
+ MSM_EXT("VoiceMute Ext",
+ msm_v_mute_info, msm_v_mute_get, msm_v_mute_put, 0),
+ MSM_EXT("Voice Call Ext",
+ msm_v_call_info, msm_v_call_get, msm_v_call_put, 0),
+ MSM_EXT("Voice session",
+ msm_voc_session_info, msm_voice_session_get, NULL, 0),
+ MSM_EXT("VoIP session",
+ msm_voc_session_info, msm_voip_session_get, NULL, 0),
+};
+
+static int msm_new_mixer(struct snd_soc_codec *codec)
+{
+ unsigned int idx;
+ int err;
+ int dev_cnt;
+
+ strcpy(codec->card->snd_card->mixername, "MSM Mixer");
+ for (idx = 0; idx < ARRAY_SIZE(snd_msm_controls); idx++) {
+ err = snd_ctl_add(codec->card->snd_card,
+ snd_ctl_new1(&snd_msm_controls[idx],
+ NULL));
+ if (err < 0)
+ pr_err("%s:ERR adding ctl\n", __func__);
+ }
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_msm_secondary_controls); idx++) {
+ err = snd_ctl_add(codec->card->snd_card,
+ snd_ctl_new1(&snd_msm_secondary_controls[idx],
+ NULL));
+ if (err < 0)
+ pr_err("%s:ERR adding secondary ctl\n", __func__);
+ }
+ dev_cnt = msm_snddev_devcount();
+
+ for (idx = 0; idx < dev_cnt; idx++) {
+ if (!snd_dev_ctl_index(idx)) {
+ err = snd_ctl_add(codec->card->snd_card,
+ snd_ctl_new1(&snd_dev_controls[idx],
+ NULL));
+ if (err < 0)
+ pr_err("%s:ERR adding ctl\n", __func__);
+ } else
+ return 0;
+ }
+ simple_control = ARRAY_SIZE(snd_msm_controls)
+ + ARRAY_SIZE(snd_msm_secondary_controls);
+ device_index = simple_control + 1;
+ return 0;
+}
+
+static int msm_soc_dai_init(
+ struct snd_soc_pcm_runtime *rtd)
+{
+
+ int ret = 0;
+ struct snd_soc_codec *codec = rtd->codec;
+
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+ memset(&session_route, DEVICE_IGNORE, sizeof(struct pcm_session));
+
+ ret = msm_new_mixer(codec);
+ if (ret < 0)
+ pr_err("%s: ALSA MSM Mixer Fail\n", __func__);
+
+ return ret;
+}
+
+static struct snd_soc_dai_link msm_dai[] = {
+{
+ .name = "MSM Primary I2S",
+ .stream_name = "DSP 1",
+ .cpu_dai_name = "msm-cpu-dai.0",
+ .platform_name = "msm-dsp-audio.0",
+ .codec_name = "msm-codec-dai.0",
+ .codec_dai_name = "msm-codec-dai",
+ .init = &msm_soc_dai_init,
+},
+#ifdef CONFIG_MSM_8x60_VOIP
+{
+ .name = "MSM Primary Voip",
+ .stream_name = "MVS",
+ .cpu_dai_name = "mvs-cpu-dai.0",
+ .platform_name = "msm-mvs-audio.0",
+ .codec_name = "mvs-codec-dai.0",
+ .codec_dai_name = "mvs-codec-dai",
+},
+#endif
+};
+
+static struct snd_soc_card snd_soc_card_msm = {
+ .name = "msm-audio",
+ .dai_link = msm_dai,
+ .num_links = ARRAY_SIZE(msm_dai),
+};
+
+static int __init msm_audio_init(void)
+{
+ int ret;
+
+ msm_audio_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!msm_audio_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(msm_audio_snd_device, &snd_soc_card_msm);
+ ret = platform_device_add(msm_audio_snd_device);
+ if (ret) {
+ platform_device_put(msm_audio_snd_device);
+ return ret;
+ }
+
+ src_dev = DEVICE_IGNORE;
+ dst_dev = DEVICE_IGNORE;
+
+ return ret;
+}
+
+static void __exit msm_audio_exit(void)
+{
+ platform_device_unregister(msm_audio_snd_device);
+}
+
+module_init(msm_audio_init);
+module_exit(msm_audio_exit);
+
+MODULE_DESCRIPTION("PCM module");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/msm_audio_mvs.h b/sound/soc/msm/msm_audio_mvs.h
new file mode 100644
index 0000000..728621a
--- /dev/null
+++ b/sound/soc/msm/msm_audio_mvs.h
@@ -0,0 +1,368 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+
+#ifndef __MSM_AUDIO_MVS_H
+#define __MSM_AUDIO_MVS_H
+#include <linux/msm_audio.h>
+#include <linux/wakelock.h>
+#include <mach/msm_rpcrouter.h>
+#include <mach/debug_mm.h>
+#include <linux/slab.h>
+
+
+#define AUDIO_GET_MVS_CONFIG _IOW(AUDIO_IOCTL_MAGIC, \
+ (AUDIO_MAX_COMMON_IOCTL_NUM + 0), unsigned)
+#define AUDIO_SET_MVS_CONFIG _IOR(AUDIO_IOCTL_MAGIC, \
+ (AUDIO_MAX_COMMON_IOCTL_NUM + 1), unsigned)
+#define AUDIO_SET_SCR_CONFIG _IOR(AUDIO_IOCTL_MAGIC, \
+ (AUDIO_MAX_COMMON_IOCTL_NUM + 2), unsigned)
+#define AUDIO_SET_DTX_CONFIG _IOR(AUDIO_IOCTL_MAGIC, \
+ (AUDIO_MAX_COMMON_IOCTL_NUM + 3), unsigned)
+/* MVS modes */
+#define MVS_MODE_LINEAR_PCM 9
+
+#define MVS_PROG 0x30000014
+#define MVS_VERS 0x00030001
+
+#define MVS_CLIENT_ID_VOIP 0x00000003 /* MVS_CLIENT_VOIP */
+
+#define MVS_ACQUIRE_PROC 4
+#define MVS_ENABLE_PROC 5
+#define MVS_RELEASE_PROC 6
+#define MVS_SET_PCM_MODE_PROC 9
+
+#define MVS_EVENT_CB_TYPE_PROC 1
+#define MVS_PACKET_UL_FN_TYPE_PROC 2
+#define MVS_PACKET_DL_FN_TYPE_PROC 3
+
+#define MVS_CB_FUNC_ID 0xAAAABBBB
+#define MVS_UL_CB_FUNC_ID 0xBBBBCCCC
+#define MVS_DL_CB_FUNC_ID 0xCCCCDDDD
+
+/* MVS frame modes */
+
+#define MVS_FRAME_MODE_PCM_UL 13
+#define MVS_FRAME_MODE_PCM_DL 14
+
+/* MVS context */
+#define MVS_PKT_CONTEXT_ISR 0x00000001
+
+/* Max voc packet size */
+#define MVS_MAX_VOC_PKT_SIZE 320
+
+#define VOIP_MAX_Q_LEN 20
+#define MVS_MAX_Q_LEN 8
+#define RPC_TYPE_REQUEST 0
+#define RPC_TYPE_REPLY 1
+
+#define RPC_STATUS_FAILURE 0
+#define RPC_STATUS_SUCCESS 1
+#define RPC_STATUS_REJECT 1
+
+
+#define RPC_COMMON_HDR_SZ (sizeof(uint32_t) * 2)
+#define RPC_REQUEST_HDR_SZ (sizeof(struct rpc_request_hdr))
+#define RPC_REPLY_HDR_SZ (sizeof(uint32_t) * 3)
+
+
+enum audio_mvs_state_type { AUDIO_MVS_CLOSED, AUDIO_MVS_OPENED,
+ AUDIO_MVS_PREPARING, AUDIO_MVS_ACQUIRE, AUDIO_MVS_ENABLED,
+ AUDIO_MVS_CLOSING
+};
+
+enum audio_mvs_event_type { AUDIO_MVS_COMMAND, AUDIO_MVS_MODE,
+ AUDIO_MVS_NOTIFY
+};
+
+enum audio_mvs_cmd_status_type { AUDIO_MVS_CMD_FAILURE, AUDIO_MVS_CMD_BUSY,
+ AUDIO_MVS_CMD_SUCCESS
+};
+
+enum audio_mvs_mode_status_type { AUDIO_MVS_MODE_NOT_AVAIL,
+ AUDIO_MVS_MODE_INIT, AUDIO_MVS_MODE_READY
+};
+
+enum audio_mvs_pkt_status_type { AUDIO_MVS_PKT_NORMAL, AUDIO_MVS_PKT_FAST,
+ AUDIO_MVS_PKT_SLOW
+};
+
+struct rpc_audio_mvs_acquire_args {
+ uint32_t client_id;
+ uint32_t cb_func_id;
+};
+
+struct audio_mvs_acquire_msg {
+ struct rpc_request_hdr rpc_hdr;
+ struct rpc_audio_mvs_acquire_args acquire_args;
+};
+
+struct rpc_audio_mvs_enable_args {
+ uint32_t client_id;
+ uint32_t mode;
+ uint32_t ul_cb_func_id;
+ uint32_t dl_cb_func_id;
+ uint32_t context;
+};
+
+struct audio_mvs_enable_msg {
+ struct rpc_request_hdr rpc_hdr;
+ struct rpc_audio_mvs_enable_args enable_args;
+};
+
+struct audio_mvs_release_msg {
+ struct rpc_request_hdr rpc_hdr;
+ uint32_t client_id;
+};
+
+struct audio_mvs_set_pcm_mode_msg {
+ struct rpc_request_hdr rpc_hdr;
+ uint32_t pcm_mode;
+};
+
+struct audio_mvs_set_pcmwb_mode_msg {
+ struct rpc_request_hdr rpc_hdr;
+ uint32_t pcmwb_mode;
+};
+
+struct audio_mvs_buffer {
+ uint8_t *voc_pkt;
+ uint32_t len;
+};
+
+union audio_mvs_event_data {
+ struct mvs_ev_command_type {
+ uint32_t event;
+ uint32_t client_id;
+ uint32_t cmd_status;
+ } mvs_ev_command_type;
+
+ struct mvs_ev_mode_type {
+ uint32_t event;
+ uint32_t client_id;
+ uint32_t mode_status;
+ uint32_t mode;
+ } mvs_ev_mode_type;
+
+ struct mvs_ev_notify_type {
+ uint32_t event;
+ uint32_t client_id;
+ uint32_t buf_dir;
+ uint32_t max_frames;
+ } mvs_ev_notify_type;
+};
+
+struct audio_mvs_cb_func_args {
+ uint32_t cb_func_id;
+ uint32_t valid_ptr;
+ uint32_t event;
+ union audio_mvs_event_data event_data;
+};
+
+struct audio_mvs_frame_info_hdr {
+ uint32_t frame_mode;
+ uint32_t mvs_mode;
+ uint32_t buf_free_cnt;
+};
+
+struct audio_mvs_ul_cb_func_args {
+ uint32_t cb_func_id;
+ uint32_t pkt_len;
+ uint32_t voc_pkt[MVS_MAX_VOC_PKT_SIZE / 4];
+
+ uint32_t valid_ptr;
+
+ uint32_t frame_mode;
+ uint32_t frame_mode_ignore;
+
+ struct audio_mvs_frame_info_hdr frame_info_hdr;
+
+ uint32_t pcm_frame;
+ uint32_t pcm_mode;
+
+ uint32_t pkt_len_ignore;
+};
+
+struct audio_mvs_ul_reply {
+ struct rpc_reply_hdr reply_hdr;
+ uint32_t valid_pkt_status_ptr;
+ uint32_t pkt_status;
+};
+
+struct audio_mvs_dl_cb_func_args {
+ uint32_t cb_func_id;
+ uint32_t valid_ptr;
+
+ uint32_t frame_mode;
+ uint32_t frame_mode_ignore;
+
+ struct audio_mvs_frame_info_hdr frame_info_hdr;
+
+ uint32_t pcm_frame;
+ uint32_t pcm_mode;
+
+};
+
+struct audio_mvs_dl_reply {
+ struct rpc_reply_hdr reply_hdr;
+ uint32_t voc_pkt[MVS_MAX_VOC_PKT_SIZE / 4];
+ uint32_t valid_frame_info_ptr;
+
+ uint32_t frame_mode;
+ uint32_t frame_mode_again;
+
+ struct audio_mvs_frame_info_hdr frame_info_hdr;
+
+ uint32_t pcm_frame;
+ uint32_t pcm_mode;
+
+ uint32_t valid_pkt_status_ptr;
+ uint32_t pkt_status;
+};
+
+struct audio_mvs_info_type {
+ enum audio_mvs_state_type state;
+ uint32_t frame_mode;
+ uint32_t mvs_mode;
+ uint32_t buf_free_cnt;
+ uint32_t pcm_frame;
+ uint32_t pcm_mode;
+ uint32_t out_sample_rate;
+ uint32_t out_channel_mode;
+ uint32_t out_weight;
+ uint32_t out_buffer_size;
+ int dl_play;
+ struct msm_rpc_endpoint *rpc_endpt;
+ uint32_t rpc_prog;
+ uint32_t rpc_ver;
+ uint32_t rpc_status;
+
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_playback_irq_pos; /* IRQ position */
+ unsigned int pcm_playback_buf_pos; /* position in buffer */
+
+ unsigned int pcm_capture_size;
+ unsigned int pcm_capture_count;
+ unsigned int pcm_capture_irq_pos; /* IRQ position */
+ unsigned int pcm_capture_buf_pos; /* position in buffer */
+
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+
+ uint8_t *mem_chunk;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ struct audio_mvs_buffer in[MVS_MAX_Q_LEN];
+ uint32_t in_read;
+ uint32_t in_write;
+
+ struct audio_mvs_buffer out[MVS_MAX_Q_LEN];
+ uint32_t out_read;
+ uint32_t out_write;
+
+ struct task_struct *task;
+
+ wait_queue_head_t wait;
+ wait_queue_head_t prepare_wait;
+ wait_queue_head_t out_wait;
+ wait_queue_head_t in_wait;
+
+
+ struct mutex lock;
+ struct mutex prepare_lock;
+ struct mutex in_lock;
+ struct mutex out_lock;
+
+ struct wake_lock suspend_lock;
+ struct wake_lock idle_lock;
+ struct timer_list timer;
+ unsigned long expiry;
+ int ack_dl_count;
+ int ack_ul_count;
+ int prepare_ack;
+ int playback_start;
+ int capture_start;
+ unsigned long expiry_delta;
+ int mvs_enable;
+ int playback_enable;
+ int capture_enable;
+ int instance;
+
+};
+
+struct audio_voip_info_type {
+ enum audio_mvs_state_type state;
+ enum audio_mvs_state_type playback_state;
+ enum audio_mvs_state_type capture_state;
+
+ unsigned int pcm_playback_size;
+ unsigned int pcm_count;
+ unsigned int pcm_playback_irq_pos; /* IRQ position */
+ unsigned int pcm_playback_buf_pos; /* position in buffer */
+
+ unsigned int pcm_capture_size;
+ unsigned int pcm_capture_count;
+ unsigned int pcm_capture_irq_pos; /* IRQ position */
+ unsigned int pcm_capture_buf_pos; /* position in buffer */
+
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ struct audio_mvs_buffer in[VOIP_MAX_Q_LEN];
+ uint32_t in_read;
+ uint32_t in_write;
+
+ struct audio_mvs_buffer out[VOIP_MAX_Q_LEN];
+ uint32_t out_read;
+ uint32_t out_write;
+
+ wait_queue_head_t out_wait;
+ wait_queue_head_t in_wait;
+
+ struct mutex lock;
+ struct mutex prepare_lock;
+
+ struct wake_lock suspend_lock;
+ struct wake_lock idle_lock;
+ int playback_start;
+ int capture_start;
+ int instance;
+};
+
+enum msm_audio_pcm_frame_type {
+ MVS_AMR_SPEECH_GOOD, /* Good speech frame */
+ MVS_AMR_SPEECH_DEGRADED, /* Speech degraded */
+ MVS_AMR_ONSET, /* onset */
+ MVS_AMR_SPEECH_BAD, /* Corrupt speech frame (bad CRC) */
+ MVS_AMR_SID_FIRST, /* First silence descriptor */
+ MVS_AMR_SID_UPDATE, /* Comfort noise frame */
+ MVS_AMR_SID_BAD, /* Corrupt SID frame (bad CRC) */
+ MVS_AMR_NO_DATA, /* Nothing to transmit */
+ MVS_AMR_SPEECH_LOST, /* downlink speech lost */
+};
+
+enum msm_audio_dtx_mode_type { MVS_DTX_OFF, MVS_DTX_ON
+};
+
+struct msm_audio_mvs_config {
+ uint32_t mvs_mode;
+ uint32_t bit_rate;
+};
+
+extern struct snd_soc_dai_driver msm_mvs_dais[2];
+extern struct snd_soc_codec_device soc_codec_dev_msm_mvs;
+extern struct snd_soc_platform_driver msm_mvs_soc_platform;
+extern struct snd_soc_platform_driver msm_voip_soc_platform;
+#endif /* __MSM_AUDIO_MVS_H */
diff --git a/sound/soc/msm/mvs-dai.c b/sound/soc/msm/mvs-dai.c
new file mode 100644
index 0000000..521c5e5
--- /dev/null
+++ b/sound/soc/msm/mvs-dai.c
@@ -0,0 +1,141 @@
+/* Copyright (c) 2010, Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include "msm_audio_mvs.h"
+
+static struct snd_soc_dai_driver msm_mvs_codec_dais[] = {
+{
+ .name = "mvs-codec-dai",
+ .playback = {
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_8000),
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_8000),
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+};
+static struct snd_soc_dai_driver msm_mvs_cpu_dais[] = {
+{
+ .name = "mvs-cpu-dai",
+ .playback = {
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_8000),
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_max = 2,
+ .rates = (SNDRV_PCM_RATE_8000),
+ .rate_min = 8000,
+ .rate_max = 8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_msm = {
+ .compress_type = SND_SOC_FLAT_COMPRESSION,
+};
+
+static __devinit int asoc_mvs_codec_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_msm,
+ msm_mvs_codec_dais, ARRAY_SIZE(msm_mvs_codec_dais));
+}
+
+static int __devexit asoc_mvs_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static __devinit int asoc_mvs_cpu_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_dai(&pdev->dev, msm_mvs_cpu_dais);
+}
+
+static int __devexit asoc_mvs_cpu_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver asoc_mvs_codec_driver = {
+ .probe = asoc_mvs_codec_probe,
+ .remove = __devexit_p(asoc_mvs_codec_remove),
+ .driver = {
+ .name = "mvs-codec-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static struct platform_driver asoc_mvs_cpu_driver = {
+ .probe = asoc_mvs_cpu_probe,
+ .remove = __devexit_p(asoc_mvs_cpu_remove),
+ .driver = {
+ .name = "mvs-cpu-dai",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init mvs_codec_dai_init(void)
+{
+ return platform_driver_register(&asoc_mvs_codec_driver);
+}
+
+static void __exit mvs_codec_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_mvs_codec_driver);
+}
+
+static int __init mvs_cpu_dai_init(void)
+{
+ return platform_driver_register(&asoc_mvs_cpu_driver);
+}
+
+static void __exit mvs_cpu_dai_exit(void)
+{
+ platform_driver_unregister(&asoc_mvs_cpu_driver);
+}
+
+module_init(mvs_codec_dai_init);
+module_exit(mvs_codec_dai_exit);
+module_init(mvs_cpu_dai_init);
+module_exit(mvs_cpu_dai_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM Codec/Cpu Dai driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6/Makefile b/sound/soc/msm/qdsp6/Makefile
new file mode 100644
index 0000000..f45bd33
--- /dev/null
+++ b/sound/soc/msm/qdsp6/Makefile
@@ -0,0 +1,2 @@
+obj-y := q6asm.o q6adm.o q6afe.o
+obj-$(CONFIG_SND_SOC_VOICE) += q6voice.o
\ No newline at end of file
diff --git a/sound/soc/msm/qdsp6/q6adm.c b/sound/soc/msm/qdsp6/q6adm.c
new file mode 100644
index 0000000..bf6f743
--- /dev/null
+++ b/sound/soc/msm/qdsp6/q6adm.c
@@ -0,0 +1,1088 @@
+/* Copyright (c) 2010-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/sched.h>
+#include <linux/jiffies.h>
+#include <linux/uaccess.h>
+#include <linux/atomic.h>
+
+#include <mach/qdsp6v2/audio_dev_ctl.h>
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <mach/qdsp6v2/rtac.h>
+
+#include <sound/apr_audio.h>
+#include <sound/q6afe.h>
+
+#define TIMEOUT_MS 1000
+#define AUDIO_RX 0x0
+#define AUDIO_TX 0x1
+
+#define ASM_MAX_SESSION 0x8 /* To do: define in a header */
+#define RESET_COPP_ID 99
+#define INVALID_COPP_ID 0xFF
+
+struct adm_ctl {
+ void *apr;
+ atomic_t copp_id[AFE_MAX_PORTS];
+ atomic_t copp_cnt[AFE_MAX_PORTS];
+ atomic_t copp_stat[AFE_MAX_PORTS];
+ wait_queue_head_t wait;
+};
+
+static struct acdb_cal_block mem_addr_audproc[MAX_AUDPROC_TYPES];
+static struct acdb_cal_block mem_addr_audvol[MAX_AUDPROC_TYPES];
+
+static struct adm_ctl this_adm;
+
+int srs_trumedia_open(int port_id, int srs_tech_id, void *srs_params)
+{
+ struct asm_pp_params_command *open = NULL;
+ int ret = 0, sz = 0;
+ int index;
+
+ pr_debug("SRS - %s", __func__);
+ switch (srs_tech_id) {
+ case SRS_ID_GLOBAL: {
+ struct srs_trumedia_params_GLOBAL *glb_params = NULL;
+ sz = sizeof(struct asm_pp_params_command) +
+ sizeof(struct srs_trumedia_params_GLOBAL);
+ open = kzalloc(sz, GFP_KERNEL);
+ open->payload_size = sizeof(struct srs_trumedia_params_GLOBAL) +
+ sizeof(struct asm_pp_param_data_hdr);
+ open->params.param_id = SRS_TRUMEDIA_PARAMS;
+ open->params.param_size =
+ sizeof(struct srs_trumedia_params_GLOBAL);
+ glb_params = (struct srs_trumedia_params_GLOBAL *)((u8 *)open +
+ sizeof(struct asm_pp_params_command));
+ memcpy(glb_params, srs_params,
+ sizeof(struct srs_trumedia_params_GLOBAL));
+ pr_debug("SRS - %s: Global params - 1 = %x, 2 = %x, 3 = %x,"
+ " 4 = %x, 5 = %x, 6 = %x, 7 = %x, 8 = %x\n",
+ __func__, (int)glb_params->v1,
+ (int)glb_params->v2, (int)glb_params->v3,
+ (int)glb_params->v4, (int)glb_params->v5,
+ (int)glb_params->v6, (int)glb_params->v7,
+ (int)glb_params->v8);
+ break;
+ }
+ case SRS_ID_WOWHD: {
+ struct srs_trumedia_params_WOWHD *whd_params = NULL;
+ sz = sizeof(struct asm_pp_params_command) +
+ sizeof(struct srs_trumedia_params_WOWHD);
+ open = kzalloc(sz, GFP_KERNEL);
+ open->payload_size = sizeof(struct srs_trumedia_params_WOWHD) +
+ sizeof(struct asm_pp_param_data_hdr);
+ open->params.param_id = SRS_TRUMEDIA_PARAMS_WOWHD;
+ open->params.param_size =
+ sizeof(struct srs_trumedia_params_WOWHD);
+ whd_params = (struct srs_trumedia_params_WOWHD *)((u8 *)open +
+ sizeof(struct asm_pp_params_command));
+ memcpy(whd_params, srs_params,
+ sizeof(struct srs_trumedia_params_WOWHD));
+ pr_debug("SRS - %s: WOWHD params - 1 = %x, 2 = %x, 3 = %x,"
+ " 4 = %x, 5 = %x, 6 = %x, 7 = %x, 8 = %x, 9 = %x,"
+ " 10 = %x, 11 = %x\n", __func__, (int)whd_params->v1,
+ (int)whd_params->v2, (int)whd_params->v3,
+ (int)whd_params->v4, (int)whd_params->v5,
+ (int)whd_params->v6, (int)whd_params->v7,
+ (int)whd_params->v8, (int)whd_params->v9,
+ (int)whd_params->v10, (int)whd_params->v11);
+ break;
+ }
+ case SRS_ID_CSHP: {
+ struct srs_trumedia_params_CSHP *chp_params = NULL;
+ sz = sizeof(struct asm_pp_params_command) +
+ sizeof(struct srs_trumedia_params_CSHP);
+ open = kzalloc(sz, GFP_KERNEL);
+ open->payload_size = sizeof(struct srs_trumedia_params_CSHP) +
+ sizeof(struct asm_pp_param_data_hdr);
+ open->params.param_id = SRS_TRUMEDIA_PARAMS_CSHP;
+ open->params.param_size =
+ sizeof(struct srs_trumedia_params_CSHP);
+ chp_params = (struct srs_trumedia_params_CSHP *)((u8 *)open +
+ sizeof(struct asm_pp_params_command));
+ memcpy(chp_params, srs_params,
+ sizeof(struct srs_trumedia_params_CSHP));
+ pr_debug("SRS - %s: CSHP params - 1 = %x, 2 = %x, 3 = %x,"
+ " 4 = %x, 5 = %x, 6 = %x, 7 = %x, 8 = %x,"
+ " 9 = %x\n", __func__, (int)chp_params->v1,
+ (int)chp_params->v2, (int)chp_params->v3,
+ (int)chp_params->v4, (int)chp_params->v5,
+ (int)chp_params->v6, (int)chp_params->v7,
+ (int)chp_params->v8, (int)chp_params->v9);
+ break;
+ }
+ case SRS_ID_HPF: {
+ struct srs_trumedia_params_HPF *hpf_params = NULL;
+ sz = sizeof(struct asm_pp_params_command) +
+ sizeof(struct srs_trumedia_params_HPF);
+ open = kzalloc(sz, GFP_KERNEL);
+ open->payload_size = sizeof(struct srs_trumedia_params_HPF) +
+ sizeof(struct asm_pp_param_data_hdr);
+ open->params.param_id = SRS_TRUMEDIA_PARAMS_HPF;
+ open->params.param_size =
+ sizeof(struct srs_trumedia_params_HPF);
+ hpf_params = (struct srs_trumedia_params_HPF *)((u8 *)open +
+ sizeof(struct asm_pp_params_command));
+ memcpy(hpf_params, srs_params,
+ sizeof(struct srs_trumedia_params_HPF));
+ pr_debug("SRS - %s: HPF params - 1 = %x\n", __func__,
+ (int)hpf_params->v1);
+ break;
+ }
+ case SRS_ID_PEQ: {
+ struct srs_trumedia_params_PEQ *peq_params = NULL;
+ sz = sizeof(struct asm_pp_params_command) +
+ sizeof(struct srs_trumedia_params_PEQ);
+ open = kzalloc(sz, GFP_KERNEL);
+ open->payload_size = sizeof(struct srs_trumedia_params_PEQ) +
+ sizeof(struct asm_pp_param_data_hdr);
+ open->params.param_id = SRS_TRUMEDIA_PARAMS_PEQ;
+ open->params.param_size =
+ sizeof(struct srs_trumedia_params_PEQ);
+ peq_params = (struct srs_trumedia_params_PEQ *)((u8 *)open +
+ sizeof(struct asm_pp_params_command));
+ memcpy(peq_params, srs_params,
+ sizeof(struct srs_trumedia_params_PEQ));
+ pr_debug("SRS - %s: PEQ params - 1 = %x 2 = %x, 3 = %x,"
+ " 4 = %x\n", __func__, (int)peq_params->v1,
+ (int)peq_params->v2, (int)peq_params->v3,
+ (int)peq_params->v4);
+ break;
+ }
+ case SRS_ID_HL: {
+ struct srs_trumedia_params_HL *hl_params = NULL;
+ sz = sizeof(struct asm_pp_params_command) +
+ sizeof(struct srs_trumedia_params_HL);
+ open = kzalloc(sz, GFP_KERNEL);
+ open->payload_size = sizeof(struct srs_trumedia_params_HL) +
+ sizeof(struct asm_pp_param_data_hdr);
+ open->params.param_id = SRS_TRUMEDIA_PARAMS_HL;
+ open->params.param_size = sizeof(struct srs_trumedia_params_HL);
+ hl_params = (struct srs_trumedia_params_HL *)((u8 *)open +
+ sizeof(struct asm_pp_params_command));
+ memcpy(hl_params, srs_params,
+ sizeof(struct srs_trumedia_params_HL));
+ pr_debug("SRS - %s: HL params - 1 = %x, 2 = %x, 3 = %x, 4 = %x,"
+ " 5 = %x, 6 = %x, 7 = %x\n", __func__,
+ (int)hl_params->v1, (int)hl_params->v2,
+ (int)hl_params->v3, (int)hl_params->v4,
+ (int)hl_params->v5, (int)hl_params->v6,
+ (int)hl_params->v7);
+ break;
+ }
+ default:
+ goto fail_cmd;
+ }
+
+ open->payload = NULL;
+ open->params.module_id = SRS_TRUMEDIA_MODULE_ID;
+ open->params.reserved = 0;
+ open->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ open->hdr.pkt_size = sz;
+ open->hdr.src_svc = APR_SVC_ADM;
+ open->hdr.src_domain = APR_DOMAIN_APPS;
+ open->hdr.src_port = port_id;
+ open->hdr.dest_svc = APR_SVC_ADM;
+ open->hdr.dest_domain = APR_DOMAIN_ADSP;
+ index = afe_get_port_index(port_id);
+ open->hdr.dest_port = atomic_read(&this_adm.copp_id[index]);
+ open->hdr.token = port_id;
+ open->hdr.opcode = ADM_CMD_SET_PARAMS;
+ pr_debug("SRS - %s: Command was sent now check Q6 - port id = %d,"
+ " size %d, module id %x, param id %x.\n", __func__,
+ open->hdr.dest_port, open->payload_size,
+ open->params.module_id, open->params.param_id);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)open);
+ if (ret < 0) {
+ pr_err("SRS - %s: ADM enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ /* Wait for the callback with copp id */
+ ret = wait_event_timeout(this_adm.wait, 1,
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("SRS - %s: ADM open failed for port %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+fail_cmd:
+ kfree(open);
+ return ret;
+}
+
+static int32_t adm_callback(struct apr_client_data *data, void *priv)
+{
+ uint32_t *payload;
+ int i, index;
+ payload = data->payload;
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("adm_callback: Reset event is received: %d %d apr[%p]\n",
+ data->reset_event, data->reset_proc,
+ this_adm.apr);
+ if (this_adm.apr) {
+ apr_reset(this_adm.apr);
+ for (i = 0; i < AFE_MAX_PORTS; i++) {
+ atomic_set(&this_adm.copp_id[i],
+ RESET_COPP_ID);
+ atomic_set(&this_adm.copp_cnt[i], 0);
+ atomic_set(&this_adm.copp_stat[i], 0);
+ }
+ this_adm.apr = NULL;
+ }
+ return 0;
+ }
+
+ pr_debug("%s: code = 0x%x %x %x size = %d\n", __func__,
+ data->opcode, payload[0], payload[1],
+ data->payload_size);
+
+ if (data->payload_size) {
+ index = afe_get_port_index(data->token);
+ pr_debug("%s: Port ID %d, index %d\n", __func__,
+ data->token, index);
+ if (index < 0 || index >= AFE_MAX_PORTS) {
+ pr_err("%s: invalid port idx %d token %d\n",
+ __func__, index, data->token);
+ return 0;
+ }
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ pr_debug("APR_BASIC_RSP_RESULT id %x\n", payload[0]);
+ switch (payload[0]) {
+ case ADM_CMD_SET_PARAMS:
+ if (rtac_make_adm_callback(payload,
+ data->payload_size))
+ break;
+ case ADM_CMD_COPP_CLOSE:
+ case ADM_CMD_MEMORY_MAP:
+ case ADM_CMD_MEMORY_UNMAP:
+ case ADM_CMD_MEMORY_MAP_REGIONS:
+ case ADM_CMD_MEMORY_UNMAP_REGIONS:
+ case ADM_CMD_MATRIX_MAP_ROUTINGS:
+ case ADM_CMD_CONNECT_AFE_PORT:
+ atomic_set(&this_adm.copp_stat[index], 1);
+ wake_up(&this_adm.wait);
+ break;
+ default:
+ pr_err("%s: Unknown Cmd: 0x%x\n", __func__,
+ payload[0]);
+ break;
+ }
+ return 0;
+ }
+
+ switch (data->opcode) {
+ case ADM_CMDRSP_COPP_OPEN:
+ case ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN: {
+ struct adm_copp_open_respond *open = data->payload;
+ if (open->copp_id == INVALID_COPP_ID) {
+ pr_err("%s: invalid coppid rxed %d\n",
+ __func__, open->copp_id);
+ atomic_set(&this_adm.copp_stat[index], 1);
+ wake_up(&this_adm.wait);
+ break;
+ }
+ atomic_set(&this_adm.copp_id[index], open->copp_id);
+ atomic_set(&this_adm.copp_stat[index], 1);
+ pr_debug("%s: coppid rxed=%d\n", __func__,
+ open->copp_id);
+ wake_up(&this_adm.wait);
+ }
+ break;
+ case ADM_CMDRSP_GET_PARAMS:
+ pr_debug("%s: ADM_CMDRSP_GET_PARAMS\n", __func__);
+ rtac_make_adm_callback(payload,
+ data->payload_size);
+ break;
+ default:
+ pr_err("%s: Unknown cmd:0x%x\n", __func__,
+ data->opcode);
+ break;
+ }
+ }
+ return 0;
+}
+
+static int send_adm_cal_block(int port_id, struct acdb_cal_block *aud_cal)
+{
+ s32 result = 0;
+ struct adm_set_params_command adm_params;
+ int index = afe_get_port_index(port_id);
+ if (index < 0 || index >= AFE_MAX_PORTS) {
+ pr_err("%s: invalid port idx %d portid %d\n",
+ __func__, index, port_id);
+ return 0;
+ }
+
+ pr_debug("%s: Port id %d, index %d\n", __func__, port_id, index);
+
+ if (!aud_cal || aud_cal->cal_size == 0) {
+ pr_debug("%s: No ADM cal to send for port_id = %d!\n",
+ __func__, port_id);
+ result = -EINVAL;
+ goto done;
+ }
+
+ adm_params.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(20), APR_PKT_VER);
+ adm_params.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(adm_params));
+ adm_params.hdr.src_svc = APR_SVC_ADM;
+ adm_params.hdr.src_domain = APR_DOMAIN_APPS;
+ adm_params.hdr.src_port = port_id;
+ adm_params.hdr.dest_svc = APR_SVC_ADM;
+ adm_params.hdr.dest_domain = APR_DOMAIN_ADSP;
+ adm_params.hdr.dest_port = atomic_read(&this_adm.copp_id[index]);
+ adm_params.hdr.token = port_id;
+ adm_params.hdr.opcode = ADM_CMD_SET_PARAMS;
+ adm_params.payload = aud_cal->cal_paddr;
+ adm_params.payload_size = aud_cal->cal_size;
+
+ atomic_set(&this_adm.copp_stat[index], 0);
+ pr_debug("%s: Sending SET_PARAMS payload = 0x%x, size = %d\n",
+ __func__, adm_params.payload, adm_params.payload_size);
+ result = apr_send_pkt(this_adm.apr, (uint32_t *)&adm_params);
+ if (result < 0) {
+ pr_err("%s: Set params failed port = %d payload = 0x%x\n",
+ __func__, port_id, aud_cal->cal_paddr);
+ result = -EINVAL;
+ goto done;
+ }
+ /* Wait for the callback */
+ result = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!result) {
+ pr_err("%s: Set params timed out port = %d, payload = 0x%x\n",
+ __func__, port_id, aud_cal->cal_paddr);
+ result = -EINVAL;
+ goto done;
+ }
+
+ result = 0;
+done:
+ return result;
+}
+
+static void send_adm_cal(int port_id, int path)
+{
+ int result = 0;
+ s32 acdb_path;
+ struct acdb_cal_block aud_cal;
+
+ pr_debug("%s\n", __func__);
+
+ /* Maps audio_dev_ctrl path definition to ACDB definition */
+ acdb_path = path - 1;
+
+ pr_debug("%s: Sending audproc cal\n", __func__);
+ get_audproc_cal(acdb_path, &aud_cal);
+
+ /* map & cache buffers used */
+ if (((mem_addr_audproc[acdb_path].cal_paddr != aud_cal.cal_paddr) &&
+ (aud_cal.cal_size > 0)) ||
+ (aud_cal.cal_size > mem_addr_audproc[acdb_path].cal_size)) {
+
+ if (mem_addr_audproc[acdb_path].cal_paddr != 0)
+ adm_memory_unmap_regions(
+ &mem_addr_audproc[acdb_path].cal_paddr,
+ &mem_addr_audproc[acdb_path].cal_size, 1);
+
+ result = adm_memory_map_regions(&aud_cal.cal_paddr, 0,
+ &aud_cal.cal_size, 1);
+ if (result < 0)
+ pr_err("ADM audproc mmap did not work! path = %d, "
+ "addr = 0x%x, size = %d\n", acdb_path,
+ aud_cal.cal_paddr, aud_cal.cal_size);
+ else
+ mem_addr_audproc[acdb_path] = aud_cal;
+ }
+
+ if (!send_adm_cal_block(port_id, &aud_cal))
+ pr_debug("%s: Audproc cal sent for port id: %d, path %d\n",
+ __func__, port_id, acdb_path);
+ else
+ pr_debug("%s: Audproc cal not sent for port id: %d, path %d\n",
+ __func__, port_id, acdb_path);
+
+ pr_debug("%s: Sending audvol cal\n", __func__);
+ get_audvol_cal(acdb_path, &aud_cal);
+
+ /* map & cache buffers used */
+ if (((mem_addr_audvol[acdb_path].cal_paddr != aud_cal.cal_paddr) &&
+ (aud_cal.cal_size > 0)) ||
+ (aud_cal.cal_size > mem_addr_audvol[acdb_path].cal_size)) {
+ if (mem_addr_audvol[acdb_path].cal_paddr != 0)
+ adm_memory_unmap_regions(
+ &mem_addr_audvol[acdb_path].cal_paddr,
+ &mem_addr_audvol[acdb_path].cal_size, 1);
+
+ result = adm_memory_map_regions(&aud_cal.cal_paddr, 0,
+ &aud_cal.cal_size, 1);
+ if (result < 0)
+ pr_err("ADM audvol mmap did not work! path = %d, "
+ "addr = 0x%x, size = %d\n", acdb_path,
+ aud_cal.cal_paddr, aud_cal.cal_size);
+ else
+ mem_addr_audvol[acdb_path] = aud_cal;
+ }
+
+ if (!send_adm_cal_block(port_id, &aud_cal))
+ pr_debug("%s: Audvol cal sent for port id: %d, path %d\n",
+ __func__, port_id, acdb_path);
+ else
+ pr_debug("%s: Audvol cal not sent for port id: %d, path %d\n",
+ __func__, port_id, acdb_path);
+}
+
+int adm_connect_afe_port(int mode, int session_id, int port_id)
+{
+ struct adm_cmd_connect_afe_port cmd;
+ int ret = 0;
+ int index;
+
+ pr_debug("%s: port %d session id:%d mode:%d\n", __func__,
+ port_id, session_id, mode);
+
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ if (afe_validate_port(port_id) < 0) {
+ pr_err("%s port idi[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+ index = afe_get_port_index(port_id);
+ pr_debug("%s: Port ID %d, index %d\n", __func__, port_id, index);
+
+ cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cmd.hdr.pkt_size = sizeof(cmd);
+ cmd.hdr.src_svc = APR_SVC_ADM;
+ cmd.hdr.src_domain = APR_DOMAIN_APPS;
+ cmd.hdr.src_port = port_id;
+ cmd.hdr.dest_svc = APR_SVC_ADM;
+ cmd.hdr.dest_domain = APR_DOMAIN_ADSP;
+ cmd.hdr.dest_port = port_id;
+ cmd.hdr.token = port_id;
+ cmd.hdr.opcode = ADM_CMD_CONNECT_AFE_PORT;
+
+ cmd.mode = mode;
+ cmd.session_id = session_id;
+ cmd.afe_port_id = port_id;
+
+ atomic_set(&this_adm.copp_stat[index], 0);
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&cmd);
+ if (ret < 0) {
+ pr_err("%s:ADM enable for port %d failed\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ /* Wait for the callback with copp id */
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s ADM connect AFE failed for port %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ atomic_inc(&this_adm.copp_cnt[index]);
+ return 0;
+
+fail_cmd:
+
+ return ret;
+}
+
+int adm_open(int port_id, int path, int rate, int channel_mode, int topology)
+{
+ struct adm_copp_open_command open;
+ int ret = 0;
+ int index;
+
+ pr_debug("%s: port %d path:%d rate:%d mode:%d\n", __func__,
+ port_id, path, rate, channel_mode);
+
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ if (afe_validate_port(port_id) < 0) {
+ pr_err("%s port idi[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+
+ index = afe_get_port_index(port_id);
+ pr_debug("%s: Port ID %d, index %d\n", __func__, port_id, index);
+
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+
+
+ /* Create a COPP if port id are not enabled */
+ if (atomic_read(&this_adm.copp_cnt[index]) == 0) {
+
+ open.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ open.hdr.pkt_size = sizeof(open);
+ open.hdr.src_svc = APR_SVC_ADM;
+ open.hdr.src_domain = APR_DOMAIN_APPS;
+ open.hdr.src_port = port_id;
+ open.hdr.dest_svc = APR_SVC_ADM;
+ open.hdr.dest_domain = APR_DOMAIN_ADSP;
+ open.hdr.dest_port = port_id;
+ open.hdr.token = port_id;
+ open.hdr.opcode = ADM_CMD_COPP_OPEN;
+
+ open.mode = path;
+ open.endpoint_id1 = port_id;
+ open.endpoint_id2 = 0xFFFF;
+
+ /* convert path to acdb path */
+ if (path == ADM_PATH_PLAYBACK)
+ open.topology_id = get_adm_rx_topology();
+ else {
+ open.topology_id = get_adm_tx_topology();
+ if ((open.topology_id ==
+ VPM_TX_SM_ECNS_COPP_TOPOLOGY) ||
+ (open.topology_id ==
+ VPM_TX_DM_FLUENCE_COPP_TOPOLOGY))
+ rate = 16000;
+ }
+
+ if (open.topology_id == 0)
+ open.topology_id = topology;
+
+ open.channel_config = channel_mode & 0x00FF;
+ open.rate = rate;
+
+ pr_debug("%s: channel_config=%d port_id=%d rate=%d"
+ "topology_id=0x%X\n", __func__, open.channel_config,\
+ open.endpoint_id1, open.rate,\
+ open.topology_id);
+
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&open);
+ if (ret < 0) {
+ pr_err("%s:ADM enable for port %d failed\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ /* Wait for the callback with copp id */
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s ADM open failed for port %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ }
+ atomic_inc(&this_adm.copp_cnt[index]);
+ return 0;
+
+fail_cmd:
+
+ return ret;
+}
+
+
+int adm_multi_ch_copp_open(int port_id, int path, int rate, int channel_mode,
+ int topology)
+{
+ struct adm_multi_ch_copp_open_command open;
+ int ret = 0;
+ int index;
+
+ pr_debug("%s: port %d path:%d rate:%d channel :%d\n", __func__,
+ port_id, path, rate, channel_mode);
+
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ if (afe_validate_port(port_id) < 0) {
+ pr_err("%s port idi[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+
+ index = afe_get_port_index(port_id);
+ pr_debug("%s: Port ID %d, index %d\n", __func__, port_id, index);
+
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+
+ /* Create a COPP if port id are not enabled */
+ if (atomic_read(&this_adm.copp_cnt[index]) == 0) {
+
+ open.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+
+ open.hdr.pkt_size =
+ sizeof(struct adm_multi_ch_copp_open_command);
+ open.hdr.opcode = ADM_CMD_MULTI_CHANNEL_COPP_OPEN;
+ memset(open.dev_channel_mapping, 0, 8);
+
+ if (channel_mode == 1) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FC;
+ } else if (channel_mode == 2) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channel_mode == 4) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ open.dev_channel_mapping[2] = PCM_CHANNEL_RB;
+ open.dev_channel_mapping[3] = PCM_CHANNEL_LB;
+ } else if (channel_mode == 6) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ open.dev_channel_mapping[2] = PCM_CHANNEL_LFE;
+ open.dev_channel_mapping[3] = PCM_CHANNEL_FC;
+ open.dev_channel_mapping[4] = PCM_CHANNEL_LB;
+ open.dev_channel_mapping[5] = PCM_CHANNEL_RB;
+ } else {
+ pr_err("%s invalid num_chan %d\n", __func__,
+ channel_mode);
+ return -EINVAL;
+ }
+
+
+ open.hdr.src_svc = APR_SVC_ADM;
+ open.hdr.src_domain = APR_DOMAIN_APPS;
+ open.hdr.src_port = port_id;
+ open.hdr.dest_svc = APR_SVC_ADM;
+ open.hdr.dest_domain = APR_DOMAIN_ADSP;
+ open.hdr.dest_port = port_id;
+ open.hdr.token = port_id;
+
+ open.mode = path;
+ open.endpoint_id1 = port_id;
+ open.endpoint_id2 = 0xFFFF;
+
+ /* convert path to acdb path */
+ if (path == ADM_PATH_PLAYBACK)
+ open.topology_id = get_adm_rx_topology();
+ else {
+ open.topology_id = get_adm_tx_topology();
+ if ((open.topology_id ==
+ VPM_TX_SM_ECNS_COPP_TOPOLOGY) ||
+ (open.topology_id ==
+ VPM_TX_DM_FLUENCE_COPP_TOPOLOGY))
+ rate = 16000;
+ }
+
+ if (open.topology_id == 0)
+ open.topology_id = topology;
+
+ open.channel_config = channel_mode & 0x00FF;
+ open.rate = rate;
+
+ pr_debug("%s: channel_config=%d port_id=%d rate=%d"
+ " topology_id=0x%X\n", __func__, open.channel_config,
+ open.endpoint_id1, open.rate,
+ open.topology_id);
+
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&open);
+ if (ret < 0) {
+ pr_err("%s:ADM enable for port %d failed\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ /* Wait for the callback with copp id */
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s ADM open failed for port %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ }
+ atomic_inc(&this_adm.copp_cnt[index]);
+ return 0;
+
+fail_cmd:
+
+ return ret;
+}
+
+int adm_matrix_map(int session_id, int path, int num_copps,
+ unsigned int *port_id, int copp_id)
+{
+ struct adm_routings_command route;
+ int ret = 0, i = 0;
+ /* Assumes port_ids have already been validated during adm_open */
+ int index = afe_get_port_index(copp_id);
+ if (index < 0 || index >= AFE_MAX_PORTS) {
+ pr_err("%s: invalid port idx %d token %d\n",
+ __func__, index, copp_id);
+ return 0;
+ }
+
+ pr_debug("%s: session 0x%x path:%d num_copps:%d port_id[0]:%d\n",
+ __func__, session_id, path, num_copps, port_id[0]);
+
+ route.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ route.hdr.pkt_size = sizeof(route);
+ route.hdr.src_svc = 0;
+ route.hdr.src_domain = APR_DOMAIN_APPS;
+ route.hdr.src_port = copp_id;
+ route.hdr.dest_svc = APR_SVC_ADM;
+ route.hdr.dest_domain = APR_DOMAIN_ADSP;
+ route.hdr.dest_port = atomic_read(&this_adm.copp_id[index]);
+ route.hdr.token = copp_id;
+ route.hdr.opcode = ADM_CMD_MATRIX_MAP_ROUTINGS;
+ route.num_sessions = 1;
+ route.session[0].id = session_id;
+ route.session[0].num_copps = num_copps;
+
+ for (i = 0; i < num_copps; i++) {
+ int tmp;
+ port_id[i] = afe_convert_virtual_to_portid(port_id[i]);
+
+ tmp = afe_get_port_index(port_id[i]);
+
+ pr_debug("%s: port_id[%d]: %d, index: %d\n", __func__, i,
+ port_id[i], tmp);
+
+ if (tmp >= 0 && tmp < AFE_MAX_PORTS)
+ route.session[0].copp_id[i] =
+ atomic_read(&this_adm.copp_id[tmp]);
+ }
+ if (num_copps % 2)
+ route.session[0].copp_id[i] = 0;
+
+ switch (path) {
+ case 0x1:
+ route.path = AUDIO_RX;
+ break;
+ case 0x2:
+ case 0x3:
+ route.path = AUDIO_TX;
+ break;
+ default:
+ pr_err("%s: Wrong path set[%d]\n", __func__, path);
+ break;
+ }
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&route);
+ if (ret < 0) {
+ pr_err("%s: ADM routing for port %d failed\n",
+ __func__, port_id[0]);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: ADM cmd Route failed for port %d\n",
+ __func__, port_id[0]);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ for (i = 0; i < num_copps; i++)
+ send_adm_cal(port_id[i], path);
+
+ for (i = 0; i < num_copps; i++)
+ rtac_add_adm_device(port_id[i], atomic_read(&this_adm.copp_id
+ [afe_get_port_index(port_id[i])]),
+ path, session_id);
+ return 0;
+
+fail_cmd:
+
+ return ret;
+}
+
+int adm_memory_map_regions(uint32_t *buf_add, uint32_t mempool_id,
+ uint32_t *bufsz, uint32_t bufcnt)
+{
+ struct adm_cmd_memory_map_regions *mmap_regions = NULL;
+ struct adm_memory_map_regions *mregions = NULL;
+ void *mmap_region_cmd = NULL;
+ void *payload = NULL;
+ int ret = 0;
+ int i = 0;
+ int cmd_size = 0;
+
+ pr_debug("%s\n", __func__);
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+
+ cmd_size = sizeof(struct adm_cmd_memory_map_regions)
+ + sizeof(struct adm_memory_map_regions) * bufcnt;
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (!mmap_region_cmd) {
+ pr_err("%s: allocate mmap_region_cmd failed\n", __func__);
+ return -ENOMEM;
+ }
+ mmap_regions = (struct adm_cmd_memory_map_regions *)mmap_region_cmd;
+ mmap_regions->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mmap_regions->hdr.pkt_size = cmd_size;
+ mmap_regions->hdr.src_port = 0;
+ mmap_regions->hdr.dest_port = 0;
+ mmap_regions->hdr.token = 0;
+ mmap_regions->hdr.opcode = ADM_CMD_MEMORY_MAP_REGIONS;
+ mmap_regions->mempool_id = mempool_id & 0x00ff;
+ mmap_regions->nregions = bufcnt & 0x00ff;
+ pr_debug("%s: map_regions->nregions = %d\n", __func__,
+ mmap_regions->nregions);
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct adm_cmd_memory_map_regions));
+ mregions = (struct adm_memory_map_regions *)payload;
+
+ for (i = 0; i < bufcnt; i++) {
+ mregions->phys = buf_add[i];
+ mregions->buf_size = bufsz[i];
+ ++mregions;
+ }
+
+ atomic_set(&this_adm.copp_stat[0], 0);
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *) mmap_region_cmd);
+ if (ret < 0) {
+ pr_err("%s: mmap_regions op[0x%x]rc[%d]\n", __func__,
+ mmap_regions->hdr.opcode, ret);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[0]), 5 * HZ);
+ if (!ret) {
+ pr_err("%s: timeout. waited for memory_map\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+fail_cmd:
+ kfree(mmap_region_cmd);
+ return ret;
+}
+
+int adm_memory_unmap_regions(uint32_t *buf_add, uint32_t *bufsz,
+ uint32_t bufcnt)
+{
+ struct adm_cmd_memory_unmap_regions *unmap_regions = NULL;
+ struct adm_memory_unmap_regions *mregions = NULL;
+ void *unmap_region_cmd = NULL;
+ void *payload = NULL;
+ int ret = 0;
+ int i = 0;
+ int cmd_size = 0;
+
+ pr_debug("%s\n", __func__);
+
+ if (this_adm.apr == NULL) {
+ pr_err("%s APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ cmd_size = sizeof(struct adm_cmd_memory_unmap_regions)
+ + sizeof(struct adm_memory_unmap_regions) * bufcnt;
+
+ unmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (!unmap_region_cmd) {
+ pr_err("%s: allocate unmap_region_cmd failed\n", __func__);
+ return -ENOMEM;
+ }
+ unmap_regions = (struct adm_cmd_memory_unmap_regions *)
+ unmap_region_cmd;
+ unmap_regions->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ unmap_regions->hdr.pkt_size = cmd_size;
+ unmap_regions->hdr.src_port = 0;
+ unmap_regions->hdr.dest_port = 0;
+ unmap_regions->hdr.token = 0;
+ unmap_regions->hdr.opcode = ADM_CMD_MEMORY_UNMAP_REGIONS;
+ unmap_regions->nregions = bufcnt & 0x00ff;
+ unmap_regions->reserved = 0;
+ pr_debug("%s: unmap_regions->nregions = %d\n", __func__,
+ unmap_regions->nregions);
+ payload = ((u8 *) unmap_region_cmd +
+ sizeof(struct adm_cmd_memory_unmap_regions));
+ mregions = (struct adm_memory_unmap_regions *)payload;
+
+ for (i = 0; i < bufcnt; i++) {
+ mregions->phys = buf_add[i];
+ ++mregions;
+ }
+ atomic_set(&this_adm.copp_stat[0], 0);
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *) unmap_region_cmd);
+ if (ret < 0) {
+ pr_err("%s: mmap_regions op[0x%x]rc[%d]\n", __func__,
+ unmap_regions->hdr.opcode, ret);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[0]), 5 * HZ);
+ if (!ret) {
+ pr_err("%s: timeout. waited for memory_unmap\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+fail_cmd:
+ kfree(unmap_region_cmd);
+ return ret;
+}
+
+int adm_get_copp_id(int port_index)
+{
+ pr_debug("%s\n", __func__);
+
+ if (port_index < 0) {
+ pr_err("%s: invalid port_id = %d\n", __func__, port_index);
+ return -EINVAL;
+ }
+
+ return atomic_read(&this_adm.copp_id[port_index]);
+}
+
+int adm_close(int port_id)
+{
+ struct apr_hdr close;
+
+ int ret = 0;
+ int index = 0;
+
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ index = afe_get_port_index(port_id);
+ if (afe_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ pr_debug("%s port_id=%d index %d\n", __func__, port_id, index);
+
+ if (!(atomic_read(&this_adm.copp_cnt[index]))) {
+ pr_err("%s: copp count for port[%d]is 0\n", __func__, port_id);
+
+ goto fail_cmd;
+ }
+ atomic_dec(&this_adm.copp_cnt[index]);
+ if (!(atomic_read(&this_adm.copp_cnt[index]))) {
+
+ close.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ close.pkt_size = sizeof(close);
+ close.src_svc = APR_SVC_ADM;
+ close.src_domain = APR_DOMAIN_APPS;
+ close.src_port = port_id;
+ close.dest_svc = APR_SVC_ADM;
+ close.dest_domain = APR_DOMAIN_ADSP;
+ close.dest_port = atomic_read(&this_adm.copp_id[index]);
+ close.token = port_id;
+ close.opcode = ADM_CMD_COPP_CLOSE;
+
+ atomic_set(&this_adm.copp_id[index], RESET_COPP_ID);
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+
+ pr_debug("%s:coppid %d portid=%d index=%d coppcnt=%d\n",
+ __func__,
+ atomic_read(&this_adm.copp_id[index]),
+ port_id, index,
+ atomic_read(&this_adm.copp_cnt[index]));
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&close);
+ if (ret < 0) {
+ pr_err("%s ADM close failed\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_adm.wait,
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: ADM cmd Route failed for port %d\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rtac_remove_adm_device(port_id);
+ }
+
+fail_cmd:
+ return ret;
+}
+
+static int __init adm_init(void)
+{
+ int i = 0;
+ init_waitqueue_head(&this_adm.wait);
+ this_adm.apr = NULL;
+
+ for (i = 0; i < AFE_MAX_PORTS; i++) {
+ atomic_set(&this_adm.copp_id[i], RESET_COPP_ID);
+ atomic_set(&this_adm.copp_cnt[i], 0);
+ atomic_set(&this_adm.copp_stat[i], 0);
+ }
+ return 0;
+}
+
+device_initcall(adm_init);
diff --git a/sound/soc/msm/qdsp6/q6afe.c b/sound/soc/msm/qdsp6/q6afe.c
new file mode 100644
index 0000000..51ef359
--- /dev/null
+++ b/sound/soc/msm/qdsp6/q6afe.c
@@ -0,0 +1,1726 @@
+/* Copyright (c) 2010-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/debugfs.h>
+#include <linux/kernel.h>
+#include <linux/kthread.h>
+#include <linux/uaccess.h>
+#include <linux/wait.h>
+#include <linux/jiffies.h>
+#include <linux/sched.h>
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <sound/apr_audio.h>
+#include <sound/q6afe.h>
+
+struct afe_ctl {
+ void *apr;
+ atomic_t state;
+ atomic_t status;
+ wait_queue_head_t wait;
+ struct task_struct *task;
+ void (*tx_cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv);
+ void (*rx_cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv);
+ void *tx_private_data;
+ void *rx_private_data;
+};
+
+static struct afe_ctl this_afe;
+
+static struct acdb_cal_block afe_cal_addr[MAX_AUDPROC_TYPES];
+
+#define TIMEOUT_MS 1000
+#define Q6AFE_MAX_VOLUME 0x3FFF
+
+#define SIZEOF_CFG_CMD(y) \
+ (sizeof(struct apr_hdr) + sizeof(u16) + (sizeof(struct y)))
+
+static int32_t afe_callback(struct apr_client_data *data, void *priv)
+{
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("q6afe: reset event = %d %d apr[%p]\n",
+ data->reset_event, data->reset_proc, this_afe.apr);
+ if (this_afe.apr) {
+ apr_reset(this_afe.apr);
+ atomic_set(&this_afe.state, 0);
+ this_afe.apr = NULL;
+ }
+ /* send info to user */
+ pr_debug("task_name = %s pid = %d\n",
+ this_afe.task->comm, this_afe.task->pid);
+ send_sig(SIGUSR1, this_afe.task, 0);
+ return 0;
+ }
+ if (data->payload_size) {
+ uint32_t *payload;
+ uint16_t port_id = 0;
+ payload = data->payload;
+ pr_debug("%s:opcode = 0x%x cmd = 0x%x status = 0x%x\n",
+ __func__, data->opcode,
+ payload[0], payload[1]);
+ /* payload[1] contains the error status for response */
+ if (payload[1] != 0) {
+ atomic_set(&this_afe.status, -1);
+ pr_err("%s: cmd = 0x%x returned error = 0x%x\n",
+ __func__, payload[0], payload[1]);
+ }
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ switch (payload[0]) {
+ case AFE_PORT_AUDIO_IF_CONFIG:
+ case AFE_PORT_CMD_I2S_CONFIG:
+ case AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG:
+ case AFE_PORT_AUDIO_SLIM_SCH_CONFIG:
+ case AFE_PORT_CMD_STOP:
+ case AFE_PORT_CMD_START:
+ case AFE_PORT_CMD_LOOPBACK:
+ case AFE_PORT_CMD_SIDETONE_CTL:
+ case AFE_PORT_CMD_SET_PARAM:
+ case AFE_PSEUDOPORT_CMD_START:
+ case AFE_PSEUDOPORT_CMD_STOP:
+ case AFE_PORT_CMD_APPLY_GAIN:
+ case AFE_SERVICE_CMD_MEMORY_MAP:
+ case AFE_SERVICE_CMD_MEMORY_UNMAP:
+ case AFE_SERVICE_CMD_UNREG_RTPORT:
+ atomic_set(&this_afe.state, 0);
+ wake_up(&this_afe.wait);
+ break;
+ case AFE_SERVICE_CMD_REG_RTPORT:
+ break;
+ case AFE_SERVICE_CMD_RTPORT_WR:
+ port_id = RT_PROXY_PORT_001_TX;
+ break;
+ case AFE_SERVICE_CMD_RTPORT_RD:
+ port_id = RT_PROXY_PORT_001_RX;
+ break;
+ default:
+ pr_err("Unknown cmd 0x%x\n",
+ payload[0]);
+ break;
+ }
+ } else if (data->opcode == AFE_EVENT_RT_PROXY_PORT_STATUS) {
+ port_id = (uint16_t)(0x0000FFFF & payload[0]);
+ }
+ pr_debug("%s:port_id = %x\n", __func__, port_id);
+ switch (port_id) {
+ case RT_PROXY_PORT_001_TX: {
+ if (this_afe.tx_cb) {
+ this_afe.tx_cb(data->opcode, data->token,
+ data->payload,
+ this_afe.tx_private_data);
+ }
+ break;
+ }
+ case RT_PROXY_PORT_001_RX: {
+ if (this_afe.rx_cb) {
+ this_afe.rx_cb(data->opcode, data->token,
+ data->payload,
+ this_afe.rx_private_data);
+ }
+ break;
+ }
+ default:
+ break;
+ }
+ }
+ return 0;
+}
+
+int afe_get_port_type(u16 port_id)
+{
+ int ret;
+
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PCM_RX:
+ case SECONDARY_I2S_RX:
+ case MI2S_RX:
+ case HDMI_RX:
+ case SLIMBUS_0_RX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_3_RX:
+ case INT_BT_SCO_RX:
+ case INT_BT_A2DP_RX:
+ case INT_FM_RX:
+ case VOICE_PLAYBACK_TX:
+ case RT_PROXY_PORT_001_RX:
+ case SLIMBUS_4_RX:
+ ret = MSM_AFE_PORT_TYPE_RX;
+ break;
+
+ case PRIMARY_I2S_TX:
+ case PCM_TX:
+ case SECONDARY_I2S_TX:
+ case MI2S_TX:
+ case DIGI_MIC_TX:
+ case VOICE_RECORD_TX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_TX:
+ case INT_FM_TX:
+ case VOICE_RECORD_RX:
+ case INT_BT_SCO_TX:
+ case RT_PROXY_PORT_001_TX:
+ case SLIMBUS_4_TX:
+ ret = MSM_AFE_PORT_TYPE_TX;
+ break;
+
+ default:
+ pr_err("%s: invalid port id %d\n", __func__, port_id);
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+int afe_validate_port(u16 port_id)
+{
+ int ret;
+
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ case PCM_RX:
+ case PCM_TX:
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case MI2S_RX:
+ case MI2S_TX:
+ case HDMI_RX:
+ case RSVD_2:
+ case RSVD_3:
+ case DIGI_MIC_TX:
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ case VOICE_PLAYBACK_TX:
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_3_RX:
+ case INT_BT_SCO_RX:
+ case INT_BT_SCO_TX:
+ case INT_BT_A2DP_RX:
+ case INT_FM_RX:
+ case INT_FM_TX:
+ case RT_PROXY_PORT_001_RX:
+ case RT_PROXY_PORT_001_TX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ {
+ ret = 0;
+ break;
+ }
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+int afe_convert_virtual_to_portid(u16 port_id)
+{
+ int ret;
+
+ /* if port_id is virtual, convert to physical..
+ * if port_id is already physical, return physical
+ */
+ if (afe_validate_port(port_id) < 0) {
+ if (port_id == RT_PROXY_DAI_001_RX ||
+ port_id == RT_PROXY_DAI_001_TX ||
+ port_id == RT_PROXY_DAI_002_RX ||
+ port_id == RT_PROXY_DAI_002_TX)
+ ret = VIRTUAL_ID_TO_PORTID(port_id);
+ else
+ ret = -EINVAL;
+ } else
+ ret = port_id;
+
+ return ret;
+}
+
+int afe_get_port_index(u16 port_id)
+{
+ switch (port_id) {
+ case PRIMARY_I2S_RX: return IDX_PRIMARY_I2S_RX;
+ case PRIMARY_I2S_TX: return IDX_PRIMARY_I2S_TX;
+ case PCM_RX: return IDX_PCM_RX;
+ case PCM_TX: return IDX_PCM_TX;
+ case SECONDARY_I2S_RX: return IDX_SECONDARY_I2S_RX;
+ case SECONDARY_I2S_TX: return IDX_SECONDARY_I2S_TX;
+ case MI2S_RX: return IDX_MI2S_RX;
+ case MI2S_TX: return IDX_MI2S_TX;
+ case HDMI_RX: return IDX_HDMI_RX;
+ case RSVD_2: return IDX_RSVD_2;
+ case RSVD_3: return IDX_RSVD_3;
+ case DIGI_MIC_TX: return IDX_DIGI_MIC_TX;
+ case VOICE_RECORD_RX: return IDX_VOICE_RECORD_RX;
+ case VOICE_RECORD_TX: return IDX_VOICE_RECORD_TX;
+ case VOICE_PLAYBACK_TX: return IDX_VOICE_PLAYBACK_TX;
+ case SLIMBUS_0_RX: return IDX_SLIMBUS_0_RX;
+ case SLIMBUS_0_TX: return IDX_SLIMBUS_0_TX;
+ case SLIMBUS_1_RX: return IDX_SLIMBUS_1_RX;
+ case SLIMBUS_1_TX: return IDX_SLIMBUS_1_TX;
+ case SLIMBUS_2_RX: return IDX_SLIMBUS_2_RX;
+ case SLIMBUS_2_TX: return IDX_SLIMBUS_2_TX;
+ case SLIMBUS_3_RX: return IDX_SLIMBUS_3_RX;
+ case INT_BT_SCO_RX: return IDX_INT_BT_SCO_RX;
+ case INT_BT_SCO_TX: return IDX_INT_BT_SCO_TX;
+ case INT_BT_A2DP_RX: return IDX_INT_BT_A2DP_RX;
+ case INT_FM_RX: return IDX_INT_FM_RX;
+ case INT_FM_TX: return IDX_INT_FM_TX;
+ case RT_PROXY_PORT_001_RX: return IDX_RT_PROXY_PORT_001_RX;
+ case RT_PROXY_PORT_001_TX: return IDX_RT_PROXY_PORT_001_TX;
+ case SLIMBUS_4_RX: return IDX_SLIMBUS_4_RX;
+ case SLIMBUS_4_TX: return IDX_SLIMBUS_4_TX;
+
+ default: return -EINVAL;
+ }
+}
+
+int afe_sizeof_cfg_cmd(u16 port_id)
+{
+ int ret_size;
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case MI2S_RX:
+ case MI2S_TX:
+ ret_size = SIZEOF_CFG_CMD(afe_port_mi2s_cfg);
+ break;
+ case HDMI_RX:
+ ret_size = SIZEOF_CFG_CMD(afe_port_hdmi_multi_ch_cfg);
+ break;
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ ret_size = SIZEOF_CFG_CMD(afe_port_slimbus_sch_cfg);
+ break;
+ case RT_PROXY_PORT_001_RX:
+ case RT_PROXY_PORT_001_TX:
+ ret_size = SIZEOF_CFG_CMD(afe_port_rtproxy_cfg);
+ break;
+ case PCM_RX:
+ case PCM_TX:
+ default:
+ ret_size = SIZEOF_CFG_CMD(afe_port_pcm_cfg);
+ break;
+ }
+ return ret_size;
+}
+
+int afe_q6_interface_prepare(void)
+{
+ int ret = 0;
+
+ pr_debug("%s:", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ }
+ }
+ return ret;
+}
+
+static void afe_send_cal_block(int32_t path, u16 port_id)
+{
+ int result = 0;
+ struct acdb_cal_block cal_block;
+ struct afe_port_cmd_set_param_no_payload afe_cal;
+ pr_debug("%s: path %d\n", __func__, path);
+
+ get_afe_cal(path, &cal_block);
+ if (cal_block.cal_size <= 0) {
+ pr_debug("%s: No AFE cal to send!\n", __func__);
+ goto done;
+ }
+
+ if ((afe_cal_addr[path].cal_paddr != cal_block.cal_paddr) ||
+ (cal_block.cal_size > afe_cal_addr[path].cal_size)) {
+ if (afe_cal_addr[path].cal_paddr != 0)
+ afe_cmd_memory_unmap_nowait(
+ afe_cal_addr[path].cal_paddr);
+
+ afe_cmd_memory_map_nowait(cal_block.cal_paddr,
+ cal_block.cal_size);
+ afe_cal_addr[path].cal_paddr = cal_block.cal_paddr;
+ afe_cal_addr[path].cal_size = cal_block.cal_size;
+ }
+
+ afe_cal.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ afe_cal.hdr.pkt_size = sizeof(afe_cal);
+ afe_cal.hdr.src_port = 0;
+ afe_cal.hdr.dest_port = 0;
+ afe_cal.hdr.token = 0;
+ afe_cal.hdr.opcode = AFE_PORT_CMD_SET_PARAM;
+ afe_cal.port_id = port_id;
+ afe_cal.payload_size = cal_block.cal_size;
+ afe_cal.payload_address = cal_block.cal_paddr;
+
+ pr_debug("%s: AFE cal sent for device port = %d, path = %d, "
+ "cal size = %d, cal addr = 0x%x\n", __func__,
+ port_id, path, cal_block.cal_size, cal_block.cal_paddr);
+
+ result = apr_send_pkt(this_afe.apr, (uint32_t *) &afe_cal);
+ if (result < 0) {
+ pr_err("%s: AFE cal for port %d failed\n",
+ __func__, port_id);
+ }
+
+ pr_debug("%s: AFE cal sent for path %d device!\n", __func__, path);
+done:
+ return;
+}
+
+void afe_send_cal(u16 port_id)
+{
+ pr_debug("%s\n", __func__);
+
+ if (afe_get_port_type(port_id) == MSM_AFE_PORT_TYPE_TX)
+ afe_send_cal_block(TX_CAL, port_id);
+ else if (afe_get_port_type(port_id) == MSM_AFE_PORT_TYPE_RX)
+ afe_send_cal_block(RX_CAL, port_id);
+}
+
+int afe_port_start_nowait(u16 port_id, union afe_port_config *afe_config,
+ u32 rate) /* This function is no blocking */
+{
+ struct afe_port_start_command start;
+ struct afe_audioif_config_command config;
+ int ret;
+
+ if (!afe_config) {
+ pr_err("%s: Error, no configuration data\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ pr_debug("%s: %d %d\n", __func__, port_id, rate);
+
+ if ((port_id == RT_PROXY_DAI_001_RX) ||
+ (port_id == RT_PROXY_DAI_002_TX))
+ return -EINVAL;
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE APR is not registered\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+
+ if (port_id == HDMI_RX) {
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+ config.hdr.token = 0;
+ config.hdr.opcode = AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG;
+ } else {
+
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+ config.hdr.token = 0;
+ switch (port_id) {
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_3_TX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ config.hdr.opcode = AFE_PORT_AUDIO_SLIM_SCH_CONFIG;
+ break;
+ case MI2S_TX:
+ case MI2S_RX:
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ /* AFE_PORT_CMD_I2S_CONFIG command is not supported
+ * in the LPASS EL 1.0. So we have to distiguish
+ * which AFE command, AFE_PORT_CMD_I2S_CONFIG or
+ * AFE_PORT_AUDIO_IF_CONFIG to use. If the format
+ * is L-PCM, the AFE_PORT_AUDIO_IF_CONFIG is used
+ * to make the backward compatible.
+ */
+ pr_debug("%s: afe_config->mi2s.format = %d\n", __func__,
+ afe_config->mi2s.format);
+ if (afe_config->mi2s.format == MSM_AFE_I2S_FORMAT_LPCM)
+ config.hdr.opcode = AFE_PORT_AUDIO_IF_CONFIG;
+ else
+ config.hdr.opcode = AFE_PORT_CMD_I2S_CONFIG;
+ break;
+ default:
+ config.hdr.opcode = AFE_PORT_AUDIO_IF_CONFIG;
+ break;
+ }
+ }
+
+ if (afe_validate_port(port_id) < 0) {
+
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ config.port_id = port_id;
+ config.port = *afe_config;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &config);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ /* send AFE cal */
+ afe_send_cal(port_id);
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PORT_CMD_START;
+ start.port_id = port_id;
+ start.gain = 0x2000;
+ start.sample_rate = rate;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+
+ if (IS_ERR_VALUE(ret)) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ if (this_afe.task != current)
+ this_afe.task = current;
+
+ pr_debug("task_name = %s pid = %d\n",
+ this_afe.task->comm, this_afe.task->pid);
+ return 0;
+
+fail_cmd:
+ return ret;
+}
+
+int afe_open(u16 port_id, union afe_port_config *afe_config, int rate)
+{
+ struct afe_port_start_command start;
+ struct afe_audioif_config_command config;
+ int ret = 0;
+
+ if (!afe_config) {
+ pr_err("%s: Error, no configuration data\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ pr_debug("%s: %d %d\n", __func__, port_id, rate);
+
+ if ((port_id == RT_PROXY_DAI_001_RX) ||
+ (port_id == RT_PROXY_DAI_002_TX))
+ return -EINVAL;
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+ config.hdr.token = 0;
+ switch (port_id) {
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_3_TX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ config.hdr.opcode = AFE_PORT_AUDIO_SLIM_SCH_CONFIG;
+ break;
+ case MI2S_TX:
+ case MI2S_RX:
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ /* AFE_PORT_CMD_I2S_CONFIG command is not supported
+ * in the LPASS EL 1.0. So we have to distiguish
+ * which AFE command, AFE_PORT_CMD_I2S_CONFIG or
+ * AFE_PORT_AUDIO_IF_CONFIG to use. If the format
+ * is L-PCM, the AFE_PORT_AUDIO_IF_CONFIG is used
+ * to make the backward compatible.
+ */
+ pr_debug("%s: afe_config->mi2s.format = %d\n", __func__,
+ afe_config->mi2s.format);
+ if (afe_config->mi2s.format == MSM_AFE_I2S_FORMAT_LPCM)
+ config.hdr.opcode = AFE_PORT_AUDIO_IF_CONFIG;
+ else
+ config.hdr.opcode = AFE_PORT_CMD_I2S_CONFIG;
+ break;
+ default:
+ config.hdr.opcode = AFE_PORT_AUDIO_IF_CONFIG;
+ break;
+ }
+
+ if (afe_validate_port(port_id) < 0) {
+
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ config.port_id = port_id;
+ config.port = *afe_config;
+
+ atomic_set(&this_afe.state, 1);
+ atomic_set(&this_afe.status, 0);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &config);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ if (atomic_read(&this_afe.status) != 0) {
+ pr_err("%s: config cmd failed\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PORT_CMD_START;
+ start.port_id = port_id;
+ start.gain = 0x2000;
+ start.sample_rate = rate;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ if (this_afe.task != current)
+ this_afe.task = current;
+
+ pr_debug("task_name = %s pid = %d\n",
+ this_afe.task->comm, this_afe.task->pid);
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_loopback(u16 enable, u16 dst_port, u16 src_port)
+{
+ struct afe_loopback_command lb_cmd;
+ int ret = 0;
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ if ((afe_get_port_type(dst_port) == MSM_AFE_PORT_TYPE_RX) &&
+ (afe_get_port_type(src_port) == MSM_AFE_PORT_TYPE_RX))
+ return afe_loopback_cfg(enable, dst_port, src_port,
+ LB_MODE_EC_REF_VOICE_AUDIO);
+
+ lb_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(20), APR_PKT_VER);
+ lb_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(lb_cmd) - APR_HDR_SIZE);
+ lb_cmd.hdr.src_port = 0;
+ lb_cmd.hdr.dest_port = 0;
+ lb_cmd.hdr.token = 0;
+ lb_cmd.hdr.opcode = AFE_PORT_CMD_LOOPBACK;
+ lb_cmd.tx_port_id = src_port;
+ lb_cmd.rx_port_id = dst_port;
+ lb_cmd.mode = 0xFFFF;
+ lb_cmd.enable = (enable ? 1 : 0);
+ atomic_set(&this_afe.state, 1);
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &lb_cmd);
+ if (ret < 0) {
+ pr_err("%s: AFE loopback failed\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ }
+done:
+ return ret;
+}
+
+int afe_loopback_cfg(u16 enable, u16 dst_port, u16 src_port, u16 mode)
+{
+ struct afe_port_cmd_set_param lp_cfg;
+ int ret = 0;
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ pr_debug("%s: src_port %d, dst_port %d\n",
+ __func__, src_port, dst_port);
+
+ lp_cfg.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ lp_cfg.hdr.pkt_size = sizeof(lp_cfg);
+ lp_cfg.hdr.src_port = 0;
+ lp_cfg.hdr.dest_port = 0;
+ lp_cfg.hdr.token = 0;
+ lp_cfg.hdr.opcode = AFE_PORT_CMD_SET_PARAM;
+
+ lp_cfg.port_id = src_port;
+ lp_cfg.payload_size = sizeof(struct afe_param_payload);
+ lp_cfg.payload_address = 0;
+
+ lp_cfg.payload.module_id = AFE_MODULE_LOOPBACK;
+ lp_cfg.payload.param_id = AFE_PARAM_ID_LOOPBACK_CONFIG;
+ lp_cfg.payload.param_size = sizeof(struct afe_param_loopback_cfg);
+ lp_cfg.payload.reserved = 0;
+
+ lp_cfg.payload.param.loopback_cfg.loopback_cfg_minor_version =
+ AFE_API_VERSION_LOOPBACK_CONFIG;
+ lp_cfg.payload.param.loopback_cfg.dst_port_id = dst_port;
+ lp_cfg.payload.param.loopback_cfg.routing_mode = mode;
+ lp_cfg.payload.param.loopback_cfg.enable = enable;
+ lp_cfg.payload.param.loopback_cfg.reserved = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &lp_cfg);
+ if (ret < 0) {
+ pr_err("%s: AFE loopback config failed for src_port %d, dst_port %d\n",
+ __func__, src_port, dst_port);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (ret < 0) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_loopback_gain(u16 port_id, u16 volume)
+{
+ struct afe_port_cmd_set_param set_param;
+ int ret = 0;
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ if (afe_validate_port(port_id) < 0) {
+
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ /* RX ports numbers are even .TX ports numbers are odd. */
+ if (port_id % 2 == 0) {
+ pr_err("%s: Failed : afe loopback gain only for TX ports."
+ " port_id %d\n", __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ pr_debug("%s: %d %hX\n", __func__, port_id, volume);
+
+ set_param.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ set_param.hdr.pkt_size = sizeof(set_param);
+ set_param.hdr.src_port = 0;
+ set_param.hdr.dest_port = 0;
+ set_param.hdr.token = 0;
+ set_param.hdr.opcode = AFE_PORT_CMD_SET_PARAM;
+
+ set_param.port_id = port_id;
+ set_param.payload_size = sizeof(struct afe_param_payload);
+ set_param.payload_address = 0;
+
+ set_param.payload.module_id = AFE_MODULE_ID_PORT_INFO;
+ set_param.payload.param_id = AFE_PARAM_ID_LOOPBACK_GAIN;
+ set_param.payload.param_size = sizeof(struct afe_param_loopback_gain);
+ set_param.payload.reserved = 0;
+
+ set_param.payload.param.loopback_gain.gain = volume;
+ set_param.payload.param.loopback_gain.reserved = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &set_param);
+ if (ret < 0) {
+ pr_err("%s: AFE param set failed for port %d\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (ret < 0) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_apply_gain(u16 port_id, u16 gain)
+{
+ struct afe_port_gain_command set_gain;
+ int ret = 0;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE is not opened\n", __func__);
+ ret = -EPERM;
+ goto fail_cmd;
+ }
+
+ if (afe_validate_port(port_id) < 0) {
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ /* RX ports numbers are even .TX ports numbers are odd. */
+ if (port_id % 2 == 0) {
+ pr_err("%s: Failed : afe apply gain only for TX ports."
+ " port_id %d\n", __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ pr_debug("%s: %d %hX\n", __func__, port_id, gain);
+
+ set_gain.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ set_gain.hdr.pkt_size = sizeof(set_gain);
+ set_gain.hdr.src_port = 0;
+ set_gain.hdr.dest_port = 0;
+ set_gain.hdr.token = 0;
+ set_gain.hdr.opcode = AFE_PORT_CMD_APPLY_GAIN;
+
+ set_gain.port_id = port_id;
+ set_gain.gain = gain;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &set_gain);
+ if (ret < 0) {
+ pr_err("%s: AFE Gain set failed for port %d\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (ret < 0) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_pseudo_port_start_nowait(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_start_command start;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE APR is not registered\n", __func__);
+ return -ENODEV;
+ }
+
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PSEUDOPORT_CMD_START;
+ start.port_id = port_id;
+ start.timing = 1;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed %d\n",
+ __func__, port_id, ret);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+int afe_start_pseudo_port(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_start_command start;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PSEUDOPORT_CMD_START;
+ start.port_id = port_id;
+ start.timing = 1;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed %d\n",
+ __func__, port_id, ret);
+ return -EINVAL;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int afe_pseudo_port_stop_nowait(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_stop_command stop;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE is already closed\n", __func__);
+ return -EINVAL;
+ }
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PSEUDOPORT_CMD_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+ if (ret < 0) {
+ pr_err("%s: AFE close failed %d\n", __func__, ret);
+ return -EINVAL;
+ }
+
+ return 0;
+
+}
+
+int afe_stop_pseudo_port(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_stop_command stop;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE is already closed\n", __func__);
+ return -EINVAL;
+ }
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PSEUDOPORT_CMD_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+ if (ret < 0) {
+ pr_err("%s: AFE close failed %d\n", __func__, ret);
+ return -EINVAL;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int afe_cmd_memory_map(u32 dma_addr_p, u32 dma_buf_sz)
+{
+ int ret = 0;
+ struct afe_cmd_memory_map mregion;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ mregion.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion.hdr.pkt_size = sizeof(mregion);
+ mregion.hdr.src_port = 0;
+ mregion.hdr.dest_port = 0;
+ mregion.hdr.token = 0;
+ mregion.hdr.opcode = AFE_SERVICE_CMD_MEMORY_MAP;
+ mregion.phy_addr = dma_addr_p;
+ mregion.mem_sz = dma_buf_sz;
+ mregion.mem_id = 0;
+ mregion.rsvd = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &mregion);
+ if (ret < 0) {
+ pr_err("%s: AFE memory map cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ return 0;
+}
+
+int afe_cmd_memory_map_nowait(u32 dma_addr_p, u32 dma_buf_sz)
+{
+ int ret = 0;
+ struct afe_cmd_memory_map mregion;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ mregion.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion.hdr.pkt_size = sizeof(mregion);
+ mregion.hdr.src_port = 0;
+ mregion.hdr.dest_port = 0;
+ mregion.hdr.token = 0;
+ mregion.hdr.opcode = AFE_SERVICE_CMD_MEMORY_MAP;
+ mregion.phy_addr = dma_addr_p;
+ mregion.mem_sz = dma_buf_sz;
+ mregion.mem_id = 0;
+ mregion.rsvd = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &mregion);
+ if (ret < 0) {
+ pr_err("%s: AFE memory map cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ }
+ return 0;
+}
+
+int afe_cmd_memory_unmap(u32 dma_addr_p)
+{
+ int ret = 0;
+ struct afe_cmd_memory_unmap mregion;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ mregion.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion.hdr.pkt_size = sizeof(mregion);
+ mregion.hdr.src_port = 0;
+ mregion.hdr.dest_port = 0;
+ mregion.hdr.token = 0;
+ mregion.hdr.opcode = AFE_SERVICE_CMD_MEMORY_UNMAP;
+ mregion.phy_addr = dma_addr_p;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &mregion);
+ if (ret < 0) {
+ pr_err("%s: AFE memory unmap cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_cmd_memory_unmap_nowait(u32 dma_addr_p)
+{
+ int ret = 0;
+ struct afe_cmd_memory_unmap mregion;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ mregion.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion.hdr.pkt_size = sizeof(mregion);
+ mregion.hdr.src_port = 0;
+ mregion.hdr.dest_port = 0;
+ mregion.hdr.token = 0;
+ mregion.hdr.opcode = AFE_SERVICE_CMD_MEMORY_UNMAP;
+ mregion.phy_addr = dma_addr_p;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &mregion);
+ if (ret < 0) {
+ pr_err("%s: AFE memory unmap cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ }
+ return 0;
+}
+
+int afe_register_get_events(u16 port_id,
+ void (*cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv),
+ void *private_data)
+{
+ int ret = 0;
+ struct afe_cmd_reg_rtport rtproxy;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+ else
+ return -EINVAL;
+
+ if (port_id == RT_PROXY_PORT_001_TX) {
+ this_afe.tx_cb = cb;
+ this_afe.tx_private_data = private_data;
+ } else if (port_id == RT_PROXY_PORT_001_RX) {
+ this_afe.rx_cb = cb;
+ this_afe.rx_private_data = private_data;
+ }
+
+ rtproxy.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ rtproxy.hdr.pkt_size = sizeof(rtproxy);
+ rtproxy.hdr.src_port = 1;
+ rtproxy.hdr.dest_port = 1;
+ rtproxy.hdr.token = 0;
+ rtproxy.hdr.opcode = AFE_SERVICE_CMD_REG_RTPORT;
+ rtproxy.port_id = port_id;
+ rtproxy.rsvd = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &rtproxy);
+ if (ret < 0) {
+ pr_err("%s: AFE reg. rtproxy_event failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_unregister_get_events(u16 port_id)
+{
+ int ret = 0;
+ struct afe_cmd_unreg_rtport rtproxy;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+ else
+ return -EINVAL;
+
+ rtproxy.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ rtproxy.hdr.pkt_size = sizeof(rtproxy);
+ rtproxy.hdr.src_port = 0;
+ rtproxy.hdr.dest_port = 0;
+ rtproxy.hdr.token = 0;
+ rtproxy.hdr.opcode = AFE_SERVICE_CMD_UNREG_RTPORT;
+ rtproxy.port_id = port_id;
+ rtproxy.rsvd = 0;
+
+ if (port_id == RT_PROXY_PORT_001_TX) {
+ this_afe.tx_cb = NULL;
+ this_afe.tx_private_data = NULL;
+ } else if (port_id == RT_PROXY_PORT_001_RX) {
+ this_afe.rx_cb = NULL;
+ this_afe.rx_private_data = NULL;
+ }
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &rtproxy);
+ if (ret < 0) {
+ pr_err("%s: AFE enable Unreg. rtproxy_event failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_rt_proxy_port_write(u32 buf_addr_p, int bytes)
+{
+ int ret = 0;
+ struct afe_cmd_rtport_wr afecmd_wr;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s:register to AFE is not done\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ pr_debug("%s: buf_addr_p = 0x%08x bytes = %d\n", __func__,
+ buf_addr_p, bytes);
+
+ afecmd_wr.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ afecmd_wr.hdr.pkt_size = sizeof(afecmd_wr);
+ afecmd_wr.hdr.src_port = 0;
+ afecmd_wr.hdr.dest_port = 0;
+ afecmd_wr.hdr.token = 0;
+ afecmd_wr.hdr.opcode = AFE_SERVICE_CMD_RTPORT_WR;
+ afecmd_wr.buf_addr = (uint32_t)buf_addr_p;
+ afecmd_wr.port_id = RT_PROXY_PORT_001_TX;
+ afecmd_wr.bytes_avail = bytes;
+ afecmd_wr.rsvd = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &afecmd_wr);
+ if (ret < 0) {
+ pr_err("%s: AFE rtproxy write to port 0x%x failed %d\n",
+ __func__, afecmd_wr.port_id, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+
+}
+
+int afe_rt_proxy_port_read(u32 buf_addr_p, int bytes)
+{
+ int ret = 0;
+ struct afe_cmd_rtport_rd afecmd_rd;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: register to AFE is not done\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ pr_debug("%s: buf_addr_p = 0x%08x bytes = %d\n", __func__,
+ buf_addr_p, bytes);
+
+ afecmd_rd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ afecmd_rd.hdr.pkt_size = sizeof(afecmd_rd);
+ afecmd_rd.hdr.src_port = 0;
+ afecmd_rd.hdr.dest_port = 0;
+ afecmd_rd.hdr.token = 0;
+ afecmd_rd.hdr.opcode = AFE_SERVICE_CMD_RTPORT_RD;
+ afecmd_rd.buf_addr = (uint32_t)buf_addr_p;
+ afecmd_rd.port_id = RT_PROXY_PORT_001_RX;
+ afecmd_rd.bytes_avail = bytes;
+ afecmd_rd.rsvd = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &afecmd_rd);
+ if (ret < 0) {
+ pr_err("%s: AFE rtproxy read cmd to port 0x%x failed %d\n",
+ __func__, afecmd_rd.port_id, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_afelb;
+static struct dentry *debugfs_afelb_gain;
+
+static int afe_debug_open(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ pr_info("debug intf %s\n", (char *) file->private_data);
+ return 0;
+}
+
+static int afe_get_parameters(char *buf, long int *param1, int num_of_par)
+{
+ char *token;
+ int base, cnt;
+
+ token = strsep(&buf, " ");
+
+ for (cnt = 0; cnt < num_of_par; cnt++) {
+ if (token != NULL) {
+ if ((token[1] == 'x') || (token[1] == 'X'))
+ base = 16;
+ else
+ base = 10;
+
+ if (strict_strtoul(token, base, ¶m1[cnt]) != 0)
+ return -EINVAL;
+
+ token = strsep(&buf, " ");
+ } else
+ return -EINVAL;
+ }
+ return 0;
+}
+#define AFE_LOOPBACK_ON (1)
+#define AFE_LOOPBACK_OFF (0)
+static ssize_t afe_debug_write(struct file *filp,
+ const char __user *ubuf, size_t cnt, loff_t *ppos)
+{
+ char *lb_str = filp->private_data;
+ char lbuf[32];
+ int rc;
+ unsigned long param[5];
+
+ if (cnt > sizeof(lbuf) - 1)
+ return -EINVAL;
+
+ rc = copy_from_user(lbuf, ubuf, cnt);
+ if (rc)
+ return -EFAULT;
+
+ lbuf[cnt] = '\0';
+
+ if (!strcmp(lb_str, "afe_loopback")) {
+ rc = afe_get_parameters(lbuf, param, 3);
+ if (!rc) {
+ pr_info("%s %lu %lu %lu\n", lb_str, param[0], param[1],
+ param[2]);
+
+ if ((param[0] != AFE_LOOPBACK_ON) && (param[0] !=
+ AFE_LOOPBACK_OFF)) {
+ pr_err("%s: Error, parameter 0 incorrect\n",
+ __func__);
+ rc = -EINVAL;
+ goto afe_error;
+ }
+ if ((afe_validate_port(param[1]) < 0) ||
+ (afe_validate_port(param[2])) < 0) {
+ pr_err("%s: Error, invalid afe port\n",
+ __func__);
+ }
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Error, AFE not opened\n", __func__);
+ rc = -EINVAL;
+ } else {
+ rc = afe_loopback(param[0], param[1], param[2]);
+ }
+ } else {
+ pr_err("%s: Error, invalid parameters\n", __func__);
+ rc = -EINVAL;
+ }
+
+ } else if (!strcmp(lb_str, "afe_loopback_gain")) {
+ rc = afe_get_parameters(lbuf, param, 2);
+ if (!rc) {
+ pr_info("%s %lu %lu\n", lb_str, param[0], param[1]);
+
+ if (afe_validate_port(param[0]) < 0) {
+ pr_err("%s: Error, invalid afe port\n",
+ __func__);
+ rc = -EINVAL;
+ goto afe_error;
+ }
+
+ if (param[1] < 0 || param[1] > 100) {
+ pr_err("%s: Error, volume shoud be 0 to 100"
+ " percentage param = %lu\n",
+ __func__, param[1]);
+ rc = -EINVAL;
+ goto afe_error;
+ }
+
+ param[1] = (Q6AFE_MAX_VOLUME * param[1]) / 100;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Error, AFE not opened\n", __func__);
+ rc = -EINVAL;
+ } else {
+ rc = afe_loopback_gain(param[0], param[1]);
+ }
+ } else {
+ pr_err("%s: Error, invalid parameters\n", __func__);
+ rc = -EINVAL;
+ }
+ }
+
+afe_error:
+ if (rc == 0)
+ rc = cnt;
+ else
+ pr_err("%s: rc = %d\n", __func__, rc);
+
+ return rc;
+}
+
+static const struct file_operations afe_debug_fops = {
+ .open = afe_debug_open,
+ .write = afe_debug_write
+};
+#endif
+int afe_sidetone(u16 tx_port_id, u16 rx_port_id, u16 enable, uint16_t gain)
+{
+ struct afe_port_sidetone_command cmd_sidetone;
+ int ret = 0;
+
+ pr_info("%s: tx_port_id:%d rx_port_id:%d enable:%d gain:%d\n", __func__,
+ tx_port_id, rx_port_id, enable, gain);
+ cmd_sidetone.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cmd_sidetone.hdr.pkt_size = sizeof(cmd_sidetone);
+ cmd_sidetone.hdr.src_port = 0;
+ cmd_sidetone.hdr.dest_port = 0;
+ cmd_sidetone.hdr.token = 0;
+ cmd_sidetone.hdr.opcode = AFE_PORT_CMD_SIDETONE_CTL;
+ cmd_sidetone.tx_port_id = tx_port_id;
+ cmd_sidetone.rx_port_id = rx_port_id;
+ cmd_sidetone.gain = gain;
+ cmd_sidetone.enable = enable;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &cmd_sidetone);
+ if (ret < 0) {
+ pr_err("%s: AFE sidetone failed for tx_port:%d rx_port:%d\n",
+ __func__, tx_port_id, rx_port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (ret < 0) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_port_stop_nowait(int port_id)
+{
+ struct afe_port_stop_command stop;
+ int ret = 0;
+
+ if (this_afe.apr == NULL) {
+ pr_err("AFE is already closed\n");
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PORT_CMD_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+
+ if (ret == -ENETRESET) {
+ pr_info("%s: Need to reset, calling APR deregister", __func__);
+ return apr_deregister(this_afe.apr);
+ } else if (IS_ERR_VALUE(ret)) {
+ pr_err("%s: AFE close failed\n", __func__);
+ ret = -EINVAL;
+ }
+
+fail_cmd:
+ return ret;
+
+}
+
+int afe_close(int port_id)
+{
+ struct afe_port_stop_command stop;
+ int ret = 0;
+
+ if (this_afe.apr == NULL) {
+ pr_err("AFE is already closed\n");
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+ port_id = afe_convert_virtual_to_portid(port_id);
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PORT_CMD_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+
+ if (ret == -ENETRESET) {
+ pr_info("%s: Need to reset, calling APR deregister", __func__);
+ return apr_deregister(this_afe.apr);
+ }
+
+ if (ret < 0) {
+ pr_err("%s: AFE close failed\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait,
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+fail_cmd:
+ return ret;
+}
+
+static int __init afe_init(void)
+{
+ init_waitqueue_head(&this_afe.wait);
+ atomic_set(&this_afe.state, 0);
+ atomic_set(&this_afe.status, 0);
+ this_afe.apr = NULL;
+#ifdef CONFIG_DEBUG_FS
+ debugfs_afelb = debugfs_create_file("afe_loopback",
+ S_IFREG | S_IWUGO, NULL, (void *) "afe_loopback",
+ &afe_debug_fops);
+
+ debugfs_afelb_gain = debugfs_create_file("afe_loopback_gain",
+ S_IFREG | S_IWUGO, NULL, (void *) "afe_loopback_gain",
+ &afe_debug_fops);
+
+
+#endif
+ return 0;
+}
+
+static void __exit afe_exit(void)
+{
+ int i;
+#ifdef CONFIG_DEBUG_FS
+ if (debugfs_afelb)
+ debugfs_remove(debugfs_afelb);
+ if (debugfs_afelb_gain)
+ debugfs_remove(debugfs_afelb_gain);
+#endif
+ for (i = 0; i < MAX_AUDPROC_TYPES; i++) {
+ if (afe_cal_addr[i].cal_paddr != 0)
+ afe_cmd_memory_unmap_nowait(
+ afe_cal_addr[i].cal_paddr);
+ }
+}
+
+device_initcall(afe_init);
+__exitcall(afe_exit);
diff --git a/sound/soc/msm/qdsp6/q6asm.c b/sound/soc/msm/qdsp6/q6asm.c
new file mode 100644
index 0000000..09bfd94
--- /dev/null
+++ b/sound/soc/msm/qdsp6/q6asm.c
@@ -0,0 +1,3533 @@
+
+/*
+ * Copyright (c) 2010-2012, Code Aurora Forum. All rights reserved.
+ * Author: Brian Swetland <swetland@google.com>
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+#include <linux/fs.h>
+#include <linux/mutex.h>
+#include <linux/wait.h>
+#include <linux/miscdevice.h>
+#include <linux/uaccess.h>
+#include <linux/sched.h>
+#include <linux/dma-mapping.h>
+#include <linux/miscdevice.h>
+#include <linux/delay.h>
+#include <linux/spinlock.h>
+#include <linux/slab.h>
+#include <linux/msm_audio.h>
+#include <linux/android_pmem.h>
+#include <linux/memory_alloc.h>
+#include <linux/debugfs.h>
+#include <linux/time.h>
+#include <linux/atomic.h>
+
+#include <asm/ioctls.h>
+
+#include <mach/memory.h>
+#include <mach/debug_mm.h>
+#include <mach/peripheral-loader.h>
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <mach/qdsp6v2/rtac.h>
+
+#include <sound/apr_audio.h>
+#include <sound/q6asm.h>
+
+
+#define TRUE 0x01
+#define FALSE 0x00
+#define READDONE_IDX_STATUS 0
+#define READDONE_IDX_BUFFER 1
+#define READDONE_IDX_SIZE 2
+#define READDONE_IDX_OFFSET 3
+#define READDONE_IDX_MSW_TS 4
+#define READDONE_IDX_LSW_TS 5
+#define READDONE_IDX_FLAGS 6
+#define READDONE_IDX_NUMFRAMES 7
+#define READDONE_IDX_ID 8
+#ifdef CONFIG_DEBUG_FS
+#define OUT_BUFFER_SIZE 56
+#define IN_BUFFER_SIZE 24
+#endif
+static DEFINE_MUTEX(session_lock);
+
+/* session id: 0 reserved */
+static struct audio_client *session[SESSION_MAX+1];
+static int32_t q6asm_mmapcallback(struct apr_client_data *data, void *priv);
+static int32_t q6asm_callback(struct apr_client_data *data, void *priv);
+static void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg);
+static void q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg);
+static int q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt);
+static int q6asm_memory_unmap_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt);
+
+static void q6asm_reset_buf_state(struct audio_client *ac);
+
+#ifdef CONFIG_DEBUG_FS
+static struct timeval out_cold_tv;
+static struct timeval out_warm_tv;
+static struct timeval out_cont_tv;
+static struct timeval in_cont_tv;
+static long out_enable_flag;
+static long in_enable_flag;
+static struct dentry *out_dentry;
+static struct dentry *in_dentry;
+static int in_cont_index;
+/*This var is used to keep track of first write done for cold output latency */
+static int out_cold_index;
+static char *out_buffer;
+static char *in_buffer;
+static int audio_output_latency_dbgfs_open(struct inode *inode,
+ struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+static ssize_t audio_output_latency_dbgfs_read(struct file *file,
+ char __user *buf, size_t count, loff_t *ppos)
+{
+ snprintf(out_buffer, OUT_BUFFER_SIZE, "%ld,%ld,%ld,%ld,%ld,%ld,",\
+ out_cold_tv.tv_sec, out_cold_tv.tv_usec, out_warm_tv.tv_sec,\
+ out_warm_tv.tv_usec, out_cont_tv.tv_sec, out_cont_tv.tv_usec);
+ return simple_read_from_buffer(buf, OUT_BUFFER_SIZE, ppos,
+ out_buffer, OUT_BUFFER_SIZE);
+}
+static ssize_t audio_output_latency_dbgfs_write(struct file *file,
+ const char __user *buf, size_t count, loff_t *ppos)
+{
+ char *temp;
+
+ if (count > 2*sizeof(char))
+ return -EINVAL;
+ else
+ temp = kmalloc(2*sizeof(char), GFP_KERNEL);
+
+ out_cold_index = 0;
+
+ if (temp) {
+ if (copy_from_user(temp, buf, 2*sizeof(char))) {
+ kfree(temp);
+ return -EFAULT;
+ }
+ if (!strict_strtol(temp, 10, &out_enable_flag)) {
+ kfree(temp);
+ return count;
+ }
+ kfree(temp);
+ }
+ return -EINVAL;
+}
+static const struct file_operations audio_output_latency_debug_fops = {
+ .open = audio_output_latency_dbgfs_open,
+ .read = audio_output_latency_dbgfs_read,
+ .write = audio_output_latency_dbgfs_write
+};
+
+static int audio_input_latency_dbgfs_open(struct inode *inode,
+ struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+static ssize_t audio_input_latency_dbgfs_read(struct file *file,
+ char __user *buf, size_t count, loff_t *ppos)
+{
+ snprintf(in_buffer, IN_BUFFER_SIZE, "%ld,%ld,",\
+ in_cont_tv.tv_sec, in_cont_tv.tv_usec);
+ return simple_read_from_buffer(buf, IN_BUFFER_SIZE, ppos,
+ in_buffer, IN_BUFFER_SIZE);
+}
+static ssize_t audio_input_latency_dbgfs_write(struct file *file,
+ const char __user *buf, size_t count, loff_t *ppos)
+{
+ char *temp;
+
+ if (count > 2*sizeof(char))
+ return -EINVAL;
+ else
+ temp = kmalloc(2*sizeof(char), GFP_KERNEL);
+ if (temp) {
+ if (copy_from_user(temp, buf, 2*sizeof(char))) {
+ kfree(temp);
+ return -EFAULT;
+ }
+ if (!strict_strtol(temp, 10, &in_enable_flag)) {
+ kfree(temp);
+ return count;
+ }
+ kfree(temp);
+ }
+ return -EINVAL;
+}
+static const struct file_operations audio_input_latency_debug_fops = {
+ .open = audio_input_latency_dbgfs_open,
+ .read = audio_input_latency_dbgfs_read,
+ .write = audio_input_latency_dbgfs_write
+};
+#endif
+struct asm_mmap {
+ atomic_t ref_cnt;
+ atomic_t cmd_state;
+ wait_queue_head_t cmd_wait;
+ void *apr;
+};
+
+static struct asm_mmap this_mmap;
+
+static int q6asm_session_alloc(struct audio_client *ac)
+{
+ int n;
+ mutex_lock(&session_lock);
+ for (n = 1; n <= SESSION_MAX; n++) {
+ if (!session[n]) {
+ session[n] = ac;
+ mutex_unlock(&session_lock);
+ return n;
+ }
+ }
+ mutex_unlock(&session_lock);
+ return -ENOMEM;
+}
+
+static void q6asm_session_free(struct audio_client *ac)
+{
+ pr_debug("%s: sessionid[%d]\n", __func__, ac->session);
+ rtac_remove_popp_from_adm_devices(ac->session);
+ mutex_lock(&session_lock);
+ session[ac->session] = 0;
+ mutex_unlock(&session_lock);
+ ac->session = 0;
+ return;
+}
+
+int q6asm_audio_client_buf_free(unsigned int dir,
+ struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+ pr_debug("%s: Session id %d\n", __func__, ac->session);
+ mutex_lock(&ac->cmd_lock);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[dir];
+ if (!port->buf) {
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+ }
+ cnt = port->max_buf_cnt - 1;
+
+ if (cnt >= 0) {
+ rc = q6asm_memory_unmap_regions(ac, dir,
+ port->buf[0].size,
+ port->max_buf_cnt);
+ if (rc < 0)
+ pr_err("%s CMD Memory_unmap_regions failed\n",
+ __func__);
+ }
+
+ while (cnt >= 0) {
+ if (port->buf[cnt].data) {
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ ion_unmap_kernel(port->buf[cnt].client,
+ port->buf[cnt].handle);
+ ion_free(port->buf[cnt].client,
+ port->buf[cnt].handle);
+ ion_client_destroy(port->buf[cnt].client);
+#else
+ pr_debug("%s:data[%p]phys[%p][%p] cnt[%d]"
+ "mem_buffer[%p]\n",
+ __func__, (void *)port->buf[cnt].data,
+ (void *)port->buf[cnt].phys,
+ (void *)&port->buf[cnt].phys, cnt,
+ (void *)port->buf[cnt].mem_buffer);
+ if (IS_ERR((void *)port->buf[cnt].mem_buffer))
+ pr_err("%s:mem buffer invalid, error ="
+ "%ld\n", __func__,
+ PTR_ERR((void *)port->buf[cnt].mem_buffer));
+ else {
+ if (iounmap(
+ port->buf[cnt].mem_buffer) < 0)
+ pr_err("%s: unmap buffer"
+ " failed\n", __func__);
+ }
+ free_contiguous_memory_by_paddr(
+ port->buf[cnt].phys);
+
+#endif
+ port->buf[cnt].data = NULL;
+ port->buf[cnt].phys = 0;
+ --(port->max_buf_cnt);
+ }
+ --cnt;
+ }
+ kfree(port->buf);
+ port->buf = NULL;
+ }
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+}
+
+int q6asm_audio_client_buf_free_contiguous(unsigned int dir,
+ struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+ pr_debug("%s: Session id %d\n", __func__, ac->session);
+ mutex_lock(&ac->cmd_lock);
+ port = &ac->port[dir];
+ if (!port->buf) {
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+ }
+ cnt = port->max_buf_cnt - 1;
+
+ if (cnt >= 0) {
+ rc = q6asm_memory_unmap(ac, port->buf[0].phys, dir);
+ if (rc < 0)
+ pr_err("%s CMD Memory_unmap_regions failed\n",
+ __func__);
+ }
+
+ if (port->buf[0].data) {
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ ion_unmap_kernel(port->buf[0].client, port->buf[0].handle);
+ ion_free(port->buf[0].client, port->buf[0].handle);
+ ion_client_destroy(port->buf[0].client);
+ pr_debug("%s:data[%p]phys[%p][%p]"
+ ", client[%p] handle[%p]\n",
+ __func__,
+ (void *)port->buf[0].data,
+ (void *)port->buf[0].phys,
+ (void *)&port->buf[0].phys,
+ (void *)port->buf[0].client,
+ (void *)port->buf[0].handle);
+#else
+ pr_debug("%s:data[%p]phys[%p][%p]"
+ "mem_buffer[%p]\n",
+ __func__,
+ (void *)port->buf[0].data,
+ (void *)port->buf[0].phys,
+ (void *)&port->buf[0].phys,
+ (void *)port->buf[0].mem_buffer);
+ if (IS_ERR((void *)port->buf[0].mem_buffer))
+ pr_err("%s:mem buffer invalid, error ="
+ "%ld\n", __func__,
+ PTR_ERR((void *)port->buf[0].mem_buffer));
+ else {
+ if (iounmap(
+ port->buf[0].mem_buffer) < 0)
+ pr_err("%s: unmap buffer"
+ " failed\n", __func__);
+ }
+ free_contiguous_memory_by_paddr(port->buf[0].phys);
+#endif
+ }
+
+ while (cnt >= 0) {
+ port->buf[cnt].data = NULL;
+ port->buf[cnt].phys = 0;
+ cnt--;
+ }
+ port->max_buf_cnt = 0;
+ kfree(port->buf);
+ port->buf = NULL;
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+}
+
+void q6asm_audio_client_free(struct audio_client *ac)
+{
+ int loopcnt;
+ struct audio_port_data *port;
+ if (!ac || !ac->session)
+ return;
+ pr_debug("%s: Session id %d\n", __func__, ac->session);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
+ port = &ac->port[loopcnt];
+ if (!port->buf)
+ continue;
+ pr_debug("%s:loopcnt = %d\n", __func__, loopcnt);
+ q6asm_audio_client_buf_free(loopcnt, ac);
+ }
+ }
+
+ apr_deregister(ac->apr);
+ q6asm_session_free(ac);
+
+ pr_debug("%s: APR De-Register\n", __func__);
+ if (atomic_read(&this_mmap.ref_cnt) <= 0) {
+ pr_err("%s: APR Common Port Already Closed\n", __func__);
+ goto done;
+ }
+
+ atomic_dec(&this_mmap.ref_cnt);
+ if (atomic_read(&this_mmap.ref_cnt) == 0) {
+ apr_deregister(this_mmap.apr);
+ pr_debug("%s:APR De-Register common port\n", __func__);
+ }
+done:
+ kfree(ac);
+ return;
+}
+
+int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode)
+{
+ if (ac == NULL) {
+ pr_err("%s APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ if ((mode == ASYNC_IO_MODE) || (mode == SYNC_IO_MODE)) {
+ ac->io_mode = mode;
+ pr_debug("%s:Set Mode to %d\n", __func__, ac->io_mode);
+ return 0;
+ } else {
+ pr_err("%s:Not an valid IO Mode:%d\n", __func__, ac->io_mode);
+ return -EINVAL;
+ }
+}
+
+struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv)
+{
+ struct audio_client *ac;
+ int n;
+ int lcnt = 0;
+
+ ac = kzalloc(sizeof(struct audio_client), GFP_KERNEL);
+ if (!ac)
+ return NULL;
+ n = q6asm_session_alloc(ac);
+ if (n <= 0)
+ goto fail_session;
+ ac->session = n;
+ ac->cb = cb;
+ ac->priv = priv;
+ ac->io_mode = SYNC_IO_MODE;
+ ac->apr = apr_register("ADSP", "ASM", \
+ (apr_fn)q6asm_callback,\
+ ((ac->session) << 8 | 0x0001),\
+ ac);
+
+ if (ac->apr == NULL) {
+ pr_err("%s Registration with APR failed\n", __func__);
+ goto fail;
+ }
+ rtac_set_asm_handle(n, ac->apr);
+
+ pr_debug("%s Registering the common port with APR\n", __func__);
+ if (atomic_read(&this_mmap.ref_cnt) == 0) {
+ this_mmap.apr = apr_register("ADSP", "ASM", \
+ (apr_fn)q6asm_mmapcallback,\
+ 0x0FFFFFFFF, &this_mmap);
+ if (this_mmap.apr == NULL) {
+ pr_debug("%s Unable to register \
+ APR ASM common port \n", __func__);
+ goto fail;
+ }
+ }
+
+ atomic_inc(&this_mmap.ref_cnt);
+ init_waitqueue_head(&ac->cmd_wait);
+ init_waitqueue_head(&ac->time_wait);
+ atomic_set(&ac->time_flag, 1);
+ mutex_init(&ac->cmd_lock);
+ for (lcnt = 0; lcnt <= OUT; lcnt++) {
+ mutex_init(&ac->port[lcnt].lock);
+ spin_lock_init(&ac->port[lcnt].dsp_lock);
+ }
+ atomic_set(&ac->cmd_state, 0);
+
+ pr_debug("%s: session[%d]\n", __func__, ac->session);
+
+ return ac;
+fail:
+ q6asm_audio_client_free(ac);
+ return NULL;
+fail_session:
+ kfree(ac);
+ return NULL;
+}
+
+struct audio_client *q6asm_get_audio_client(int session_id)
+{
+ if ((session_id <= 0) || (session_id > SESSION_MAX)) {
+ pr_err("%s: invalid session: %d\n", __func__, session_id);
+ goto err;
+ }
+
+ if (!session[session_id]) {
+ pr_err("%s: session not active: %d\n", __func__, session_id);
+ goto err;
+ }
+
+ return session[session_id];
+err:
+ return NULL;
+}
+
+int q6asm_audio_client_buf_alloc(unsigned int dir,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt)
+{
+ int cnt = 0;
+ int rc = 0;
+ struct audio_buffer *buf;
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ int len;
+#endif
+
+ if (!(ac) || ((dir != IN) && (dir != OUT)))
+ return -EINVAL;
+
+ pr_debug("%s: session[%d]bufsz[%d]bufcnt[%d]\n", __func__, ac->session,
+ bufsz, bufcnt);
+
+ if (ac->session <= 0 || ac->session > 8)
+ goto fail;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ if (ac->port[dir].buf) {
+ pr_debug("%s: buffer already allocated\n", __func__);
+ return 0;
+ }
+ mutex_lock(&ac->cmd_lock);
+ buf = kzalloc(((sizeof(struct audio_buffer))*bufcnt),
+ GFP_KERNEL);
+
+ if (!buf) {
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ ac->port[dir].buf = buf;
+
+ while (cnt < bufcnt) {
+ if (bufsz > 0) {
+ if (!buf[cnt].data) {
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ buf[cnt].client = msm_ion_client_create
+ (UINT_MAX, "audio_client");
+ if (IS_ERR_OR_NULL((void *)
+ buf[cnt].client)) {
+ pr_err("%s: ION create client"
+ " for AUDIO failed\n",
+ __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[cnt].handle = ion_alloc
+ (buf[cnt].client, bufsz, SZ_4K,
+ (0x1 << ION_AUDIO_HEAP_ID));
+ if (IS_ERR_OR_NULL((void *)
+ buf[cnt].handle)) {
+ pr_err("%s: ION memory"
+ " allocation for AUDIO failed\n",
+ __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ rc = ion_phys(buf[cnt].client,
+ buf[cnt].handle,
+ (ion_phys_addr_t *)
+ &buf[cnt].phys,
+ (size_t *)&len);
+ if (rc) {
+ pr_err("%s: ION Get Physical"
+ " for AUDIO failed, rc = %d\n",
+ __func__, rc);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ buf[cnt].data = ion_map_kernel
+ (buf[cnt].client, buf[cnt].handle,
+ 0);
+ if (IS_ERR_OR_NULL((void *)
+ buf[cnt].data)) {
+ pr_err("%s: ION memory"
+ " mapping for AUDIO failed\n", __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ memset((void *)buf[cnt].data, 0, bufsz);
+#else
+ unsigned int flags = 0;
+ buf[cnt].phys =
+ allocate_contiguous_ebi_nomap(bufsz,
+ SZ_4K);
+ if (!buf[cnt].phys) {
+ pr_err("%s:Buf alloc failed "
+ " size=%d\n", __func__,
+ bufsz);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[cnt].mem_buffer =
+ ioremap(buf[cnt].phys, bufsz);
+ if (IS_ERR(
+ (void *)buf[cnt].mem_buffer)) {
+ pr_err("%s:map_buffer failed,"
+ "error = %ld\n",
+ __func__, PTR_ERR((void *)buf[cnt].mem_buffer));
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[cnt].data =
+ buf[cnt].mem_buffer;
+ if (!buf[cnt].data) {
+ pr_err("%s:invalid vaddr,"
+ " iomap failed\n", __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+#endif
+ buf[cnt].used = 1;
+ buf[cnt].size = bufsz;
+ buf[cnt].actual_size = bufsz;
+ pr_debug("%s data[%p]phys[%p][%p]\n",
+ __func__,
+ (void *)buf[cnt].data,
+ (void *)buf[cnt].phys,
+ (void *)&buf[cnt].phys);
+ cnt++;
+ }
+ }
+ }
+ ac->port[dir].max_buf_cnt = cnt;
+
+ mutex_unlock(&ac->cmd_lock);
+ rc = q6asm_memory_map_regions(ac, dir, bufsz, cnt);
+ if (rc < 0) {
+ pr_err("%s:CMD Memory_map_regions failed\n", __func__);
+ goto fail;
+ }
+ }
+ return 0;
+fail:
+ q6asm_audio_client_buf_free(dir, ac);
+ return -EINVAL;
+}
+
+int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt)
+{
+ int cnt = 0;
+ int rc = 0;
+ struct audio_buffer *buf;
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ int len;
+#else
+ int flags = 0;
+#endif
+ if (!(ac) || ((dir != IN) && (dir != OUT)))
+ return -EINVAL;
+
+ pr_debug("%s: session[%d]bufsz[%d]bufcnt[%d]\n",
+ __func__, ac->session,
+ bufsz, bufcnt);
+
+ if (ac->session <= 0 || ac->session > 8)
+ goto fail;
+
+ if (ac->port[dir].buf) {
+ pr_debug("%s: buffer already allocated\n", __func__);
+ return 0;
+ }
+ mutex_lock(&ac->cmd_lock);
+ buf = kzalloc(((sizeof(struct audio_buffer))*bufcnt),
+ GFP_KERNEL);
+
+ if (!buf) {
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ ac->port[dir].buf = buf;
+
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ buf[0].client = msm_ion_client_create(UINT_MAX, "audio_client");
+ if (IS_ERR_OR_NULL((void *)buf[0].client)) {
+ pr_err("%s: ION create client for AUDIO failed\n", __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[0].handle = ion_alloc(buf[0].client, bufsz * bufcnt, SZ_4K,
+ (0x1 << ION_AUDIO_HEAP_ID));
+ if (IS_ERR_OR_NULL((void *) buf[0].handle)) {
+ pr_err("%s: ION memory allocation for AUDIO failed\n",
+ __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ rc = ion_phys(buf[0].client, buf[0].handle,
+ (ion_phys_addr_t *)&buf[0].phys, (size_t *)&len);
+ if (rc) {
+ pr_err("%s: ION Get Physical for AUDIO failed, rc = %d\n",
+ __func__, rc);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ buf[0].data = ion_map_kernel(buf[0].client, buf[0].handle, 0);
+ if (IS_ERR_OR_NULL((void *) buf[0].data)) {
+ pr_err("%s: ION memory mapping for AUDIO failed\n", __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ memset((void *)buf[0].data, 0, (bufsz * bufcnt));
+#else
+ buf[0].phys = allocate_contiguous_ebi_nomap(bufsz * bufcnt,
+ SZ_4K);
+ if (!buf[0].phys) {
+ pr_err("%s:Buf alloc failed "
+ " size=%d, bufcnt=%d\n", __func__,
+ bufsz, bufcnt);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ buf[0].mem_buffer = ioremap(buf[0].phys, bufsz * bufcnt);
+ if (IS_ERR((void *)buf[cnt].mem_buffer)) {
+ pr_err("%s:map_buffer failed,"
+ "error = %ld\n",
+ __func__, PTR_ERR((void *)buf[0].mem_buffer));
+
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[0].data = buf[0].mem_buffer;
+#endif
+ if (!buf[0].data) {
+ pr_err("%s:invalid vaddr,"
+ " iomap failed\n", __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ buf[0].used = dir ^ 1;
+ buf[0].size = bufsz;
+ buf[0].actual_size = bufsz;
+ cnt = 1;
+ while (cnt < bufcnt) {
+ if (bufsz > 0) {
+ buf[cnt].data = buf[0].data + (cnt * bufsz);
+ buf[cnt].phys = buf[0].phys + (cnt * bufsz);
+ if (!buf[cnt].data) {
+ pr_err("%s Buf alloc failed\n",
+ __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[cnt].used = dir ^ 1;
+ buf[cnt].size = bufsz;
+ buf[cnt].actual_size = bufsz;
+ pr_debug("%s data[%p]phys[%p][%p]\n", __func__,
+ (void *)buf[cnt].data,
+ (void *)buf[cnt].phys,
+ (void *)&buf[cnt].phys);
+ }
+ cnt++;
+ }
+ ac->port[dir].max_buf_cnt = cnt;
+ mutex_unlock(&ac->cmd_lock);
+ rc = q6asm_memory_map(ac, buf[0].phys, dir, bufsz, cnt);
+ if (rc < 0) {
+ pr_err("%s:CMD Memory_map_regions failed\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ q6asm_audio_client_buf_free_contiguous(dir, ac);
+ return -EINVAL;
+}
+
+static int32_t q6asm_mmapcallback(struct apr_client_data *data, void *priv)
+{
+ uint32_t token;
+ uint32_t *payload = data->payload;
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("%s: Reset event is received: %d %d apr[%p]\n",
+ __func__,
+ data->reset_event,
+ data->reset_proc,
+ this_mmap.apr);
+ apr_reset(this_mmap.apr);
+ this_mmap.apr = NULL;
+ atomic_set(&this_mmap.cmd_state, 0);
+ return 0;
+ }
+
+ pr_debug("%s:ptr0[0x%x]ptr1[0x%x]opcode[0x%x]"
+ "token[0x%x]payload_s[%d] src[%d] dest[%d]\n", __func__,
+ payload[0], payload[1], data->opcode, data->token,
+ data->payload_size, data->src_port, data->dest_port);
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ token = data->token;
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_MEMORY_MAP:
+ case ASM_SESSION_CMD_MEMORY_UNMAP:
+ case ASM_SESSION_CMD_MEMORY_MAP_REGIONS:
+ case ASM_SESSION_CMD_MEMORY_UNMAP_REGIONS:
+ pr_debug("%s:command[0x%x]success [0x%x]\n",
+ __func__, payload[0], payload[1]);
+ if (atomic_read(&this_mmap.cmd_state)) {
+ atomic_set(&this_mmap.cmd_state, 0);
+ wake_up(&this_mmap.cmd_wait);
+ }
+ break;
+ default:
+ pr_debug("%s:command[0x%x] not expecting rsp\n",
+ __func__, payload[0]);
+ break;
+ }
+ }
+ return 0;
+}
+
+
+static int32_t q6asm_callback(struct apr_client_data *data, void *priv)
+{
+ int i = 0;
+ struct audio_client *ac = (struct audio_client *)priv;
+ uint32_t token;
+ unsigned long dsp_flags;
+ uint32_t *payload;
+
+
+ if ((ac == NULL) || (data == NULL)) {
+ pr_err("ac or priv NULL\n");
+ return -EINVAL;
+ }
+ if (ac->session <= 0 || ac->session > 8) {
+ pr_err("%s:Session ID is invalid, session = %d\n", __func__,
+ ac->session);
+ return -EINVAL;
+ }
+
+ payload = data->payload;
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("q6asm_callback: Reset event is received: %d %d apr[%p]\n",
+ data->reset_event, data->reset_proc, ac->apr);
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ (uint32_t *)data->payload, ac->priv);
+ apr_reset(ac->apr);
+ return 0;
+ }
+
+ pr_debug("%s: session[%d]opcode[0x%x] \
+ token[0x%x]payload_s[%d] src[%d] dest[%d]\n", __func__,
+ ac->session, data->opcode,
+ data->token, data->payload_size, data->src_port,
+ data->dest_port);
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ token = data->token;
+ switch (payload[0]) {
+ case ASM_STREAM_CMD_SET_PP_PARAMS:
+ if (rtac_make_asm_callback(ac->session, payload,
+ data->payload_size))
+ break;
+ case ASM_SESSION_CMD_PAUSE:
+ case ASM_DATA_CMD_EOS:
+ case ASM_STREAM_CMD_CLOSE:
+ case ASM_STREAM_CMD_FLUSH:
+ case ASM_SESSION_CMD_RUN:
+ case ASM_SESSION_CMD_REGISTER_FOR_TX_OVERFLOW_EVENTS:
+ case ASM_STREAM_CMD_FLUSH_READBUFS:
+ pr_debug("%s:Payload = [0x%x]\n", __func__, payload[0]);
+ if (token != ac->session) {
+ pr_err("%s:Invalid session[%d] rxed expected[%d]",
+ __func__, token, ac->session);
+ return -EINVAL;
+ }
+ case ASM_STREAM_CMD_OPEN_READ:
+ case ASM_STREAM_CMD_OPEN_WRITE:
+ case ASM_STREAM_CMD_OPEN_READWRITE:
+ case ASM_DATA_CMD_MEDIA_FORMAT_UPDATE:
+ case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+ case ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED:
+ if (atomic_read(&ac->cmd_state)) {
+ atomic_set(&ac->cmd_state, 0);
+ wake_up(&ac->cmd_wait);
+ }
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ (uint32_t *)data->payload, ac->priv);
+ break;
+ default:
+ pr_debug("%s:command[0x%x] not expecting rsp\n",
+ __func__, payload[0]);
+ break;
+ }
+ return 0;
+ }
+
+ switch (data->opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE:{
+ struct audio_port_data *port = &ac->port[IN];
+ pr_debug("%s: Rxed opcode[0x%x] status[0x%x] token[%d]",
+ __func__, payload[0], payload[1],
+ data->token);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ if (port->buf == NULL) {
+ pr_err("%s: Unexpected Write Done\n",
+ __func__);
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&port->dsp_lock, dsp_flags);
+ if (port->buf[data->token].phys !=
+ payload[0]) {
+ pr_err("Buf expected[%p]rxed[%p]\n",\
+ (void *)port->buf[data->token].phys,\
+ (void *)payload[0]);
+ spin_unlock_irqrestore(&port->dsp_lock,
+ dsp_flags);
+ return -EINVAL;
+ }
+ token = data->token;
+ port->buf[token].used = 1;
+ spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
+#ifdef CONFIG_DEBUG_FS
+ if (out_enable_flag) {
+ /* For first Write done log the time and reset
+ out_cold_index*/
+ if (out_cold_index != 1) {
+ do_gettimeofday(&out_cold_tv);
+ pr_debug("COLD: apr_send_pkt at %ld \
+ sec %ld microsec\n",\
+ out_cold_tv.tv_sec,\
+ out_cold_tv.tv_usec);
+ out_cold_index = 1;
+ }
+ pr_debug("out_enable_flag %ld",\
+ out_enable_flag);
+ }
+#endif
+ for (i = 0; i < port->max_buf_cnt; i++)
+ pr_debug("%d ", port->buf[i].used);
+
+ }
+ break;
+ }
+ case ASM_STREAM_CMDRSP_GET_PP_PARAMS:
+ rtac_make_asm_callback(ac->session, payload,
+ data->payload_size);
+ break;
+ case ASM_DATA_EVENT_READ_DONE:{
+
+ struct audio_port_data *port = &ac->port[OUT];
+#ifdef CONFIG_DEBUG_FS
+ if (in_enable_flag) {
+ /* when in_cont_index == 7, DSP would be
+ * writing into the 8th 512 byte buffer and this
+ * timestamp is tapped here.Once done it then writes
+ * to 9th 512 byte buffer.These two buffers(8th, 9th)
+ * reach the test application in 5th iteration and that
+ * timestamp is tapped at user level. The difference
+ * of these two timestamps gives us the time between
+ * the time at which dsp started filling the sample
+ * required and when it reached the test application.
+ * Hence continuous input latency
+ */
+ if (in_cont_index == 7) {
+ do_gettimeofday(&in_cont_tv);
+ pr_err("In_CONT:previous read buffer done \
+ at %ld sec %ld microsec\n",\
+ in_cont_tv.tv_sec, in_cont_tv.tv_usec);
+ }
+ }
+#endif
+ pr_debug("%s:R-D: status=%d buff_add=%x act_size=%d offset=%d\n",
+ __func__, payload[READDONE_IDX_STATUS],
+ payload[READDONE_IDX_BUFFER],
+ payload[READDONE_IDX_SIZE],
+ payload[READDONE_IDX_OFFSET]);
+ pr_debug("%s:R-D:msw_ts=%d lsw_ts=%d flags=%d id=%d num=%d\n",
+ __func__, payload[READDONE_IDX_MSW_TS],
+ payload[READDONE_IDX_LSW_TS],
+ payload[READDONE_IDX_FLAGS],
+ payload[READDONE_IDX_ID],
+ payload[READDONE_IDX_NUMFRAMES]);
+#ifdef CONFIG_DEBUG_FS
+ if (in_enable_flag) {
+ in_cont_index++;
+ }
+#endif
+ if (ac->io_mode == SYNC_IO_MODE) {
+ if (port->buf == NULL) {
+ pr_err("%s: Unexpected Write Done\n", __func__);
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&port->dsp_lock, dsp_flags);
+ token = data->token;
+ port->buf[token].used = 0;
+ if (port->buf[token].phys !=
+ payload[READDONE_IDX_BUFFER]) {
+ pr_err("Buf expected[%p]rxed[%p]\n",\
+ (void *)port->buf[token].phys,\
+ (void *)payload[READDONE_IDX_BUFFER]);
+ spin_unlock_irqrestore(&port->dsp_lock,
+ dsp_flags);
+ break;
+ }
+ port->buf[token].actual_size =
+ payload[READDONE_IDX_SIZE];
+ spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
+ }
+ break;
+ }
+ case ASM_DATA_EVENT_EOS:
+ case ASM_DATA_CMDRSP_EOS:
+ pr_debug("%s:EOS ACK received: rxed opcode[0x%x]\n",
+ __func__, data->opcode);
+ break;
+ case ASM_STREAM_CMDRSP_GET_ENCDEC_PARAM:
+ break;
+ case ASM_SESSION_EVENT_TX_OVERFLOW:
+ pr_err("ASM_SESSION_EVENT_TX_OVERFLOW\n");
+ break;
+ case ASM_SESSION_CMDRSP_GET_SESSION_TIME:
+ pr_debug("%s: ASM_SESSION_CMDRSP_GET_SESSION_TIME, "
+ "payload[0] = %d, payload[1] = %d, "
+ "payload[2] = %d\n", __func__,
+ payload[0], payload[1], payload[2]);
+ ac->time_stamp = (uint64_t)(((uint64_t)payload[1] << 32) |
+ payload[2]);
+ if (atomic_read(&ac->time_flag)) {
+ atomic_set(&ac->time_flag, 0);
+ wake_up(&ac->time_wait);
+ }
+ break;
+ case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
+ case ASM_DATA_EVENT_ENC_SR_CM_NOTIFY:
+ pr_debug("%s: ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY, "
+ "payload[0] = %d, payload[1] = %d, "
+ "payload[2] = %d, payload[3] = %d\n", __func__,
+ payload[0], payload[1], payload[2],
+ payload[3]);
+ break;
+ }
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ data->payload, ac->priv);
+
+ return 0;
+}
+
+void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac, uint32_t *size,
+ uint32_t *index)
+{
+ void *data;
+ unsigned char idx;
+ struct audio_port_data *port;
+
+ if (!ac || ((dir != IN) && (dir != OUT)))
+ return NULL;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[dir];
+
+ mutex_lock(&port->lock);
+ idx = port->cpu_buf;
+ if (port->buf == NULL) {
+ pr_debug("%s:Buffer pointer null\n", __func__);
+ mutex_unlock(&port->lock);
+ return NULL;
+ }
+ /* dir 0: used = 0 means buf in use
+ dir 1: used = 1 means buf in use */
+ if (port->buf[idx].used == dir) {
+ /* To make it more robust, we could loop and get the
+ next avail buf, its risky though */
+ pr_debug("%s:Next buf idx[0x%x] not available,\
+ dir[%d]\n", __func__, idx, dir);
+ mutex_unlock(&port->lock);
+ return NULL;
+ }
+ *size = port->buf[idx].actual_size;
+ *index = port->cpu_buf;
+ data = port->buf[idx].data;
+ pr_debug("%s:session[%d]index[%d] data[%p]size[%d]\n",
+ __func__,
+ ac->session,
+ port->cpu_buf,
+ data, *size);
+ /* By default increase the cpu_buf cnt
+ user accesses this function,increase cpu
+ buf(to avoid another api)*/
+ port->buf[idx].used = dir;
+ port->cpu_buf = ((port->cpu_buf + 1) & (port->max_buf_cnt - 1));
+ mutex_unlock(&port->lock);
+ return data;
+ }
+ return NULL;
+}
+
+void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac,
+ uint32_t *size, uint32_t *index)
+{
+ void *data;
+ unsigned char idx;
+ struct audio_port_data *port;
+
+ if (!ac || ((dir != IN) && (dir != OUT)))
+ return NULL;
+
+ port = &ac->port[dir];
+
+ idx = port->cpu_buf;
+ if (port->buf == NULL) {
+ pr_debug("%s:Buffer pointer null\n", __func__);
+ return NULL;
+ }
+ /*
+ * dir 0: used = 0 means buf in use
+ * dir 1: used = 1 means buf in use
+ */
+ if (port->buf[idx].used == dir) {
+ /*
+ * To make it more robust, we could loop and get the
+ * next avail buf, its risky though
+ */
+ pr_debug("%s:Next buf idx[0x%x] not available,\
+ dir[%d]\n", __func__, idx, dir);
+ return NULL;
+ }
+ *size = port->buf[idx].actual_size;
+ *index = port->cpu_buf;
+ data = port->buf[idx].data;
+ pr_debug("%s:session[%d]index[%d] data[%p]size[%d]\n",
+ __func__, ac->session, port->cpu_buf,
+ data, *size);
+ /*
+ * By default increase the cpu_buf cnt
+ * user accesses this function,increase cpu
+ * buf(to avoid another api)
+ */
+ port->buf[idx].used = dir;
+ port->cpu_buf = ((port->cpu_buf + 1) & (port->max_buf_cnt - 1));
+ return data;
+}
+
+int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac)
+{
+ int ret = -1;
+ struct audio_port_data *port;
+ uint32_t idx;
+
+ if (!ac || (dir != OUT))
+ return ret;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[dir];
+
+ mutex_lock(&port->lock);
+ idx = port->dsp_buf;
+
+ if (port->buf[idx].used == (dir ^ 1)) {
+ /* To make it more robust, we could loop and get the
+ next avail buf, its risky though */
+ pr_err("Next buf idx[0x%x] not available, dir[%d]\n",
+ idx, dir);
+ mutex_unlock(&port->lock);
+ return ret;
+ }
+ pr_debug("%s: session[%d]dsp_buf=%d cpu_buf=%d\n", __func__,
+ ac->session, port->dsp_buf, port->cpu_buf);
+ ret = ((port->dsp_buf != port->cpu_buf) ? 0 : -1);
+ mutex_unlock(&port->lock);
+ }
+ return ret;
+}
+
+static void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg)
+{
+ pr_debug("%s:session=%d pkt size=%d cmd_flg=%d\n", __func__, pkt_size,
+ cmd_flg, ac->session);
+ mutex_lock(&ac->cmd_lock);
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(sizeof(struct apr_hdr)),\
+ APR_PKT_VER);
+ hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
+ hdr->src_domain = APR_DOMAIN_APPS;
+ hdr->dest_svc = APR_SVC_ASM;
+ hdr->dest_domain = APR_DOMAIN_ADSP;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ if (cmd_flg) {
+ hdr->token = ac->session;
+ atomic_set(&ac->cmd_state, 1);
+ }
+ hdr->pkt_size = pkt_size;
+ mutex_unlock(&ac->cmd_lock);
+ return;
+}
+
+static void q6asm_add_mmaphdr(struct apr_hdr *hdr, uint32_t pkt_size,
+ uint32_t cmd_flg)
+{
+ pr_debug("%s:pkt size=%d cmd_flg=%d\n", __func__, pkt_size, cmd_flg);
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ hdr->src_port = 0;
+ hdr->dest_port = 0;
+ if (cmd_flg) {
+ hdr->token = 0;
+ atomic_set(&this_mmap.cmd_state, 1);
+ }
+ hdr->pkt_size = pkt_size;
+ return;
+}
+
+int q6asm_open_read(struct audio_client *ac,
+ uint32_t format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_read open;
+#ifdef CONFIG_DEBUG_FS
+ in_cont_index = 0;
+#endif
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s:session[%d]", __func__, ac->session);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ;
+ /* Stream prio : High, provide meta info with encoded frames */
+ open.src_endpoint = ASM_END_POINT_DEVICE_MATRIX;
+
+ open.pre_proc_top = get_asm_topology();
+ if (open.pre_proc_top == 0)
+ open.pre_proc_top = DEFAULT_POPP_TOPOLOGY;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open.uMode = STREAM_PRIORITY_HIGH;
+ open.format = LINEAR_PCM;
+ break;
+ case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
+ open.uMode = STREAM_PRIORITY_HIGH;
+ open.format = MULTI_CHANNEL_PCM;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.uMode = BUFFER_META_ENABLE | STREAM_PRIORITY_HIGH;
+ open.format = MPEG4_AAC;
+ break;
+ case FORMAT_V13K:
+ open.uMode = BUFFER_META_ENABLE | STREAM_PRIORITY_HIGH;
+ open.format = V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ open.uMode = BUFFER_META_ENABLE | STREAM_PRIORITY_HIGH;
+ open.format = EVRC_FS;
+ break;
+ case FORMAT_AMRNB:
+ open.uMode = BUFFER_META_ENABLE | STREAM_PRIORITY_HIGH;
+ open.format = AMRNB_FS;
+ break;
+ case FORMAT_AMRWB:
+ open.uMode = BUFFER_META_ENABLE | STREAM_PRIORITY_HIGH;
+ open.format = AMRWB_FS;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", format);
+ goto fail_cmd;
+ }
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("open failed op[0x%x]rc[%d]\n", \
+ open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout. waited for OPEN_WRITE rc[%d]\n", __func__,
+ rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_write_compressed open;
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] wr_format[0x%x]", __func__, ac->session,
+ format);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED;
+
+ switch (format) {
+ case FORMAT_AC3:
+ open.format = AC3_DECODER;
+ break;
+ case FORMAT_EAC3:
+ open.format = EAC3_DECODER;
+ break;
+ case FORMAT_MP3:
+ open.format = MP3;
+ break;
+ case FORMAT_DTS:
+ open.format = DTS;
+ break;
+ case FORMAT_AAC:
+ open.format = MPEG4_AAC;
+ break;
+ case FORMAT_ATRAC:
+ open.format = ATRAC;
+ break;
+ case FORMAT_WMA_V10PRO:
+ open.format = WMA_V10PRO;
+ break;
+ case FORMAT_MAT:
+ open.format = MAT;
+ break;
+ default:
+ pr_err("%s: Invalid format[%d]\n", __func__, format);
+ goto fail_cmd;
+ }
+ /*Below flag indicates the DSP that Compressed audio input
+ stream is not IEC 61937 or IEC 60958 packetizied*/
+ open.flags = 0x00000000;
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("%s: open failed op[0x%x]rc[%d]\n", \
+ __func__, open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout. waited for OPEN_WRITE rc[%d]\n", __func__,
+ rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_open_write(struct audio_client *ac, uint32_t format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_write open;
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] wr_format[0x%x]", __func__, ac->session,
+ format);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE;
+ open.uMode = STREAM_PRIORITY_HIGH;
+ /* source endpoint : matrix */
+ open.sink_endpoint = ASM_END_POINT_DEVICE_MATRIX;
+ open.stream_handle = 0x00;
+
+ open.post_proc_top = get_asm_topology();
+ if (open.post_proc_top == 0)
+ open.post_proc_top = DEFAULT_POPP_TOPOLOGY;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open.format = LINEAR_PCM;
+ break;
+ case FORMAT_MULTI_CHANNEL_LINEAR_PCM:
+ open.format = MULTI_CHANNEL_PCM;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.format = MPEG4_AAC;
+ break;
+ case FORMAT_MPEG4_MULTI_AAC:
+ open.format = MPEG4_MULTI_AAC;
+ break;
+ case FORMAT_WMA_V9:
+ open.format = WMA_V9;
+ break;
+ case FORMAT_WMA_V10PRO:
+ open.format = WMA_V10PRO;
+ break;
+ case FORMAT_MP3:
+ open.format = MP3;
+ break;
+ default:
+ pr_err("%s: Invalid format[%d]\n", __func__, format);
+ goto fail_cmd;
+ }
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("%s: open failed op[0x%x]rc[%d]\n", \
+ __func__, open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout. waited for OPEN_WRITE rc[%d]\n", __func__,
+ rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_open_read_write(struct audio_client *ac,
+ uint32_t rd_format,
+ uint32_t wr_format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_read_write open;
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d]", __func__, ac->session);
+ pr_debug("wr_format[0x%x]rd_format[0x%x]",
+ wr_format, rd_format);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_READWRITE;
+
+ open.uMode = BUFFER_META_ENABLE | STREAM_PRIORITY_NORMAL;
+ /* source endpoint : matrix */
+ open.post_proc_top = get_asm_topology();
+ if (open.post_proc_top == 0)
+ open.post_proc_top = DEFAULT_POPP_TOPOLOGY;
+
+ switch (wr_format) {
+ case FORMAT_LINEAR_PCM:
+ open.write_format = LINEAR_PCM;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.write_format = MPEG4_AAC;
+ break;
+ case FORMAT_MPEG4_MULTI_AAC:
+ open.write_format = MPEG4_MULTI_AAC;
+ break;
+ case FORMAT_WMA_V9:
+ open.write_format = WMA_V9;
+ break;
+ case FORMAT_WMA_V10PRO:
+ open.write_format = WMA_V10PRO;
+ break;
+ case FORMAT_AMRNB:
+ open.write_format = AMRNB_FS;
+ break;
+ case FORMAT_AMRWB:
+ open.write_format = AMRWB_FS;
+ break;
+ case FORMAT_V13K:
+ open.write_format = V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ open.write_format = EVRC_FS;
+ break;
+ case FORMAT_EVRCB:
+ open.write_format = EVRCB_FS;
+ break;
+ case FORMAT_EVRCWB:
+ open.write_format = EVRCWB_FS;
+ break;
+ case FORMAT_MP3:
+ open.write_format = MP3;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", wr_format);
+ goto fail_cmd;
+ }
+
+ switch (rd_format) {
+ case FORMAT_LINEAR_PCM:
+ open.read_format = LINEAR_PCM;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.read_format = MPEG4_AAC;
+ break;
+ case FORMAT_V13K:
+ open.read_format = V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ open.read_format = EVRC_FS;
+ break;
+ case FORMAT_AMRNB:
+ open.read_format = AMRNB_FS;
+ break;
+ case FORMAT_AMRWB:
+ open.read_format = AMRWB_FS;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", rd_format);
+ goto fail_cmd;
+ }
+ pr_debug("%s:rdformat[0x%x]wrformat[0x%x]\n", __func__,
+ open.read_format, open.write_format);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("open failed op[0x%x]rc[%d]\n", \
+ open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for OPEN_WRITE rc[%d]\n", rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts)
+{
+ struct asm_stream_cmd_run run;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s session[%d]", __func__, ac->session);
+ q6asm_add_hdr(ac, &run.hdr, sizeof(run), TRUE);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN;
+ run.flags = flags;
+ run.msw_ts = msw_ts;
+ run.lsw_ts = lsw_ts;
+#ifdef CONFIG_DEBUG_FS
+ if (out_enable_flag) {
+ do_gettimeofday(&out_cold_tv);
+ pr_debug("COLD: apr_send_pkt at %ld sec %ld microsec\n",\
+ out_cold_tv.tv_sec, out_cold_tv.tv_usec);
+ }
+#endif
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &run);
+ if (rc < 0) {
+ pr_err("Commmand run failed[%d]", rc);
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for run success rc[%d]", rc);
+ goto fail_cmd;
+ }
+
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts)
+{
+ struct asm_stream_cmd_run run;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s:APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("session[%d]", ac->session);
+ q6asm_add_hdr_async(ac, &run.hdr, sizeof(run), TRUE);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN;
+ run.flags = flags;
+ run.msw_ts = msw_ts;
+ run.lsw_ts = lsw_ts;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &run);
+ if (rc < 0) {
+ pr_err("%s:Commmand run failed[%d]", __func__, rc);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+int q6asm_enc_cfg_blk_aac(struct audio_client *ac,
+ uint32_t frames_per_buf,
+ uint32_t sample_rate, uint32_t channels,
+ uint32_t bit_rate, uint32_t mode, uint32_t format)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]SR[%d]ch[%d]bitrate[%d]mode[%d]"
+ "format[%d]", __func__, ac->session, frames_per_buf,
+ sample_rate, channels, bit_rate, mode, format);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+ enc_cfg.enc_blk.frames_per_buf = frames_per_buf;
+ enc_cfg.enc_blk.format_id = MPEG4_AAC;
+ enc_cfg.enc_blk.cfg_size = sizeof(struct asm_aac_read_cfg);
+ enc_cfg.enc_blk.cfg.aac.bitrate = bit_rate;
+ enc_cfg.enc_blk.cfg.aac.enc_mode = mode;
+ enc_cfg.enc_blk.cfg.aac.format = format;
+ enc_cfg.enc_blk.cfg.aac.ch_cfg = channels;
+ enc_cfg.enc_blk.cfg.aac.sample_rate = sample_rate;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d, rate = %d, channels = %d\n", __func__,
+ ac->session, rate, channels);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+ enc_cfg.enc_blk.frames_per_buf = 1;
+ enc_cfg.enc_blk.format_id = LINEAR_PCM;
+ enc_cfg.enc_blk.cfg_size = sizeof(struct asm_pcm_cfg);
+ enc_cfg.enc_blk.cfg.pcm.ch_cfg = channels;
+ enc_cfg.enc_blk.cfg.pcm.bits_per_sample = 16;
+ enc_cfg.enc_blk.cfg.pcm.sample_rate = rate;
+ enc_cfg.enc_blk.cfg.pcm.is_signed = 1;
+ enc_cfg.enc_blk.cfg.pcm.interleaved = 1;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd open failed\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout opcode[0x%x] ", enc_cfg.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d, rate = %d, channels = %d\n", __func__,
+ ac->session, rate, channels);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+ enc_cfg.enc_blk.frames_per_buf = 1;
+ enc_cfg.enc_blk.format_id = MULTI_CHANNEL_PCM;
+ enc_cfg.enc_blk.cfg_size =
+ sizeof(struct asm_multi_channel_pcm_fmt_blk);
+ enc_cfg.enc_blk.cfg.mpcm.num_channels = channels;
+ enc_cfg.enc_blk.cfg.mpcm.bits_per_sample = 16;
+ enc_cfg.enc_blk.cfg.mpcm.sample_rate = rate;
+ enc_cfg.enc_blk.cfg.mpcm.is_signed = 1;
+ enc_cfg.enc_blk.cfg.mpcm.is_interleaved = 1;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[0] = PCM_CHANNEL_FL;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[1] = PCM_CHANNEL_FR;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[2] = PCM_CHANNEL_RB;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[3] = PCM_CHANNEL_LB;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[4] = 0;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[5] = 0;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[6] = 0;
+ enc_cfg.enc_blk.cfg.mpcm.channel_mapping[7] = 0;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd open failed\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout opcode[0x%x] ", enc_cfg.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enable_sbrps(struct audio_client *ac,
+ uint32_t sbr_ps_enable)
+{
+ struct asm_stream_cmd_encdec_sbr sbrps;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d\n", __func__, ac->session);
+
+ q6asm_add_hdr(ac, &sbrps.hdr, sizeof(sbrps), TRUE);
+
+ sbrps.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ sbrps.param_id = ASM_ENABLE_SBR_PS;
+ sbrps.param_size = sizeof(struct asm_sbr_ps);
+ sbrps.sbr_ps.enable = sbr_ps_enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &sbrps);
+ if (rc < 0) {
+ pr_err("Command opcode[0x%x]paramid[0x%x] failed\n",
+ ASM_STREAM_CMD_SET_ENCDEC_PARAM,
+ ASM_ENABLE_SBR_PS);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout opcode[0x%x] ", sbrps.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_cfg_dual_mono_aac(struct audio_client *ac,
+ uint16_t sce_left, uint16_t sce_right)
+{
+ struct asm_stream_cmd_encdec_dualmono dual_mono;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d, sce_left = %d, sce_right = %d\n",
+ __func__, ac->session, sce_left, sce_right);
+
+ q6asm_add_hdr(ac, &dual_mono.hdr, sizeof(dual_mono), TRUE);
+
+ dual_mono.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ dual_mono.param_id = ASM_CONFIGURE_DUAL_MONO;
+ dual_mono.param_size = sizeof(struct asm_dual_mono);
+ dual_mono.channel_map.sce_left = sce_left;
+ dual_mono.channel_map.sce_right = sce_right;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &dual_mono);
+ if (rc < 0) {
+ pr_err("%s:Command opcode[0x%x]paramid[0x%x] failed\n",
+ __func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM,
+ ASM_CONFIGURE_DUAL_MONO);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout opcode[0x%x]\n", __func__,
+ dual_mono.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_set_encdec_chan_map(struct audio_client *ac,
+ uint32_t num_channels)
+{
+ struct asm_stream_cmd_encdec_channelmap chan_map;
+ u8 *channel_mapping;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d, num_channels = %d\n",
+ __func__, ac->session, num_channels);
+
+ q6asm_add_hdr(ac, &chan_map.hdr, sizeof(chan_map), TRUE);
+
+ chan_map.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ chan_map.param_id = ASM_ENCDEC_DEC_CHAN_MAP;
+ chan_map.param_size = sizeof(struct asm_dec_chan_map);
+ chan_map.chan_map.num_channels = num_channels;
+
+ channel_mapping =
+ chan_map.chan_map.channel_mapping;
+
+ memset(channel_mapping, PCM_CHANNEL_NULL, MAX_CHAN_MAP_CHANNELS);
+ if (num_channels == 1) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ } else if (num_channels == 2) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (num_channels == 6) {
+ channel_mapping[0] = PCM_CHANNEL_FC;
+ channel_mapping[1] = PCM_CHANNEL_FL;
+ channel_mapping[2] = PCM_CHANNEL_FR;
+ channel_mapping[3] = PCM_CHANNEL_LB;
+ channel_mapping[4] = PCM_CHANNEL_RB;
+ channel_mapping[5] = PCM_CHANNEL_LFE;
+ } else {
+ pr_err("%s: ERROR.unsupported num_ch = %u\n", __func__,
+ num_channels);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &chan_map);
+ if (rc < 0) {
+ pr_err("%s:Command opcode[0x%x]paramid[0x%x] failed\n",
+ __func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM,
+ ASM_ENCDEC_DEC_CHAN_MAP);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout opcode[0x%x]\n", __func__,
+ chan_map.hdr.opcode);
+ rc = -ETIMEDOUT;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t reduced_rate_level, uint16_t rate_modulation_cmd)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]min_rate[0x%4x]max_rate[0x%4x] \
+ reduced_rate_level[0x%4x]rate_modulation_cmd[0x%4x]", __func__,
+ ac->session, frames_per_buf, min_rate, max_rate,
+ reduced_rate_level, rate_modulation_cmd);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+
+ enc_cfg.enc_blk.frames_per_buf = frames_per_buf;
+ enc_cfg.enc_blk.format_id = V13K_FS;
+ enc_cfg.enc_blk.cfg_size = sizeof(struct asm_qcelp13_read_cfg);
+ enc_cfg.enc_blk.cfg.qcelp13.min_rate = min_rate;
+ enc_cfg.enc_blk.cfg.qcelp13.max_rate = max_rate;
+ enc_cfg.enc_blk.cfg.qcelp13.reduced_rate_level = reduced_rate_level;
+ enc_cfg.enc_blk.cfg.qcelp13.rate_modulation_cmd = rate_modulation_cmd;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t rate_modulation_cmd)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]min_rate[0x%4x]max_rate[0x%4x] \
+ rate_modulation_cmd[0x%4x]", __func__, ac->session,
+ frames_per_buf, min_rate, max_rate, rate_modulation_cmd);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+
+ enc_cfg.enc_blk.frames_per_buf = frames_per_buf;
+ enc_cfg.enc_blk.format_id = EVRC_FS;
+ enc_cfg.enc_blk.cfg_size = sizeof(struct asm_evrc_read_cfg);
+ enc_cfg.enc_blk.cfg.evrc.min_rate = min_rate;
+ enc_cfg.enc_blk.cfg.evrc.max_rate = max_rate;
+ enc_cfg.enc_blk.cfg.evrc.rate_modulation_cmd = rate_modulation_cmd;
+ enc_cfg.enc_blk.cfg.evrc.reserved = 0;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]band_mode[0x%4x]dtx_enable[0x%4x]",
+ __func__, ac->session, frames_per_buf, band_mode, dtx_enable);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+
+ enc_cfg.enc_blk.frames_per_buf = frames_per_buf;
+ enc_cfg.enc_blk.format_id = AMRNB_FS;
+ enc_cfg.enc_blk.cfg_size = sizeof(struct asm_amrnb_read_cfg);
+ enc_cfg.enc_blk.cfg.amrnb.mode = band_mode;
+ enc_cfg.enc_blk.cfg.amrnb.dtx_mode = dtx_enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable)
+{
+ struct asm_stream_cmd_encdec_cfg_blk enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]band_mode[0x%4x]dtx_enable[0x%4x]",
+ __func__, ac->session, frames_per_buf, band_mode, dtx_enable);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+
+ enc_cfg.param_id = ASM_ENCDEC_CFG_BLK_ID;
+ enc_cfg.param_size = sizeof(struct asm_encode_cfg_blk);
+
+ enc_cfg.enc_blk.frames_per_buf = frames_per_buf;
+ enc_cfg.enc_blk.format_id = AMRWB_FS;
+ enc_cfg.enc_blk.cfg_size = sizeof(struct asm_amrwb_read_cfg);
+ enc_cfg.enc_blk.cfg.amrwb.mode = band_mode;
+ enc_cfg.enc_blk.cfg.amrwb.dtx_mode = dtx_enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_stream_media_format_update fmt;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session, rate,
+ channels);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = LINEAR_PCM;
+ fmt.cfg_size = sizeof(struct asm_pcm_cfg);
+ fmt.write_cfg.pcm_cfg.ch_cfg = channels;
+ fmt.write_cfg.pcm_cfg.bits_per_sample = 16;
+ fmt.write_cfg.pcm_cfg.sample_rate = rate;
+ fmt.write_cfg.pcm_cfg.is_signed = 1;
+ fmt.write_cfg.pcm_cfg.interleaved = 1;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_stream_media_format_update fmt;
+ u8 *channel_mapping;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session, rate,
+ channels);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = MULTI_CHANNEL_PCM;
+ fmt.cfg_size = sizeof(struct asm_multi_channel_pcm_fmt_blk);
+ fmt.write_cfg.multi_ch_pcm_cfg.num_channels = channels;
+ fmt.write_cfg.multi_ch_pcm_cfg.bits_per_sample = 16;
+ fmt.write_cfg.multi_ch_pcm_cfg.sample_rate = rate;
+ fmt.write_cfg.multi_ch_pcm_cfg.is_signed = 1;
+ fmt.write_cfg.multi_ch_pcm_cfg.is_interleaved = 1;
+ channel_mapping =
+ fmt.write_cfg.multi_ch_pcm_cfg.channel_mapping;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ if (channels == 1) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ } else if (channels == 2) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channels == 6) {
+ channel_mapping[0] = PCM_CHANNEL_FC;
+ channel_mapping[1] = PCM_CHANNEL_FL;
+ channel_mapping[2] = PCM_CHANNEL_FR;
+ channel_mapping[3] = PCM_CHANNEL_LB;
+ channel_mapping[4] = PCM_CHANNEL_RB;
+ channel_mapping[5] = PCM_CHANNEL_LFE;
+ } else {
+ pr_err("%s: ERROR.unsupported num_ch = %u\n", __func__,
+ channels);
+ return -EINVAL;
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg)
+{
+ struct asm_stream_media_format_update fmt;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session,
+ cfg->sample_rate, cfg->ch_cfg);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = MPEG4_AAC;
+ fmt.cfg_size = sizeof(struct asm_aac_cfg);
+ fmt.write_cfg.aac_cfg.format = cfg->format;
+ fmt.write_cfg.aac_cfg.aot = cfg->aot;
+ fmt.write_cfg.aac_cfg.ep_config = cfg->ep_config;
+ fmt.write_cfg.aac_cfg.section_data_resilience =
+ cfg->section_data_resilience;
+ fmt.write_cfg.aac_cfg.scalefactor_data_resilience =
+ cfg->scalefactor_data_resilience;
+ fmt.write_cfg.aac_cfg.spectral_data_resilience =
+ cfg->spectral_data_resilience;
+ fmt.write_cfg.aac_cfg.ch_cfg = cfg->ch_cfg;
+ fmt.write_cfg.aac_cfg.sample_rate = cfg->sample_rate;
+ pr_info("%s:format=%x cfg_size=%d aac-cfg=%x aot=%d ch=%d sr=%d\n",
+ __func__, fmt.format, fmt.cfg_size,
+ fmt.write_cfg.aac_cfg.format,
+ fmt.write_cfg.aac_cfg.aot,
+ fmt.write_cfg.aac_cfg.ch_cfg,
+ fmt.write_cfg.aac_cfg.sample_rate);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+
+int q6asm_media_format_block_multi_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg)
+{
+ struct asm_stream_media_format_update fmt;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session,
+ cfg->sample_rate, cfg->ch_cfg);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = MPEG4_MULTI_AAC;
+ fmt.cfg_size = sizeof(struct asm_aac_cfg);
+ fmt.write_cfg.aac_cfg.format = cfg->format;
+ fmt.write_cfg.aac_cfg.aot = cfg->aot;
+ fmt.write_cfg.aac_cfg.ep_config = cfg->ep_config;
+ fmt.write_cfg.aac_cfg.section_data_resilience =
+ cfg->section_data_resilience;
+ fmt.write_cfg.aac_cfg.scalefactor_data_resilience =
+ cfg->scalefactor_data_resilience;
+ fmt.write_cfg.aac_cfg.spectral_data_resilience =
+ cfg->spectral_data_resilience;
+ fmt.write_cfg.aac_cfg.ch_cfg = cfg->ch_cfg;
+ fmt.write_cfg.aac_cfg.sample_rate = cfg->sample_rate;
+ pr_info("%s:format=%x cfg_size=%d aac-cfg=%x aot=%d ch=%d sr=%d\n",
+ __func__, fmt.format, fmt.cfg_size,
+ fmt.write_cfg.aac_cfg.format,
+ fmt.write_cfg.aac_cfg.aot,
+ fmt.write_cfg.aac_cfg.ch_cfg,
+ fmt.write_cfg.aac_cfg.sample_rate);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+
+
+int q6asm_media_format_block(struct audio_client *ac, uint32_t format)
+{
+
+ struct asm_stream_media_format_update fmt;
+ int rc = 0;
+
+ pr_debug("%s:session[%d] format[0x%x]\n", __func__,
+ ac->session, format);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+ switch (format) {
+ case FORMAT_V13K:
+ fmt.format = V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ fmt.format = EVRC_FS;
+ break;
+ case FORMAT_AMRWB:
+ fmt.format = AMRWB_FS;
+ break;
+ case FORMAT_AMRNB:
+ fmt.format = AMRNB_FS;
+ break;
+ case FORMAT_MP3:
+ fmt.format = MP3;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", format);
+ goto fail_cmd;
+ }
+ fmt.cfg_size = 0;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_wma(struct audio_client *ac,
+ void *cfg)
+{
+ struct asm_stream_media_format_update fmt;
+ struct asm_wma_cfg *wma_cfg = (struct asm_wma_cfg *)cfg;
+ int rc = 0;
+
+ pr_debug("session[%d]format_tag[0x%4x] rate[%d] ch[0x%4x] bps[%d],\
+ balign[0x%4x], bit_sample[0x%4x], ch_msk[%d], enc_opt[0x%4x]\n",
+ ac->session, wma_cfg->format_tag, wma_cfg->sample_rate,
+ wma_cfg->ch_cfg, wma_cfg->avg_bytes_per_sec,
+ wma_cfg->block_align, wma_cfg->valid_bits_per_sample,
+ wma_cfg->ch_mask, wma_cfg->encode_opt);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = WMA_V9;
+ fmt.cfg_size = sizeof(struct asm_wma_cfg);
+ fmt.write_cfg.wma_cfg.format_tag = wma_cfg->format_tag;
+ fmt.write_cfg.wma_cfg.ch_cfg = wma_cfg->ch_cfg;
+ fmt.write_cfg.wma_cfg.sample_rate = wma_cfg->sample_rate;
+ fmt.write_cfg.wma_cfg.avg_bytes_per_sec = wma_cfg->avg_bytes_per_sec;
+ fmt.write_cfg.wma_cfg.block_align = wma_cfg->block_align;
+ fmt.write_cfg.wma_cfg.valid_bits_per_sample =
+ wma_cfg->valid_bits_per_sample;
+ fmt.write_cfg.wma_cfg.ch_mask = wma_cfg->ch_mask;
+ fmt.write_cfg.wma_cfg.encode_opt = wma_cfg->encode_opt;
+ fmt.write_cfg.wma_cfg.adv_encode_opt = 0;
+ fmt.write_cfg.wma_cfg.adv_encode_opt2 = 0;
+ fmt.write_cfg.wma_cfg.drc_peak_ref = 0;
+ fmt.write_cfg.wma_cfg.drc_peak_target = 0;
+ fmt.write_cfg.wma_cfg.drc_ave_ref = 0;
+ fmt.write_cfg.wma_cfg.drc_ave_target = 0;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_wmapro(struct audio_client *ac,
+ void *cfg)
+{
+ struct asm_stream_media_format_update fmt;
+ struct asm_wmapro_cfg *wmapro_cfg = (struct asm_wmapro_cfg *)cfg;
+ int rc = 0;
+
+ pr_debug("session[%d]format_tag[0x%4x] rate[%d] ch[0x%4x] bps[%d],"
+ "balign[0x%4x], bit_sample[0x%4x], ch_msk[%d], enc_opt[0x%4x],\
+ adv_enc_opt[0x%4x], adv_enc_opt2[0x%8x]\n",
+ ac->session, wmapro_cfg->format_tag, wmapro_cfg->sample_rate,
+ wmapro_cfg->ch_cfg, wmapro_cfg->avg_bytes_per_sec,
+ wmapro_cfg->block_align, wmapro_cfg->valid_bits_per_sample,
+ wmapro_cfg->ch_mask, wmapro_cfg->encode_opt,
+ wmapro_cfg->adv_encode_opt, wmapro_cfg->adv_encode_opt2);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
+
+ fmt.format = WMA_V10PRO;
+ fmt.cfg_size = sizeof(struct asm_wmapro_cfg);
+ fmt.write_cfg.wmapro_cfg.format_tag = wmapro_cfg->format_tag;
+ fmt.write_cfg.wmapro_cfg.ch_cfg = wmapro_cfg->ch_cfg;
+ fmt.write_cfg.wmapro_cfg.sample_rate = wmapro_cfg->sample_rate;
+ fmt.write_cfg.wmapro_cfg.avg_bytes_per_sec =
+ wmapro_cfg->avg_bytes_per_sec;
+ fmt.write_cfg.wmapro_cfg.block_align = wmapro_cfg->block_align;
+ fmt.write_cfg.wmapro_cfg.valid_bits_per_sample =
+ wmapro_cfg->valid_bits_per_sample;
+ fmt.write_cfg.wmapro_cfg.ch_mask = wmapro_cfg->ch_mask;
+ fmt.write_cfg.wmapro_cfg.encode_opt = wmapro_cfg->encode_opt;
+ fmt.write_cfg.wmapro_cfg.adv_encode_opt = wmapro_cfg->adv_encode_opt;
+ fmt.write_cfg.wmapro_cfg.adv_encode_opt2 = wmapro_cfg->adv_encode_opt2;
+ fmt.write_cfg.wmapro_cfg.drc_peak_ref = 0;
+ fmt.write_cfg.wmapro_cfg.drc_peak_target = 0;
+ fmt.write_cfg.wmapro_cfg.drc_ave_ref = 0;
+ fmt.write_cfg.wmapro_cfg.drc_ave_target = 0;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_memory_map(struct audio_client *ac, uint32_t buf_add, int dir,
+ uint32_t bufsz, uint32_t bufcnt)
+{
+ struct asm_stream_cmd_memory_map mem_map;
+ int rc = 0;
+
+ if (!ac || ac->apr == NULL || this_mmap.apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ mem_map.hdr.opcode = ASM_SESSION_CMD_MEMORY_MAP;
+
+ mem_map.buf_add = buf_add;
+ mem_map.buf_size = bufsz * bufcnt;
+ mem_map.mempool_id = 0; /* EBI */
+ mem_map.reserved = 0;
+
+ q6asm_add_mmaphdr(&mem_map.hdr,
+ sizeof(struct asm_stream_cmd_memory_map), TRUE);
+
+ pr_debug("buf add[%x] buf_add_parameter[%x]\n",
+ mem_map.buf_add, buf_add);
+
+ rc = apr_send_pkt(this_mmap.apr, (uint32_t *) &mem_map);
+ if (rc < 0) {
+ pr_err("mem_map op[0x%x]rc[%d]\n",
+ mem_map.hdr.opcode, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(this_mmap.cmd_wait,
+ (atomic_read(&this_mmap.cmd_state) == 0), 5 * HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_map\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_memory_unmap(struct audio_client *ac, uint32_t buf_add, int dir)
+{
+ struct asm_stream_cmd_memory_unmap mem_unmap;
+ int rc = 0;
+
+ if (!ac || ac->apr == NULL || this_mmap.apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ q6asm_add_mmaphdr(&mem_unmap.hdr,
+ sizeof(struct asm_stream_cmd_memory_unmap), TRUE);
+ mem_unmap.hdr.opcode = ASM_SESSION_CMD_MEMORY_UNMAP;
+ mem_unmap.buf_add = buf_add;
+
+ rc = apr_send_pkt(this_mmap.apr, (uint32_t *) &mem_unmap);
+ if (rc < 0) {
+ pr_err("mem_unmap op[0x%x]rc[%d]\n",
+ mem_unmap.hdr.opcode, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(this_mmap.cmd_wait,
+ (atomic_read(&this_mmap.cmd_state) == 0), 5 * HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_map\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain)
+{
+ void *vol_cmd = NULL;
+ void *payload = NULL;
+ struct asm_pp_params_command *cmd = NULL;
+ struct asm_lrchannel_gain_params *lrgain = NULL;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_pp_params_command) +
+ + sizeof(struct asm_lrchannel_gain_params);
+ vol_cmd = kzalloc(sz, GFP_KERNEL);
+ if (vol_cmd == NULL) {
+ pr_err("%s[%d]: Mem alloc failed\n", __func__, ac->session);
+ rc = -EINVAL;
+ return rc;
+ }
+ cmd = (struct asm_pp_params_command *)vol_cmd;
+ q6asm_add_hdr_async(ac, &cmd->hdr, sz, TRUE);
+ cmd->hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS;
+ cmd->payload = NULL;
+ cmd->payload_size = sizeof(struct asm_pp_param_data_hdr) +
+ sizeof(struct asm_lrchannel_gain_params);
+ cmd->params.module_id = VOLUME_CONTROL_MODULE_ID;
+ cmd->params.param_id = L_R_CHANNEL_GAIN_PARAM_ID;
+ cmd->params.param_size = sizeof(struct asm_lrchannel_gain_params);
+ cmd->params.reserved = 0;
+
+ payload = (u8 *)(vol_cmd + sizeof(struct asm_pp_params_command));
+ lrgain = (struct asm_lrchannel_gain_params *)payload;
+
+ lrgain->left_gain = left_gain;
+ lrgain->right_gain = right_gain;
+ rc = apr_send_pkt(ac->apr, (uint32_t *) vol_cmd);
+ if (rc < 0) {
+ pr_err("%s: Volume Command failed\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in sending volume command to apr\n",
+ __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(vol_cmd);
+ return rc;
+}
+
+static int q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt)
+{
+ struct asm_stream_cmd_memory_map_regions *mmap_regions = NULL;
+ struct asm_memory_map_regions *mregions = NULL;
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ void *mmap_region_cmd = NULL;
+ void *payload = NULL;
+ int rc = 0;
+ int i = 0;
+ int cmd_size = 0;
+
+ if (!ac || ac->apr == NULL || this_mmap.apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ cmd_size = sizeof(struct asm_stream_cmd_memory_map_regions)
+ + sizeof(struct asm_memory_map_regions) * bufcnt;
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (mmap_region_cmd == NULL) {
+ pr_err("%s: Mem alloc failed\n", __func__);
+ rc = -EINVAL;
+ return rc;
+ }
+ mmap_regions = (struct asm_stream_cmd_memory_map_regions *)
+ mmap_region_cmd;
+ q6asm_add_mmaphdr(&mmap_regions->hdr, cmd_size, TRUE);
+ mmap_regions->hdr.opcode = ASM_SESSION_CMD_MEMORY_MAP_REGIONS;
+ mmap_regions->mempool_id = 0;
+ mmap_regions->nregions = bufcnt & 0x00ff;
+ pr_debug("map_regions->nregions = %d\n", mmap_regions->nregions);
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct asm_stream_cmd_memory_map_regions));
+ mregions = (struct asm_memory_map_regions *)payload;
+
+ port = &ac->port[dir];
+ for (i = 0; i < bufcnt; i++) {
+ ab = &port->buf[i];
+ mregions->phys = ab->phys;
+ mregions->buf_size = ab->size;
+ ++mregions;
+ }
+
+ rc = apr_send_pkt(this_mmap.apr, (uint32_t *) mmap_region_cmd);
+ if (rc < 0) {
+ pr_err("mmap_regions op[0x%x]rc[%d]\n",
+ mmap_regions->hdr.opcode, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(this_mmap.cmd_wait,
+ (atomic_read(&this_mmap.cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_map\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(mmap_region_cmd);
+ return rc;
+}
+
+static int q6asm_memory_unmap_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt)
+{
+ struct asm_stream_cmd_memory_unmap_regions *unmap_regions = NULL;
+ struct asm_memory_unmap_regions *mregions = NULL;
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ void *unmap_region_cmd = NULL;
+ void *payload = NULL;
+ int rc = 0;
+ int i = 0;
+ int cmd_size = 0;
+
+ if (!ac || ac->apr == NULL || this_mmap.apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ cmd_size = sizeof(struct asm_stream_cmd_memory_unmap_regions) +
+ sizeof(struct asm_memory_unmap_regions) * bufcnt;
+
+ unmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (unmap_region_cmd == NULL) {
+ pr_err("%s: Mem alloc failed\n", __func__);
+ rc = -EINVAL;
+ return rc;
+ }
+ unmap_regions = (struct asm_stream_cmd_memory_unmap_regions *)
+ unmap_region_cmd;
+ q6asm_add_mmaphdr(&unmap_regions->hdr, cmd_size, TRUE);
+ unmap_regions->hdr.opcode = ASM_SESSION_CMD_MEMORY_UNMAP_REGIONS;
+ unmap_regions->nregions = bufcnt & 0x00ff;
+ pr_debug("unmap_regions->nregions = %d\n", unmap_regions->nregions);
+ payload = ((u8 *) unmap_region_cmd +
+ sizeof(struct asm_stream_cmd_memory_unmap_regions));
+ mregions = (struct asm_memory_unmap_regions *)payload;
+ port = &ac->port[dir];
+ for (i = 0; i < bufcnt; i++) {
+ ab = &port->buf[i];
+ mregions->phys = ab->phys;
+ ++mregions;
+ }
+
+ rc = apr_send_pkt(this_mmap.apr, (uint32_t *) unmap_region_cmd);
+ if (rc < 0) {
+ pr_err("mmap_regions op[0x%x]rc[%d]\n",
+ unmap_regions->hdr.opcode, rc);
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(this_mmap.cmd_wait,
+ (atomic_read(&this_mmap.cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_unmap\n");
+ goto fail_cmd;
+ }
+ rc = 0;
+
+fail_cmd:
+ kfree(unmap_region_cmd);
+ return rc;
+}
+
+int q6asm_set_mute(struct audio_client *ac, int muteflag)
+{
+ void *vol_cmd = NULL;
+ void *payload = NULL;
+ struct asm_pp_params_command *cmd = NULL;
+ struct asm_mute_params *mute = NULL;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_pp_params_command) +
+ + sizeof(struct asm_mute_params);
+ vol_cmd = kzalloc(sz, GFP_KERNEL);
+ if (vol_cmd == NULL) {
+ pr_err("%s[%d]: Mem alloc failed\n", __func__, ac->session);
+ rc = -EINVAL;
+ return rc;
+ }
+ cmd = (struct asm_pp_params_command *)vol_cmd;
+ q6asm_add_hdr_async(ac, &cmd->hdr, sz, TRUE);
+ cmd->hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS;
+ cmd->payload = NULL;
+ cmd->payload_size = sizeof(struct asm_pp_param_data_hdr) +
+ sizeof(struct asm_mute_params);
+ cmd->params.module_id = VOLUME_CONTROL_MODULE_ID;
+ cmd->params.param_id = MUTE_CONFIG_PARAM_ID;
+ cmd->params.param_size = sizeof(struct asm_mute_params);
+ cmd->params.reserved = 0;
+
+ payload = (u8 *)(vol_cmd + sizeof(struct asm_pp_params_command));
+ mute = (struct asm_mute_params *)payload;
+
+ mute->muteflag = muteflag;
+ rc = apr_send_pkt(ac->apr, (uint32_t *) vol_cmd);
+ if (rc < 0) {
+ pr_err("%s: Mute Command failed\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in sending mute command to apr\n",
+ __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(vol_cmd);
+ return rc;
+}
+
+int q6asm_set_volume(struct audio_client *ac, int volume)
+{
+ void *vol_cmd = NULL;
+ void *payload = NULL;
+ struct asm_pp_params_command *cmd = NULL;
+ struct asm_master_gain_params *mgain = NULL;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_pp_params_command) +
+ + sizeof(struct asm_master_gain_params);
+ vol_cmd = kzalloc(sz, GFP_KERNEL);
+ if (vol_cmd == NULL) {
+ pr_err("%s[%d]: Mem alloc failed\n", __func__, ac->session);
+ rc = -EINVAL;
+ return rc;
+ }
+ cmd = (struct asm_pp_params_command *)vol_cmd;
+ q6asm_add_hdr_async(ac, &cmd->hdr, sz, TRUE);
+ cmd->hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS;
+ cmd->payload = NULL;
+ cmd->payload_size = sizeof(struct asm_pp_param_data_hdr) +
+ sizeof(struct asm_master_gain_params);
+ cmd->params.module_id = VOLUME_CONTROL_MODULE_ID;
+ cmd->params.param_id = MASTER_GAIN_PARAM_ID;
+ cmd->params.param_size = sizeof(struct asm_master_gain_params);
+ cmd->params.reserved = 0;
+
+ payload = (u8 *)(vol_cmd + sizeof(struct asm_pp_params_command));
+ mgain = (struct asm_master_gain_params *)payload;
+
+ mgain->master_gain = volume;
+ mgain->padding = 0x00;
+ rc = apr_send_pkt(ac->apr, (uint32_t *) vol_cmd);
+ if (rc < 0) {
+ pr_err("%s: Volume Command failed\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in sending volume command to apr\n",
+ __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(vol_cmd);
+ return rc;
+}
+
+int q6asm_set_softpause(struct audio_client *ac,
+ struct asm_softpause_params *pause_param)
+{
+ void *vol_cmd = NULL;
+ void *payload = NULL;
+ struct asm_pp_params_command *cmd = NULL;
+ struct asm_softpause_params *params = NULL;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_pp_params_command) +
+ + sizeof(struct asm_softpause_params);
+ vol_cmd = kzalloc(sz, GFP_KERNEL);
+ if (vol_cmd == NULL) {
+ pr_err("%s[%d]: Mem alloc failed\n", __func__, ac->session);
+ rc = -EINVAL;
+ return rc;
+ }
+ cmd = (struct asm_pp_params_command *)vol_cmd;
+ q6asm_add_hdr_async(ac, &cmd->hdr, sz, TRUE);
+ cmd->hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS;
+ cmd->payload = NULL;
+ cmd->payload_size = sizeof(struct asm_pp_param_data_hdr) +
+ sizeof(struct asm_softpause_params);
+ cmd->params.module_id = VOLUME_CONTROL_MODULE_ID;
+ cmd->params.param_id = SOFT_PAUSE_PARAM_ID;
+ cmd->params.param_size = sizeof(struct asm_softpause_params);
+ cmd->params.reserved = 0;
+
+ payload = (u8 *)(vol_cmd + sizeof(struct asm_pp_params_command));
+ params = (struct asm_softpause_params *)payload;
+
+ params->enable = pause_param->enable;
+ params->period = pause_param->period;
+ params->step = pause_param->step;
+ params->rampingcurve = pause_param->rampingcurve;
+ pr_debug("%s: soft Pause Command: enable = %d, period = %d,"
+ "step = %d, curve = %d\n", __func__, params->enable,
+ params->period, params->step, params->rampingcurve);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) vol_cmd);
+ if (rc < 0) {
+ pr_err("%s: Volume Command(soft_pause) failed\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in sending volume command(soft_pause)"
+ "to apr\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(vol_cmd);
+ return rc;
+}
+
+int q6asm_set_softvolume(struct audio_client *ac,
+ struct asm_softvolume_params *softvol_param)
+{
+ void *vol_cmd = NULL;
+ void *payload = NULL;
+ struct asm_pp_params_command *cmd = NULL;
+ struct asm_softvolume_params *params = NULL;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_pp_params_command) +
+ + sizeof(struct asm_softvolume_params);
+ vol_cmd = kzalloc(sz, GFP_KERNEL);
+ if (vol_cmd == NULL) {
+ pr_err("%s[%d]: Mem alloc failed\n", __func__, ac->session);
+ rc = -EINVAL;
+ return rc;
+ }
+ cmd = (struct asm_pp_params_command *)vol_cmd;
+ q6asm_add_hdr_async(ac, &cmd->hdr, sz, TRUE);
+ cmd->hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS;
+ cmd->payload = NULL;
+ cmd->payload_size = sizeof(struct asm_pp_param_data_hdr) +
+ sizeof(struct asm_softvolume_params);
+ cmd->params.module_id = VOLUME_CONTROL_MODULE_ID;
+ cmd->params.param_id = SOFT_VOLUME_PARAM_ID;
+ cmd->params.param_size = sizeof(struct asm_softvolume_params);
+ cmd->params.reserved = 0;
+
+ payload = (u8 *)(vol_cmd + sizeof(struct asm_pp_params_command));
+ params = (struct asm_softvolume_params *)payload;
+
+ params->period = softvol_param->period;
+ params->step = softvol_param->step;
+ params->rampingcurve = softvol_param->rampingcurve;
+ pr_debug("%s: soft Volume:opcode = %d,payload_sz =%d,module_id =%d,"
+ "param_id = %d, param_sz = %d\n", __func__,
+ cmd->hdr.opcode, cmd->payload_size,
+ cmd->params.module_id, cmd->params.param_id,
+ cmd->params.param_size);
+ pr_debug("%s: soft Volume Command: period = %d,"
+ "step = %d, curve = %d\n", __func__, params->period,
+ params->step, params->rampingcurve);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) vol_cmd);
+ if (rc < 0) {
+ pr_err("%s: Volume Command(soft_volume) failed\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in sending volume command(soft_volume)"
+ "to apr\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(vol_cmd);
+ return rc;
+}
+
+int q6asm_equalizer(struct audio_client *ac, void *eq)
+{
+ void *eq_cmd = NULL;
+ void *payload = NULL;
+ struct asm_pp_params_command *cmd = NULL;
+ struct asm_equalizer_params *equalizer = NULL;
+ struct msm_audio_eq_stream_config *eq_params = NULL;
+ int i = 0;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_pp_params_command) +
+ + sizeof(struct asm_equalizer_params);
+ eq_cmd = kzalloc(sz, GFP_KERNEL);
+ if (eq_cmd == NULL) {
+ pr_err("%s[%d]: Mem alloc failed\n", __func__, ac->session);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ eq_params = (struct msm_audio_eq_stream_config *) eq;
+ cmd = (struct asm_pp_params_command *)eq_cmd;
+ q6asm_add_hdr(ac, &cmd->hdr, sz, TRUE);
+ cmd->hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS;
+ cmd->payload = NULL;
+ cmd->payload_size = sizeof(struct asm_pp_param_data_hdr) +
+ sizeof(struct asm_equalizer_params);
+ cmd->params.module_id = EQUALIZER_MODULE_ID;
+ cmd->params.param_id = EQUALIZER_PARAM_ID;
+ cmd->params.param_size = sizeof(struct asm_equalizer_params);
+ cmd->params.reserved = 0;
+ payload = (u8 *)(eq_cmd + sizeof(struct asm_pp_params_command));
+ equalizer = (struct asm_equalizer_params *)payload;
+
+ equalizer->enable = eq_params->enable;
+ equalizer->num_bands = eq_params->num_bands;
+ pr_debug("%s: enable:%d numbands:%d\n", __func__, eq_params->enable,
+ eq_params->num_bands);
+ for (i = 0; i < eq_params->num_bands; i++) {
+ equalizer->eq_bands[i].band_idx =
+ eq_params->eq_bands[i].band_idx;
+ equalizer->eq_bands[i].filter_type =
+ eq_params->eq_bands[i].filter_type;
+ equalizer->eq_bands[i].center_freq_hz =
+ eq_params->eq_bands[i].center_freq_hz;
+ equalizer->eq_bands[i].filter_gain =
+ eq_params->eq_bands[i].filter_gain;
+ equalizer->eq_bands[i].q_factor =
+ eq_params->eq_bands[i].q_factor;
+ pr_debug("%s: filter_type:%u bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].filter_type, i);
+ pr_debug("%s: center_freq_hz:%u bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].center_freq_hz, i);
+ pr_debug("%s: filter_gain:%d bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].filter_gain, i);
+ pr_debug("%s: q_factor:%d bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].q_factor, i);
+ }
+ rc = apr_send_pkt(ac->apr, (uint32_t *) eq_cmd);
+ if (rc < 0) {
+ pr_err("%s: Equalizer Command failed\n", __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in sending equalizer command to apr\n",
+ __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ kfree(eq_cmd);
+ return rc;
+}
+
+int q6asm_read(struct audio_client *ac)
+{
+ struct asm_stream_cmd_read read;
+ struct audio_buffer *ab;
+ int dsp_buf;
+ struct audio_port_data *port;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[OUT];
+
+ q6asm_add_hdr(ac, &read.hdr, sizeof(read), FALSE);
+
+ mutex_lock(&port->lock);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ pr_debug("%s:session[%d]dsp-buf[%d][%p]cpu_buf[%d][%p]\n",
+ __func__,
+ ac->session,
+ dsp_buf,
+ (void *)port->buf[dsp_buf].data,
+ port->cpu_buf,
+ (void *)port->buf[port->cpu_buf].phys);
+
+ read.hdr.opcode = ASM_DATA_CMD_READ;
+ read.buf_add = ab->phys;
+ read.buf_size = ab->size;
+ read.uid = port->dsp_buf;
+ read.hdr.token = port->dsp_buf;
+
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+ mutex_unlock(&port->lock);
+ pr_debug("%s:buf add[0x%x] token[%d] uid[%d]\n", __func__,
+ read.buf_add,
+ read.hdr.token,
+ read.uid);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
+ if (rc < 0) {
+ pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_read_nolock(struct audio_client *ac)
+{
+ struct asm_stream_cmd_read read;
+ struct audio_buffer *ab;
+ int dsp_buf;
+ struct audio_port_data *port;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[OUT];
+
+ q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
+
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ pr_debug("%s:session[%d]dsp-buf[%d][%p]cpu_buf[%d][%p]\n",
+ __func__,
+ ac->session,
+ dsp_buf,
+ (void *)port->buf[dsp_buf].data,
+ port->cpu_buf,
+ (void *)port->buf[port->cpu_buf].phys);
+
+ read.hdr.opcode = ASM_DATA_CMD_READ;
+ read.buf_add = ab->phys;
+ read.buf_size = ab->size;
+ read.uid = port->dsp_buf;
+ read.hdr.token = port->dsp_buf;
+
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+ pr_debug("%s:buf add[0x%x] token[%d] uid[%d]\n", __func__,
+ read.buf_add,
+ read.hdr.token,
+ read.uid);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
+ if (rc < 0) {
+ pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+
+static void q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg)
+{
+ pr_debug("session=%d pkt size=%d cmd_flg=%d\n", pkt_size, cmd_flg,
+ ac->session);
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(sizeof(struct apr_hdr)),\
+ APR_PKT_VER);
+ hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
+ hdr->src_domain = APR_DOMAIN_APPS;
+ hdr->dest_svc = APR_SVC_ASM;
+ hdr->dest_domain = APR_DOMAIN_ADSP;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ if (cmd_flg) {
+ hdr->token = ac->session;
+ atomic_set(&ac->cmd_state, 1);
+ }
+ hdr->pkt_size = pkt_size;
+ return;
+}
+
+int q6asm_async_write(struct audio_client *ac,
+ struct audio_aio_write_param *param)
+{
+ int rc = 0;
+ struct asm_stream_cmd_write write;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ q6asm_add_hdr_async(ac, &write.hdr, sizeof(write), FALSE);
+
+ /* Pass physical address as token for AIO scheme */
+ write.hdr.token = param->uid;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE;
+ write.buf_add = param->paddr;
+ write.avail_bytes = param->len;
+ write.uid = param->uid;
+ write.msw_ts = param->msw_ts;
+ write.lsw_ts = param->lsw_ts;
+ /* Use 0xFF00 for disabling timestamps */
+ if (param->flags == 0xFF00)
+ write.uflags = (0x00000000 | (param->flags & 0x800000FF));
+ else
+ write.uflags = (0x80000000 | param->flags);
+
+ pr_debug("%s: session[%d] bufadd[0x%x]len[0x%x]", __func__, ac->session,
+ write.buf_add, write.avail_bytes);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
+ if (rc < 0) {
+ pr_debug("[%s] write op[0x%x]rc[%d]\n", __func__,
+ write.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_async_read(struct audio_client *ac,
+ struct audio_aio_read_param *param)
+{
+ int rc = 0;
+ struct asm_stream_cmd_read read;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
+
+ /* Pass physical address as token for AIO scheme */
+ read.hdr.token = param->paddr;
+ read.hdr.opcode = ASM_DATA_CMD_READ;
+ read.buf_add = param->paddr;
+ read.buf_size = param->len;
+ read.uid = param->uid;
+
+ pr_debug("%s: session[%d] bufadd[0x%x]len[0x%x]", __func__, ac->session,
+ read.buf_add, read.buf_size);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
+ if (rc < 0) {
+ pr_debug("[%s] read op[0x%x]rc[%d]\n", __func__,
+ read.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags)
+{
+ int rc = 0;
+ struct asm_stream_cmd_write write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int dsp_buf = 0;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] len=%d", __func__, ac->session, len);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[IN];
+
+ q6asm_add_hdr(ac, &write.hdr, sizeof(write),
+ FALSE);
+ mutex_lock(&port->lock);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE;
+ write.buf_add = ab->phys;
+ write.avail_bytes = len;
+ write.uid = port->dsp_buf;
+ write.msw_ts = msw_ts;
+ write.lsw_ts = lsw_ts;
+ /* Use 0xFF00 for disabling timestamps */
+ if (flags == 0xFF00)
+ write.uflags = (0x00000000 | (flags & 0x800000FF));
+ else
+ write.uflags = (0x80000000 | flags);
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+
+ pr_debug("%s:ab->phys[0x%x]bufadd[0x%x]token[0x%x]buf_id[0x%x]"
+ , __func__,
+ ab->phys,
+ write.buf_add,
+ write.hdr.token,
+ write.uid);
+ mutex_unlock(&port->lock);
+#ifdef CONFIG_DEBUG_FS
+ if (out_enable_flag) {
+ char zero_pattern[2] = {0x00, 0x00};
+ /* If First two byte is non zero and last two byte
+ is zero then it is warm output pattern */
+ if ((strncmp(((char *)ab->data), zero_pattern, 2)) &&
+ (!strncmp(((char *)ab->data + 2), zero_pattern, 2))) {
+ do_gettimeofday(&out_warm_tv);
+ pr_debug("WARM:apr_send_pkt at \
+ %ld sec %ld microsec\n", out_warm_tv.tv_sec,\
+ out_warm_tv.tv_usec);
+ pr_debug("Warm Pattern Matched");
+ }
+ /* If First two byte is zero and last two byte is
+ non zero then it is cont ouput pattern */
+ else if ((!strncmp(((char *)ab->data), zero_pattern, 2))
+ && (strncmp(((char *)ab->data + 2), zero_pattern, 2))) {
+ do_gettimeofday(&out_cont_tv);
+ pr_debug("CONT:apr_send_pkt at \
+ %ld sec %ld microsec\n", out_cont_tv.tv_sec,\
+ out_cont_tv.tv_usec);
+ pr_debug("Cont Pattern Matched");
+ }
+ }
+#endif
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
+ if (rc < 0) {
+ pr_err("write op[0x%x]rc[%d]\n", write.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ pr_debug("%s: WRITE SUCCESS\n", __func__);
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags)
+{
+ int rc = 0;
+ struct asm_stream_cmd_write write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int dsp_buf = 0;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] len=%d", __func__, ac->session, len);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[IN];
+
+ q6asm_add_hdr_async(ac, &write.hdr, sizeof(write),
+ FALSE);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE;
+ write.buf_add = ab->phys;
+ write.avail_bytes = len;
+ write.uid = port->dsp_buf;
+ write.msw_ts = msw_ts;
+ write.lsw_ts = lsw_ts;
+ /* Use 0xFF00 for disabling timestamps */
+ if (flags == 0xFF00)
+ write.uflags = (0x00000000 | (flags & 0x800000FF));
+ else
+ write.uflags = (0x80000000 | flags);
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+
+ pr_debug("%s:ab->phys[0x%x]bufadd[0x%x]token[0x%x]buf_id[0x%x]"
+ , __func__,
+ ab->phys,
+ write.buf_add,
+ write.hdr.token,
+ write.uid);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
+ if (rc < 0) {
+ pr_err("write op[0x%x]rc[%d]\n", write.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ pr_debug("%s: WRITE SUCCESS\n", __func__);
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+uint64_t q6asm_get_session_time(struct audio_client *ac)
+{
+ struct apr_hdr hdr;
+ int rc;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), TRUE);
+ hdr.opcode = ASM_SESSION_CMD_GET_SESSION_TIME;
+ atomic_set(&ac->time_flag, 1);
+
+ pr_debug("%s: session[%d]opcode[0x%x]\n", __func__,
+ ac->session,
+ hdr.opcode);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
+ if (rc < 0) {
+ pr_err("Commmand 0x%x failed\n", hdr.opcode);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->time_wait,
+ (atomic_read(&ac->time_flag) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in getting session time from DSP\n",
+ __func__);
+ goto fail_cmd;
+ }
+ return ac->time_stamp;
+
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_cmd(struct audio_client *ac, int cmd)
+{
+ struct apr_hdr hdr;
+ int rc;
+ atomic_t *state;
+ int cnt = 0;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), TRUE);
+ switch (cmd) {
+ case CMD_PAUSE:
+ pr_debug("%s:CMD_PAUSE\n", __func__);
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ state = &ac->cmd_state;
+ break;
+ case CMD_FLUSH:
+ pr_debug("%s:CMD_FLUSH\n", __func__);
+ hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ state = &ac->cmd_state;
+ break;
+ case CMD_OUT_FLUSH:
+ pr_debug("%s:CMD_OUT_FLUSH\n", __func__);
+ hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ state = &ac->cmd_state;
+ break;
+ case CMD_EOS:
+ pr_debug("%s:CMD_EOS\n", __func__);
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ atomic_set(&ac->cmd_state, 0);
+ state = &ac->cmd_state;
+ break;
+ case CMD_CLOSE:
+ pr_debug("%s:CMD_CLOSE\n", __func__);
+ hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ state = &ac->cmd_state;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", cmd);
+ goto fail_cmd;
+ }
+ pr_debug("%s:session[%d]opcode[0x%x] ", __func__,
+ ac->session,
+ hdr.opcode);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
+ if (rc < 0) {
+ pr_err("Commmand 0x%x failed\n", hdr.opcode);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait, (atomic_read(state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for response opcode[0x%x]\n",
+ hdr.opcode);
+ goto fail_cmd;
+ }
+ if (cmd == CMD_FLUSH)
+ q6asm_reset_buf_state(ac);
+ if (cmd == CMD_CLOSE) {
+ /* check if DSP return all buffers */
+ if (ac->port[IN].buf) {
+ for (cnt = 0; cnt < ac->port[IN].max_buf_cnt;
+ cnt++) {
+ if (ac->port[IN].buf[cnt].used == IN) {
+ pr_debug("Write Buf[%d] not returned\n",
+ cnt);
+ }
+ }
+ }
+ if (ac->port[OUT].buf) {
+ for (cnt = 0; cnt < ac->port[OUT].max_buf_cnt; cnt++) {
+ if (ac->port[OUT].buf[cnt].used == OUT) {
+ pr_debug("Read Buf[%d] not returned\n",
+ cnt);
+ }
+ }
+ }
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
+{
+ struct apr_hdr hdr;
+ int rc;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s:APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ q6asm_add_hdr_async(ac, &hdr, sizeof(hdr), TRUE);
+ switch (cmd) {
+ case CMD_PAUSE:
+ pr_debug("%s:CMD_PAUSE\n", __func__);
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_EOS:
+ pr_debug("%s:CMD_EOS\n", __func__);
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ break;
+ default:
+ pr_err("%s:Invalid format[%d]\n", __func__, cmd);
+ goto fail_cmd;
+ }
+ pr_debug("%s:session[%d]opcode[0x%x] ", __func__,
+ ac->session,
+ hdr.opcode);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
+ if (rc < 0) {
+ pr_err("%s:Commmand 0x%x failed\n", __func__, hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+ int cnt = 0;
+ int loopcnt = 0;
+ struct audio_port_data *port = NULL;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ mutex_lock(&ac->cmd_lock);
+ for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
+ port = &ac->port[loopcnt];
+ cnt = port->max_buf_cnt - 1;
+ port->dsp_buf = 0;
+ port->cpu_buf = 0;
+ while (cnt >= 0) {
+ if (!port->buf)
+ continue;
+ port->buf[cnt].used = 1;
+ cnt--;
+ }
+ }
+ mutex_unlock(&ac->cmd_lock);
+ }
+}
+
+int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable)
+{
+ struct asm_stream_cmd_reg_tx_overflow_event tx_overflow;
+ int rc;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s:session[%d]enable[%d]\n", __func__,
+ ac->session, enable);
+ q6asm_add_hdr(ac, &tx_overflow.hdr, sizeof(tx_overflow), TRUE);
+
+ tx_overflow.hdr.opcode = \
+ ASM_SESSION_CMD_REGISTER_FOR_TX_OVERFLOW_EVENTS;
+ /* tx overflow event: enable */
+ tx_overflow.enable = enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &tx_overflow);
+ if (rc < 0) {
+ pr_err("tx overflow op[0x%x]rc[%d]\n", \
+ tx_overflow.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for tx overflow\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_get_apr_service_id(int session_id)
+{
+ pr_debug("%s\n", __func__);
+
+ if (session_id < 0 || session_id > SESSION_MAX) {
+ pr_err("%s: invalid session_id = %d\n", __func__, session_id);
+ return -EINVAL;
+ }
+
+ return ((struct apr_svc *)session[session_id]->apr)->id;
+}
+
+
+static int __init q6asm_init(void)
+{
+ pr_debug("%s\n", __func__);
+ init_waitqueue_head(&this_mmap.cmd_wait);
+ memset(session, 0, sizeof(session));
+#ifdef CONFIG_DEBUG_FS
+ out_buffer = kmalloc(OUT_BUFFER_SIZE, GFP_KERNEL);
+ out_dentry = debugfs_create_file("audio_out_latency_measurement_node",\
+ S_IFREG | S_IRUGO | S_IWUGO,\
+ NULL, NULL, &audio_output_latency_debug_fops);
+ if (IS_ERR(out_dentry))
+ pr_err("debugfs_create_file failed\n");
+ in_buffer = kmalloc(IN_BUFFER_SIZE, GFP_KERNEL);
+ in_dentry = debugfs_create_file("audio_in_latency_measurement_node",\
+ S_IFREG | S_IRUGO | S_IWUGO,\
+ NULL, NULL, &audio_input_latency_debug_fops);
+ if (IS_ERR(in_dentry))
+ pr_err("debugfs_create_file failed\n");
+#endif
+ return 0;
+}
+
+device_initcall(q6asm_init);
diff --git a/sound/soc/msm/qdsp6/q6voice.c b/sound/soc/msm/qdsp6/q6voice.c
new file mode 100644
index 0000000..0c30dc9
--- /dev/null
+++ b/sound/soc/msm/qdsp6/q6voice.c
@@ -0,0 +1,4013 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#include <linux/slab.h>
+#include <linux/kthread.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/uaccess.h>
+#include <linux/wait.h>
+#include <linux/mutex.h>
+
+#include <asm/mach-types.h>
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <mach/qdsp6v2/rtac.h>
+
+#include "sound/apr_audio.h"
+#include "sound/q6afe.h"
+
+#include "q6voice.h"
+
+#define TIMEOUT_MS 3000
+
+#define CMD_STATUS_SUCCESS 0
+#define CMD_STATUS_FAIL 1
+
+#define VOC_PATH_PASSIVE 0
+#define VOC_PATH_FULL 1
+#define VOC_PATH_VOLTE_PASSIVE 2
+
+/* CVP CAL Size: 245760 = 240 * 1024 */
+#define CVP_CAL_SIZE 245760
+/* CVS CAL Size: 49152 = 48 * 1024 */
+#define CVS_CAL_SIZE 49152
+
+static struct common_data common;
+
+static int voice_send_enable_vocproc_cmd(struct voice_data *v);
+static int voice_send_netid_timing_cmd(struct voice_data *v);
+static int voice_send_attach_vocproc_cmd(struct voice_data *v);
+static int voice_send_set_device_cmd(struct voice_data *v);
+static int voice_send_disable_vocproc_cmd(struct voice_data *v);
+static int voice_send_vol_index_cmd(struct voice_data *v);
+static int voice_send_cvp_map_memory_cmd(struct voice_data *v);
+static int voice_send_cvp_unmap_memory_cmd(struct voice_data *v);
+static int voice_send_cvs_map_memory_cmd(struct voice_data *v);
+static int voice_send_cvs_unmap_memory_cmd(struct voice_data *v);
+static int voice_send_cvs_register_cal_cmd(struct voice_data *v);
+static int voice_send_cvs_deregister_cal_cmd(struct voice_data *v);
+static int voice_send_cvp_register_cal_cmd(struct voice_data *v);
+static int voice_send_cvp_deregister_cal_cmd(struct voice_data *v);
+static int voice_send_cvp_register_vol_cal_table_cmd(struct voice_data *v);
+static int voice_send_cvp_deregister_vol_cal_table_cmd(struct voice_data *v);
+static int voice_send_set_widevoice_enable_cmd(struct voice_data *v);
+static int voice_send_set_pp_enable_cmd(struct voice_data *v,
+ uint32_t module_id, int enable);
+static int voice_cvs_stop_playback(struct voice_data *v);
+static int voice_cvs_start_playback(struct voice_data *v);
+static int voice_cvs_start_record(struct voice_data *v, uint32_t rec_mode);
+static int voice_cvs_stop_record(struct voice_data *v);
+
+static int32_t qdsp_mvm_callback(struct apr_client_data *data, void *priv);
+static int32_t qdsp_cvs_callback(struct apr_client_data *data, void *priv);
+static int32_t qdsp_cvp_callback(struct apr_client_data *data, void *priv);
+
+static u16 voice_get_mvm_handle(struct voice_data *v)
+{
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return 0;
+ }
+
+ pr_debug("%s: mvm_handle %d\n", __func__, v->mvm_handle);
+
+ return v->mvm_handle;
+}
+
+static void voice_set_mvm_handle(struct voice_data *v, u16 mvm_handle)
+{
+ pr_debug("%s: mvm_handle %d\n", __func__, mvm_handle);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return;
+ }
+
+ v->mvm_handle = mvm_handle;
+}
+
+static u16 voice_get_cvs_handle(struct voice_data *v)
+{
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return 0;
+ }
+
+ pr_debug("%s: cvs_handle %d\n", __func__, v->cvs_handle);
+
+ return v->cvs_handle;
+}
+
+static void voice_set_cvs_handle(struct voice_data *v, u16 cvs_handle)
+{
+ pr_debug("%s: cvs_handle %d\n", __func__, cvs_handle);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return;
+ }
+
+ v->cvs_handle = cvs_handle;
+}
+
+static u16 voice_get_cvp_handle(struct voice_data *v)
+{
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return 0;
+ }
+
+ pr_debug("%s: cvp_handle %d\n", __func__, v->cvp_handle);
+
+ return v->cvp_handle;
+}
+
+static void voice_set_cvp_handle(struct voice_data *v, u16 cvp_handle)
+{
+ pr_debug("%s: cvp_handle %d\n", __func__, cvp_handle);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return;
+ }
+
+ v->cvp_handle = cvp_handle;
+}
+
+uint16_t voc_get_session_id(char *name)
+{
+ u16 session_id = 0;
+
+ if (name != NULL) {
+ if (!strncmp(name, "Voice session", 13))
+ session_id = common.voice[VOC_PATH_PASSIVE].session_id;
+ else if (!strncmp(name, "VoLTE session", 13))
+ session_id =
+ common.voice[VOC_PATH_VOLTE_PASSIVE].session_id;
+ else
+ session_id = common.voice[VOC_PATH_FULL].session_id;
+
+ pr_debug("%s: %s has session id 0x%x\n", __func__, name,
+ session_id);
+ }
+
+ return session_id;
+}
+
+static struct voice_data *voice_get_session(u16 session_id)
+{
+ struct voice_data *v = NULL;
+
+ if ((session_id >= SESSION_ID_BASE) &&
+ (session_id < SESSION_ID_BASE + MAX_VOC_SESSIONS)) {
+ v = &common.voice[session_id - SESSION_ID_BASE];
+ }
+
+ pr_debug("%s: session_id 0x%x session handle 0x%x\n",
+ __func__, session_id, (unsigned int)v);
+
+ return v;
+}
+
+static bool is_voice_session(u16 session_id)
+{
+ return (session_id == common.voice[VOC_PATH_PASSIVE].session_id);
+}
+
+static bool is_voip_session(u16 session_id)
+{
+ return (session_id == common.voice[VOC_PATH_FULL].session_id);
+}
+
+static bool is_volte_session(u16 session_id)
+{
+ return (session_id == common.voice[VOC_PATH_VOLTE_PASSIVE].session_id);
+}
+
+static int voice_apr_register(void)
+{
+ pr_debug("%s\n", __func__);
+
+ mutex_lock(&common.common_lock);
+
+ /* register callback to APR */
+ if (common.apr_q6_mvm == NULL) {
+ pr_debug("%s: Start to register MVM callback\n", __func__);
+
+ common.apr_q6_mvm = apr_register("ADSP", "MVM",
+ qdsp_mvm_callback,
+ 0xFFFFFFFF, &common);
+
+ if (common.apr_q6_mvm == NULL) {
+ pr_err("%s: Unable to register MVM\n", __func__);
+ goto err;
+ }
+ }
+
+ if (common.apr_q6_cvs == NULL) {
+ pr_debug("%s: Start to register CVS callback\n", __func__);
+
+ common.apr_q6_cvs = apr_register("ADSP", "CVS",
+ qdsp_cvs_callback,
+ 0xFFFFFFFF, &common);
+
+ if (common.apr_q6_cvs == NULL) {
+ pr_err("%s: Unable to register CVS\n", __func__);
+ goto err;
+ }
+
+ rtac_set_voice_handle(RTAC_CVS, common.apr_q6_cvs);
+ }
+
+ if (common.apr_q6_cvp == NULL) {
+ pr_debug("%s: Start to register CVP callback\n", __func__);
+
+ common.apr_q6_cvp = apr_register("ADSP", "CVP",
+ qdsp_cvp_callback,
+ 0xFFFFFFFF, &common);
+
+ if (common.apr_q6_cvp == NULL) {
+ pr_err("%s: Unable to register CVP\n", __func__);
+ goto err;
+ }
+
+ rtac_set_voice_handle(RTAC_CVP, common.apr_q6_cvp);
+ }
+
+ mutex_unlock(&common.common_lock);
+
+ return 0;
+
+err:
+ if (common.apr_q6_cvs != NULL) {
+ apr_deregister(common.apr_q6_cvs);
+ common.apr_q6_cvs = NULL;
+ rtac_set_voice_handle(RTAC_CVS, NULL);
+ }
+ if (common.apr_q6_mvm != NULL) {
+ apr_deregister(common.apr_q6_mvm);
+ common.apr_q6_mvm = NULL;
+ }
+
+ mutex_unlock(&common.common_lock);
+
+ return -ENODEV;
+}
+
+static int voice_send_dual_control_cmd(struct voice_data *v)
+{
+ int ret = 0;
+ struct mvm_modem_dual_control_session_cmd mvm_voice_ctl_cmd;
+ void *apr_mvm;
+ u16 mvm_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s: VoLTE command to MVM\n", __func__);
+ if (is_volte_session(v->session_id)) {
+ mvm_handle = voice_get_mvm_handle(v);
+ mvm_voice_ctl_cmd.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_voice_ctl_cmd.hdr.pkt_size = APR_PKT_SIZE(
+ APR_HDR_SIZE,
+ sizeof(mvm_voice_ctl_cmd) -
+ APR_HDR_SIZE);
+ pr_debug("%s: send mvm Voice Ctl pkt size = %d\n",
+ __func__, mvm_voice_ctl_cmd.hdr.pkt_size);
+ mvm_voice_ctl_cmd.hdr.src_port = v->session_id;
+ mvm_voice_ctl_cmd.hdr.dest_port = mvm_handle;
+ mvm_voice_ctl_cmd.hdr.token = 0;
+ mvm_voice_ctl_cmd.hdr.opcode =
+ VSS_IMVM_CMD_SET_POLICY_DUAL_CONTROL;
+ mvm_voice_ctl_cmd.voice_ctl.enable_flag = true;
+ v->mvm_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_voice_ctl_cmd);
+ if (ret < 0) {
+ pr_err("%s: Error sending MVM Voice CTL CMD\n",
+ __func__);
+ ret = -EINVAL;
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail;
+ }
+ }
+ ret = 0;
+fail:
+ return ret;
+}
+
+
+static int voice_create_mvm_cvs_session(struct voice_data *v)
+{
+ int ret = 0;
+ struct mvm_create_ctl_session_cmd mvm_session_cmd;
+ struct cvs_create_passive_ctl_session_cmd cvs_session_cmd;
+ struct cvs_create_full_ctl_session_cmd cvs_full_ctl_cmd;
+ struct mvm_attach_stream_cmd attach_stream_cmd;
+ void *apr_mvm, *apr_cvs, *apr_cvp;
+ u16 mvm_handle, cvs_handle, cvp_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+ apr_cvs = common.apr_q6_cvs;
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_mvm || !apr_cvs || !apr_cvp) {
+ pr_err("%s: apr_mvm or apr_cvs or apr_cvp is NULL\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+ cvs_handle = voice_get_cvs_handle(v);
+ cvp_handle = voice_get_cvp_handle(v);
+
+ pr_debug("%s: mvm_hdl=%d, cvs_hdl=%d\n", __func__,
+ mvm_handle, cvs_handle);
+ /* send cmd to create mvm session and wait for response */
+
+ if (!mvm_handle) {
+ if (is_voice_session(v->session_id) ||
+ is_volte_session(v->session_id)) {
+ mvm_session_cmd.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_session_cmd.hdr.pkt_size = APR_PKT_SIZE(
+ APR_HDR_SIZE,
+ sizeof(mvm_session_cmd) -
+ APR_HDR_SIZE);
+ pr_debug("%s: send mvm create session pkt size = %d\n",
+ __func__, mvm_session_cmd.hdr.pkt_size);
+ mvm_session_cmd.hdr.src_port = v->session_id;
+ mvm_session_cmd.hdr.dest_port = 0;
+ mvm_session_cmd.hdr.token = 0;
+ mvm_session_cmd.hdr.opcode =
+ VSS_IMVM_CMD_CREATE_PASSIVE_CONTROL_SESSION;
+ if (is_volte_session(v->session_id)) {
+ strlcpy(mvm_session_cmd.mvm_session.name,
+ "default volte voice",
+ sizeof(mvm_session_cmd.mvm_session.name));
+ } else {
+ strlcpy(mvm_session_cmd.mvm_session.name,
+ "default modem voice",
+ sizeof(mvm_session_cmd.mvm_session.name));
+ }
+
+ v->mvm_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_mvm,
+ (uint32_t *) &mvm_session_cmd);
+ if (ret < 0) {
+ pr_err("%s: Error sending MVM_CONTROL_SESSION\n",
+ __func__);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ } else {
+ pr_debug("%s: creating MVM full ctrl\n", __func__);
+ mvm_session_cmd.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mvm_session_cmd.hdr.pkt_size =
+ APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_session_cmd) -
+ APR_HDR_SIZE);
+ mvm_session_cmd.hdr.src_port = v->session_id;
+ mvm_session_cmd.hdr.dest_port = 0;
+ mvm_session_cmd.hdr.token = 0;
+ mvm_session_cmd.hdr.opcode =
+ VSS_IMVM_CMD_CREATE_FULL_CONTROL_SESSION;
+ strlcpy(mvm_session_cmd.mvm_session.name,
+ "default voip",
+ sizeof(mvm_session_cmd.mvm_session.name));
+
+ v->mvm_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_mvm,
+ (uint32_t *) &mvm_session_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending MVM_CONTROL_SESSION\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ }
+ /* Get the created MVM handle. */
+ mvm_handle = voice_get_mvm_handle(v);
+ }
+ /* send cmd to create cvs session */
+ if (!cvs_handle) {
+ if (is_voice_session(v->session_id) ||
+ is_volte_session(v->session_id)) {
+ pr_debug("%s: creating CVS passive session\n",
+ __func__);
+
+ cvs_session_cmd.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_session_cmd.hdr.pkt_size =
+ APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_session_cmd) -
+ APR_HDR_SIZE);
+ cvs_session_cmd.hdr.src_port = v->session_id;
+ cvs_session_cmd.hdr.dest_port = 0;
+ cvs_session_cmd.hdr.token = 0;
+ cvs_session_cmd.hdr.opcode =
+ VSS_ISTREAM_CMD_CREATE_PASSIVE_CONTROL_SESSION;
+ if (is_volte_session(v->session_id)) {
+ strlcpy(mvm_session_cmd.mvm_session.name,
+ "default volte voice",
+ sizeof(mvm_session_cmd.mvm_session.name));
+ } else {
+ strlcpy(cvs_session_cmd.cvs_session.name,
+ "default modem voice",
+ sizeof(cvs_session_cmd.cvs_session.name));
+ }
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs,
+ (uint32_t *) &cvs_session_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending STREAM_CONTROL_SESSION\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ /* Get the created CVS handle. */
+ cvs_handle = voice_get_cvs_handle(v);
+
+ } else {
+ pr_debug("%s: creating CVS full session\n", __func__);
+
+ cvs_full_ctl_cmd.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+
+ cvs_full_ctl_cmd.hdr.pkt_size =
+ APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_full_ctl_cmd) -
+ APR_HDR_SIZE);
+
+ cvs_full_ctl_cmd.hdr.src_port = v->session_id;
+ cvs_full_ctl_cmd.hdr.dest_port = 0;
+ cvs_full_ctl_cmd.hdr.token = 0;
+ cvs_full_ctl_cmd.hdr.opcode =
+ VSS_ISTREAM_CMD_CREATE_FULL_CONTROL_SESSION;
+ cvs_full_ctl_cmd.cvs_session.direction = 2;
+ cvs_full_ctl_cmd.cvs_session.enc_media_type =
+ common.mvs_info.media_type;
+ cvs_full_ctl_cmd.cvs_session.dec_media_type =
+ common.mvs_info.media_type;
+ cvs_full_ctl_cmd.cvs_session.network_id =
+ common.mvs_info.network_type;
+ strlcpy(cvs_full_ctl_cmd.cvs_session.name,
+ "default q6 voice",
+ sizeof(cvs_full_ctl_cmd.cvs_session.name));
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs,
+ (uint32_t *) &cvs_full_ctl_cmd);
+
+ if (ret < 0) {
+ pr_err("%s: Err %d sending CREATE_FULL_CTRL\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ /* Get the created CVS handle. */
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* Attach MVM to CVS. */
+ pr_debug("%s: Attach MVM to stream\n", __func__);
+
+ attach_stream_cmd.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ attach_stream_cmd.hdr.pkt_size =
+ APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(attach_stream_cmd) -
+ APR_HDR_SIZE);
+ attach_stream_cmd.hdr.src_port = v->session_id;
+ attach_stream_cmd.hdr.dest_port = mvm_handle;
+ attach_stream_cmd.hdr.token = 0;
+ attach_stream_cmd.hdr.opcode =
+ VSS_IMVM_CMD_ATTACH_STREAM;
+ attach_stream_cmd.attach_stream.handle = cvs_handle;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm,
+ (uint32_t *) &attach_stream_cmd);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending ATTACH_STREAM\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ }
+ }
+ return 0;
+
+fail:
+ return -EINVAL;
+}
+
+static int voice_destroy_mvm_cvs_session(struct voice_data *v)
+{
+ int ret = 0;
+ struct mvm_detach_stream_cmd detach_stream;
+ struct apr_hdr mvm_destroy;
+ struct apr_hdr cvs_destroy;
+ void *apr_mvm, *apr_cvs;
+ u16 mvm_handle, cvs_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_mvm || !apr_cvs) {
+ pr_err("%s: apr_mvm or apr_cvs is NULL\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* MVM, CVS sessions are destroyed only for Full control sessions. */
+ if (is_voip_session(v->session_id)) {
+ pr_debug("%s: MVM detach stream\n", __func__);
+
+ /* Detach voice stream. */
+ detach_stream.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ detach_stream.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(detach_stream) - APR_HDR_SIZE);
+ detach_stream.hdr.src_port = v->session_id;
+ detach_stream.hdr.dest_port = mvm_handle;
+ detach_stream.hdr.token = 0;
+ detach_stream.hdr.opcode = VSS_IMVM_CMD_DETACH_STREAM;
+ detach_stream.detach_stream.handle = cvs_handle;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &detach_stream);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending DETACH_STREAM\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait event timeout\n", __func__);
+ goto fail;
+ }
+ /* Destroy CVS. */
+ pr_debug("%s: CVS destroy session\n", __func__);
+
+ cvs_destroy.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_destroy.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_destroy) - APR_HDR_SIZE);
+ cvs_destroy.src_port = v->session_id;
+ cvs_destroy.dest_port = cvs_handle;
+ cvs_destroy.token = 0;
+ cvs_destroy.opcode = APRV2_IBASIC_CMD_DESTROY_SESSION;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_destroy);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending CVS DESTROY\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait event timeout\n", __func__);
+
+ goto fail;
+ }
+ cvs_handle = 0;
+ voice_set_cvs_handle(v, cvs_handle);
+
+ /* Destroy MVM. */
+ pr_debug("MVM destroy session\n");
+
+ mvm_destroy.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_destroy.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_destroy) - APR_HDR_SIZE);
+ mvm_destroy.src_port = v->session_id;
+ mvm_destroy.dest_port = mvm_handle;
+ mvm_destroy.token = 0;
+ mvm_destroy.opcode = APRV2_IBASIC_CMD_DESTROY_SESSION;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_destroy);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending MVM DESTROY\n",
+ __func__, ret);
+
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait event timeout\n", __func__);
+
+ goto fail;
+ }
+ mvm_handle = 0;
+ voice_set_mvm_handle(v, mvm_handle);
+ }
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_tty_mode_cmd(struct voice_data *v)
+{
+ int ret = 0;
+ struct mvm_set_tty_mode_cmd mvm_tty_mode_cmd;
+ void *apr_mvm;
+ u16 mvm_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+
+ if (v->tty_mode) {
+ /* send tty mode cmd to mvm */
+ mvm_tty_mode_cmd.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_tty_mode_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_tty_mode_cmd) -
+ APR_HDR_SIZE);
+ pr_debug("%s: pkt size = %d\n",
+ __func__, mvm_tty_mode_cmd.hdr.pkt_size);
+ mvm_tty_mode_cmd.hdr.src_port = v->session_id;
+ mvm_tty_mode_cmd.hdr.dest_port = mvm_handle;
+ mvm_tty_mode_cmd.hdr.token = 0;
+ mvm_tty_mode_cmd.hdr.opcode = VSS_ISTREAM_CMD_SET_TTY_MODE;
+ mvm_tty_mode_cmd.tty_mode.mode = v->tty_mode;
+ pr_debug("tty mode =%d\n", mvm_tty_mode_cmd.tty_mode.mode);
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_tty_mode_cmd);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_TTY_MODE\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ }
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_set_dtx(struct voice_data *v)
+{
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ struct cvs_set_enc_dtx_mode_cmd cvs_set_dtx;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* Set DTX */
+ cvs_set_dtx.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_set_dtx.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_set_dtx) - APR_HDR_SIZE);
+ cvs_set_dtx.hdr.src_port = v->session_id;
+ cvs_set_dtx.hdr.dest_port = cvs_handle;
+ cvs_set_dtx.hdr.token = 0;
+ cvs_set_dtx.hdr.opcode = VSS_ISTREAM_CMD_SET_ENC_DTX_MODE;
+ cvs_set_dtx.dtx_mode.enable = common.mvs_info.dtx_mode;
+
+ pr_debug("%s: Setting DTX %d\n", __func__, common.mvs_info.dtx_mode);
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_set_dtx);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_DTX\n", __func__, ret);
+ return -EINVAL;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int voice_config_cvs_vocoder(struct voice_data *v)
+{
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ /* Set media type. */
+ struct cvs_set_media_type_cmd cvs_set_media_cmd;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ cvs_set_media_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_set_media_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_set_media_cmd) - APR_HDR_SIZE);
+ cvs_set_media_cmd.hdr.src_port = v->session_id;
+ cvs_set_media_cmd.hdr.dest_port = cvs_handle;
+ cvs_set_media_cmd.hdr.token = 0;
+ cvs_set_media_cmd.hdr.opcode = VSS_ISTREAM_CMD_SET_MEDIA_TYPE;
+ cvs_set_media_cmd.media_type.tx_media_id = common.mvs_info.media_type;
+ cvs_set_media_cmd.media_type.rx_media_id = common.mvs_info.media_type;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_set_media_cmd);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_MEDIA_TYPE\n",
+ __func__, ret);
+
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+ /* Set encoder properties. */
+ switch (common.mvs_info.media_type) {
+ case VSS_MEDIA_ID_EVRC_MODEM: {
+ struct cvs_set_cdma_enc_minmax_rate_cmd cvs_set_cdma_rate;
+
+ pr_debug("Setting EVRC min-max rate\n");
+
+ cvs_set_cdma_rate.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_set_cdma_rate.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_set_cdma_rate) - APR_HDR_SIZE);
+ cvs_set_cdma_rate.hdr.src_port = v->session_id;
+ cvs_set_cdma_rate.hdr.dest_port = cvs_handle;
+ cvs_set_cdma_rate.hdr.token = 0;
+ cvs_set_cdma_rate.hdr.opcode =
+ VSS_ISTREAM_CMD_CDMA_SET_ENC_MINMAX_RATE;
+ cvs_set_cdma_rate.cdma_rate.min_rate = common.mvs_info.rate;
+ cvs_set_cdma_rate.cdma_rate.max_rate = common.mvs_info.rate;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_set_cdma_rate);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_EVRC_MINMAX_RATE\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+ break;
+ }
+ case VSS_MEDIA_ID_AMR_NB_MODEM: {
+ struct cvs_set_amr_enc_rate_cmd cvs_set_amr_rate;
+
+ pr_debug("Setting AMR rate\n");
+
+ cvs_set_amr_rate.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_set_amr_rate.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_set_amr_rate) - APR_HDR_SIZE);
+ cvs_set_amr_rate.hdr.src_port = v->session_id;
+ cvs_set_amr_rate.hdr.dest_port = cvs_handle;
+ cvs_set_amr_rate.hdr.token = 0;
+ cvs_set_amr_rate.hdr.opcode =
+ VSS_ISTREAM_CMD_VOC_AMR_SET_ENC_RATE;
+ cvs_set_amr_rate.amr_rate.mode = common.mvs_info.rate;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_set_amr_rate);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_AMR_RATE\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ ret = voice_set_dtx(v);
+ if (ret < 0)
+ goto fail;
+
+ break;
+ }
+ case VSS_MEDIA_ID_AMR_WB_MODEM: {
+ struct cvs_set_amrwb_enc_rate_cmd cvs_set_amrwb_rate;
+
+ pr_debug("Setting AMR WB rate\n");
+
+ cvs_set_amrwb_rate.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_set_amrwb_rate.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_set_amrwb_rate) -
+ APR_HDR_SIZE);
+ cvs_set_amrwb_rate.hdr.src_port = v->session_id;
+ cvs_set_amrwb_rate.hdr.dest_port = cvs_handle;
+ cvs_set_amrwb_rate.hdr.token = 0;
+ cvs_set_amrwb_rate.hdr.opcode =
+ VSS_ISTREAM_CMD_VOC_AMRWB_SET_ENC_RATE;
+ cvs_set_amrwb_rate.amrwb_rate.mode = common.mvs_info.rate;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_set_amrwb_rate);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_AMRWB_RATE\n",
+ __func__, ret);
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ ret = voice_set_dtx(v);
+ if (ret < 0)
+ goto fail;
+
+ break;
+ }
+ case VSS_MEDIA_ID_G729:
+ case VSS_MEDIA_ID_G711_ALAW:
+ case VSS_MEDIA_ID_G711_MULAW: {
+ ret = voice_set_dtx(v);
+
+ break;
+ }
+ default:
+ /* Do nothing. */
+ break;
+ }
+ return 0;
+
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_start_voice_cmd(struct voice_data *v)
+{
+ struct apr_hdr mvm_start_voice_cmd;
+ int ret = 0;
+ void *apr_mvm;
+ u16 mvm_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+
+ mvm_start_voice_cmd.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_start_voice_cmd.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_start_voice_cmd) - APR_HDR_SIZE);
+ pr_debug("send mvm_start_voice_cmd pkt size = %d\n",
+ mvm_start_voice_cmd.pkt_size);
+ mvm_start_voice_cmd.src_port = v->session_id;
+ mvm_start_voice_cmd.dest_port = mvm_handle;
+ mvm_start_voice_cmd.token = 0;
+ mvm_start_voice_cmd.opcode = VSS_IMVM_CMD_START_VOICE;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_start_voice_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IMVM_CMD_START_VOICE\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_disable_vocproc_cmd(struct voice_data *v)
+{
+ struct apr_hdr cvp_disable_cmd;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr regist failed\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* disable vocproc and wait for respose */
+ cvp_disable_cmd.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_disable_cmd.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_disable_cmd) - APR_HDR_SIZE);
+ pr_debug("cvp_disable_cmd pkt size = %d, cvp_handle=%d\n",
+ cvp_disable_cmd.pkt_size, cvp_handle);
+ cvp_disable_cmd.src_port = v->session_id;
+ cvp_disable_cmd.dest_port = cvp_handle;
+ cvp_disable_cmd.token = 0;
+ cvp_disable_cmd.opcode = VSS_IVOCPROC_CMD_DISABLE;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_disable_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IVOCPROC_CMD_DISABLE\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_set_device_cmd(struct voice_data *v)
+{
+ struct cvp_set_device_cmd cvp_setdev_cmd;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* set device and wait for response */
+ cvp_setdev_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_setdev_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_setdev_cmd) - APR_HDR_SIZE);
+ pr_debug(" send create cvp setdev, pkt size = %d\n",
+ cvp_setdev_cmd.hdr.pkt_size);
+ cvp_setdev_cmd.hdr.src_port = v->session_id;
+ cvp_setdev_cmd.hdr.dest_port = cvp_handle;
+ cvp_setdev_cmd.hdr.token = 0;
+ cvp_setdev_cmd.hdr.opcode = VSS_IVOCPROC_CMD_SET_DEVICE;
+
+ /* Use default topology if invalid value in ACDB */
+ cvp_setdev_cmd.cvp_set_device.tx_topology_id =
+ get_voice_tx_topology();
+ if (cvp_setdev_cmd.cvp_set_device.tx_topology_id == 0)
+ cvp_setdev_cmd.cvp_set_device.tx_topology_id =
+ VSS_IVOCPROC_TOPOLOGY_ID_TX_SM_ECNS;
+
+ cvp_setdev_cmd.cvp_set_device.rx_topology_id =
+ get_voice_rx_topology();
+ if (cvp_setdev_cmd.cvp_set_device.rx_topology_id == 0)
+ cvp_setdev_cmd.cvp_set_device.rx_topology_id =
+ VSS_IVOCPROC_TOPOLOGY_ID_RX_DEFAULT;
+ cvp_setdev_cmd.cvp_set_device.tx_port_id = v->dev_tx.port_id;
+ cvp_setdev_cmd.cvp_set_device.rx_port_id = v->dev_rx.port_id;
+ pr_debug("topology=%d , tx_port_id=%d, rx_port_id=%d\n",
+ cvp_setdev_cmd.cvp_set_device.tx_topology_id,
+ cvp_setdev_cmd.cvp_set_device.tx_port_id,
+ cvp_setdev_cmd.cvp_set_device.rx_port_id);
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_setdev_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VOCPROC_FULL_CONTROL_SESSION\n");
+ goto fail;
+ }
+ pr_debug("wait for cvp create session event\n");
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_stop_voice_cmd(struct voice_data *v)
+{
+ struct apr_hdr mvm_stop_voice_cmd;
+ int ret = 0;
+ void *apr_mvm;
+ u16 mvm_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+
+ mvm_stop_voice_cmd.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_stop_voice_cmd.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_stop_voice_cmd) - APR_HDR_SIZE);
+ pr_debug("send mvm_stop_voice_cmd pkt size = %d\n",
+ mvm_stop_voice_cmd.pkt_size);
+ mvm_stop_voice_cmd.src_port = v->session_id;
+ mvm_stop_voice_cmd.dest_port = mvm_handle;
+ mvm_stop_voice_cmd.token = 0;
+ mvm_stop_voice_cmd.opcode = VSS_IMVM_CMD_STOP_VOICE;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_stop_voice_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IMVM_CMD_STOP_VOICE\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_cvs_register_cal_cmd(struct voice_data *v)
+{
+ struct cvs_register_cal_data_cmd cvs_reg_cal_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ uint32_t cal_paddr;
+
+ /* get the cvs cal data */
+ get_all_vocstrm_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ goto fail;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id)) {
+ if (common.cvs_cal.buf) {
+ cal_paddr = common.cvs_cal.phy;
+
+ memcpy(common.cvs_cal.buf,
+ (void *) cal_block.cal_kvaddr,
+ cal_block.cal_size);
+ } else {
+ return -EINVAL;
+ }
+ } else {
+ cal_paddr = cal_block.cal_paddr;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* fill in the header */
+ cvs_reg_cal_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_reg_cal_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_reg_cal_cmd) - APR_HDR_SIZE);
+ cvs_reg_cal_cmd.hdr.src_port = v->session_id;
+ cvs_reg_cal_cmd.hdr.dest_port = cvs_handle;
+ cvs_reg_cal_cmd.hdr.token = 0;
+ cvs_reg_cal_cmd.hdr.opcode = VSS_ISTREAM_CMD_REGISTER_CALIBRATION_DATA;
+
+ cvs_reg_cal_cmd.cvs_cal_data.phys_addr = cal_paddr;
+ cvs_reg_cal_cmd.cvs_cal_data.mem_size = cal_block.cal_size;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_reg_cal_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvs cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvs_deregister_cal_cmd(struct voice_data *v)
+{
+ struct cvs_deregister_cal_data_cmd cvs_dereg_cal_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ get_all_vocstrm_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ return 0;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* fill in the header */
+ cvs_dereg_cal_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_dereg_cal_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_dereg_cal_cmd) - APR_HDR_SIZE);
+ cvs_dereg_cal_cmd.hdr.src_port = v->session_id;
+ cvs_dereg_cal_cmd.hdr.dest_port = cvs_handle;
+ cvs_dereg_cal_cmd.hdr.token = 0;
+ cvs_dereg_cal_cmd.hdr.opcode =
+ VSS_ISTREAM_CMD_DEREGISTER_CALIBRATION_DATA;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_dereg_cal_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvs cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvp_map_memory_cmd(struct voice_data *v)
+{
+ struct vss_map_memory_cmd cvp_map_mem_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+ uint32_t cal_paddr;
+
+ /* get all cvp cal data */
+ get_all_cvp_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ goto fail;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id)) {
+ if (common.cvp_cal.buf)
+ cal_paddr = common.cvp_cal.phy;
+ else
+ return -EINVAL;
+ } else {
+ cal_paddr = cal_block.cal_paddr;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* fill in the header */
+ cvp_map_mem_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvp_map_mem_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_map_mem_cmd) - APR_HDR_SIZE);
+ cvp_map_mem_cmd.hdr.src_port = v->session_id;
+ cvp_map_mem_cmd.hdr.dest_port = cvp_handle;
+ cvp_map_mem_cmd.hdr.token = 0;
+ cvp_map_mem_cmd.hdr.opcode = VSS_ICOMMON_CMD_MAP_MEMORY;
+
+ pr_debug("%s, phy_addr:0x%x, mem_size:%d\n", __func__,
+ cal_paddr, cal_block.cal_size);
+ cvp_map_mem_cmd.vss_map_mem.phys_addr = cal_paddr;
+ cvp_map_mem_cmd.vss_map_mem.mem_size = cal_block.cal_size;
+ cvp_map_mem_cmd.vss_map_mem.mem_pool_id =
+ VSS_ICOMMON_MAP_MEMORY_SHMEM8_4K_POOL;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_map_mem_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvp cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvp_unmap_memory_cmd(struct voice_data *v)
+{
+ struct vss_unmap_memory_cmd cvp_unmap_mem_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+ uint32_t cal_paddr;
+
+ get_all_cvp_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ return 0;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id))
+ cal_paddr = common.cvp_cal.phy;
+ else
+ cal_paddr = cal_block.cal_paddr;
+
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* fill in the header */
+ cvp_unmap_mem_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvp_unmap_mem_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_unmap_mem_cmd) - APR_HDR_SIZE);
+ cvp_unmap_mem_cmd.hdr.src_port = v->session_id;
+ cvp_unmap_mem_cmd.hdr.dest_port = cvp_handle;
+ cvp_unmap_mem_cmd.hdr.token = 0;
+ cvp_unmap_mem_cmd.hdr.opcode = VSS_ICOMMON_CMD_UNMAP_MEMORY;
+
+ cvp_unmap_mem_cmd.vss_unmap_mem.phys_addr = cal_paddr;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_unmap_mem_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvp cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvs_map_memory_cmd(struct voice_data *v)
+{
+ struct vss_map_memory_cmd cvs_map_mem_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ uint32_t cal_paddr;
+
+ /* get all cvs cal data */
+ get_all_vocstrm_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ goto fail;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id)) {
+ if (common.cvs_cal.buf)
+ cal_paddr = common.cvs_cal.phy;
+ else
+ return -EINVAL;
+ } else {
+ cal_paddr = cal_block.cal_paddr;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* fill in the header */
+ cvs_map_mem_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_map_mem_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_map_mem_cmd) - APR_HDR_SIZE);
+ cvs_map_mem_cmd.hdr.src_port = v->session_id;
+ cvs_map_mem_cmd.hdr.dest_port = cvs_handle;
+ cvs_map_mem_cmd.hdr.token = 0;
+ cvs_map_mem_cmd.hdr.opcode = VSS_ICOMMON_CMD_MAP_MEMORY;
+
+ pr_debug("%s, phys_addr: 0x%x, mem_size: %d\n", __func__,
+ cal_paddr, cal_block.cal_size);
+ cvs_map_mem_cmd.vss_map_mem.phys_addr = cal_paddr;
+ cvs_map_mem_cmd.vss_map_mem.mem_size = cal_block.cal_size;
+ cvs_map_mem_cmd.vss_map_mem.mem_pool_id =
+ VSS_ICOMMON_MAP_MEMORY_SHMEM8_4K_POOL;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_map_mem_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvs cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvs_unmap_memory_cmd(struct voice_data *v)
+{
+ struct vss_unmap_memory_cmd cvs_unmap_mem_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ uint32_t cal_paddr;
+
+ get_all_vocstrm_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ return 0;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id))
+ cal_paddr = common.cvs_cal.phy;
+ else
+ cal_paddr = cal_block.cal_paddr;
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* fill in the header */
+ cvs_unmap_mem_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_unmap_mem_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_unmap_mem_cmd) - APR_HDR_SIZE);
+ cvs_unmap_mem_cmd.hdr.src_port = v->session_id;
+ cvs_unmap_mem_cmd.hdr.dest_port = cvs_handle;
+ cvs_unmap_mem_cmd.hdr.token = 0;
+ cvs_unmap_mem_cmd.hdr.opcode = VSS_ICOMMON_CMD_UNMAP_MEMORY;
+
+ cvs_unmap_mem_cmd.vss_unmap_mem.phys_addr = cal_paddr;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_unmap_mem_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvs cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvp_register_cal_cmd(struct voice_data *v)
+{
+ struct cvp_register_cal_data_cmd cvp_reg_cal_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+ uint32_t cal_paddr;
+
+ /* get the cvp cal data */
+ get_all_vocproc_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ goto fail;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id)) {
+ if (common.cvp_cal.buf) {
+ cal_paddr = common.cvp_cal.phy;
+
+ memcpy(common.cvp_cal.buf,
+ (void *)cal_block.cal_kvaddr,
+ cal_block.cal_size);
+ } else {
+ return -EINVAL;
+ }
+ } else {
+ cal_paddr = cal_block.cal_paddr;
+ }
+
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* fill in the header */
+ cvp_reg_cal_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvp_reg_cal_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_reg_cal_cmd) - APR_HDR_SIZE);
+ cvp_reg_cal_cmd.hdr.src_port = v->session_id;
+ cvp_reg_cal_cmd.hdr.dest_port = cvp_handle;
+ cvp_reg_cal_cmd.hdr.token = 0;
+ cvp_reg_cal_cmd.hdr.opcode = VSS_IVOCPROC_CMD_REGISTER_CALIBRATION_DATA;
+
+ cvp_reg_cal_cmd.cvp_cal_data.phys_addr = cal_paddr;
+ cvp_reg_cal_cmd.cvp_cal_data.mem_size = cal_block.cal_size;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_reg_cal_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvp cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvp_deregister_cal_cmd(struct voice_data *v)
+{
+ struct cvp_deregister_cal_data_cmd cvp_dereg_cal_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+
+ get_all_vocproc_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ return 0;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* fill in the header */
+ cvp_dereg_cal_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvp_dereg_cal_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_dereg_cal_cmd) - APR_HDR_SIZE);
+ cvp_dereg_cal_cmd.hdr.src_port = v->session_id;
+ cvp_dereg_cal_cmd.hdr.dest_port = cvp_handle;
+ cvp_dereg_cal_cmd.hdr.token = 0;
+ cvp_dereg_cal_cmd.hdr.opcode =
+ VSS_IVOCPROC_CMD_DEREGISTER_CALIBRATION_DATA;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_dereg_cal_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvp cal,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvp_register_vol_cal_table_cmd(struct voice_data *v)
+{
+ struct cvp_register_vol_cal_table_cmd cvp_reg_cal_tbl_cmd;
+ struct acdb_cal_block vol_block;
+ struct acdb_cal_block voc_block;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+ uint32_t cal_paddr;
+
+ /* get the cvp vol cal data */
+ get_all_vocvol_cal(&vol_block);
+ get_all_vocproc_cal(&voc_block);
+
+ if (vol_block.cal_size == 0)
+ goto fail;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ if (is_voip_session(v->session_id)) {
+ if (common.cvp_cal.buf) {
+ cal_paddr = common.cvp_cal.phy + voc_block.cal_size;
+
+ memcpy(common.cvp_cal.buf + voc_block.cal_size,
+ (void *) vol_block.cal_kvaddr,
+ vol_block.cal_size);
+ } else {
+ return -EINVAL;
+ }
+ } else {
+ cal_paddr = vol_block.cal_paddr;
+ }
+
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* fill in the header */
+ cvp_reg_cal_tbl_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvp_reg_cal_tbl_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_reg_cal_tbl_cmd) - APR_HDR_SIZE);
+ cvp_reg_cal_tbl_cmd.hdr.src_port = v->session_id;
+ cvp_reg_cal_tbl_cmd.hdr.dest_port = cvp_handle;
+ cvp_reg_cal_tbl_cmd.hdr.token = 0;
+ cvp_reg_cal_tbl_cmd.hdr.opcode =
+ VSS_IVOCPROC_CMD_REGISTER_VOLUME_CAL_TABLE;
+
+ cvp_reg_cal_tbl_cmd.cvp_vol_cal_tbl.phys_addr = cal_paddr;
+ cvp_reg_cal_tbl_cmd.cvp_vol_cal_tbl.mem_size = vol_block.cal_size;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_reg_cal_tbl_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvp cal table,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_cvp_deregister_vol_cal_table_cmd(struct voice_data *v)
+{
+ struct cvp_deregister_vol_cal_table_cmd cvp_dereg_cal_tbl_cmd;
+ struct acdb_cal_block cal_block;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+
+ get_all_vocvol_cal(&cal_block);
+ if (cal_block.cal_size == 0)
+ return 0;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* fill in the header */
+ cvp_dereg_cal_tbl_cmd.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_dereg_cal_tbl_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_dereg_cal_tbl_cmd) - APR_HDR_SIZE);
+ cvp_dereg_cal_tbl_cmd.hdr.src_port = v->session_id;
+ cvp_dereg_cal_tbl_cmd.hdr.dest_port = cvp_handle;
+ cvp_dereg_cal_tbl_cmd.hdr.token = 0;
+ cvp_dereg_cal_tbl_cmd.hdr.opcode =
+ VSS_IVOCPROC_CMD_DEREGISTER_VOLUME_CAL_TABLE;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_dereg_cal_tbl_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvp cal table,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+
+}
+
+static int voice_send_set_widevoice_enable_cmd(struct voice_data *v)
+{
+ struct mvm_set_widevoice_enable_cmd mvm_set_wv_cmd;
+ int ret = 0;
+ void *apr_mvm;
+ u16 mvm_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+
+ /* fill in the header */
+ mvm_set_wv_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mvm_set_wv_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_set_wv_cmd) - APR_HDR_SIZE);
+ mvm_set_wv_cmd.hdr.src_port = v->session_id;
+ mvm_set_wv_cmd.hdr.dest_port = mvm_handle;
+ mvm_set_wv_cmd.hdr.token = 0;
+ mvm_set_wv_cmd.hdr.opcode = VSS_IWIDEVOICE_CMD_SET_WIDEVOICE;
+
+ mvm_set_wv_cmd.vss_set_wv.enable = v->wv_enable;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_set_wv_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending mvm set widevoice enable,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_set_pp_enable_cmd(struct voice_data *v,
+ uint32_t module_id, int enable)
+{
+ struct cvs_set_pp_enable_cmd cvs_set_pp_cmd;
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* fill in the header */
+ cvs_set_pp_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_set_pp_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_set_pp_cmd) - APR_HDR_SIZE);
+ cvs_set_pp_cmd.hdr.src_port = v->session_id;
+ cvs_set_pp_cmd.hdr.dest_port = cvs_handle;
+ cvs_set_pp_cmd.hdr.token = 0;
+ cvs_set_pp_cmd.hdr.opcode = VSS_ICOMMON_CMD_SET_UI_PROPERTY;
+
+ cvs_set_pp_cmd.vss_set_pp.module_id = module_id;
+ cvs_set_pp_cmd.vss_set_pp.param_id = VOICE_PARAM_MOD_ENABLE;
+ cvs_set_pp_cmd.vss_set_pp.param_size = MOD_ENABLE_PARAM_LEN;
+ cvs_set_pp_cmd.vss_set_pp.reserved = 0;
+ cvs_set_pp_cmd.vss_set_pp.enable = enable;
+ cvs_set_pp_cmd.vss_set_pp.reserved_field = 0;
+ pr_debug("voice_send_set_pp_enable_cmd, module_id=%d, enable=%d\n",
+ module_id, enable);
+
+ v->cvs_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_set_pp_cmd);
+ if (ret < 0) {
+ pr_err("Fail: sending cvs set slowtalk enable,\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_setup_vocproc(struct voice_data *v)
+{
+ struct cvp_create_full_ctl_session_cmd cvp_session_cmd;
+ int ret = 0;
+ void *apr_cvp;
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ /* create cvp session and wait for response */
+ cvp_session_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_session_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_session_cmd) - APR_HDR_SIZE);
+ pr_debug(" send create cvp session, pkt size = %d\n",
+ cvp_session_cmd.hdr.pkt_size);
+ cvp_session_cmd.hdr.src_port = v->session_id;
+ cvp_session_cmd.hdr.dest_port = 0;
+ cvp_session_cmd.hdr.token = 0;
+ cvp_session_cmd.hdr.opcode =
+ VSS_IVOCPROC_CMD_CREATE_FULL_CONTROL_SESSION;
+
+ /* Use default topology if invalid value in ACDB */
+ cvp_session_cmd.cvp_session.tx_topology_id =
+ get_voice_tx_topology();
+ if (cvp_session_cmd.cvp_session.tx_topology_id == 0)
+ cvp_session_cmd.cvp_session.tx_topology_id =
+ VSS_IVOCPROC_TOPOLOGY_ID_TX_SM_ECNS;
+
+ cvp_session_cmd.cvp_session.rx_topology_id =
+ get_voice_rx_topology();
+ if (cvp_session_cmd.cvp_session.rx_topology_id == 0)
+ cvp_session_cmd.cvp_session.rx_topology_id =
+ VSS_IVOCPROC_TOPOLOGY_ID_RX_DEFAULT;
+
+ cvp_session_cmd.cvp_session.direction = 2; /*tx and rx*/
+ cvp_session_cmd.cvp_session.network_id = VSS_NETWORK_ID_DEFAULT;
+ cvp_session_cmd.cvp_session.tx_port_id = v->dev_tx.port_id;
+ cvp_session_cmd.cvp_session.rx_port_id = v->dev_rx.port_id;
+
+ pr_debug("topology=%d net_id=%d, dir=%d tx_port_id=%d, rx_port_id=%d\n",
+ cvp_session_cmd.cvp_session.tx_topology_id,
+ cvp_session_cmd.cvp_session.network_id,
+ cvp_session_cmd.cvp_session.direction,
+ cvp_session_cmd.cvp_session.tx_port_id,
+ cvp_session_cmd.cvp_session.rx_port_id);
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_session_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VOCPROC_FULL_CONTROL_SESSION\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ /* send cvs cal */
+ ret = voice_send_cvs_map_memory_cmd(v);
+ if (!ret)
+ voice_send_cvs_register_cal_cmd(v);
+
+ /* send cvp and vol cal */
+ ret = voice_send_cvp_map_memory_cmd(v);
+ if (!ret) {
+ voice_send_cvp_register_cal_cmd(v);
+ voice_send_cvp_register_vol_cal_table_cmd(v);
+ }
+
+ /* enable vocproc */
+ ret = voice_send_enable_vocproc_cmd(v);
+ if (ret < 0)
+ goto fail;
+
+ /* attach vocproc */
+ ret = voice_send_attach_vocproc_cmd(v);
+ if (ret < 0)
+ goto fail;
+
+ /* send tty mode if tty device is used */
+ voice_send_tty_mode_cmd(v);
+
+ /* enable widevoice if wv_enable is set */
+ if (v->wv_enable)
+ voice_send_set_widevoice_enable_cmd(v);
+
+ /* enable slowtalk if st_enable is set */
+ if (v->st_enable)
+ voice_send_set_pp_enable_cmd(v, MODULE_ID_VOICE_MODULE_ST,
+ v->st_enable);
+ voice_send_set_pp_enable_cmd(v, MODULE_ID_VOICE_MODULE_FENS,
+ v->fens_enable);
+
+ if (is_voip_session(v->session_id))
+ voice_send_netid_timing_cmd(v);
+
+ /* Start in-call music delivery if this feature is enabled */
+ if (v->music_info.play_enable)
+ voice_cvs_start_playback(v);
+
+ /* Start in-call recording if this feature is enabled */
+ if (v->rec_info.rec_enable)
+ voice_cvs_start_record(v, v->rec_info.rec_mode);
+
+ rtac_add_voice(voice_get_cvs_handle(v),
+ voice_get_cvp_handle(v),
+ v->dev_rx.port_id, v->dev_tx.port_id,
+ v->session_id);
+
+ return 0;
+
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_enable_vocproc_cmd(struct voice_data *v)
+{
+ int ret = 0;
+ struct apr_hdr cvp_enable_cmd;
+ void *apr_cvp;
+ u16 cvp_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* enable vocproc and wait for respose */
+ cvp_enable_cmd.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_enable_cmd.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_enable_cmd) - APR_HDR_SIZE);
+ pr_debug("cvp_enable_cmd pkt size = %d, cvp_handle=%d\n",
+ cvp_enable_cmd.pkt_size, cvp_handle);
+ cvp_enable_cmd.src_port = v->session_id;
+ cvp_enable_cmd.dest_port = cvp_handle;
+ cvp_enable_cmd.token = 0;
+ cvp_enable_cmd.opcode = VSS_IVOCPROC_CMD_ENABLE;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_enable_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IVOCPROC_CMD_ENABLE\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_netid_timing_cmd(struct voice_data *v)
+{
+ int ret = 0;
+ void *apr_mvm;
+ u16 mvm_handle;
+ struct mvm_set_network_cmd mvm_set_network;
+ struct mvm_set_voice_timing_cmd mvm_set_voice_timing;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+
+ ret = voice_config_cvs_vocoder(v);
+ if (ret < 0) {
+ pr_err("%s: Error %d configuring CVS voc",
+ __func__, ret);
+ goto fail;
+ }
+ /* Set network ID. */
+ pr_debug("Setting network ID\n");
+
+ mvm_set_network.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_set_network.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_set_network) - APR_HDR_SIZE);
+ mvm_set_network.hdr.src_port = v->session_id;
+ mvm_set_network.hdr.dest_port = mvm_handle;
+ mvm_set_network.hdr.token = 0;
+ mvm_set_network.hdr.opcode = VSS_ICOMMON_CMD_SET_NETWORK;
+ mvm_set_network.network.network_id = common.mvs_info.network_type;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_set_network);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_NETWORK\n", __func__, ret);
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ /* Set voice timing. */
+ pr_debug("Setting voice timing\n");
+
+ mvm_set_voice_timing.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_set_voice_timing.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_set_voice_timing) -
+ APR_HDR_SIZE);
+ mvm_set_voice_timing.hdr.src_port = v->session_id;
+ mvm_set_voice_timing.hdr.dest_port = mvm_handle;
+ mvm_set_voice_timing.hdr.token = 0;
+ mvm_set_voice_timing.hdr.opcode = VSS_ICOMMON_CMD_SET_VOICE_TIMING;
+ mvm_set_voice_timing.timing.mode = 0;
+ mvm_set_voice_timing.timing.enc_offset = 8000;
+ mvm_set_voice_timing.timing.dec_req_offset = 3300;
+ mvm_set_voice_timing.timing.dec_offset = 8300;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_set_voice_timing);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending SET_TIMING\n", __func__, ret);
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_attach_vocproc_cmd(struct voice_data *v)
+{
+ int ret = 0;
+ struct mvm_attach_vocproc_cmd mvm_a_vocproc_cmd;
+ void *apr_mvm;
+ u16 mvm_handle, cvp_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* attach vocproc and wait for response */
+ mvm_a_vocproc_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_a_vocproc_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_a_vocproc_cmd) - APR_HDR_SIZE);
+ pr_debug("send mvm_a_vocproc_cmd pkt size = %d\n",
+ mvm_a_vocproc_cmd.hdr.pkt_size);
+ mvm_a_vocproc_cmd.hdr.src_port = v->session_id;
+ mvm_a_vocproc_cmd.hdr.dest_port = mvm_handle;
+ mvm_a_vocproc_cmd.hdr.token = 0;
+ mvm_a_vocproc_cmd.hdr.opcode = VSS_IMVM_CMD_ATTACH_VOCPROC;
+ mvm_a_vocproc_cmd.mvm_attach_cvp_handle.handle = cvp_handle;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_a_vocproc_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IMVM_CMD_ATTACH_VOCPROC\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_destroy_vocproc(struct voice_data *v)
+{
+ struct mvm_detach_vocproc_cmd mvm_d_vocproc_cmd;
+ struct apr_hdr cvp_destroy_session_cmd;
+ int ret = 0;
+ void *apr_mvm, *apr_cvp;
+ u16 mvm_handle, cvp_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_mvm || !apr_cvp) {
+ pr_err("%s: apr_mvm or apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* stop playback or recording */
+ v->music_info.force = 1;
+ voice_cvs_stop_playback(v);
+ voice_cvs_stop_record(v);
+ /* send stop voice cmd */
+ voice_send_stop_voice_cmd(v);
+
+ /* detach VOCPROC and wait for response from mvm */
+ mvm_d_vocproc_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_d_vocproc_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_d_vocproc_cmd) - APR_HDR_SIZE);
+ pr_debug("mvm_d_vocproc_cmd pkt size = %d\n",
+ mvm_d_vocproc_cmd.hdr.pkt_size);
+ mvm_d_vocproc_cmd.hdr.src_port = v->session_id;
+ mvm_d_vocproc_cmd.hdr.dest_port = mvm_handle;
+ mvm_d_vocproc_cmd.hdr.token = 0;
+ mvm_d_vocproc_cmd.hdr.opcode = VSS_IMVM_CMD_DETACH_VOCPROC;
+ mvm_d_vocproc_cmd.mvm_detach_cvp_handle.handle = cvp_handle;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_d_vocproc_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IMVM_CMD_DETACH_VOCPROC\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->mvm_wait,
+ (v->mvm_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ /* deregister cvp and vol cal */
+ voice_send_cvp_deregister_vol_cal_table_cmd(v);
+ voice_send_cvp_deregister_cal_cmd(v);
+ voice_send_cvp_unmap_memory_cmd(v);
+
+ /* deregister cvs cal */
+ voice_send_cvs_deregister_cal_cmd(v);
+ voice_send_cvs_unmap_memory_cmd(v);
+
+ /* destrop cvp session */
+ cvp_destroy_session_cmd.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_destroy_session_cmd.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_destroy_session_cmd) - APR_HDR_SIZE);
+ pr_debug("cvp_destroy_session_cmd pkt size = %d\n",
+ cvp_destroy_session_cmd.pkt_size);
+ cvp_destroy_session_cmd.src_port = v->session_id;
+ cvp_destroy_session_cmd.dest_port = cvp_handle;
+ cvp_destroy_session_cmd.token = 0;
+ cvp_destroy_session_cmd.opcode = APRV2_IBASIC_CMD_DESTROY_SESSION;
+
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_destroy_session_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending APRV2_IBASIC_CMD_DESTROY_SESSION\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ goto fail;
+ }
+
+ rtac_remove_voice(voice_get_cvs_handle(v));
+ cvp_handle = 0;
+ voice_set_cvp_handle(v, cvp_handle);
+
+ return 0;
+
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_mute_cmd(struct voice_data *v)
+{
+ struct cvs_set_mute_cmd cvs_mute_cmd;
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvs_handle = voice_get_cvs_handle(v);
+
+ /* send mute/unmute to cvs */
+ cvs_mute_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_mute_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_mute_cmd) - APR_HDR_SIZE);
+ cvs_mute_cmd.hdr.src_port = v->session_id;
+ cvs_mute_cmd.hdr.dest_port = cvs_handle;
+ cvs_mute_cmd.hdr.token = 0;
+ cvs_mute_cmd.hdr.opcode = VSS_ISTREAM_CMD_SET_MUTE;
+ cvs_mute_cmd.cvs_set_mute.direction = 0; /*tx*/
+ cvs_mute_cmd.cvs_set_mute.mute_flag = v->dev_tx.mute;
+
+ pr_info(" mute value =%d\n", cvs_mute_cmd.cvs_set_mute.mute_flag);
+ v->cvs_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_mute_cmd);
+ if (ret < 0) {
+ pr_err("Fail: send STREAM SET MUTE\n");
+ goto fail;
+ }
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret)
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+static int voice_send_rx_device_mute_cmd(struct voice_data *v)
+{
+ struct cvp_set_mute_cmd cvp_mute_cmd;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ cvp_mute_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_mute_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_mute_cmd) - APR_HDR_SIZE);
+ cvp_mute_cmd.hdr.src_port = v->session_id;
+ cvp_mute_cmd.hdr.dest_port = cvp_handle;
+ cvp_mute_cmd.hdr.token = 0;
+ cvp_mute_cmd.hdr.opcode = VSS_IVOCPROC_CMD_SET_MUTE;
+ cvp_mute_cmd.cvp_set_mute.direction = 1;
+ cvp_mute_cmd.cvp_set_mute.mute_flag = v->dev_rx.mute;
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_mute_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending RX device mute cmd\n");
+ return -EINVAL;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int voice_send_vol_index_cmd(struct voice_data *v)
+{
+ struct cvp_set_rx_volume_index_cmd cvp_vol_cmd;
+ int ret = 0;
+ void *apr_cvp;
+ u16 cvp_handle;
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvp = common.apr_q6_cvp;
+
+ if (!apr_cvp) {
+ pr_err("%s: apr_cvp is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ cvp_handle = voice_get_cvp_handle(v);
+
+ /* send volume index to cvp */
+ cvp_vol_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvp_vol_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvp_vol_cmd) - APR_HDR_SIZE);
+ cvp_vol_cmd.hdr.src_port = v->session_id;
+ cvp_vol_cmd.hdr.dest_port = cvp_handle;
+ cvp_vol_cmd.hdr.token = 0;
+ cvp_vol_cmd.hdr.opcode = VSS_IVOCPROC_CMD_SET_RX_VOLUME_INDEX;
+ cvp_vol_cmd.cvp_set_vol_idx.vol_index = v->dev_rx.volume;
+ v->cvp_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_cvp, (uint32_t *) &cvp_vol_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending RX VOL INDEX\n");
+ return -EINVAL;
+ }
+ ret = wait_event_timeout(v->cvp_wait,
+ (v->cvp_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int voice_cvs_start_record(struct voice_data *v, uint32_t rec_mode)
+{
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ struct cvs_start_record_cmd cvs_start_record;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (!v->rec_info.recording) {
+ cvs_start_record.hdr.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_start_record.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_start_record) - APR_HDR_SIZE);
+ cvs_start_record.hdr.src_port = v->session_id;
+ cvs_start_record.hdr.dest_port = cvs_handle;
+ cvs_start_record.hdr.token = 0;
+ cvs_start_record.hdr.opcode = VSS_ISTREAM_CMD_START_RECORD;
+
+ if (rec_mode == VOC_REC_UPLINK) {
+ cvs_start_record.rec_mode.rx_tap_point =
+ VSS_TAP_POINT_NONE;
+ cvs_start_record.rec_mode.tx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ } else if (rec_mode == VOC_REC_DOWNLINK) {
+ cvs_start_record.rec_mode.rx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ cvs_start_record.rec_mode.tx_tap_point =
+ VSS_TAP_POINT_NONE;
+ } else if (rec_mode == VOC_REC_BOTH) {
+ cvs_start_record.rec_mode.rx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ cvs_start_record.rec_mode.tx_tap_point =
+ VSS_TAP_POINT_STREAM_END;
+ } else {
+ pr_err("%s: Invalid in-call rec_mode %d\n", __func__,
+ rec_mode);
+
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_start_record);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending START_RECORD\n", __func__,
+ ret);
+
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+ v->rec_info.recording = 1;
+ } else {
+ pr_debug("%s: Start record already sent\n", __func__);
+ }
+
+ return 0;
+
+fail:
+ return ret;
+}
+
+static int voice_cvs_stop_record(struct voice_data *v)
+{
+ int ret = 0;
+ void *apr_cvs;
+ u16 cvs_handle;
+ struct apr_hdr cvs_stop_record;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (v->rec_info.recording) {
+ cvs_stop_record.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_stop_record.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_stop_record) - APR_HDR_SIZE);
+ cvs_stop_record.src_port = v->session_id;
+ cvs_stop_record.dest_port = cvs_handle;
+ cvs_stop_record.token = 0;
+ cvs_stop_record.opcode = VSS_ISTREAM_CMD_STOP_RECORD;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_stop_record);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending STOP_RECORD\n",
+ __func__, ret);
+
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+ v->rec_info.recording = 0;
+ } else {
+ pr_debug("%s: Stop record already sent\n", __func__);
+ }
+
+ return 0;
+
+fail:
+
+ return ret;
+}
+
+int voc_start_record(uint32_t port_id, uint32_t set)
+{
+ int ret = 0;
+ int rec_mode = 0;
+ u16 cvs_handle;
+ int i, rec_set = 0;
+
+ for (i = 0; i < MAX_VOC_SESSIONS; i++) {
+ struct voice_data *v = &common.voice[i];
+ pr_debug("%s: i:%d port_id: %d, set: %d\n",
+ __func__, i, port_id, set);
+
+ mutex_lock(&v->lock);
+ rec_mode = v->rec_info.rec_mode;
+ rec_set = set;
+ if (set) {
+ if ((v->rec_route_state.ul_flag != 0) &&
+ (v->rec_route_state.dl_flag != 0)) {
+ pr_debug("%s: i=%d, rec mode already set.\n",
+ __func__, i);
+ mutex_unlock(&v->lock);
+ if (i < MAX_VOC_SESSIONS)
+ continue;
+ else
+ return 0;
+ }
+
+ if (port_id == VOICE_RECORD_TX) {
+ if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ rec_mode = VOC_REC_UPLINK;
+ v->rec_route_state.ul_flag = 1;
+ } else if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ voice_cvs_stop_record(v);
+ rec_mode = VOC_REC_BOTH;
+ v->rec_route_state.ul_flag = 1;
+ }
+ } else if (port_id == VOICE_RECORD_RX) {
+ if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ rec_mode = VOC_REC_DOWNLINK;
+ v->rec_route_state.dl_flag = 1;
+ } else if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ voice_cvs_stop_record(v);
+ rec_mode = VOC_REC_BOTH;
+ v->rec_route_state.dl_flag = 1;
+ }
+ }
+ rec_set = 1;
+ } else {
+ if ((v->rec_route_state.ul_flag == 0) &&
+ (v->rec_route_state.dl_flag == 0)) {
+ pr_debug("%s: i=%d, rec already stops.\n",
+ __func__, i);
+ mutex_unlock(&v->lock);
+ if (i < MAX_VOC_SESSIONS)
+ continue;
+ else
+ return 0;
+ }
+
+ if (port_id == VOICE_RECORD_TX) {
+ if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag == 0)) {
+ v->rec_route_state.ul_flag = 0;
+ rec_set = 0;
+ } else if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ voice_cvs_stop_record(v);
+ v->rec_route_state.ul_flag = 0;
+ rec_mode = VOC_REC_DOWNLINK;
+ rec_set = 1;
+ }
+ } else if (port_id == VOICE_RECORD_RX) {
+ if ((v->rec_route_state.ul_flag == 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ v->rec_route_state.dl_flag = 0;
+ rec_set = 0;
+ } else if ((v->rec_route_state.ul_flag != 0)
+ && (v->rec_route_state.dl_flag != 0)) {
+ voice_cvs_stop_record(v);
+ v->rec_route_state.dl_flag = 0;
+ rec_mode = VOC_REC_UPLINK;
+ rec_set = 1;
+ }
+ }
+ }
+ pr_debug("%s: i=%d, mode =%d, set =%d\n", __func__,
+ i, rec_mode, rec_set);
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (cvs_handle != 0) {
+ if (rec_set)
+ ret = voice_cvs_start_record(v, rec_mode);
+ else
+ ret = voice_cvs_stop_record(v);
+ }
+
+ /* Cache the value */
+ v->rec_info.rec_enable = rec_set;
+ v->rec_info.rec_mode = rec_mode;
+
+ mutex_unlock(&v->lock);
+ }
+
+ return ret;
+}
+
+static int voice_cvs_start_playback(struct voice_data *v)
+{
+ int ret = 0;
+ struct apr_hdr cvs_start_playback;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (!v->music_info.playing && v->music_info.count) {
+ cvs_start_playback.hdr_field = APR_HDR_FIELD(
+ APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ cvs_start_playback.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_start_playback) - APR_HDR_SIZE);
+ cvs_start_playback.src_port = v->session_id;
+ cvs_start_playback.dest_port = cvs_handle;
+ cvs_start_playback.token = 0;
+ cvs_start_playback.opcode = VSS_ISTREAM_CMD_START_PLAYBACK;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_start_playback);
+
+ if (ret < 0) {
+ pr_err("%s: Error %d sending START_PLAYBACK\n",
+ __func__, ret);
+
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+
+ v->music_info.playing = 1;
+ } else {
+ pr_debug("%s: Start playback already sent\n", __func__);
+ }
+
+ return 0;
+
+fail:
+ return ret;
+}
+
+static int voice_cvs_stop_playback(struct voice_data *v)
+{
+ int ret = 0;
+ struct apr_hdr cvs_stop_playback;
+ void *apr_cvs;
+ u16 cvs_handle;
+
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_cvs = common.apr_q6_cvs;
+
+ if (!apr_cvs) {
+ pr_err("%s: apr_cvs is NULL.\n", __func__);
+ return -EINVAL;
+ }
+
+ cvs_handle = voice_get_cvs_handle(v);
+
+ if (v->music_info.playing && ((!v->music_info.count) ||
+ (v->music_info.force))) {
+ cvs_stop_playback.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cvs_stop_playback.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(cvs_stop_playback) - APR_HDR_SIZE);
+ cvs_stop_playback.src_port = v->session_id;
+ cvs_stop_playback.dest_port = cvs_handle;
+ cvs_stop_playback.token = 0;
+
+ cvs_stop_playback.opcode = VSS_ISTREAM_CMD_STOP_PLAYBACK;
+
+ v->cvs_state = CMD_STATUS_FAIL;
+
+ ret = apr_send_pkt(apr_cvs, (uint32_t *) &cvs_stop_playback);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending STOP_PLAYBACK\n",
+ __func__, ret);
+
+
+ goto fail;
+ }
+
+ ret = wait_event_timeout(v->cvs_wait,
+ (v->cvs_state == CMD_STATUS_SUCCESS),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+
+ goto fail;
+ }
+
+ v->music_info.playing = 0;
+ v->music_info.force = 0;
+ } else {
+ pr_debug("%s: Stop playback already sent\n", __func__);
+ }
+
+ return 0;
+
+fail:
+ return ret;
+}
+
+int voc_start_playback(uint32_t set)
+{
+ int ret = 0;
+ u16 cvs_handle;
+ int i;
+
+
+ for (i = 0; i < MAX_VOC_SESSIONS; i++) {
+ struct voice_data *v = &common.voice[i];
+
+ mutex_lock(&v->lock);
+ v->music_info.play_enable = set;
+ if (set)
+ v->music_info.count++;
+ else
+ v->music_info.count--;
+ pr_debug("%s: music_info count =%d\n", __func__,
+ v->music_info.count);
+
+ cvs_handle = voice_get_cvs_handle(v);
+ if (cvs_handle != 0) {
+ if (set)
+ ret = voice_cvs_start_playback(v);
+ else
+ ret = voice_cvs_stop_playback(v);
+ }
+
+ mutex_unlock(&v->lock);
+ }
+
+ return ret;
+}
+
+int voc_disable_cvp(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ if (v->voc_state == VOC_RUN) {
+ if (v->dev_tx.port_id != RT_PROXY_PORT_001_TX &&
+ v->dev_rx.port_id != RT_PROXY_PORT_001_RX)
+ afe_sidetone(v->dev_tx.port_id, v->dev_rx.port_id,
+ 0, 0);
+
+ rtac_remove_voice(voice_get_cvs_handle(v));
+ /* send cmd to dsp to disable vocproc */
+ ret = voice_send_disable_vocproc_cmd(v);
+ if (ret < 0) {
+ pr_err("%s: disable vocproc failed\n", __func__);
+ goto fail;
+ }
+
+ /* deregister cvp and vol cal */
+ voice_send_cvp_deregister_vol_cal_table_cmd(v);
+ voice_send_cvp_deregister_cal_cmd(v);
+ voice_send_cvp_unmap_memory_cmd(v);
+
+ v->voc_state = VOC_CHANGE;
+ }
+
+fail: mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_enable_cvp(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ struct sidetone_cal sidetone_cal_data;
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ if (v->voc_state == VOC_CHANGE) {
+ ret = voice_send_set_device_cmd(v);
+ if (ret < 0) {
+ pr_err("%s: set device failed\n", __func__);
+ goto fail;
+ }
+ /* send cvp and vol cal */
+ ret = voice_send_cvp_map_memory_cmd(v);
+ if (!ret) {
+ voice_send_cvp_register_cal_cmd(v);
+ voice_send_cvp_register_vol_cal_table_cmd(v);
+ }
+ ret = voice_send_enable_vocproc_cmd(v);
+ if (ret < 0) {
+ pr_err("%s: enable vocproc failed\n", __func__);
+ goto fail;
+
+ }
+ /* send tty mode if tty device is used */
+ voice_send_tty_mode_cmd(v);
+
+ /* enable widevoice if wv_enable is set */
+ if (v->wv_enable)
+ voice_send_set_widevoice_enable_cmd(v);
+
+ /* enable slowtalk */
+ if (v->st_enable)
+ voice_send_set_pp_enable_cmd(v,
+ MODULE_ID_VOICE_MODULE_ST,
+ v->st_enable);
+ /* enable FENS */
+ if (v->fens_enable)
+ voice_send_set_pp_enable_cmd(v,
+ MODULE_ID_VOICE_MODULE_FENS,
+ v->fens_enable);
+
+ get_sidetone_cal(&sidetone_cal_data);
+ if (v->dev_tx.port_id != RT_PROXY_PORT_001_TX &&
+ v->dev_rx.port_id != RT_PROXY_PORT_001_RX) {
+ ret = afe_sidetone(v->dev_tx.port_id,
+ v->dev_rx.port_id,
+ sidetone_cal_data.enable,
+ sidetone_cal_data.gain);
+
+ if (ret < 0)
+ pr_err("%s: AFE command sidetone failed\n",
+ __func__);
+ }
+
+ rtac_add_voice(voice_get_cvs_handle(v),
+ voice_get_cvp_handle(v),
+ v->dev_rx.port_id, v->dev_tx.port_id,
+ v->session_id);
+ v->voc_state = VOC_RUN;
+ }
+
+fail:
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_tx_mute(uint16_t session_id, uint32_t dir, uint32_t mute)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ v->dev_tx.mute = mute;
+
+ if (v->voc_state == VOC_RUN)
+ ret = voice_send_mute_cmd(v);
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_rx_device_mute(uint16_t session_id, uint32_t mute)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ v->dev_rx.mute = mute;
+
+ if (v->voc_state == VOC_RUN)
+ ret = voice_send_rx_device_mute_cmd(v);
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_get_rx_device_mute(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ ret = v->dev_rx.mute;
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_tty_mode(uint16_t session_id, uint8_t tty_mode)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ v->tty_mode = tty_mode;
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+uint8_t voc_get_tty_mode(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ ret = v->tty_mode;
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_widevoice_enable(uint16_t session_id, uint32_t wv_enable)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ u16 mvm_handle;
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ v->wv_enable = wv_enable;
+
+ mvm_handle = voice_get_mvm_handle(v);
+
+ if (mvm_handle != 0)
+ voice_send_set_widevoice_enable_cmd(v);
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+uint32_t voc_get_widevoice_enable(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ ret = v->wv_enable;
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_pp_enable(uint16_t session_id, uint32_t module_id, uint32_t enable)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+ if (module_id == MODULE_ID_VOICE_MODULE_ST)
+ v->st_enable = enable;
+ else if (module_id == MODULE_ID_VOICE_MODULE_FENS)
+ v->fens_enable = enable;
+
+ if (v->voc_state == VOC_RUN) {
+ if (module_id == MODULE_ID_VOICE_MODULE_ST)
+ ret = voice_send_set_pp_enable_cmd(v,
+ MODULE_ID_VOICE_MODULE_ST,
+ enable);
+ else if (module_id == MODULE_ID_VOICE_MODULE_FENS)
+ ret = voice_send_set_pp_enable_cmd(v,
+ MODULE_ID_VOICE_MODULE_FENS,
+ enable);
+ }
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_get_pp_enable(uint16_t session_id, uint32_t module_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+ if (module_id == MODULE_ID_VOICE_MODULE_ST)
+ ret = v->st_enable;
+ else if (module_id == MODULE_ID_VOICE_MODULE_FENS)
+ ret = v->fens_enable;
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_rx_vol_index(uint16_t session_id, uint32_t dir, uint32_t vol_idx)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ v->dev_rx.volume = vol_idx;
+
+ if (v->voc_state == VOC_RUN)
+ ret = voice_send_vol_index_cmd(v);
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_set_rxtx_port(uint16_t session_id, uint32_t port_id, uint32_t dev_type)
+{
+ struct voice_data *v = voice_get_session(session_id);
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ pr_debug("%s: port_id=%d, type=%d\n", __func__, port_id, dev_type);
+
+ mutex_lock(&v->lock);
+
+ if (dev_type == DEV_RX)
+ v->dev_rx.port_id = port_id;
+ else
+ v->dev_tx.port_id = port_id;
+
+ mutex_unlock(&v->lock);
+
+ return 0;
+}
+
+int voc_set_route_flag(uint16_t session_id, uint8_t path_dir, uint8_t set)
+{
+ struct voice_data *v = voice_get_session(session_id);
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ pr_debug("%s: path_dir=%d, set=%d\n", __func__, path_dir, set);
+
+ mutex_lock(&v->lock);
+
+ if (path_dir == RX_PATH)
+ v->voc_route_state.rx_route_flag = set;
+ else
+ v->voc_route_state.tx_route_flag = set;
+
+ mutex_unlock(&v->lock);
+
+ return 0;
+}
+
+uint8_t voc_get_route_flag(uint16_t session_id, uint8_t path_dir)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return 0;
+ }
+
+ mutex_lock(&v->lock);
+
+ if (path_dir == RX_PATH)
+ ret = v->voc_route_state.rx_route_flag;
+ else
+ ret = v->voc_route_state.tx_route_flag;
+
+ mutex_unlock(&v->lock);
+
+ return ret;
+}
+
+int voc_end_voice_call(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ if (v->voc_state == VOC_RUN || v->voc_state == VOC_STANDBY) {
+ if (v->dev_tx.port_id != RT_PROXY_PORT_001_TX &&
+ v->dev_rx.port_id != RT_PROXY_PORT_001_RX)
+ afe_sidetone(v->dev_tx.port_id, v->dev_rx.port_id,
+ 0, 0);
+ ret = voice_destroy_vocproc(v);
+ if (ret < 0)
+ pr_err("%s: destroy voice failed\n", __func__);
+ voice_destroy_mvm_cvs_session(v);
+
+ v->voc_state = VOC_RELEASE;
+ }
+ mutex_unlock(&v->lock);
+ return ret;
+}
+
+int voc_resume_voice_call(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ struct apr_hdr mvm_start_voice_cmd;
+ int ret = 0;
+ void *apr_mvm;
+ u16 mvm_handle;
+
+ pr_debug("%s:\n", __func__);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ apr_mvm = common.apr_q6_mvm;
+
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ return -EINVAL;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+
+ mvm_start_voice_cmd.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mvm_start_voice_cmd.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_start_voice_cmd) - APR_HDR_SIZE);
+ pr_debug("send mvm_start_voice_cmd pkt size = %d\n",
+ mvm_start_voice_cmd.pkt_size);
+ mvm_start_voice_cmd.src_port = v->session_id;
+ mvm_start_voice_cmd.dest_port = mvm_handle;
+ mvm_start_voice_cmd.token = 0;
+ mvm_start_voice_cmd.opcode = VSS_IMVM_CMD_START_VOICE;
+
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm, (uint32_t *) &mvm_start_voice_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IMVM_CMD_START_VOICE\n");
+ goto fail;
+ }
+ v->voc_state = VOC_RUN;
+ return 0;
+fail:
+ return -EINVAL;
+}
+
+int voc_start_voice_call(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ struct sidetone_cal sidetone_cal_data;
+ int ret = 0;
+
+ if (v == NULL) {
+ pr_err("%s: invalid session_id 0x%x\n", __func__, session_id);
+
+ return -EINVAL;
+ }
+
+ mutex_lock(&v->lock);
+
+ if ((v->voc_state == VOC_INIT) ||
+ (v->voc_state == VOC_RELEASE)) {
+ ret = voice_apr_register();
+ if (ret < 0) {
+ pr_err("%s: apr register failed\n", __func__);
+ goto fail;
+ }
+ ret = voice_create_mvm_cvs_session(v);
+ if (ret < 0) {
+ pr_err("create mvm and cvs failed\n");
+ goto fail;
+ }
+ ret = voice_send_dual_control_cmd(v);
+ if (ret < 0) {
+ pr_err("Err Dual command failed\n");
+ goto fail;
+ }
+ ret = voice_setup_vocproc(v);
+ if (ret < 0) {
+ pr_err("setup voice failed\n");
+ goto fail;
+ }
+ ret = voice_send_start_voice_cmd(v);
+ if (ret < 0) {
+ pr_err("start voice failed\n");
+ goto fail;
+ }
+ get_sidetone_cal(&sidetone_cal_data);
+ if (v->dev_tx.port_id != RT_PROXY_PORT_001_TX &&
+ v->dev_rx.port_id != RT_PROXY_PORT_001_RX) {
+ ret = afe_sidetone(v->dev_tx.port_id,
+ v->dev_rx.port_id,
+ sidetone_cal_data.enable,
+ sidetone_cal_data.gain);
+ if (ret < 0)
+ pr_err("AFE command sidetone failed\n");
+ }
+
+ v->voc_state = VOC_RUN;
+ } else if (v->voc_state == VOC_STANDBY) {
+ pr_err("Error: PCM Prepare when in Standby\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+fail: mutex_unlock(&v->lock);
+ return ret;
+}
+
+int voc_standby_voice_call(uint16_t session_id)
+{
+ struct voice_data *v = voice_get_session(session_id);
+ struct apr_hdr mvm_standby_voice_cmd;
+ void *apr_mvm;
+ u16 mvm_handle;
+ int ret = 0;
+
+ pr_debug("%s: voc state=%d", __func__, v->voc_state);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+ return -EINVAL;
+ }
+ if (v->voc_state == VOC_RUN) {
+ apr_mvm = common.apr_q6_mvm;
+ if (!apr_mvm) {
+ pr_err("%s: apr_mvm is NULL.\n", __func__);
+ ret = -EINVAL;
+ goto fail;
+ }
+ mvm_handle = voice_get_mvm_handle(v);
+ mvm_standby_voice_cmd.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mvm_standby_voice_cmd.pkt_size =
+ APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(mvm_standby_voice_cmd) - APR_HDR_SIZE);
+ pr_debug("send mvm_standby_voice_cmd pkt size = %d\n",
+ mvm_standby_voice_cmd.pkt_size);
+ mvm_standby_voice_cmd.src_port = v->session_id;
+ mvm_standby_voice_cmd.dest_port = mvm_handle;
+ mvm_standby_voice_cmd.token = 0;
+ mvm_standby_voice_cmd.opcode = VSS_IMVM_CMD_STANDBY_VOICE;
+ v->mvm_state = CMD_STATUS_FAIL;
+ ret = apr_send_pkt(apr_mvm,
+ (uint32_t *)&mvm_standby_voice_cmd);
+ if (ret < 0) {
+ pr_err("Fail in sending VSS_IMVM_CMD_STANDBY_VOICE\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+ v->voc_state = VOC_STANDBY;
+ }
+fail:
+ return ret;
+}
+
+void voc_register_mvs_cb(ul_cb_fn ul_cb,
+ dl_cb_fn dl_cb,
+ void *private_data)
+{
+ common.mvs_info.ul_cb = ul_cb;
+ common.mvs_info.dl_cb = dl_cb;
+ common.mvs_info.private_data = private_data;
+}
+
+void voc_config_vocoder(uint32_t media_type,
+ uint32_t rate,
+ uint32_t network_type,
+ uint32_t dtx_mode)
+{
+ common.mvs_info.media_type = media_type;
+ common.mvs_info.rate = rate;
+ common.mvs_info.network_type = network_type;
+ common.mvs_info.dtx_mode = dtx_mode;
+}
+
+static int32_t qdsp_mvm_callback(struct apr_client_data *data, void *priv)
+{
+ uint32_t *ptr = NULL;
+ struct common_data *c = NULL;
+ struct voice_data *v = NULL;
+ int i = 0;
+
+ if ((data == NULL) || (priv == NULL)) {
+ pr_err("%s: data or priv is NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ c = priv;
+
+ pr_debug("%s: session_id 0x%x\n", __func__, data->dest_port);
+
+ v = voice_get_session(data->dest_port);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+
+ return -EINVAL;
+ }
+
+ pr_debug("%s: Payload Length = %d, opcode=%x\n", __func__,
+ data->payload_size, data->opcode);
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("%s: Reset event received in Voice service\n",
+ __func__);
+
+ apr_reset(c->apr_q6_mvm);
+ c->apr_q6_mvm = NULL;
+
+ /* Sub-system restart is applicable to all sessions. */
+ for (i = 0; i < MAX_VOC_SESSIONS; i++)
+ c->voice[i].mvm_handle = 0;
+
+ return 0;
+ }
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ if (data->payload_size) {
+ ptr = data->payload;
+
+ pr_info("%x %x\n", ptr[0], ptr[1]);
+ /* ping mvm service ACK */
+ switch (ptr[0]) {
+ case VSS_IMVM_CMD_CREATE_PASSIVE_CONTROL_SESSION:
+ case VSS_IMVM_CMD_CREATE_FULL_CONTROL_SESSION:
+ /* Passive session is used for CS call
+ * Full session is used for VoIP call. */
+ pr_debug("%s: cmd = 0x%x\n", __func__, ptr[0]);
+ if (!ptr[1]) {
+ pr_debug("%s: MVM handle is %d\n",
+ __func__, data->src_port);
+ voice_set_mvm_handle(v, data->src_port);
+ } else
+ pr_err("got NACK for sending \
+ MVM create session \n");
+ v->mvm_state = CMD_STATUS_SUCCESS;
+ wake_up(&v->mvm_wait);
+ break;
+ case VSS_IMVM_CMD_START_VOICE:
+ case VSS_IMVM_CMD_ATTACH_VOCPROC:
+ case VSS_IMVM_CMD_STOP_VOICE:
+ case VSS_IMVM_CMD_DETACH_VOCPROC:
+ case VSS_ISTREAM_CMD_SET_TTY_MODE:
+ case APRV2_IBASIC_CMD_DESTROY_SESSION:
+ case VSS_IMVM_CMD_ATTACH_STREAM:
+ case VSS_IMVM_CMD_DETACH_STREAM:
+ case VSS_ICOMMON_CMD_SET_NETWORK:
+ case VSS_ICOMMON_CMD_SET_VOICE_TIMING:
+ case VSS_IWIDEVOICE_CMD_SET_WIDEVOICE:
+ case VSS_IMVM_CMD_SET_POLICY_DUAL_CONTROL:
+ case VSS_IMVM_CMD_STANDBY_VOICE:
+ pr_debug("%s: cmd = 0x%x\n", __func__, ptr[0]);
+ v->mvm_state = CMD_STATUS_SUCCESS;
+ wake_up(&v->mvm_wait);
+ break;
+ default:
+ pr_debug("%s: not match cmd = 0x%x\n",
+ __func__, ptr[0]);
+ break;
+ }
+ }
+ }
+
+ return 0;
+}
+
+static int32_t qdsp_cvs_callback(struct apr_client_data *data, void *priv)
+{
+ uint32_t *ptr = NULL;
+ struct common_data *c = NULL;
+ struct voice_data *v = NULL;
+ int i = 0;
+
+ if ((data == NULL) || (priv == NULL)) {
+ pr_err("%s: data or priv is NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ c = priv;
+
+ pr_debug("%s: session_id 0x%x\n", __func__, data->dest_port);
+
+ v = voice_get_session(data->dest_port);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+
+ return -EINVAL;
+ }
+
+ pr_debug("%s: Payload Length = %d, opcode=%x\n", __func__,
+ data->payload_size, data->opcode);
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("%s: Reset event received in Voice service\n",
+ __func__);
+
+ apr_reset(c->apr_q6_cvs);
+ c->apr_q6_cvs = NULL;
+
+ /* Sub-system restart is applicable to all sessions. */
+ for (i = 0; i < MAX_VOC_SESSIONS; i++)
+ c->voice[i].cvs_handle = 0;
+
+ return 0;
+ }
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ if (data->payload_size) {
+ ptr = data->payload;
+
+ pr_info("%x %x\n", ptr[0], ptr[1]);
+ /*response from CVS */
+ switch (ptr[0]) {
+ case VSS_ISTREAM_CMD_CREATE_PASSIVE_CONTROL_SESSION:
+ case VSS_ISTREAM_CMD_CREATE_FULL_CONTROL_SESSION:
+ if (!ptr[1]) {
+ pr_debug("%s: CVS handle is %d\n",
+ __func__, data->src_port);
+ voice_set_cvs_handle(v, data->src_port);
+ } else
+ pr_err("got NACK for sending \
+ CVS create session \n");
+ v->cvs_state = CMD_STATUS_SUCCESS;
+ wake_up(&v->cvs_wait);
+ break;
+ case VSS_ISTREAM_CMD_SET_MUTE:
+ case VSS_ISTREAM_CMD_SET_MEDIA_TYPE:
+ case VSS_ISTREAM_CMD_VOC_AMR_SET_ENC_RATE:
+ case VSS_ISTREAM_CMD_VOC_AMRWB_SET_ENC_RATE:
+ case VSS_ISTREAM_CMD_SET_ENC_DTX_MODE:
+ case VSS_ISTREAM_CMD_CDMA_SET_ENC_MINMAX_RATE:
+ case APRV2_IBASIC_CMD_DESTROY_SESSION:
+ case VSS_ISTREAM_CMD_REGISTER_CALIBRATION_DATA:
+ case VSS_ISTREAM_CMD_DEREGISTER_CALIBRATION_DATA:
+ case VSS_ICOMMON_CMD_MAP_MEMORY:
+ case VSS_ICOMMON_CMD_UNMAP_MEMORY:
+ case VSS_ICOMMON_CMD_SET_UI_PROPERTY:
+ case VSS_ISTREAM_CMD_START_PLAYBACK:
+ case VSS_ISTREAM_CMD_STOP_PLAYBACK:
+ case VSS_ISTREAM_CMD_START_RECORD:
+ case VSS_ISTREAM_CMD_STOP_RECORD:
+ pr_debug("%s: cmd = 0x%x\n", __func__, ptr[0]);
+ v->cvs_state = CMD_STATUS_SUCCESS;
+ wake_up(&v->cvs_wait);
+ break;
+ case VOICE_CMD_SET_PARAM:
+ rtac_make_voice_callback(RTAC_CVS, ptr,
+ data->payload_size);
+ break;
+ default:
+ pr_debug("%s: cmd = 0x%x\n", __func__, ptr[0]);
+ break;
+ }
+ }
+ } else if (data->opcode == VSS_ISTREAM_EVT_SEND_ENC_BUFFER) {
+ uint32_t *voc_pkt = data->payload;
+ uint32_t pkt_len = data->payload_size;
+
+ if (voc_pkt != NULL && c->mvs_info.ul_cb != NULL) {
+ pr_debug("%s: Media type is 0x%x\n",
+ __func__, voc_pkt[0]);
+
+ /* Remove media ID from payload. */
+ voc_pkt++;
+ pkt_len = pkt_len - 4;
+
+ c->mvs_info.ul_cb((uint8_t *)voc_pkt,
+ pkt_len,
+ c->mvs_info.private_data);
+ } else
+ pr_err("%s: voc_pkt is 0x%x ul_cb is 0x%x\n",
+ __func__, (unsigned int)voc_pkt,
+ (unsigned int) c->mvs_info.ul_cb);
+ } else if (data->opcode == VSS_ISTREAM_EVT_REQUEST_DEC_BUFFER) {
+ struct cvs_send_dec_buf_cmd send_dec_buf;
+ int ret = 0;
+ uint32_t pkt_len = 0;
+
+ if (c->mvs_info.dl_cb != NULL) {
+ send_dec_buf.dec_buf.media_id = c->mvs_info.media_type;
+
+ c->mvs_info.dl_cb(
+ (uint8_t *)&send_dec_buf.dec_buf.packet_data,
+ &pkt_len,
+ c->mvs_info.private_data);
+
+ send_dec_buf.hdr.hdr_field =
+ APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ send_dec_buf.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(send_dec_buf.dec_buf.media_id) + pkt_len);
+ send_dec_buf.hdr.src_port = v->session_id;
+ send_dec_buf.hdr.dest_port = voice_get_cvs_handle(v);
+ send_dec_buf.hdr.token = 0;
+ send_dec_buf.hdr.opcode =
+ VSS_ISTREAM_EVT_SEND_DEC_BUFFER;
+
+ ret = apr_send_pkt(c->apr_q6_cvs,
+ (uint32_t *) &send_dec_buf);
+ if (ret < 0) {
+ pr_err("%s: Error %d sending DEC_BUF\n",
+ __func__, ret);
+ goto fail;
+ }
+ } else
+ pr_debug("%s: dl_cb is NULL\n", __func__);
+ } else if (data->opcode == VSS_ISTREAM_EVT_SEND_DEC_BUFFER) {
+ pr_debug("Send dec buf resp\n");
+ } else if (data->opcode == VOICE_EVT_GET_PARAM_ACK) {
+ rtac_make_voice_callback(RTAC_CVS, data->payload,
+ data->payload_size);
+ } else
+ pr_debug("Unknown opcode 0x%x\n", data->opcode);
+
+fail:
+ return 0;
+}
+
+static int32_t qdsp_cvp_callback(struct apr_client_data *data, void *priv)
+{
+ uint32_t *ptr = NULL;
+ struct common_data *c = NULL;
+ struct voice_data *v = NULL;
+ int i = 0;
+
+ if ((data == NULL) || (priv == NULL)) {
+ pr_err("%s: data or priv is NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ c = priv;
+
+ v = voice_get_session(data->dest_port);
+ if (v == NULL) {
+ pr_err("%s: v is NULL\n", __func__);
+
+ return -EINVAL;
+ }
+
+ pr_debug("%s: Payload Length = %d, opcode=%x\n", __func__,
+ data->payload_size, data->opcode);
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("%s: Reset event received in Voice service\n",
+ __func__);
+
+ apr_reset(c->apr_q6_cvp);
+ c->apr_q6_cvp = NULL;
+
+ /* Sub-system restart is applicable to all sessions. */
+ for (i = 0; i < MAX_VOC_SESSIONS; i++)
+ c->voice[i].cvp_handle = 0;
+
+ return 0;
+ }
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ if (data->payload_size) {
+ ptr = data->payload;
+
+ pr_info("%x %x\n", ptr[0], ptr[1]);
+
+ switch (ptr[0]) {
+ case VSS_IVOCPROC_CMD_CREATE_FULL_CONTROL_SESSION:
+ /*response from CVP */
+ pr_debug("%s: cmd = 0x%x\n", __func__, ptr[0]);
+ if (!ptr[1]) {
+ voice_set_cvp_handle(v, data->src_port);
+ pr_debug("cvphdl=%d\n", data->src_port);
+ } else
+ pr_err("got NACK from CVP create \
+ session response\n");
+ v->cvp_state = CMD_STATUS_SUCCESS;
+ wake_up(&v->cvp_wait);
+ break;
+ case VSS_IVOCPROC_CMD_SET_DEVICE:
+ case VSS_IVOCPROC_CMD_SET_RX_VOLUME_INDEX:
+ case VSS_IVOCPROC_CMD_ENABLE:
+ case VSS_IVOCPROC_CMD_DISABLE:
+ case APRV2_IBASIC_CMD_DESTROY_SESSION:
+ case VSS_IVOCPROC_CMD_REGISTER_VOLUME_CAL_TABLE:
+ case VSS_IVOCPROC_CMD_DEREGISTER_VOLUME_CAL_TABLE:
+ case VSS_IVOCPROC_CMD_REGISTER_CALIBRATION_DATA:
+ case VSS_IVOCPROC_CMD_DEREGISTER_CALIBRATION_DATA:
+ case VSS_ICOMMON_CMD_MAP_MEMORY:
+ case VSS_ICOMMON_CMD_UNMAP_MEMORY:
+ case VSS_IVOCPROC_CMD_SET_MUTE:
+ v->cvp_state = CMD_STATUS_SUCCESS;
+ wake_up(&v->cvp_wait);
+ break;
+ case VOICE_CMD_SET_PARAM:
+ rtac_make_voice_callback(RTAC_CVP, ptr,
+ data->payload_size);
+ break;
+ default:
+ pr_debug("%s: not match cmd = 0x%x\n",
+ __func__, ptr[0]);
+ break;
+ }
+ }
+ } else if (data->opcode == VOICE_EVT_GET_PARAM_ACK) {
+ rtac_make_voice_callback(RTAC_CVP, data->payload,
+ data->payload_size);
+ }
+ return 0;
+}
+
+
+static int __init voice_init(void)
+{
+ int rc = 0, i = 0;
+ int len;
+
+ memset(&common, 0, sizeof(struct common_data));
+
+ /* Allocate memory for VoIP calibration */
+ common.client = msm_ion_client_create(UINT_MAX, "voip_client");
+ if (IS_ERR_OR_NULL((void *)common.client)) {
+ pr_err("%s: ION create client for Voip failed\n", __func__);
+ goto cont;
+ }
+ common.cvp_cal.handle = ion_alloc(common.client, CVP_CAL_SIZE, SZ_4K,
+ ION_HEAP(ION_AUDIO_HEAP_ID));
+ if (IS_ERR_OR_NULL((void *) common.cvp_cal.handle)) {
+ pr_err("%s: ION memory allocation for CVP failed\n",
+ __func__);
+ ion_client_destroy(common.client);
+ goto cont;
+ }
+
+ rc = ion_phys(common.client, common.cvp_cal.handle,
+ (ion_phys_addr_t *)&common.cvp_cal.phy, (size_t *)&len);
+ if (rc) {
+ pr_err("%s: ION Get Physical for cvp failed, rc = %d\n",
+ __func__, rc);
+ ion_free(common.client, common.cvp_cal.handle);
+ ion_client_destroy(common.client);
+ goto cont;
+ }
+
+ common.cvp_cal.buf = ion_map_kernel(common.client,
+ common.cvp_cal.handle, 0);
+ if (IS_ERR_OR_NULL((void *) common.cvp_cal.buf)) {
+ pr_err("%s: ION memory mapping for cvp failed\n", __func__);
+ common.cvp_cal.buf = NULL;
+ ion_free(common.client, common.cvp_cal.handle);
+ ion_client_destroy(common.client);
+ goto cont;
+ }
+ memset((void *)common.cvp_cal.buf, 0, CVP_CAL_SIZE);
+
+ common.cvs_cal.handle = ion_alloc(common.client, CVS_CAL_SIZE, SZ_4K,
+ ION_HEAP(ION_AUDIO_HEAP_ID));
+ if (IS_ERR_OR_NULL((void *) common.cvs_cal.handle)) {
+ pr_err("%s: ION memory allocation for CVS failed\n",
+ __func__);
+ goto cont;
+ }
+
+ rc = ion_phys(common.client, common.cvs_cal.handle,
+ (ion_phys_addr_t *)&common.cvs_cal.phy, (size_t *)&len);
+ if (rc) {
+ pr_err("%s: ION Get Physical for cvs failed, rc = %d\n",
+ __func__, rc);
+ ion_free(common.client, common.cvs_cal.handle);
+ goto cont;
+ }
+
+ common.cvs_cal.buf = ion_map_kernel(common.client,
+ common.cvs_cal.handle, 0);
+ if (IS_ERR_OR_NULL((void *) common.cvs_cal.buf)) {
+ pr_err("%s: ION memory mapping for cvs failed\n", __func__);
+ common.cvs_cal.buf = NULL;
+ ion_free(common.client, common.cvs_cal.handle);
+ goto cont;
+ }
+ memset((void *)common.cvs_cal.buf, 0, CVS_CAL_SIZE);
+cont:
+ /* set default value */
+ common.default_mute_val = 1; /* default is mute */
+ common.default_vol_val = 0;
+ common.default_sample_val = 8000;
+
+ /* Initialize MVS info. */
+ common.mvs_info.network_type = VSS_NETWORK_ID_DEFAULT;
+
+ mutex_init(&common.common_lock);
+
+ for (i = 0; i < MAX_VOC_SESSIONS; i++) {
+ common.voice[i].session_id = SESSION_ID_BASE + i;
+
+ /* initialize dev_rx and dev_tx */
+ common.voice[i].dev_rx.volume = common.default_vol_val;
+ common.voice[i].dev_rx.mute = 0;
+ common.voice[i].dev_tx.mute = common.default_mute_val;
+
+ common.voice[i].dev_tx.port_id = 1;
+ common.voice[i].dev_rx.port_id = 0;
+ common.voice[i].sidetone_gain = 0x512;
+
+ common.voice[i].voc_state = VOC_INIT;
+
+ init_waitqueue_head(&common.voice[i].mvm_wait);
+ init_waitqueue_head(&common.voice[i].cvs_wait);
+ init_waitqueue_head(&common.voice[i].cvp_wait);
+
+ mutex_init(&common.voice[i].lock);
+ }
+
+ return rc;
+}
+
+device_initcall(voice_init);
diff --git a/sound/soc/msm/qdsp6/q6voice.h b/sound/soc/msm/qdsp6/q6voice.h
new file mode 100644
index 0000000..88ab0d5
--- /dev/null
+++ b/sound/soc/msm/qdsp6/q6voice.h
@@ -0,0 +1,997 @@
+/* Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef __QDSP6VOICE_H__
+#define __QDSP6VOICE_H__
+
+#include <mach/qdsp6v2/apr.h>
+#include <linux/ion.h>
+
+#define MAX_VOC_PKT_SIZE 642
+#define SESSION_NAME_LEN 20
+
+#define VOC_REC_UPLINK 0x00
+#define VOC_REC_DOWNLINK 0x01
+#define VOC_REC_BOTH 0x02
+
+struct voice_header {
+ uint32_t id;
+ uint32_t data_len;
+};
+
+struct voice_init {
+ struct voice_header hdr;
+ void *cb_handle;
+};
+
+/* Device information payload structure */
+
+struct device_data {
+ uint32_t volume; /* in index */
+ uint32_t mute;
+ uint32_t sample;
+ uint32_t enabled;
+ uint32_t dev_id;
+ uint32_t port_id;
+};
+
+struct voice_dev_route_state {
+ u16 rx_route_flag;
+ u16 tx_route_flag;
+};
+
+struct voice_rec_route_state {
+ u16 ul_flag;
+ u16 dl_flag;
+};
+
+enum {
+ VOC_INIT = 0,
+ VOC_RUN,
+ VOC_CHANGE,
+ VOC_RELEASE,
+ VOC_STANDBY,
+};
+
+/* Common */
+#define VSS_ICOMMON_CMD_SET_UI_PROPERTY 0x00011103
+/* Set a UI property */
+#define VSS_ICOMMON_CMD_MAP_MEMORY 0x00011025
+#define VSS_ICOMMON_CMD_UNMAP_MEMORY 0x00011026
+/* General shared memory; byte-accessible, 4 kB-aligned. */
+#define VSS_ICOMMON_MAP_MEMORY_SHMEM8_4K_POOL 3
+
+struct vss_icommon_cmd_map_memory_t {
+ uint32_t phys_addr;
+ /* Physical address of a memory region; must be at least
+ * 4 kB aligned.
+ */
+
+ uint32_t mem_size;
+ /* Number of bytes in the region; should be a multiple of 32. */
+
+ uint16_t mem_pool_id;
+ /* Type of memory being provided. The memory ID implicitly defines
+ * the characteristics of the memory. The characteristics might include
+ * alignment type, permissions, etc.
+ * Memory pool ID. Possible values:
+ * 3 -- VSS_ICOMMON_MEM_TYPE_SHMEM8_4K_POOL.
+ */
+} __packed;
+
+struct vss_icommon_cmd_unmap_memory_t {
+ uint32_t phys_addr;
+ /* Physical address of a memory region; must be at least
+ * 4 kB aligned.
+ */
+} __packed;
+
+struct vss_map_memory_cmd {
+ struct apr_hdr hdr;
+ struct vss_icommon_cmd_map_memory_t vss_map_mem;
+} __packed;
+
+struct vss_unmap_memory_cmd {
+ struct apr_hdr hdr;
+ struct vss_icommon_cmd_unmap_memory_t vss_unmap_mem;
+} __packed;
+
+/* TO MVM commands */
+#define VSS_IMVM_CMD_CREATE_PASSIVE_CONTROL_SESSION 0x000110FF
+/**< No payload. Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_IMVM_CMD_SET_POLICY_DUAL_CONTROL 0x00011327
+/*
+ * VSS_IMVM_CMD_SET_POLICY_DUAL_CONTROL
+ * Description: This command is required to let MVM know
+ * who is in control of session.
+ * Payload: Defined by vss_imvm_cmd_set_policy_dual_control_t.
+ * Result: Wait for APRV2_IBASIC_RSP_RESULT response.
+ */
+
+#define VSS_IMVM_CMD_CREATE_FULL_CONTROL_SESSION 0x000110FE
+/* Create a new full control MVM session. */
+
+#define APRV2_IBASIC_CMD_DESTROY_SESSION 0x0001003C
+/**< No payload. Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_IMVM_CMD_ATTACH_STREAM 0x0001123C
+/* Attach a stream to the MVM. */
+
+#define VSS_IMVM_CMD_DETACH_STREAM 0x0001123D
+/* Detach a stream from the MVM. */
+
+#define VSS_IMVM_CMD_ATTACH_VOCPROC 0x0001123E
+/* Attach a vocproc to the MVM. The MVM will symmetrically connect this vocproc
+ * to all the streams currently attached to it.
+ */
+
+#define VSS_IMVM_CMD_DETACH_VOCPROC 0x0001123F
+/* Detach a vocproc from the MVM. The MVM will symmetrically disconnect this
+ * vocproc from all the streams to which it is currently attached.
+*/
+
+#define VSS_IMVM_CMD_START_VOICE 0x00011190
+/*
+ * Start Voice call command.
+ * Wait for APRV2_IBASIC_RSP_RESULT response.
+ * No pay load.
+ */
+
+#define VSS_IMVM_CMD_STANDBY_VOICE 0x00011191
+/* No payload. Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_IMVM_CMD_STOP_VOICE 0x00011192
+/**< No payload. Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_ISTREAM_CMD_ATTACH_VOCPROC 0x000110F8
+/**< Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_ISTREAM_CMD_DETACH_VOCPROC 0x000110F9
+/**< Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+
+#define VSS_ISTREAM_CMD_SET_TTY_MODE 0x00011196
+/**< Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_ICOMMON_CMD_SET_NETWORK 0x0001119C
+/* Set the network type. */
+
+#define VSS_ICOMMON_CMD_SET_VOICE_TIMING 0x000111E0
+/* Set the voice timing parameters. */
+
+#define VSS_IWIDEVOICE_CMD_SET_WIDEVOICE 0x00011243
+/* Enable/disable WideVoice */
+
+enum msm_audio_voc_rate {
+ VOC_0_RATE, /* Blank frame */
+ VOC_8_RATE, /* 1/8 rate */
+ VOC_4_RATE, /* 1/4 rate */
+ VOC_2_RATE, /* 1/2 rate */
+ VOC_1_RATE /* Full rate */
+};
+
+struct vss_istream_cmd_set_tty_mode_t {
+ uint32_t mode;
+ /**<
+ * TTY mode.
+ *
+ * 0 : TTY disabled
+ * 1 : HCO
+ * 2 : VCO
+ * 3 : FULL
+ */
+} __packed;
+
+struct vss_istream_cmd_attach_vocproc_t {
+ uint16_t handle;
+ /**< Handle of vocproc being attached. */
+} __packed;
+
+struct vss_istream_cmd_detach_vocproc_t {
+ uint16_t handle;
+ /**< Handle of vocproc being detached. */
+} __packed;
+
+struct vss_imvm_cmd_attach_stream_t {
+ uint16_t handle;
+ /* The stream handle to attach. */
+} __packed;
+
+struct vss_imvm_cmd_detach_stream_t {
+ uint16_t handle;
+ /* The stream handle to detach. */
+} __packed;
+
+struct vss_icommon_cmd_set_network_t {
+ uint32_t network_id;
+ /* Network ID. (Refer to VSS_NETWORK_ID_XXX). */
+} __packed;
+
+struct vss_icommon_cmd_set_voice_timing_t {
+ uint16_t mode;
+ /*
+ * The vocoder frame synchronization mode.
+ *
+ * 0 : No frame sync.
+ * 1 : Hard VFR (20ms Vocoder Frame Reference interrupt).
+ */
+ uint16_t enc_offset;
+ /*
+ * The offset in microseconds from the VFR to deliver a Tx vocoder
+ * packet. The offset should be less than 20000us.
+ */
+ uint16_t dec_req_offset;
+ /*
+ * The offset in microseconds from the VFR to request for an Rx vocoder
+ * packet. The offset should be less than 20000us.
+ */
+ uint16_t dec_offset;
+ /*
+ * The offset in microseconds from the VFR to indicate the deadline to
+ * receive an Rx vocoder packet. The offset should be less than 20000us.
+ * Rx vocoder packets received after this deadline are not guaranteed to
+ * be processed.
+ */
+} __packed;
+
+struct vss_imvm_cmd_create_control_session_t {
+ char name[SESSION_NAME_LEN];
+ /*
+ * A variable-sized stream name.
+ *
+ * The stream name size is the payload size minus the size of the other
+ * fields.
+ */
+} __packed;
+
+
+struct vss_imvm_cmd_set_policy_dual_control_t {
+ bool enable_flag;
+ /* Set to TRUE to enable modem state machine control */
+} __packed;
+
+struct vss_iwidevoice_cmd_set_widevoice_t {
+ uint32_t enable;
+ /* WideVoice enable/disable; possible values:
+ * - 0 -- WideVoice disabled
+ * - 1 -- WideVoice enabled
+ */
+} __packed;
+
+struct mvm_attach_vocproc_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_attach_vocproc_t mvm_attach_cvp_handle;
+} __packed;
+
+struct mvm_detach_vocproc_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_detach_vocproc_t mvm_detach_cvp_handle;
+} __packed;
+
+struct mvm_create_ctl_session_cmd {
+ struct apr_hdr hdr;
+ struct vss_imvm_cmd_create_control_session_t mvm_session;
+} __packed;
+
+struct mvm_modem_dual_control_session_cmd {
+ struct apr_hdr hdr;
+ struct vss_imvm_cmd_set_policy_dual_control_t voice_ctl;
+} __packed;
+
+struct mvm_set_tty_mode_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_set_tty_mode_t tty_mode;
+} __packed;
+
+struct mvm_attach_stream_cmd {
+ struct apr_hdr hdr;
+ struct vss_imvm_cmd_attach_stream_t attach_stream;
+} __packed;
+
+struct mvm_detach_stream_cmd {
+ struct apr_hdr hdr;
+ struct vss_imvm_cmd_detach_stream_t detach_stream;
+} __packed;
+
+struct mvm_set_network_cmd {
+ struct apr_hdr hdr;
+ struct vss_icommon_cmd_set_network_t network;
+} __packed;
+
+struct mvm_set_voice_timing_cmd {
+ struct apr_hdr hdr;
+ struct vss_icommon_cmd_set_voice_timing_t timing;
+} __packed;
+
+struct mvm_set_widevoice_enable_cmd {
+ struct apr_hdr hdr;
+ struct vss_iwidevoice_cmd_set_widevoice_t vss_set_wv;
+} __packed;
+
+/* TO CVS commands */
+#define VSS_ISTREAM_CMD_CREATE_PASSIVE_CONTROL_SESSION 0x00011140
+/**< Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_ISTREAM_CMD_CREATE_FULL_CONTROL_SESSION 0x000110F7
+/* Create a new full control stream session. */
+
+#define APRV2_IBASIC_CMD_DESTROY_SESSION 0x0001003C
+
+#define VSS_ISTREAM_CMD_SET_MUTE 0x00011022
+
+#define VSS_ISTREAM_CMD_REGISTER_CALIBRATION_DATA 0x00011279
+
+#define VSS_ISTREAM_CMD_DEREGISTER_CALIBRATION_DATA 0x0001127A
+
+#define VSS_ISTREAM_CMD_SET_MEDIA_TYPE 0x00011186
+/* Set media type on the stream. */
+
+#define VSS_ISTREAM_EVT_SEND_ENC_BUFFER 0x00011015
+/* Event sent by the stream to its client to provide an encoded packet. */
+
+#define VSS_ISTREAM_EVT_REQUEST_DEC_BUFFER 0x00011017
+/* Event sent by the stream to its client requesting for a decoder packet.
+ * The client should respond with a VSS_ISTREAM_EVT_SEND_DEC_BUFFER event.
+ */
+
+#define VSS_ISTREAM_EVT_SEND_DEC_BUFFER 0x00011016
+/* Event sent by the client to the stream in response to a
+ * VSS_ISTREAM_EVT_REQUEST_DEC_BUFFER event, providing a decoder packet.
+ */
+
+#define VSS_ISTREAM_CMD_VOC_AMR_SET_ENC_RATE 0x0001113E
+/* Set AMR encoder rate. */
+
+#define VSS_ISTREAM_CMD_VOC_AMRWB_SET_ENC_RATE 0x0001113F
+/* Set AMR-WB encoder rate. */
+
+#define VSS_ISTREAM_CMD_CDMA_SET_ENC_MINMAX_RATE 0x00011019
+/* Set encoder minimum and maximum rate. */
+
+#define VSS_ISTREAM_CMD_SET_ENC_DTX_MODE 0x0001101D
+/* Set encoder DTX mode. */
+
+#define MODULE_ID_VOICE_MODULE_FENS 0x00010EEB
+#define MODULE_ID_VOICE_MODULE_ST 0x00010EE3
+#define VOICE_PARAM_MOD_ENABLE 0x00010E00
+#define MOD_ENABLE_PARAM_LEN 4
+
+#define VSS_ISTREAM_CMD_START_PLAYBACK 0x00011238
+/* Start in-call music delivery on the Tx voice path. */
+
+#define VSS_ISTREAM_CMD_STOP_PLAYBACK 0x00011239
+/* Stop the in-call music delivery on the Tx voice path. */
+
+#define VSS_ISTREAM_CMD_START_RECORD 0x00011236
+/* Start in-call conversation recording. */
+#define VSS_ISTREAM_CMD_STOP_RECORD 0x00011237
+/* Stop in-call conversation recording. */
+
+#define VSS_TAP_POINT_NONE 0x00010F78
+/* Indicates no tapping for specified path. */
+
+#define VSS_TAP_POINT_STREAM_END 0x00010F79
+/* Indicates that specified path should be tapped at the end of the stream. */
+
+struct vss_istream_cmd_start_record_t {
+ uint32_t rx_tap_point;
+ /* Tap point to use on the Rx path. Supported values are:
+ * VSS_TAP_POINT_NONE : Do not record Rx path.
+ * VSS_TAP_POINT_STREAM_END : Rx tap point is at the end of the stream.
+ */
+ uint32_t tx_tap_point;
+ /* Tap point to use on the Tx path. Supported values are:
+ * VSS_TAP_POINT_NONE : Do not record tx path.
+ * VSS_TAP_POINT_STREAM_END : Tx tap point is at the end of the stream.
+ */
+} __packed;
+
+struct vss_istream_cmd_create_passive_control_session_t {
+ char name[SESSION_NAME_LEN];
+ /**<
+ * A variable-sized stream name.
+ *
+ * The stream name size is the payload size minus the size of the other
+ * fields.
+ */
+} __packed;
+
+struct vss_istream_cmd_set_mute_t {
+ uint16_t direction;
+ /**<
+ * 0 : TX only
+ * 1 : RX only
+ * 2 : TX and Rx
+ */
+ uint16_t mute_flag;
+ /**<
+ * Mute, un-mute.
+ *
+ * 0 : Silence disable
+ * 1 : Silence enable
+ * 2 : CNG enable. Applicable to TX only. If set on RX behavior
+ * will be the same as 1
+ */
+} __packed;
+
+struct vss_istream_cmd_create_full_control_session_t {
+ uint16_t direction;
+ /*
+ * Stream direction.
+ *
+ * 0 : TX only
+ * 1 : RX only
+ * 2 : TX and RX
+ * 3 : TX and RX loopback
+ */
+ uint32_t enc_media_type;
+ /* Tx vocoder type. (Refer to VSS_MEDIA_ID_XXX). */
+ uint32_t dec_media_type;
+ /* Rx vocoder type. (Refer to VSS_MEDIA_ID_XXX). */
+ uint32_t network_id;
+ /* Network ID. (Refer to VSS_NETWORK_ID_XXX). */
+ char name[SESSION_NAME_LEN];
+ /*
+ * A variable-sized stream name.
+ *
+ * The stream name size is the payload size minus the size of the other
+ * fields.
+ */
+} __packed;
+
+struct vss_istream_cmd_set_media_type_t {
+ uint32_t rx_media_id;
+ /* Set the Rx vocoder type. (Refer to VSS_MEDIA_ID_XXX). */
+ uint32_t tx_media_id;
+ /* Set the Tx vocoder type. (Refer to VSS_MEDIA_ID_XXX). */
+} __packed;
+
+struct vss_istream_evt_send_enc_buffer_t {
+ uint32_t media_id;
+ /* Media ID of the packet. */
+ uint8_t packet_data[MAX_VOC_PKT_SIZE];
+ /* Packet data buffer. */
+} __packed;
+
+struct vss_istream_evt_send_dec_buffer_t {
+ uint32_t media_id;
+ /* Media ID of the packet. */
+ uint8_t packet_data[MAX_VOC_PKT_SIZE];
+ /* Packet data. */
+} __packed;
+
+struct vss_istream_cmd_voc_amr_set_enc_rate_t {
+ uint32_t mode;
+ /* Set the AMR encoder rate.
+ *
+ * 0x00000000 : 4.75 kbps
+ * 0x00000001 : 5.15 kbps
+ * 0x00000002 : 5.90 kbps
+ * 0x00000003 : 6.70 kbps
+ * 0x00000004 : 7.40 kbps
+ * 0x00000005 : 7.95 kbps
+ * 0x00000006 : 10.2 kbps
+ * 0x00000007 : 12.2 kbps
+ */
+} __packed;
+
+struct vss_istream_cmd_voc_amrwb_set_enc_rate_t {
+ uint32_t mode;
+ /* Set the AMR-WB encoder rate.
+ *
+ * 0x00000000 : 6.60 kbps
+ * 0x00000001 : 8.85 kbps
+ * 0x00000002 : 12.65 kbps
+ * 0x00000003 : 14.25 kbps
+ * 0x00000004 : 15.85 kbps
+ * 0x00000005 : 18.25 kbps
+ * 0x00000006 : 19.85 kbps
+ * 0x00000007 : 23.05 kbps
+ * 0x00000008 : 23.85 kbps
+ */
+} __packed;
+
+struct vss_istream_cmd_cdma_set_enc_minmax_rate_t {
+ uint16_t min_rate;
+ /* Set the lower bound encoder rate.
+ *
+ * 0x0000 : Blank frame
+ * 0x0001 : Eighth rate
+ * 0x0002 : Quarter rate
+ * 0x0003 : Half rate
+ * 0x0004 : Full rate
+ */
+ uint16_t max_rate;
+ /* Set the upper bound encoder rate.
+ *
+ * 0x0000 : Blank frame
+ * 0x0001 : Eighth rate
+ * 0x0002 : Quarter rate
+ * 0x0003 : Half rate
+ * 0x0004 : Full rate
+ */
+} __packed;
+
+struct vss_istream_cmd_set_enc_dtx_mode_t {
+ uint32_t enable;
+ /* Toggle DTX on or off.
+ *
+ * 0 : Disables DTX
+ * 1 : Enables DTX
+ */
+} __packed;
+
+struct vss_istream_cmd_register_calibration_data_t {
+ uint32_t phys_addr;
+ /* Phsical address to be registered with stream. The calibration data
+ * is stored at this address.
+ */
+ uint32_t mem_size;
+ /* Size of the calibration data in bytes. */
+};
+
+struct vss_icommon_cmd_set_ui_property_enable_t {
+ uint32_t module_id;
+ /* Unique ID of the module. */
+ uint32_t param_id;
+ /* Unique ID of the parameter. */
+ uint16_t param_size;
+ /* Size of the parameter in bytes: MOD_ENABLE_PARAM_LEN */
+ uint16_t reserved;
+ /* Reserved; set to 0. */
+ uint16_t enable;
+ uint16_t reserved_field;
+ /* Reserved, set to 0. */
+};
+
+struct cvs_create_passive_ctl_session_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_create_passive_control_session_t cvs_session;
+} __packed;
+
+struct cvs_create_full_ctl_session_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_create_full_control_session_t cvs_session;
+} __packed;
+
+struct cvs_destroy_session_cmd {
+ struct apr_hdr hdr;
+} __packed;
+
+struct cvs_set_mute_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_set_mute_t cvs_set_mute;
+} __packed;
+
+struct cvs_set_media_type_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_set_media_type_t media_type;
+} __packed;
+
+struct cvs_send_dec_buf_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_evt_send_dec_buffer_t dec_buf;
+} __packed;
+
+struct cvs_set_amr_enc_rate_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_voc_amr_set_enc_rate_t amr_rate;
+} __packed;
+
+struct cvs_set_amrwb_enc_rate_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_voc_amrwb_set_enc_rate_t amrwb_rate;
+} __packed;
+
+struct cvs_set_cdma_enc_minmax_rate_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_cdma_set_enc_minmax_rate_t cdma_rate;
+} __packed;
+
+struct cvs_set_enc_dtx_mode_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_set_enc_dtx_mode_t dtx_mode;
+} __packed;
+
+struct cvs_register_cal_data_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_register_calibration_data_t cvs_cal_data;
+} __packed;
+
+struct cvs_deregister_cal_data_cmd {
+ struct apr_hdr hdr;
+} __packed;
+
+struct cvs_set_pp_enable_cmd {
+ struct apr_hdr hdr;
+ struct vss_icommon_cmd_set_ui_property_enable_t vss_set_pp;
+} __packed;
+struct cvs_start_record_cmd {
+ struct apr_hdr hdr;
+ struct vss_istream_cmd_start_record_t rec_mode;
+} __packed;
+
+/* TO CVP commands */
+
+#define VSS_IVOCPROC_CMD_CREATE_FULL_CONTROL_SESSION 0x000100C3
+/**< Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define APRV2_IBASIC_CMD_DESTROY_SESSION 0x0001003C
+
+#define VSS_IVOCPROC_CMD_SET_DEVICE 0x000100C4
+
+#define VSS_IVOCPROC_CMD_SET_VP3_DATA 0x000110EB
+
+#define VSS_IVOCPROC_CMD_SET_RX_VOLUME_INDEX 0x000110EE
+
+#define VSS_IVOCPROC_CMD_ENABLE 0x000100C6
+/**< No payload. Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_IVOCPROC_CMD_DISABLE 0x000110E1
+/**< No payload. Wait for APRV2_IBASIC_RSP_RESULT response. */
+
+#define VSS_IVOCPROC_CMD_REGISTER_CALIBRATION_DATA 0x00011275
+#define VSS_IVOCPROC_CMD_DEREGISTER_CALIBRATION_DATA 0x00011276
+
+#define VSS_IVOCPROC_CMD_REGISTER_VOLUME_CAL_TABLE 0x00011277
+#define VSS_IVOCPROC_CMD_DEREGISTER_VOLUME_CAL_TABLE 0x00011278
+
+#define VSS_IVOCPROC_TOPOLOGY_ID_NONE 0x00010F70
+#define VSS_IVOCPROC_TOPOLOGY_ID_TX_SM_ECNS 0x00010F71
+#define VSS_IVOCPROC_TOPOLOGY_ID_TX_DM_FLUENCE 0x00010F72
+
+#define VSS_IVOCPROC_TOPOLOGY_ID_RX_DEFAULT 0x00010F77
+
+/* Newtwork IDs */
+#define VSS_NETWORK_ID_DEFAULT 0x00010037
+#define VSS_NETWORK_ID_VOIP_NB 0x00011240
+#define VSS_NETWORK_ID_VOIP_WB 0x00011241
+#define VSS_NETWORK_ID_VOIP_WV 0x00011242
+
+/* Media types */
+#define VSS_MEDIA_ID_EVRC_MODEM 0x00010FC2
+/* 80-VF690-47 CDMA enhanced variable rate vocoder modem format. */
+#define VSS_MEDIA_ID_AMR_NB_MODEM 0x00010FC6
+/* 80-VF690-47 UMTS AMR-NB vocoder modem format. */
+#define VSS_MEDIA_ID_AMR_WB_MODEM 0x00010FC7
+/* 80-VF690-47 UMTS AMR-WB vocoder modem format. */
+#define VSS_MEDIA_ID_PCM_NB 0x00010FCB
+#define VSS_MEDIA_ID_PCM_WB 0x00010FCC
+/* Linear PCM (16-bit, little-endian). */
+#define VSS_MEDIA_ID_G711_ALAW 0x00010FCD
+/* G.711 a-law (contains two 10ms vocoder frames). */
+#define VSS_MEDIA_ID_G711_MULAW 0x00010FCE
+/* G.711 mu-law (contains two 10ms vocoder frames). */
+#define VSS_MEDIA_ID_G729 0x00010FD0
+/* G.729AB (contains two 10ms vocoder frames. */
+#define VSS_MEDIA_ID_4GV_NB_MODEM 0x00010FC3
+/*CDMA EVRC-B vocoder modem format */
+#define VSS_MEDIA_ID_4GV_WB_MODEM 0x00010FC4
+/*CDMA EVRC-WB vocoder modem format */
+
+#define VSS_IVOCPROC_CMD_SET_MUTE 0x000110EF
+
+#define VOICE_CMD_SET_PARAM 0x00011006
+#define VOICE_CMD_GET_PARAM 0x00011007
+#define VOICE_EVT_GET_PARAM_ACK 0x00011008
+
+struct vss_ivocproc_cmd_create_full_control_session_t {
+ uint16_t direction;
+ /*
+ * stream direction.
+ * 0 : TX only
+ * 1 : RX only
+ * 2 : TX and RX
+ */
+ uint32_t tx_port_id;
+ /*
+ * TX device port ID which vocproc will connect to. If not supplying a
+ * port ID set to VSS_IVOCPROC_PORT_ID_NONE.
+ */
+ uint32_t tx_topology_id;
+ /*
+ * Tx leg topology ID. If not supplying a topology ID set to
+ * VSS_IVOCPROC_TOPOLOGY_ID_NONE.
+ */
+ uint32_t rx_port_id;
+ /*
+ * RX device port ID which vocproc will connect to. If not supplying a
+ * port ID set to VSS_IVOCPROC_PORT_ID_NONE.
+ */
+ uint32_t rx_topology_id;
+ /*
+ * Rx leg topology ID. If not supplying a topology ID set to
+ * VSS_IVOCPROC_TOPOLOGY_ID_NONE.
+ */
+ int32_t network_id;
+ /*
+ * Network ID. (Refer to VSS_NETWORK_ID_XXX). If not supplying a network
+ * ID set to VSS_NETWORK_ID_DEFAULT.
+ */
+} __packed;
+
+struct vss_ivocproc_cmd_set_volume_index_t {
+ uint16_t vol_index;
+ /**<
+ * Volume index utilized by the vocproc to index into the volume table
+ * provided in VSS_IVOCPROC_CMD_CACHE_VOLUME_CALIBRATION_TABLE and set
+ * volume on the VDSP.
+ */
+} __packed;
+
+struct vss_ivocproc_cmd_set_device_t {
+ uint32_t tx_port_id;
+ /**<
+ * TX device port ID which vocproc will connect to.
+ * VSS_IVOCPROC_PORT_ID_NONE means vocproc will not connect to any port.
+ */
+ uint32_t tx_topology_id;
+ /**<
+ * TX leg topology ID.
+ * VSS_IVOCPROC_TOPOLOGY_ID_NONE means vocproc does not contain any
+ * pre/post-processing blocks and is pass-through.
+ */
+ int32_t rx_port_id;
+ /**<
+ * RX device port ID which vocproc will connect to.
+ * VSS_IVOCPROC_PORT_ID_NONE means vocproc will not connect to any port.
+ */
+ uint32_t rx_topology_id;
+ /**<
+ * RX leg topology ID.
+ * VSS_IVOCPROC_TOPOLOGY_ID_NONE means vocproc does not contain any
+ * pre/post-processing blocks and is pass-through.
+ */
+} __packed;
+
+struct vss_ivocproc_cmd_register_calibration_data_t {
+ uint32_t phys_addr;
+ /* Phsical address to be registered with vocproc. Calibration data
+ * is stored at this address.
+ */
+ uint32_t mem_size;
+ /* Size of the calibration data in bytes. */
+} __packed;
+
+struct vss_ivocproc_cmd_register_volume_cal_table_t {
+ uint32_t phys_addr;
+ /* Phsical address to be registered with the vocproc. The volume
+ * calibration table is stored at this location.
+ */
+
+ uint32_t mem_size;
+ /* Size of the volume calibration table in bytes. */
+} __packed;
+
+struct vss_ivocproc_cmd_set_mute_t {
+ uint16_t direction;
+ /*
+ * 0 : TX only.
+ * 1 : RX only.
+ * 2 : TX and Rx.
+ */
+ uint16_t mute_flag;
+ /*
+ * Mute, un-mute.
+ *
+ * 0 : Disable.
+ * 1 : Enable.
+ */
+} __packed;
+
+struct cvp_create_full_ctl_session_cmd {
+ struct apr_hdr hdr;
+ struct vss_ivocproc_cmd_create_full_control_session_t cvp_session;
+} __packed;
+
+struct cvp_command {
+ struct apr_hdr hdr;
+} __packed;
+
+struct cvp_set_device_cmd {
+ struct apr_hdr hdr;
+ struct vss_ivocproc_cmd_set_device_t cvp_set_device;
+} __packed;
+
+struct cvp_set_vp3_data_cmd {
+ struct apr_hdr hdr;
+} __packed;
+
+struct cvp_set_rx_volume_index_cmd {
+ struct apr_hdr hdr;
+ struct vss_ivocproc_cmd_set_volume_index_t cvp_set_vol_idx;
+} __packed;
+
+struct cvp_register_cal_data_cmd {
+ struct apr_hdr hdr;
+ struct vss_ivocproc_cmd_register_calibration_data_t cvp_cal_data;
+} __packed;
+
+struct cvp_deregister_cal_data_cmd {
+ struct apr_hdr hdr;
+} __packed;
+
+struct cvp_register_vol_cal_table_cmd {
+ struct apr_hdr hdr;
+ struct vss_ivocproc_cmd_register_volume_cal_table_t cvp_vol_cal_tbl;
+} __packed;
+
+struct cvp_deregister_vol_cal_table_cmd {
+ struct apr_hdr hdr;
+} __packed;
+
+struct cvp_set_mute_cmd {
+ struct apr_hdr hdr;
+ struct vss_ivocproc_cmd_set_mute_t cvp_set_mute;
+} __packed;
+
+/* CB for up-link packets. */
+typedef void (*ul_cb_fn)(uint8_t *voc_pkt,
+ uint32_t pkt_len,
+ void *private_data);
+
+/* CB for down-link packets. */
+typedef void (*dl_cb_fn)(uint8_t *voc_pkt,
+ uint32_t *pkt_len,
+ void *private_data);
+
+
+struct mvs_driver_info {
+ uint32_t media_type;
+ uint32_t rate;
+ uint32_t network_type;
+ uint32_t dtx_mode;
+ ul_cb_fn ul_cb;
+ dl_cb_fn dl_cb;
+ void *private_data;
+};
+
+struct incall_rec_info {
+ uint32_t rec_enable;
+ uint32_t rec_mode;
+ uint32_t recording;
+};
+
+struct incall_music_info {
+ uint32_t play_enable;
+ uint32_t playing;
+ int count;
+ int force;
+};
+
+struct voice_data {
+ int voc_state;/*INIT, CHANGE, RELEASE, RUN */
+
+ wait_queue_head_t mvm_wait;
+ wait_queue_head_t cvs_wait;
+ wait_queue_head_t cvp_wait;
+
+ /* cache the values related to Rx and Tx */
+ struct device_data dev_rx;
+ struct device_data dev_tx;
+
+ u32 mvm_state;
+ u32 cvs_state;
+ u32 cvp_state;
+
+ /* Handle to MVM in the Q6 */
+ u16 mvm_handle;
+ /* Handle to CVS in the Q6 */
+ u16 cvs_handle;
+ /* Handle to CVP in the Q6 */
+ u16 cvp_handle;
+
+ struct mutex lock;
+
+ uint16_t sidetone_gain;
+ uint8_t tty_mode;
+ /* widevoice enable value */
+ uint8_t wv_enable;
+ /* slowtalk enable value */
+ uint32_t st_enable;
+ /* FENC enable value */
+ uint32_t fens_enable;
+
+ struct voice_dev_route_state voc_route_state;
+
+ u16 session_id;
+
+ struct incall_rec_info rec_info;
+
+ struct incall_music_info music_info;
+
+ struct voice_rec_route_state rec_route_state;
+};
+
+struct cal_mem {
+ struct ion_handle *handle;
+ uint32_t phy;
+ void *buf;
+};
+
+#define MAX_VOC_SESSIONS 3
+#define SESSION_ID_BASE 0xFFF0
+
+struct common_data {
+ /* these default values are for all devices */
+ uint32_t default_mute_val;
+ uint32_t default_vol_val;
+ uint32_t default_sample_val;
+
+ /* APR to MVM in the Q6 */
+ void *apr_q6_mvm;
+ /* APR to CVS in the Q6 */
+ void *apr_q6_cvs;
+ /* APR to CVP in the Q6 */
+ void *apr_q6_cvp;
+
+ struct ion_client *client;
+ struct cal_mem cvp_cal;
+ struct cal_mem cvs_cal;
+
+ struct mutex common_lock;
+
+ struct mvs_driver_info mvs_info;
+
+ struct voice_data voice[MAX_VOC_SESSIONS];
+};
+
+void voc_register_mvs_cb(ul_cb_fn ul_cb,
+ dl_cb_fn dl_cb,
+ void *private_data);
+
+void voc_config_vocoder(uint32_t media_type,
+ uint32_t rate,
+ uint32_t network_type,
+ uint32_t dtx_mode);
+
+enum {
+ DEV_RX = 0,
+ DEV_TX,
+};
+
+enum {
+ RX_PATH = 0,
+ TX_PATH,
+};
+
+/* called by alsa driver */
+int voc_set_pp_enable(uint16_t session_id, uint32_t module_id, uint32_t enable);
+int voc_get_pp_enable(uint16_t session_id, uint32_t module_id);
+int voc_set_widevoice_enable(uint16_t session_id, uint32_t wv_enable);
+uint32_t voc_get_widevoice_enable(uint16_t session_id);
+uint8_t voc_get_tty_mode(uint16_t session_id);
+int voc_set_tty_mode(uint16_t session_id, uint8_t tty_mode);
+int voc_start_voice_call(uint16_t session_id);
+int voc_standby_voice_call(uint16_t session_id);
+int voc_resume_voice_call(uint16_t session_id);
+int voc_end_voice_call(uint16_t session_id);
+int voc_set_rxtx_port(uint16_t session_id,
+ uint32_t dev_port_id,
+ uint32_t dev_type);
+int voc_set_rx_vol_index(uint16_t session_id, uint32_t dir, uint32_t voc_idx);
+int voc_set_tx_mute(uint16_t session_id, uint32_t dir, uint32_t mute);
+int voc_set_rx_device_mute(uint16_t session_id, uint32_t mute);
+int voc_get_rx_device_mute(uint16_t session_id);
+int voc_disable_cvp(uint16_t session_id);
+int voc_enable_cvp(uint16_t session_id);
+int voc_set_route_flag(uint16_t session_id, uint8_t path_dir, uint8_t set);
+uint8_t voc_get_route_flag(uint16_t session_id, uint8_t path_dir);
+
+#define VOICE_SESSION_NAME "Voice session"
+#define VOIP_SESSION_NAME "VoIP session"
+#define VOLTE_SESSION_NAME "VoLTE session"
+uint16_t voc_get_session_id(char *name);
+
+int voc_start_playback(uint32_t set);
+int voc_start_record(uint32_t port_id, uint32_t set);
+#endif
diff --git a/sound/soc/msm/qdsp6v2/Makefile b/sound/soc/msm/qdsp6v2/Makefile
new file mode 100644
index 0000000..6f765d1
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/Makefile
@@ -0,0 +1,4 @@
+snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o msm-compr-q6-v2.o msm-multi-ch-pcm-q6-v2.o
+snd-soc-qdsp6v2-objs += msm-pcm-lpa-v2.o msm-pcm-afe-v2.o
+obj-$(CONFIG_SND_SOC_QDSP6V2) += snd-soc-qdsp6v2.o
+obj-y += q6adm.o q6afe.o q6asm.o q6audio-v2.o
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
new file mode 100644
index 0000000..daba79d
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
@@ -0,0 +1,666 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+
+#include "msm-compr-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+
+struct snd_msm {
+ struct msm_audio *prtd;
+ unsigned volume;
+};
+static struct snd_msm compressed_audio = {NULL, 0x2000} ;
+
+static struct audio_locks the_locks;
+
+static struct snd_pcm_hardware msm_compr_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 1200 * 1024 * 2,
+ .period_bytes_min = 4800,
+ .period_bytes_max = 1200 * 1024,
+ .periods_min = 2,
+ .periods_max = 512,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void compr_event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct compr_audio *compr = priv;
+ struct msm_audio *prtd = &compr->prtd;
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_aio_write_param param;
+ struct audio_buffer *buf = NULL;
+ int i = 0;
+
+ pr_debug("%s opcode =%08x\n", __func__, opcode);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2: {
+ uint32_t *ptrmem = (uint32_t *)¶m;
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start)) {
+ atomic_set(&prtd->pending_buffer, 1);
+ break;
+ } else
+ atomic_set(&prtd->pending_buffer, 0);
+
+ if (runtime->status->hw_ptr >= runtime->control->appl_ptr)
+ break;
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ pr_debug("%s:writing buffer[%d] from 0x%08x\n",
+ __func__, prtd->out_head,
+ ((unsigned int)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count)));
+
+ param.paddr = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
+ i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1) & (runtime->periods - 1);
+ break;
+ }
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN_V2: {
+ if (!atomic_read(&prtd->pending_buffer))
+ break;
+ pr_debug("%s:writing %d bytes"
+ " of buffer[%d] to dsp\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s:writing buffer[%d] from 0x%08x\n",
+ __func__, prtd->out_head,
+ ((unsigned int)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count)));
+ param.paddr = (unsigned long)buf[prtd->out_head].phys;
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[prtd->out_head].phys;
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1)
+ & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ }
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ pr_debug("ASM_STREAM_CMD_FLUSH\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_compr_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct asm_aac_cfg aac_cfg;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ prtd->out_head = 0;
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ if (prtd->enabled)
+ return 0;
+
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_MP3:
+ /* No media format block for mp3 */
+ break;
+ case SND_AUDIOCODEC_AAC:
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
+ aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
+ aac_cfg.format = 0x03;
+ aac_cfg.ch_cfg = runtime->channels;
+ aac_cfg.sample_rate = runtime->rate;
+ ret = q6asm_media_format_block_aac(prtd->audio_client,
+ &aac_cfg);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->pcm_irq_pos = 0;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->start, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static void populate_codec_list(struct compr_audio *compr,
+ struct snd_pcm_runtime *runtime)
+{
+ pr_debug("%s\n", __func__);
+ /* MP3 Block */
+ compr->info.compr_cap.num_codecs = 1;
+ compr->info.compr_cap.min_fragment_size = runtime->hw.period_bytes_min;
+ compr->info.compr_cap.max_fragment_size = runtime->hw.period_bytes_max;
+ compr->info.compr_cap.min_fragments = runtime->hw.periods_min;
+ compr->info.compr_cap.max_fragments = runtime->hw.periods_max;
+ compr->info.compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
+ compr->info.compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
+ /* Add new codecs here */
+}
+
+static int msm_compr_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct compr_audio *compr;
+ struct msm_audio *prtd;
+ int ret = 0;
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return -EINVAL;
+
+ pr_debug("%s\n", __func__);
+ compr = kzalloc(sizeof(struct compr_audio), GFP_KERNEL);
+ if (compr == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd = &compr->prtd;
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)compr_event_handler, compr);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ runtime->hw = msm_compr_hardware_playback;
+
+ pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ atomic_set(&prtd->pending_buffer, 1);
+ compr->codec = FORMAT_MP3;
+ populate_codec_list(compr, runtime);
+ runtime->private_data = compr;
+ compressed_audio.prtd = &compr->prtd;
+ ret = compressed_set_volume(compressed_audio.volume);
+ if (ret < 0)
+ pr_err("%s : Set Volume failed : %d", __func__, ret);
+
+ ret = q6asm_set_softpause(compressed_audio.prtd->audio_client,
+ &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(compressed_audio.prtd->audio_client,
+ &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ return 0;
+}
+
+int compressed_set_volume(unsigned volume)
+{
+ int rc = 0;
+ if (compressed_audio.prtd && compressed_audio.prtd->audio_client) {
+ rc = q6asm_set_volume(compressed_audio.prtd->audio_client,
+ volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed"
+ " rc=%d\n", __func__, rc);
+ }
+ }
+ compressed_audio.volume = volume;
+ return rc;
+}
+
+static int msm_compr_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ int dir = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ atomic_set(&prtd->pending_buffer, 0);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ compressed_audio.prtd = NULL;
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_compr_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_compr_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = EINVAL;
+ return ret;
+}
+static int msm_compr_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_compr_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = EINVAL;
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_compr_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_compr_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+ return result;
+}
+
+static int msm_compr_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ pr_debug("%s\n", __func__);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ return -EINVAL;
+
+ ret = q6asm_open_write(prtd->audio_client, compr->codec);
+ if (ret < 0) {
+ pr_err("%s: Session out open failed\n", __func__);
+ return -ENOMEM;
+ }
+ ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed "
+ "rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static int msm_compr_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int rc = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ uint64_t timestamp;
+ uint64_t temp;
+
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP: {
+ struct snd_compr_tstamp tstamp;
+ pr_debug("SNDRV_COMPRESS_TSTAMP\n");
+
+ memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
+ timestamp = q6asm_get_session_time(prtd->audio_client);
+ if (timestamp < 0) {
+ pr_err("%s: Get Session Time return value =%lld\n",
+ __func__, timestamp);
+ return -EAGAIN;
+ }
+ temp = (timestamp * 2 * runtime->channels);
+ temp = temp * (runtime->rate/1000);
+ temp = div_u64(temp, 1000);
+ tstamp.sampling_rate = runtime->rate;
+ tstamp.rendered = (size_t)(temp & 0xFFFFFFFF);
+ tstamp.decoded = (size_t)((temp >> 32) & 0xFFFFFFFF);
+ tstamp.timestamp = timestamp;
+ pr_debug("%s: bytes_consumed:lsb = %d, msb = %d,"
+ "timestamp = %lld,\n",
+ __func__, tstamp.rendered, tstamp.decoded,
+ tstamp.timestamp);
+ if (copy_to_user((void *) arg, &tstamp,
+ sizeof(struct snd_compr_tstamp)))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_COMPRESS_GET_CAPS:
+ pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
+ if (copy_to_user((void *) arg, &compr->info.compr_cap,
+ sizeof(struct snd_compr_caps))) {
+ rc = -EFAULT;
+ pr_err("%s: ERROR: copy to user\n", __func__);
+ return rc;
+ }
+ return 0;
+ case SNDRV_COMPRESS_SET_PARAMS:
+ pr_debug("SNDRV_COMPRESS_SET_PARAMS: ");
+ if (copy_from_user(&compr->info.codec_param, (void *) arg,
+ sizeof(struct snd_compr_params))) {
+ rc = -EFAULT;
+ pr_err("%s: ERROR: copy from user\n", __func__);
+ return rc;
+ }
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_MP3:
+ /* For MP3 we dont need any other parameter */
+ pr_debug("SND_AUDIOCODEC_MP3\n");
+ compr->codec = FORMAT_MP3;
+ break;
+ case SND_AUDIOCODEC_AAC:
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ compr->codec = FORMAT_MPEG4_AAC;
+ break;
+ default:
+ pr_debug("FORMAT_LINEAR_PCM\n");
+ compr->codec = FORMAT_LINEAR_PCM;
+ break;
+ }
+ return 0;
+ case SNDRV_PCM_IOCTL1_RESET:
+ prtd->cmd_ack = 0;
+ rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ if (rc < 0)
+ pr_err("%s: flush cmd failed rc=%d\n", __func__, rc);
+ rc = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (rc < 0)
+ pr_err("Flush cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ break;
+ default:
+ break;
+ }
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static struct snd_pcm_ops msm_compr_ops = {
+ .open = msm_compr_open,
+ .hw_params = msm_compr_hw_params,
+ .close = msm_compr_close,
+ .ioctl = msm_compr_ioctl,
+ .prepare = msm_compr_prepare,
+ .trigger = msm_compr_trigger,
+ .pointer = msm_compr_pointer,
+ .mmap = msm_compr_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_compr_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_compr_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_compr_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_compr_driver = {
+ .driver = {
+ .name = "msm-compr-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_compr_probe,
+ .remove = __devexit_p(msm_compr_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_compr_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_compr_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
new file mode 100644
index 0000000..2183690
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef _MSM_COMPR_H
+#define _MSM_COMPR_H
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+#include <sound/snd_compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+
+#include "msm-pcm-q6-v2.h"
+
+struct compr_info {
+ struct snd_compr_caps compr_cap;
+ struct snd_compr_codec_caps codec_caps;
+ struct snd_compr_params codec_param;
+};
+
+struct compr_audio {
+ struct msm_audio prtd;
+ struct compr_info info;
+ uint32_t codec;
+};
+
+#endif /*_MSM_COMPR_H*/
diff --git a/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
new file mode 100644
index 0000000..1605062
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-dai-q6-v2.c
@@ -0,0 +1,1229 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wcd9xxx/core.h>
+#include <linux/bitops.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/apr_audio-v2.h>
+#include <sound/q6afe-v2.h>
+#include <sound/msm-dai-q6-v2.h>
+#include <sound/pcm_params.h>
+#include <mach/clk.h>
+
+enum {
+ STATUS_PORT_STARTED, /* track if AFE port has started */
+ STATUS_MAX
+};
+
+struct msm_dai_q6_dai_data {
+ DECLARE_BITMAP(status_mask, STATUS_MAX);
+ u32 rate;
+ u32 channels;
+ union afe_port_config port_config;
+};
+
+static struct clk *pcm_clk;
+static DEFINE_MUTEX(aux_pcm_mutex);
+static int aux_pcm_count;
+static struct msm_dai_auxpcm_pdata *auxpcm_plat_data;
+
+static u8 num_of_bits_set(u8 sd_line_mask)
+{
+ u8 num_bits_set = 0;
+
+ while (sd_line_mask) {
+ num_bits_set++;
+ sd_line_mask = sd_line_mask & (sd_line_mask - 1);
+ }
+ return num_bits_set;
+}
+
+static int msm_dai_q6_cdc_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->channels = params_channels(params);
+ switch (dai_data->channels) {
+ case 2:
+ dai_data->port_config.i2s.mono_stereo = MSM_AFE_STEREO;
+ break;
+ case 1:
+ dai_data->port_config.i2s.mono_stereo = MSM_AFE_MONO;
+ break;
+ default:
+ return -EINVAL;
+ break;
+ }
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.i2s.sample_rate = dai_data->rate;
+ dai_data->port_config.i2s.i2s_cfg_minor_version =
+ AFE_API_VERSION_I2S_CONFIG;
+ dai_data->port_config.i2s.data_format = AFE_LINEAR_PCM_DATA;
+ dev_dbg(dai->dev, " channel %d sample rate %d entered\n",
+ dai_data->channels, dai_data->rate);
+
+ /* Q6 only supports 16 as now */
+ dai_data->port_config.i2s.bit_width = 16;
+ dai_data->port_config.i2s.channel_mode = 1;
+ return 0;
+}
+
+static int msm_dai_q6_i2s_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct msm_i2s_data *i2s_pdata =
+ (struct msm_i2s_data *) dai->dev->platform_data;
+
+ dai_data->channels = params_channels(params);
+ if (num_of_bits_set(i2s_pdata->sd_lines) == 1) {
+ switch (dai_data->channels) {
+ case 2:
+ dai_data->port_config.i2s.mono_stereo = MSM_AFE_STEREO;
+ break;
+ case 1:
+ dai_data->port_config.i2s.mono_stereo = MSM_AFE_MONO;
+ break;
+ default:
+ pr_warn("greater than stereo has not been validated");
+ break;
+ }
+ }
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.i2s.sample_rate = dai_data->rate;
+ dai_data->port_config.i2s.i2s_cfg_minor_version =
+ AFE_API_VERSION_I2S_CONFIG;
+ dai_data->port_config.i2s.data_format = AFE_LINEAR_PCM_DATA;
+ /* Q6 only supports 16 as now */
+ dai_data->port_config.i2s.bit_width = 16;
+ dai_data->port_config.i2s.channel_mode = 1;
+
+ return 0;
+}
+
+static int msm_dai_q6_i2s_platform_data_validation(
+ struct snd_soc_dai *dai)
+{
+ u8 num_of_sd_lines;
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct msm_i2s_data *i2s_pdata =
+ (struct msm_i2s_data *)dai->dev->platform_data;
+ struct snd_soc_dai_driver *dai_driver =
+ (struct snd_soc_dai_driver *)dai->driver;
+
+ num_of_sd_lines = num_of_bits_set(i2s_pdata->sd_lines);
+
+ switch (num_of_sd_lines) {
+ case 1:
+ switch (i2s_pdata->sd_lines) {
+ case MSM_MI2S_SD0:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_SD0;
+ break;
+ case MSM_MI2S_SD1:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_SD1;
+ break;
+ case MSM_MI2S_SD2:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_SD2;
+ break;
+ case MSM_MI2S_SD3:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_SD3;
+ break;
+ default:
+ pr_err("%s: invalid SD line\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ case 2:
+ switch (i2s_pdata->sd_lines) {
+ case MSM_MI2S_SD0 | MSM_MI2S_SD1:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_QUAD01;
+ break;
+ case MSM_MI2S_SD2 | MSM_MI2S_SD3:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_QUAD23;
+ break;
+ default:
+ pr_err("%s: invalid SD line\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ case 3:
+ switch (i2s_pdata->sd_lines) {
+ case MSM_MI2S_SD0 | MSM_MI2S_SD1 | MSM_MI2S_SD2:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_6CHS;
+ break;
+ default:
+ pr_err("%s: invalid SD lines\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ case 4:
+ switch (i2s_pdata->sd_lines) {
+ case MSM_MI2S_SD0 | MSM_MI2S_SD1 | MSM_MI2S_SD2 | MSM_MI2S_SD3:
+ dai_data->port_config.i2s.channel_mode =
+ AFE_PORT_I2S_8CHS;
+ break;
+ default:
+ pr_err("%s: invalid SD lines\n",
+ __func__);
+ goto error_invalid_data;
+ }
+ break;
+ default:
+ pr_err("%s: invalid SD lines\n", __func__);
+ goto error_invalid_data;
+ }
+ if (i2s_pdata->capability == MSM_MI2S_CAP_RX)
+ dai_driver->playback.channels_max = num_of_sd_lines << 1;
+
+ return 0;
+
+error_invalid_data:
+ return -EINVAL;
+}
+
+static int msm_dai_q6_cdc_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ dai_data->port_config.i2s.ws_src = 1; /* CPU is master */
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ dai_data->port_config.i2s.ws_src = 0; /* CPU is slave */
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
+static int msm_dai_q6_slim_bus_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->channels = params_channels(params);
+ dai_data->rate = params_rate(params);
+
+ /* Q6 only supports 16 as now */
+ dai_data->port_config.slim_sch.sb_cfg_minor_version =
+ AFE_API_VERSION_SLIMBUS_CONFIG;
+ dai_data->port_config.slim_sch.bit_width = 16;
+ dai_data->port_config.slim_sch.data_format = 0;
+ dai_data->port_config.slim_sch.num_channels = dai_data->channels;
+ dai_data->port_config.slim_sch.sample_rate = dai_data->rate;
+
+ dev_dbg(dai->dev, "%s:slimbus_dev_id[%hu] bit_wd[%hu] format[%hu]\n"
+ "num_channel %hu shared_ch_mapping[0] %hu\n"
+ "slave_port_mapping[1] %hu slave_port_mapping[2] %hu\n"
+ "sample_rate %d\n", __func__,
+ dai_data->port_config.slim_sch.slimbus_dev_id,
+ dai_data->port_config.slim_sch.bit_width,
+ dai_data->port_config.slim_sch.data_format,
+ dai_data->port_config.slim_sch.num_channels,
+ dai_data->port_config.slim_sch.shared_ch_mapping[0],
+ dai_data->port_config.slim_sch.shared_ch_mapping[1],
+ dai_data->port_config.slim_sch.shared_ch_mapping[2],
+ dai_data->rate);
+
+ return 0;
+}
+
+static int msm_dai_q6_bt_fm_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->channels = params_channels(params);
+ dai_data->rate = params_rate(params);
+
+ dev_dbg(dai->dev, "channels %d sample rate %d entered\n",
+ dai_data->channels, dai_data->rate);
+
+ memset(&dai_data->port_config, 0, sizeof(dai_data->port_config));
+
+ return 0;
+}
+static int msm_dai_q6_auxpcm_hw_params(
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct msm_dai_auxpcm_pdata *auxpcm_pdata =
+ (struct msm_dai_auxpcm_pdata *) dai->dev->platform_data;
+
+ if (params_channels(params) != 1) {
+ dev_err(dai->dev, "AUX PCM supports only mono stream\n");
+ return -EINVAL;
+ }
+ dai_data->channels = params_channels(params);
+
+ if (params_rate(params) != 8000) {
+ dev_err(dai->dev, "AUX PCM supports only 8KHz sampling rate\n");
+ return -EINVAL;
+ }
+ dai_data->rate = params_rate(params);
+
+ dai_data->port_config.pcm.pcm_cfg_minor_version =
+ AFE_API_VERSION_PCM_CONFIG;
+ dai_data->port_config.pcm.aux_mode = auxpcm_pdata->mode;
+ dai_data->port_config.pcm.sync_src = auxpcm_pdata->sync;
+ dai_data->port_config.pcm.frame_setting = auxpcm_pdata->frame;
+ dai_data->port_config.pcm.quantype = auxpcm_pdata->quant;
+ dai_data->port_config.pcm.ctrl_data_out_enable = auxpcm_pdata->data;
+ dai_data->port_config.pcm.sample_rate = dai_data->rate;
+ dai_data->port_config.pcm.num_channels = dai_data->channels;
+ dai_data->port_config.pcm.bit_width = 16;
+ dai_data->port_config.pcm.slot_number_mapping[0] = auxpcm_pdata->slot;
+
+ return 0;
+}
+
+static int msm_dai_q6_afe_rtproxy_hw_params(struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ dai_data->rate = params_rate(params);
+ dai_data->port_config.rtproxy.num_channels = params_channels(params);
+ dai_data->port_config.rtproxy.sample_rate = params_rate(params);
+
+ pr_debug("channel %d entered,dai_id: %d,rate: %d\n",
+ dai_data->port_config.rtproxy.num_channels, dai->id, dai_data->rate);
+
+ dai_data->port_config.rtproxy.rt_proxy_cfg_minor_version =
+ AFE_API_VERSION_RT_PROXY_CONFIG;
+ dai_data->port_config.rtproxy.bit_width = 16; /* Q6 only supports 16 */
+ dai_data->port_config.rtproxy.interleaved = 1;
+ dai_data->port_config.rtproxy.frame_size = params_period_bytes(params);
+ dai_data->port_config.rtproxy.jitter_allowance =
+ dai_data->port_config.rtproxy.frame_size/2;
+ dai_data->port_config.rtproxy.low_water_mark = 0;
+ dai_data->port_config.rtproxy.high_water_mark = 0;
+
+ return 0;
+}
+
+/* Current implementation assumes hw_param is called once
+ * This may not be the case but what to do when ADM and AFE
+ * port are already opened and parameter changes
+ */
+static int msm_dai_q6_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ int rc = 0;
+
+ switch (dai->id) {
+ case PRIMARY_I2S_TX:
+ case PRIMARY_I2S_RX:
+ case SECONDARY_I2S_RX:
+ rc = msm_dai_q6_cdc_hw_params(params, dai, substream->stream);
+ break;
+ case MI2S_RX:
+ rc = msm_dai_q6_i2s_hw_params(params, dai, substream->stream);
+ break;
+ case SLIMBUS_0_RX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_TX:
+ rc = msm_dai_q6_slim_bus_hw_params(params, dai,
+ substream->stream);
+ break;
+ case INT_BT_SCO_RX:
+ case INT_BT_SCO_TX:
+ case INT_FM_RX:
+ case INT_FM_TX:
+ rc = msm_dai_q6_bt_fm_hw_params(params, dai, substream->stream);
+ break;
+ case RT_PROXY_DAI_001_TX:
+ case RT_PROXY_DAI_001_RX:
+ case RT_PROXY_DAI_002_TX:
+ case RT_PROXY_DAI_002_RX:
+ rc = msm_dai_q6_afe_rtproxy_hw_params(params, dai);
+ break;
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ rc = 0;
+ break;
+ default:
+ dev_err(dai->dev, "invalid AFE port ID\n");
+ rc = -EINVAL;
+ break;
+ }
+
+ return rc;
+}
+
+static void msm_dai_q6_auxpcm_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ int rc = 0;
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (aux_pcm_count == 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count is 0. Just"
+ " return\n", __func__, dai->id);
+ mutex_unlock(&aux_pcm_mutex);
+ return;
+ }
+
+ aux_pcm_count--;
+
+ if (aux_pcm_count > 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d\n",
+ __func__, dai->id, aux_pcm_count);
+ mutex_unlock(&aux_pcm_mutex);
+ return;
+ } else if (aux_pcm_count < 0) {
+ dev_err(dai->dev, "%s(): ERROR: dai->id %d"
+ " aux_pcm_count = %d < 0\n",
+ __func__, dai->id, aux_pcm_count);
+ aux_pcm_count = 0;
+ mutex_unlock(&aux_pcm_mutex);
+ return;
+ }
+
+ pr_debug("%s: dai->id = %d aux_pcm_count = %d\n", __func__,
+ dai->id, aux_pcm_count);
+
+ rc = afe_close(PCM_RX); /* can block */
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close PCM_RX AFE port\n");
+
+ rc = afe_close(PCM_TX);
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AUX PCM TX port\n");
+
+ mutex_unlock(&aux_pcm_mutex);
+}
+
+static void msm_dai_q6_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ pr_debug("%s, stop pseudo port:%d\n",
+ __func__, dai->id);
+ rc = afe_stop_pseudo_port(dai->id);
+ break;
+ default:
+ rc = afe_close(dai->id); /* can block */
+ break;
+ }
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+ pr_debug("%s: dai_data->status_mask = %ld\n", __func__,
+ *dai_data->status_mask);
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+}
+
+static int msm_dai_q6_auxpcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (aux_pcm_count == 2) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count is 2. Just"
+ " return.\n", __func__, dai->id);
+ mutex_unlock(&aux_pcm_mutex);
+ return 0;
+ } else if (aux_pcm_count > 2) {
+ dev_err(dai->dev, "%s(): ERROR: dai->id %d"
+ " aux_pcm_count = %d > 2\n",
+ __func__, dai->id, aux_pcm_count);
+ mutex_unlock(&aux_pcm_mutex);
+ return 0;
+ }
+
+ aux_pcm_count++;
+ if (aux_pcm_count == 2) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d after "
+ " increment\n", __func__, dai->id, aux_pcm_count);
+ mutex_unlock(&aux_pcm_mutex);
+ return 0;
+ }
+
+ pr_debug("%s:dai->id:%d aux_pcm_count = %d. opening afe\n",
+ __func__, dai->id, aux_pcm_count);
+
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+
+ /*
+ * For AUX PCM Interface the below sequence of clk
+ * settings and afe_open is a strict requirement.
+ *
+ * Also using afe_open instead of afe_port_start_nowait
+ * to make sure the port is open before deasserting the
+ * clock line. This is required because pcm register is
+ * not written before clock deassert. Hence the hw does
+ * not get updated with new setting if the below clock
+ * assert/deasset and afe_open sequence is not followed.
+ */
+
+ afe_open(PCM_RX, &dai_data->port_config, dai_data->rate);
+
+ afe_open(PCM_TX, &dai_data->port_config, dai_data->rate);
+
+ mutex_unlock(&aux_pcm_mutex);
+
+ return rc;
+}
+
+static int msm_dai_q6_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ /* PORT START should be set if prepare called in active state */
+ rc = afe_q6_interface_prepare();
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to open AFE APR\n");
+ }
+ return rc;
+}
+
+static int msm_dai_q6_auxpcm_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ int rc = 0;
+
+ pr_debug("%s:port:%d cmd:%d aux_pcm_count= %d",
+ __func__, dai->id, cmd, aux_pcm_count);
+
+ switch (cmd) {
+
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /* afe_open will be called from prepare */
+ return 0;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ return 0;
+
+ default:
+ rc = -EINVAL;
+ }
+
+ return rc;
+
+}
+
+static int msm_dai_q6_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc = 0;
+
+ /* Start/stop port without waiting for Q6 AFE response. Need to have
+ * native q6 AFE driver propagates AFE response in order to handle
+ * port start/stop command error properly if error does arise.
+ */
+ pr_debug("%s:port:%d cmd:%d dai_data->status_mask = %ld",
+ __func__, dai->id, cmd, *dai_data->status_mask);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (!test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ afe_pseudo_port_start_nowait(dai->id);
+ break;
+ default:
+ afe_port_start_nowait(dai->id,
+ &dai_data->port_config, dai_data->rate);
+ break;
+ }
+ set_bit(STATUS_PORT_STARTED,
+ dai_data->status_mask);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ afe_pseudo_port_stop_nowait(dai->id);
+ break;
+ default:
+ afe_port_stop_nowait(dai->id);
+ break;
+ }
+ clear_bit(STATUS_PORT_STARTED,
+ dai_data->status_mask);
+ }
+ break;
+
+ default:
+ rc = -EINVAL;
+ }
+
+ return rc;
+}
+static int msm_dai_q6_dai_auxpcm_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc = 0;
+
+ struct msm_dai_auxpcm_pdata *auxpcm_pdata =
+ (struct msm_dai_auxpcm_pdata *) dai->dev->platform_data;
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (!auxpcm_plat_data)
+ auxpcm_plat_data = auxpcm_pdata;
+ else if (auxpcm_plat_data != auxpcm_pdata) {
+
+ dev_err(dai->dev, "AUX PCM RX and TX devices does not have"
+ " same platform data\n");
+ return -EINVAL;
+ }
+
+ /*
+ * The clk name for AUX PCM operation is passed as platform
+ * data to the cpu driver, since cpu drive is unaware of any
+ * boarc specific configuration.
+ */
+ if (!pcm_clk)
+ pcm_clk = clk_get(dai->dev, auxpcm_pdata->clk);
+
+ mutex_unlock(&aux_pcm_mutex);
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_dai_data), GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+
+ pr_err("%s : probe done for dai->id %d\n", __func__, dai->id);
+ return rc;
+}
+
+static int msm_dai_q6_dai_auxpcm_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc;
+
+ dai_data = dev_get_drvdata(dai->dev);
+
+ mutex_lock(&aux_pcm_mutex);
+
+ if (aux_pcm_count == 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count is 0. clean"
+ " up and return\n", __func__, dai->id);
+ goto done;
+ }
+
+ aux_pcm_count--;
+
+ if (aux_pcm_count > 0) {
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d\n",
+ __func__, dai->id, aux_pcm_count);
+ goto done;
+ } else if (aux_pcm_count < 0) {
+ dev_err(dai->dev, "%s(): ERROR: dai->id %d"
+ " aux_pcm_count = %d < 0\n",
+ __func__, dai->id, aux_pcm_count);
+ goto done;
+ }
+
+ dev_dbg(dai->dev, "%s(): dai->id %d aux_pcm_count = %d."
+ "closing afe\n",
+ __func__, dai->id, aux_pcm_count);
+
+ rc = afe_close(PCM_RX); /* can block */
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AUX PCM RX AFE port\n");
+
+ rc = afe_close(PCM_TX);
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AUX PCM TX AFE port\n");
+
+done:
+ kfree(dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ mutex_unlock(&aux_pcm_mutex);
+
+ return 0;
+}
+static int msm_dai_q6_dai_i2s_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc = 0;
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_dai_data),
+ GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ goto rtn;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+
+ rc = msm_dai_q6_i2s_platform_data_validation(dai);
+ if (rc != 0) {
+ pr_err("%s: The msm_dai_q6_i2s_platform_data_validation failed\n",
+ __func__);
+ kfree(dai_data);
+ }
+rtn:
+ return rc;
+}
+
+static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc = 0;
+
+ dai_data = kzalloc(sizeof(struct msm_dai_q6_dai_data),
+ GFP_KERNEL);
+
+ if (!dai_data) {
+ dev_err(dai->dev, "DAI-%d: fail to allocate dai data\n",
+ dai->id);
+ rc = -ENOMEM;
+ } else
+ dev_set_drvdata(dai->dev, dai_data);
+
+ return rc;
+}
+
+static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai)
+{
+ struct msm_dai_q6_dai_data *dai_data;
+ int rc;
+
+ dai_data = dev_get_drvdata(dai->dev);
+
+ /* If AFE port is still up, close it */
+ if (test_bit(STATUS_PORT_STARTED, dai_data->status_mask)) {
+ switch (dai->id) {
+ case VOICE_PLAYBACK_TX:
+ case VOICE_RECORD_TX:
+ case VOICE_RECORD_RX:
+ pr_debug("%s, stop pseudo port:%d\n",
+ __func__, dai->id);
+ rc = afe_stop_pseudo_port(dai->id);
+ break;
+ default:
+ rc = afe_close(dai->id); /* can block */
+ }
+ if (IS_ERR_VALUE(rc))
+ dev_err(dai->dev, "fail to close AFE port\n");
+ clear_bit(STATUS_PORT_STARTED, dai_data->status_mask);
+ }
+ kfree(dai_data);
+ snd_soc_unregister_dai(dai->dev);
+
+ return 0;
+}
+
+static int msm_dai_q6_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ int rc = 0;
+
+ dev_dbg(dai->dev, "enter %s, id = %d fmt[%d]\n", __func__,
+ dai->id, fmt);
+ switch (dai->id) {
+ case PRIMARY_I2S_TX:
+ case PRIMARY_I2S_RX:
+ case MI2S_RX:
+ case SECONDARY_I2S_RX:
+ rc = msm_dai_q6_cdc_set_fmt(dai, fmt);
+ break;
+ default:
+ dev_err(dai->dev, "invalid cpu_dai set_fmt\n");
+ rc = -EINVAL;
+ break;
+ }
+
+ return rc;
+}
+
+static int msm_dai_q6_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+
+{
+ int rc = 0;
+ struct msm_dai_q6_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ unsigned int i = 0;
+
+ dev_dbg(dai->dev, "enter %s, id = %d\n", __func__,
+ dai->id);
+ switch (dai->id) {
+ case SLIMBUS_0_RX:
+ case SLIMBUS_1_RX:
+ /* channel number to be between 128 and 255. For RX port
+ * use channel numbers from 138 to 144, for TX port
+ * use channel numbers from 128 to 137
+ * For ports between MDM-APQ use channel numbers from 145
+ */
+ if (!rx_slot)
+ return -EINVAL;
+ for (i = 0; i < rx_num; i++) {
+ dai_data->port_config.slim_sch.shared_ch_mapping[i] =
+ rx_slot[i];
+ pr_debug("%s: find number of channels[%d] ch[%d]\n",
+ __func__, i,
+ rx_slot[i]);
+ }
+ dai_data->port_config.slim_sch.num_channels = rx_num;
+ pr_debug("%s:SLIMBUS_0_RX cnt[%d] ch[%d %d]\n", __func__,
+ rx_num, dai_data->port_config.slim_sch.shared_ch_mapping[0],
+ dai_data->port_config.slim_sch.shared_ch_mapping[1]);
+
+ break;
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_TX:
+ /* channel number to be between 128 and 255. For RX port
+ * use channel numbers from 138 to 144, for TX port
+ * use channel numbers from 128 to 137
+ * For ports between MDM-APQ use channel numbers from 145
+ */
+ if (!tx_slot)
+ return -EINVAL;
+ for (i = 0; i < tx_num; i++) {
+ dai_data->port_config.slim_sch.shared_ch_mapping[i] =
+ tx_slot[i];
+ pr_debug("%s: find number of channels[%d] ch[%d]\n",
+ __func__, i, tx_slot[i]);
+ }
+ dai_data->port_config.slim_sch.num_channels = tx_num;
+ pr_debug("%s:SLIMBUS_0_TX cnt[%d] ch[%d %d]\n", __func__,
+ tx_num, dai_data->port_config.slim_sch.shared_ch_mapping[0],
+ dai_data->port_config.slim_sch.shared_ch_mapping[1]);
+ break;
+ default:
+ dev_err(dai->dev, "invalid cpu_dai set_fmt\n");
+ rc = -EINVAL;
+ break;
+ }
+ return rc;
+}
+
+static struct snd_soc_dai_ops msm_dai_q6_ops = {
+ .prepare = msm_dai_q6_prepare,
+ .trigger = msm_dai_q6_trigger,
+ .hw_params = msm_dai_q6_hw_params,
+ .shutdown = msm_dai_q6_shutdown,
+ .set_fmt = msm_dai_q6_set_fmt,
+ .set_channel_map = msm_dai_q6_set_channel_map,
+};
+
+static struct snd_soc_dai_ops msm_dai_q6_auxpcm_ops = {
+ .prepare = msm_dai_q6_auxpcm_prepare,
+ .trigger = msm_dai_q6_auxpcm_trigger,
+ .hw_params = msm_dai_q6_auxpcm_hw_params,
+ .shutdown = msm_dai_q6_auxpcm_shutdown,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_i2s_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 4,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_i2s_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_i2s_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_afe_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_afe_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_voice_playback_tx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_incall_record_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_bt_sco_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 16000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_bt_sco_tx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 16000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_fm_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_fm_tx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 2,
+ .channels_max = 2,
+ .rate_max = 48000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_aux_pcm_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 8000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_auxpcm_ops,
+ .probe = msm_dai_q6_dai_auxpcm_probe,
+ .remove = msm_dai_q6_dai_auxpcm_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_aux_pcm_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_max = 8000,
+ .rate_min = 8000,
+ },
+ .ops = &msm_dai_q6_auxpcm_ops,
+ .probe = msm_dai_q6_dai_auxpcm_probe,
+ .remove = msm_dai_q6_dai_auxpcm_remove,
+};
+
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_1_rx_dai = {
+ .playback = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static struct snd_soc_dai_driver msm_dai_q6_slimbus_1_tx_dai = {
+ .capture = {
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 1,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ },
+ .ops = &msm_dai_q6_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+};
+
+static __devinit int msm_dai_q6_dev_probe(struct platform_device *pdev)
+{
+ int rc = 0;
+
+ dev_dbg(&pdev->dev, "dev name %s\n", dev_name(&pdev->dev));
+
+ switch (pdev->id) {
+ case PRIMARY_I2S_RX:
+ case SECONDARY_I2S_RX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_i2s_rx_dai);
+ break;
+ case PRIMARY_I2S_TX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_i2s_tx_dai);
+ break;
+ case AFE_PORT_ID_PRIMARY_PCM_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_aux_pcm_rx_dai);
+ break;
+ case AFE_PORT_ID_PRIMARY_PCM_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_aux_pcm_tx_dai);
+ break;
+ case MI2S_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_i2s_rx_dai);
+ break;
+ case SLIMBUS_0_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_rx_dai);
+ break;
+ case SLIMBUS_0_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_tx_dai);
+ break;
+
+ case SLIMBUS_1_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_1_rx_dai);
+ break;
+ case SLIMBUS_1_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_slimbus_1_tx_dai);
+ break;
+ case INT_BT_SCO_RX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_bt_sco_rx_dai);
+ break;
+ case INT_BT_SCO_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_bt_sco_tx_dai);
+ break;
+ case INT_FM_RX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_fm_rx_dai);
+ break;
+ case INT_FM_TX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_fm_tx_dai);
+ break;
+ case RT_PROXY_DAI_001_RX:
+ case RT_PROXY_DAI_002_RX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_afe_rx_dai);
+ break;
+ case RT_PROXY_DAI_001_TX:
+ case RT_PROXY_DAI_002_TX:
+ rc = snd_soc_register_dai(&pdev->dev, &msm_dai_q6_afe_tx_dai);
+ break;
+ case VOICE_PLAYBACK_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_voice_playback_tx_dai);
+ break;
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ rc = snd_soc_register_dai(&pdev->dev,
+ &msm_dai_q6_incall_record_dai);
+ break;
+ default:
+ rc = -ENODEV;
+ break;
+ }
+ return rc;
+}
+
+static __devexit int msm_dai_q6_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_dai_q6_driver = {
+ .probe = msm_dai_q6_dev_probe,
+ .remove = msm_dai_q6_dev_remove,
+ .driver = {
+ .name = "msm-dai-q6",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init msm_dai_q6_init(void)
+{
+ return platform_driver_register(&msm_dai_q6_driver);
+}
+module_init(msm_dai_q6_init);
+
+static void __exit msm_dai_q6_exit(void)
+{
+ platform_driver_unregister(&msm_dai_q6_driver);
+}
+module_exit(msm_dai_q6_exit);
+
+/* Module information */
+MODULE_DESCRIPTION("MSM DSP DAI driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c
new file mode 100644
index 0000000..cab689d
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c
@@ -0,0 +1,777 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <asm/dma.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+
+#include "msm-pcm-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+struct snd_msm_volume {
+ struct msm_audio *prtd;
+ unsigned volume;
+};
+static struct snd_msm_volume multi_ch_pcm_audio = {NULL, 0x2000};
+
+#define PLAYBACK_NUM_PERIODS 8
+#define PLAYBACK_PERIOD_SIZE 4032
+#define CAPTURE_NUM_PERIODS 16
+#define CAPTURE_PERIOD_SIZE 320
+
+static struct snd_pcm_hardware msm_pcm_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = CAPTURE_NUM_PERIODS * CAPTURE_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_PERIOD_SIZE,
+ .periods_min = CAPTURE_NUM_PERIODS,
+ .periods_max = CAPTURE_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 6,
+ .buffer_bytes_max = PLAYBACK_NUM_PERIODS * PLAYBACK_PERIOD_SIZE,
+ .period_bytes_min = PLAYBACK_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_PERIOD_SIZE,
+ .periods_min = PLAYBACK_NUM_PERIODS,
+ .periods_max = PLAYBACK_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static uint32_t in_frame_info[CAPTURE_NUM_PERIODS][2];
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ int i = 0;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ pr_debug("%s\n", __func__);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag)
+ break;
+ if (q6asm_is_cpu_buf_avail_nolock(IN,
+ prtd->audio_client,
+ &size, &idx)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ }
+ break;
+ }
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+ pr_debug("token = 0x%08x\n", token);
+ for (i = 0; i < 8; i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ in_frame_info[token][0] = payload[2];
+ in_frame_info[token][1] = payload[3];
+ prtd->pcm_irq_pos += in_frame_info[token][0];
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag
+ && q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx))
+ q6asm_read_nolock(prtd->audio_client);
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN_V2:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK) {
+ atomic_set(&prtd->start, 1);
+ break;
+ }
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ atomic_set(&prtd->start, 1);
+ break;
+ default:
+ break;
+ }
+ }
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_pcm(prtd->audio_client,
+ runtime->rate, runtime->channels);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ int i = 0;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return 0;
+
+ pr_debug("Samp_rate = %d\n", prtd->samp_rate);
+ pr_debug("Channel = %d\n", prtd->channel_mode);
+ ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client, prtd->samp_rate,
+ prtd->channel_mode);
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+ prtd->periods = runtime->periods;
+
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ int ret = 0;
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_err("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_pcm_hardware_playback;
+ ret = q6asm_open_write(prtd->audio_client,
+ FORMAT_MULTI_CHANNEL_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_hardware_capture;
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm in open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ runtime->private_data = prtd;
+ pr_debug("substream->pcm->device = %d\n", substream->pcm->device);
+ pr_debug("soc_prtd->dai_link->be_id = %d\n", soc_prtd->dai_link->be_id);
+ multi_ch_pcm_audio.prtd = prtd;
+ ret = multi_ch_pcm_set_volume(multi_ch_pcm_audio.volume);
+ if (ret < 0)
+ pr_err("%s : Set Volume failed : %d", __func__, ret);
+
+ ret = q6asm_set_softpause(multi_ch_pcm_audio.prtd->audio_client,
+ &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(multi_ch_pcm_audio.prtd->audio_client,
+ &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ return 0;
+}
+
+int multi_ch_pcm_set_volume(unsigned volume)
+{
+ int rc = 0;
+ pr_err("multi_ch_pcm_set_volume\n");
+
+ if (multi_ch_pcm_audio.prtd && multi_ch_pcm_audio.prtd->audio_client) {
+ pr_err("%s q6asm_set_volume\n", __func__);
+ rc = q6asm_set_volume(multi_ch_pcm_audio.prtd->audio_client,
+ volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed"
+ " rc=%d\n", __func__, rc);
+ }
+ }
+ multi_ch_pcm_audio.volume = volume;
+ return rc;
+}
+
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_err("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_err("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ xfer = fbytes;
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+fail:
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ ret = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (ret < 0)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ multi_ch_pcm_audio.prtd = NULL;
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%08x\n",
+ (unsigned int) data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = in_frame_info[idx][1];
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ in_frame_info[idx][1] += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&in_frame_info[idx], 0,
+ sizeof(uint32_t) * 2);
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-multi-ch-pcm-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("Multi channel PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-afe-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-afe-v2.c
new file mode 100644
index 0000000..4593784
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-afe-v2.c
@@ -0,0 +1,581 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License version 2 and
+* only version 2 as published by the Free Software Foundation.
+*
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*/
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6adm-v2.h>
+#include <asm/dma.h>
+#include "msm-pcm-afe-v2.h"
+
+#define MIN_PERIOD_SIZE (128 * 2)
+#define MAX_PERIOD_SIZE (128 * 2 * 2 * 6)
+static struct snd_pcm_hardware msm_afe_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000),
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = MAX_PERIOD_SIZE * 32,
+ .period_bytes_min = MIN_PERIOD_SIZE,
+ .period_bytes_max = MAX_PERIOD_SIZE,
+ .periods_min = 32,
+ .periods_max = 384,
+ .fifo_size = 0,
+};
+static enum hrtimer_restart afe_hrtimer_callback(struct hrtimer *hrt);
+static enum hrtimer_restart afe_hrtimer_rec_callback(struct hrtimer *hrt);
+
+static enum hrtimer_restart afe_hrtimer_callback(struct hrtimer *hrt)
+{
+ struct pcm_afe_info *prtd =
+ container_of(hrt, struct pcm_afe_info, hrt);
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ u32 mem_map_handle = 0;
+ if (prtd->start) {
+ pr_debug("sending frame to DSP: poll_time: %d\n",
+ prtd->poll_time);
+ if (prtd->dsp_cnt == runtime->periods)
+ prtd->dsp_cnt = 0;
+ pr_debug("%s: mem_map_handle 0x%x\n", __func__, mem_map_handle);
+ afe_rt_proxy_port_write(
+ (prtd->dma_addr +
+ (prtd->dsp_cnt *
+ snd_pcm_lib_period_bytes(prtd->substream))), mem_map_handle,
+ snd_pcm_lib_period_bytes(prtd->substream));
+ prtd->dsp_cnt++;
+ hrtimer_forward_now(hrt, ns_to_ktime(prtd->poll_time
+ * 1000));
+
+ return HRTIMER_RESTART;
+ } else
+ return HRTIMER_NORESTART;
+}
+static enum hrtimer_restart afe_hrtimer_rec_callback(struct hrtimer *hrt)
+{
+ struct pcm_afe_info *prtd =
+ container_of(hrt, struct pcm_afe_info, hrt);
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ u32 mem_map_handle = 0;
+ if (prtd->start) {
+ if (prtd->dsp_cnt == runtime->periods)
+ prtd->dsp_cnt = 0;
+ pr_err("%s: mem_map_handle 0x%x\n", __func__, mem_map_handle);
+ afe_rt_proxy_port_read(
+ (prtd->dma_addr + (prtd->dsp_cnt
+ * snd_pcm_lib_period_bytes(prtd->substream))), mem_map_handle,
+ snd_pcm_lib_period_bytes(prtd->substream));
+ prtd->dsp_cnt++;
+ pr_debug("sending frame rec to DSP: poll_time: %d\n",
+ prtd->poll_time);
+ hrtimer_forward_now(hrt, ns_to_ktime(prtd->poll_time
+ * 1000));
+
+ return HRTIMER_RESTART;
+ } else
+ return HRTIMER_NORESTART;
+}
+static void pcm_afe_process_tx_pkt(uint32_t opcode,
+ uint32_t token, uint32_t *payload,
+ void *priv)
+{
+ struct pcm_afe_info *prtd = priv;
+ unsigned long dsp_flags;
+ struct snd_pcm_substream *substream = NULL;
+ struct snd_pcm_runtime *runtime = NULL;
+ uint16_t event;
+
+ if (prtd == NULL)
+ return;
+ substream = prtd->substream;
+ runtime = substream->runtime;
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&prtd->dsp_lock, dsp_flags);
+ switch (opcode) {
+ case AFE_EVENT_RT_PROXY_PORT_STATUS: {
+ event = (uint16_t)((0xFFFF0000 & payload[0]) >> 0x10);
+ switch (event) {
+ case AFE_EVENT_RTPORT_START: {
+ prtd->dsp_cnt = 0;
+ prtd->poll_time = ((unsigned long)((
+ snd_pcm_lib_period_bytes
+ (prtd->substream) *
+ 1000 * 1000)/
+ (runtime->rate *
+ runtime->channels * 2)));
+ pr_debug("prtd->poll_time: %d",
+ prtd->poll_time);
+ hrtimer_start(&prtd->hrt,
+ ns_to_ktime(0),
+ HRTIMER_MODE_REL);
+ break;
+ }
+ case AFE_EVENT_RTPORT_STOP:
+ pr_debug("%s: event!=0\n", __func__);
+ prtd->start = 0;
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ break;
+ case AFE_EVENT_RTPORT_LOW_WM:
+ pr_debug("%s: Underrun\n", __func__);
+ break;
+ case AFE_EVENT_RTPORT_HI_WM:
+ pr_debug("%s: Overrun\n", __func__);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2:
+ pr_debug("write done\n");
+ prtd->pcm_irq_pos += snd_pcm_lib_period_bytes
+ (prtd->substream);
+ snd_pcm_period_elapsed(prtd->substream);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ break;
+ }
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+}
+
+static void pcm_afe_process_rx_pkt(uint32_t opcode,
+ uint32_t token, uint32_t *payload,
+ void *priv)
+{
+ struct pcm_afe_info *prtd = priv;
+ unsigned long dsp_flags;
+ struct snd_pcm_substream *substream = NULL;
+ struct snd_pcm_runtime *runtime = NULL;
+ uint16_t event;
+
+ if (prtd == NULL)
+ return;
+ substream = prtd->substream;
+ runtime = substream->runtime;
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&prtd->dsp_lock, dsp_flags);
+ switch (opcode) {
+ case AFE_EVENT_RT_PROXY_PORT_STATUS: {
+ event = (uint16_t)((0xFFFF0000 & payload[0]) >> 0x10);
+ switch (event) {
+ case AFE_EVENT_RTPORT_START: {
+ prtd->dsp_cnt = 0;
+ prtd->poll_time = ((unsigned long)((
+ snd_pcm_lib_period_bytes(prtd->substream)
+ * 1000 * 1000)/(runtime->rate
+ * runtime->channels * 2)));
+ hrtimer_start(&prtd->hrt,
+ ns_to_ktime(0),
+ HRTIMER_MODE_REL);
+ pr_debug("prtd->poll_time : %d", prtd->poll_time);
+ break;
+ }
+ case AFE_EVENT_RTPORT_STOP:
+ pr_debug("%s: event!=0\n", __func__);
+ prtd->start = 0;
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP);
+ break;
+ case AFE_EVENT_RTPORT_LOW_WM:
+ pr_debug("%s: Underrun\n", __func__);
+ break;
+ case AFE_EVENT_RTPORT_HI_WM:
+ pr_debug("%s: Overrun\n", __func__);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2:
+ pr_debug("Read done\n");
+ prtd->pcm_irq_pos += snd_pcm_lib_period_bytes
+ (prtd->substream);
+ snd_pcm_period_elapsed(prtd->substream);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ break;
+ }
+ spin_unlock_irqrestore(&prtd->dsp_lock, dsp_flags);
+}
+
+static int msm_afe_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ int ret = 0;
+
+ pr_debug("%s: sample_rate=%d\n", __func__, runtime->rate);
+
+ pr_debug("%s: dai->id =%x\n", __func__, dai->id);
+ ret = afe_register_get_events(dai->id,
+ pcm_afe_process_tx_pkt, prtd);
+ if (ret < 0) {
+ pr_err("afe-pcm:register for events failed\n");
+ return ret;
+ }
+ pr_debug("%s:success\n", __func__);
+ prtd->prepared++;
+ return ret;
+}
+
+static int msm_afe_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ pr_debug("%s: dai->id =%x\n", __func__, dai->id);
+ ret = afe_register_get_events(dai->id,
+ pcm_afe_process_rx_pkt, prtd);
+ if (ret < 0) {
+ pr_err("afe-pcm:register for events failed\n");
+ return ret;
+ }
+ pr_debug("%s:success\n", __func__);
+ prtd->prepared++;
+ return 0;
+}
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 16000, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static int msm_afe_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = NULL;
+ int ret = 0;
+
+ prtd = kzalloc(sizeof(struct pcm_afe_info), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ } else
+ pr_debug("prtd %x\n", (unsigned int)prtd);
+
+ mutex_init(&prtd->lock);
+ spin_lock_init(&prtd->dsp_lock);
+ prtd->dsp_cnt = 0;
+
+ mutex_lock(&prtd->lock);
+
+ runtime->hw = msm_afe_hardware;
+ prtd->substream = substream;
+ runtime->private_data = prtd;
+ mutex_unlock(&prtd->lock);
+ hrtimer_init(&prtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->hrt.function = afe_hrtimer_callback;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ prtd->hrt.function = afe_hrtimer_rec_callback;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_err("snd_pcm_hw_constraint_integer failed\n");
+
+ return 0;
+}
+
+static int msm_afe_close(struct snd_pcm_substream *substream)
+{
+ int rc = 0;
+ struct snd_dma_buffer *dma_buf;
+ struct snd_pcm_runtime *runtime;
+ struct pcm_afe_info *prtd;
+ struct snd_soc_pcm_runtime *rtd = NULL;
+ struct snd_soc_dai *dai = NULL;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ if (substream == NULL) {
+ pr_err("substream is NULL\n");
+ return -EINVAL;
+ }
+ rtd = substream->private_data;
+ dai = rtd->cpu_dai;
+ runtime = substream->runtime;
+ prtd = runtime->private_data;
+
+ mutex_lock(&prtd->lock);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = afe_unregister_get_events(dai->id);
+ if (ret < 0)
+ pr_err("AFE unregister for events failed\n");
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = afe_unregister_get_events(dai->id);
+ if (ret < 0)
+ pr_err("AFE unregister for events failed\n");
+ }
+ hrtimer_cancel(&prtd->hrt);
+
+ rc = afe_cmd_memory_unmap(runtime->dma_addr);
+ if (rc < 0)
+ pr_err("AFE memory unmap failed\n");
+
+ pr_debug("release all buffer\n");
+ dma_buf = &substream->dma_buffer;
+ if (dma_buf == NULL) {
+ pr_debug("dma_buf is NULL\n");
+ goto done;
+ }
+ if (dma_buf->area != NULL) {
+ dma_free_coherent(substream->pcm->card->dev,
+ runtime->hw.buffer_bytes_max, dma_buf->area,
+ dma_buf->addr);
+ dma_buf->area = NULL;
+ }
+done:
+ pr_debug("%s: dai->id =%x\n", __func__, dai->id);
+ mutex_unlock(&prtd->lock);
+ prtd->prepared--;
+ kfree(prtd);
+ return 0;
+}
+static int msm_afe_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ prtd->pcm_irq_pos = 0;
+ if (prtd->prepared)
+ return 0;
+ mutex_lock(&prtd->lock);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_afe_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_afe_capture_prepare(substream);
+ mutex_unlock(&prtd->lock);
+ return ret;
+}
+static int msm_afe_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+ dma_mmap_coherent(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+ return 0;
+}
+static int msm_afe_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__);
+ prtd->start = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("%s: SNDRV_PCM_TRIGGER_STOP\n", __func__);
+ prtd->start = 0;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+ return ret;
+}
+static int msm_afe_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct pcm_afe_info *prtd = runtime->private_data;
+ int rc;
+
+ pr_debug("%s:\n", __func__);
+
+ mutex_lock(&prtd->lock);
+
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = dma_alloc_coherent(dma_buf->dev.dev,
+ runtime->hw.buffer_bytes_max,
+ &dma_buf->addr, GFP_KERNEL);
+
+ pr_debug("%s: dma_buf->area: 0x%p, dma_buf->addr: 0x%x", __func__,
+ (unsigned int *) dma_buf->area, dma_buf->addr);
+ if (!dma_buf->area) {
+ pr_err("%s:MSM AFE memory allocation failed\n", __func__);
+ mutex_unlock(&prtd->lock);
+ return -ENOMEM;
+ }
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ memset(dma_buf->area, 0, runtime->hw.buffer_bytes_max);
+ prtd->dma_addr = (u32) dma_buf->addr;
+
+ mutex_unlock(&prtd->lock);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ rc = afe_cmd_memory_map(dma_buf->addr, dma_buf->bytes);
+ if (rc < 0)
+ pr_err("fail to map memory to DSP\n");
+
+ return rc;
+}
+static snd_pcm_uframes_t msm_afe_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct pcm_afe_info *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= snd_pcm_lib_buffer_bytes(substream))
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static struct snd_pcm_ops msm_afe_ops = {
+ .open = msm_afe_open,
+ .hw_params = msm_afe_hw_params,
+ .trigger = msm_afe_trigger,
+ .close = msm_afe_close,
+ .prepare = msm_afe_prepare,
+ .mmap = msm_afe_mmap,
+ .pointer = msm_afe_pointer,
+};
+
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static int msm_afe_afe_probe(struct snd_soc_platform *platform)
+{
+ pr_debug("%s\n", __func__);
+ return 0;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_afe_ops,
+ .pcm_new = msm_asoc_pcm_new,
+ .probe = msm_afe_afe_probe,
+};
+
+static __devinit int msm_afe_probe(struct platform_device *pdev)
+{
+ pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_afe_remove(struct platform_device *pdev)
+{
+ pr_debug("%s\n", __func__);
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_afe_driver = {
+ .driver = {
+ .name = "msm-pcm-afe",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_afe_probe,
+ .remove = __devexit_p(msm_afe_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ pr_debug("%s\n", __func__);
+ return platform_driver_register(&msm_afe_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ pr_debug("%s\n", __func__);
+ platform_driver_unregister(&msm_afe_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("AFE PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-afe-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-afe-v2.h
new file mode 100644
index 0000000..20d6377
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-afe-v2.h
@@ -0,0 +1,45 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef _MSM_PCM_AFE_H
+#define _MSM_PCM_AFE_H
+#include <sound/apr_audio-v2.h>
+#include <sound/q6afe-v2.h>
+
+
+struct pcm_afe_info {
+ unsigned long dma_addr;
+ struct snd_pcm_substream *substream;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ struct mutex lock;
+ spinlock_t dsp_lock;
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint8_t start;
+ uint32_t dsp_cnt;
+ uint32_t buf_phys;
+ int32_t mmap_flag;
+ int prepared;
+ struct hrtimer hrt;
+ int poll_time;
+};
+
+
+#define MSM_EXT(xname, fp_info, fp_get, fp_put, addr) \
+ {.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .name = xname, \
+ .info = fp_info,\
+ .get = fp_get, .put = fp_put, \
+ .private_value = addr, \
+ }
+
+#endif /*_MSM_PCM_AFE_H*/
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
new file mode 100644
index 0000000..ee92753
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
@@ -0,0 +1,609 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+#include <sound/snd_compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+#include <sound/timer.h>
+
+#include "msm-pcm-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct msm_audio *prtd;
+ unsigned volume;
+};
+static struct snd_msm lpa_audio;
+
+static struct snd_pcm_hardware msm_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = 2 * 1024 * 1024,
+ .period_bytes_min = 128 * 1024,
+ .period_bytes_max = 512 * 1024,
+ .periods_min = 4,
+ .periods_max = 16,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_aio_write_param param;
+ struct audio_buffer *buf = NULL;
+ unsigned long flag = 0;
+ int i = 0;
+
+ pr_debug("%s\n", __func__);
+ spin_lock_irqsave(&the_locks.event_lock, flag);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2: {
+ uint32_t *ptrmem = (uint32_t *)¶m;
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ else
+ if (substream->timer_running)
+ snd_timer_interrupt(substream->timer, 1);
+
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start)) {
+ atomic_set(&prtd->pending_buffer, 1);
+ break;
+ } else
+ atomic_set(&prtd->pending_buffer, 0);
+ if (runtime->status->hw_ptr >= runtime->control->appl_ptr)
+ break;
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+
+ buf = prtd->audio_client->port[IN].buf;
+ param.paddr = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count);
+ for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
+ i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1) & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ break;
+ }
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN_V2: {
+ if (!atomic_read(&prtd->pending_buffer))
+ break;
+ if (runtime->status->hw_ptr >=
+ runtime->control->appl_ptr)
+ break;
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__, prtd->pcm_count);
+ buf = prtd->audio_client->port[IN].buf;
+ param.paddr = (unsigned long)buf[prtd->out_head].phys;
+ param.len = prtd->pcm_count;
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.uid = (unsigned long)buf[prtd->out_head].phys;
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1)
+ & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ }
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ pr_debug("ASM_STREAM_CMD_FLUSH\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+ spin_unlock_irqrestore(&the_locks.event_lock, flag);
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ prtd->out_head = 0;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_pcm(prtd->audio_client, runtime->rate,
+ runtime->channels);
+ if (ret < 0)
+ pr_debug("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+ return 0;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->pcm_irq_pos = 0;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("SNDRV_PCM_TRIGGER_START\n");
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->start, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ runtime->hw = msm_pcm_hardware;
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_debug("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ return -EPERM;
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_debug("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ atomic_set(&prtd->pending_buffer, 1);
+ runtime->private_data = prtd;
+ lpa_audio.prtd = prtd;
+ lpa_set_volume(lpa_audio.volume);
+ ret = q6asm_set_softpause(lpa_audio.prtd->audio_client, &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(lpa_audio.prtd->audio_client, &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ return 0;
+}
+
+int lpa_set_volume(unsigned volume)
+{
+ int rc = 0;
+ if (lpa_audio.prtd && lpa_audio.prtd->audio_client) {
+ rc = q6asm_set_volume(lpa_audio.prtd->audio_client, volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed"
+ " rc=%d\n", __func__, rc);
+ }
+ }
+ lpa_audio.volume = volume;
+ return rc;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int rc = 0;
+
+ /*
+ If routing is still enabled, we need to issue EOS to
+ the DSP
+ To issue EOS to dsp, we need to be run state otherwise
+ EOS is not honored.
+ */
+ if (msm_routing_check_backend_enabled(soc_prtd->dai_link->be_id)) {
+ rc = q6asm_run(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->pending_buffer, 0);
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ pr_debug("%s\n", __func__);
+ rc = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (rc < 0)
+ pr_err("EOS cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ }
+
+ dir = IN;
+ atomic_set(&prtd->pending_buffer, 0);
+ lpa_audio.prtd = NULL;
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ pr_debug("%s\n", __func__);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ pr_debug("%s\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ return ret;
+}
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s: pcm_irq_pos = %d\n", __func__, prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ return -EPERM;
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed "
+ "rc = %d\n", ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+
+ if (buf == NULL || buf[0].data == NULL)
+ return -ENOMEM;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static int msm_pcm_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int rc = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ uint64_t timestamp;
+ uint64_t temp;
+
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP: {
+ struct snd_compr_tstamp tstamp;
+ pr_debug("SNDRV_COMPRESS_TSTAMP\n");
+
+ memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
+ timestamp = q6asm_get_session_time(prtd->audio_client);
+ if (timestamp < 0) {
+ pr_err("%s: Get Session Time return value =%lld\n",
+ __func__, timestamp);
+ return -EAGAIN;
+ }
+ temp = (timestamp * 2 * runtime->channels);
+ temp = temp * (runtime->rate/1000);
+ temp = div_u64(temp, 1000);
+ tstamp.sampling_rate = runtime->rate;
+ tstamp.rendered = (size_t)(temp & 0xFFFFFFFF);
+ tstamp.decoded = (size_t)((temp >> 32) & 0xFFFFFFFF);
+ tstamp.timestamp = timestamp;
+ pr_debug("%s: bytes_consumed:lsb = %d, msb = %d,"
+ "timestamp = %lld,\n",
+ __func__, tstamp.rendered, tstamp.decoded,
+ tstamp.timestamp);
+ if (copy_to_user((void *) arg, &tstamp,
+ sizeof(struct snd_compr_tstamp)))
+ return -EFAULT;
+ return 0;
+ }
+ case SNDRV_PCM_IOCTL1_RESET:
+ prtd->cmd_ack = 0;
+ rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ if (rc < 0)
+ pr_err("%s: flush cmd failed rc=%d\n", __func__, rc);
+ rc = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (rc < 0)
+ pr_err("Flush cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ break;
+ default:
+ break;
+ }
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = msm_pcm_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ dev_info(&pdev->dev, "%s: dev name %s\n",
+ __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-lpa",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ spin_lock_init(&the_locks.event_lock);
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
new file mode 100644
index 0000000..f94e6c1
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
@@ -0,0 +1,725 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/android_pmem.h>
+
+#include "msm-pcm-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+
+static struct audio_locks the_locks;
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+#define PLAYBACK_NUM_PERIODS 8
+#define PLAYBACK_PERIOD_SIZE 2048
+#define CAPTURE_NUM_PERIODS 16
+#define CAPTURE_PERIOD_SIZE 512
+
+static struct snd_pcm_hardware msm_pcm_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = CAPTURE_NUM_PERIODS * CAPTURE_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_PERIOD_SIZE,
+ .periods_min = CAPTURE_NUM_PERIODS,
+ .periods_max = CAPTURE_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = PLAYBACK_NUM_PERIODS * PLAYBACK_PERIOD_SIZE,
+ .period_bytes_min = PLAYBACK_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_PERIOD_SIZE,
+ .periods_min = PLAYBACK_NUM_PERIODS,
+ .periods_max = PLAYBACK_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+static uint32_t in_frame_info[CAPTURE_NUM_PERIODS][2];
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ pr_err("%s\n", __func__);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag)
+ break;
+ if (q6asm_is_cpu_buf_avail_nolock(IN,
+ prtd->audio_client,
+ &size, &idx)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ }
+ break;
+ }
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("ASM_DATA_EVENT_RENDERED_EOS\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE_V2\n");
+ pr_debug("token = 0x%08x\n", token);
+ in_frame_info[token][0] = payload[4];
+ in_frame_info[token][1] = payload[5];
+ prtd->pcm_irq_pos += in_frame_info[token][0];
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag
+ && q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx))
+ q6asm_read_nolock(prtd->audio_client);
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN_V2:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK) {
+ atomic_set(&prtd->start, 1);
+ break;
+ }
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes"
+ " of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ atomic_set(&prtd->start, 1);
+ break;
+ default:
+ pr_debug("%s:Payload = [0x%x]stat[0x%x]\n",
+ __func__, payload[0], payload[1]);
+ break;
+ }
+ }
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ ret = q6asm_media_format_block_pcm(prtd->audio_client, runtime->rate,
+ runtime->channels);
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ int ret = 0;
+ int i = 0;
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return 0;
+
+ pr_debug("Samp_rate = %d\n", prtd->samp_rate);
+ pr_debug("Channel = %d\n", prtd->channel_mode);
+ ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client, prtd->samp_rate,
+ prtd->channel_mode);
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+ prtd->periods = runtime->periods;
+
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ break;
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_pcm_hardware_playback;
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm out open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+ /* Capture path */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw = msm_pcm_hardware_capture;
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM);
+ if (ret < 0) {
+ pr_err("%s: pcm in open failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ }
+
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+ msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ prtd->cmd_ack = 1;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ prtd->dsp_cnt = 0;
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_err("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_err("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ xfer = fbytes;
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+fail:
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ ret = wait_event_timeout(the_locks.eos_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (ret < 0)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (ret < 0) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%08x\n",
+ (unsigned int) data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = in_frame_info[idx][1];
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ in_frame_info[idx][1] += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&in_frame_info[idx], 0,
+ sizeof(uint32_t) * 2);
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ int result = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ pr_debug("%s\n", __func__);
+ prtd->mmap_flag = 1;
+
+ if (runtime->dma_addr && runtime->dma_bytes) {
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ result = remap_pfn_range(vma, vma->vm_start,
+ runtime->dma_addr >> PAGE_SHIFT,
+ runtime->dma_bytes,
+ vma->vm_page_prot);
+ } else {
+ pr_err("Physical address or size of buf is NULL");
+ return -EINVAL;
+ }
+
+ return result;
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+pr_err("%s: before buf alloc\n", __func__);
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed "
+ "rc = %d\n", ret);
+ return -ENOMEM;
+ }
+pr_err("%s: after buf alloc\n", __func__);
+ buf = prtd->audio_client->port[dir].buf;
+ if (buf == NULL || buf[0].data == NULL)
+ return -ENOMEM;
+
+ pr_debug("%s:buf = %p\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = runtime->hw.buffer_bytes_max;
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static __devinit int msm_pcm_probe(struct platform_device *pdev)
+{
+ pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-dsp",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_pcm_probe,
+ .remove = __devexit_p(msm_pcm_remove),
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
new file mode 100644
index 0000000..44395b7
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2012 Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * See the GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, you can find it at http://www.fsf.org.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+
+
+
+/* Support unconventional sample rates 12000, 24000 as well */
+#define USE_RATE \
+ (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+
+extern int copy_count;
+
+struct buffer {
+ void *data;
+ unsigned size;
+ unsigned used;
+ unsigned addr;
+};
+
+struct buffer_rec {
+ void *data;
+ unsigned int size;
+ unsigned int read;
+ unsigned int addr;
+};
+
+struct audio_locks {
+ spinlock_t event_lock;
+ wait_queue_head_t read_wait;
+ wait_queue_head_t write_wait;
+ wait_queue_head_t eos_wait;
+ wait_queue_head_t enable_wait;
+};
+
+struct msm_audio {
+ struct snd_pcm_substream *substream;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ uint16_t source; /* Encoding source bit mask */
+
+ struct audio_client *audio_client;
+
+ uint16_t session_id;
+
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint32_t dsp_cnt;
+
+ int abort; /* set when error, like sample rate mismatch */
+
+ int enabled;
+ int close_ack;
+ int cmd_ack;
+ atomic_t start;
+ atomic_t out_count;
+ atomic_t in_count;
+ atomic_t out_needed;
+ int out_head;
+ int periods;
+ int mmap_flag;
+ atomic_t pending_buffer;
+};
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
new file mode 100644
index 0000000..2eebae5
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
@@ -0,0 +1,1834 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+#include <linux/bitops.h>
+#include <linux/mutex.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/q6adm-v2.h>
+#include <sound/q6asm-v2.h>
+#include <sound/q6afe-v2.h>
+#include <sound/tlv.h>
+#include "msm-pcm-routing-v2.h"
+#include "../qdsp6/q6voice.h"
+
+struct msm_pcm_routing_bdai_data {
+ u16 port_id; /* AFE port ID */
+ u8 active; /* track if this backend is enabled */
+ struct snd_pcm_hw_params *hw_params; /* to get freq and channel mode */
+ unsigned long fe_sessions; /* Front-end sessions */
+ unsigned long port_sessions; /* track Tx BE ports -> Rx BE */
+};
+
+#define INVALID_SESSION -1
+#define SESSION_TYPE_RX 0
+#define SESSION_TYPE_TX 1
+
+static struct mutex routing_lock;
+
+static int fm_switch_enable;
+
+#define INT_FM_RX_VOL_MAX_STEPS 100
+#define INT_FM_RX_VOL_GAIN 2000
+
+static int msm_route_fm_vol_control;
+static const DECLARE_TLV_DB_SCALE(fm_rx_vol_gain, 0,
+ INT_FM_RX_VOL_MAX_STEPS, 0);
+
+#define INT_RX_VOL_MAX_STEPS 100
+#define INT_RX_VOL_GAIN 0x2000
+
+static int msm_route_lpa_vol_control;
+static const DECLARE_TLV_DB_SCALE(lpa_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS, 0);
+
+static int msm_route_multimedia2_vol_control;
+static const DECLARE_TLV_DB_SCALE(multimedia2_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS, 0);
+
+static int msm_route_compressed_vol_control;
+static const DECLARE_TLV_DB_SCALE(compressed_rx_vol_gain, 0,
+ INT_RX_VOL_MAX_STEPS, 0);
+
+
+
+/* Equal to Frontend after last of the MULTIMEDIA SESSIONS */
+#define MAX_EQ_SESSIONS MSM_FRONTEND_DAI_CS_VOICE
+
+enum {
+ EQ_BAND1 = 0,
+ EQ_BAND2,
+ EQ_BAND3,
+ EQ_BAND4,
+ EQ_BAND5,
+ EQ_BAND6,
+ EQ_BAND7,
+ EQ_BAND8,
+ EQ_BAND9,
+ EQ_BAND10,
+ EQ_BAND11,
+ EQ_BAND12,
+ EQ_BAND_MAX,
+};
+
+struct msm_audio_eq_band {
+ uint16_t band_idx; /* The band index, 0 .. 11 */
+ uint32_t filter_type; /* Filter band type */
+ uint32_t center_freq_hz; /* Filter band center frequency */
+ uint32_t filter_gain; /* Filter band initial gain (dB) */
+ /* Range is +12 dB to -12 dB with 1dB increments. */
+ uint32_t q_factor;
+} __packed;
+
+struct msm_audio_eq_stream_config {
+ uint32_t enable; /* Number of consequtive bands specified */
+ uint32_t num_bands;
+ struct msm_audio_eq_band eq_bands[EQ_BAND_MAX];
+} __packed;
+
+struct msm_audio_eq_stream_config eq_data[MAX_EQ_SESSIONS];
+
+static void msm_send_eq_values(int eq_idx);
+/* This array is indexed by back-end DAI ID defined in msm-pcm-routing.h
+ * If new back-end is defined, add new back-end DAI ID at the end of enum
+ */
+static struct msm_pcm_routing_bdai_data msm_bedais[MSM_BACKEND_DAI_MAX] = {
+ { PRIMARY_I2S_RX, 0, NULL, 0, 0},
+ { PRIMARY_I2S_TX, 0, NULL, 0, 0},
+ { SLIMBUS_0_RX, 0, NULL, 0, 0},
+ { SLIMBUS_0_TX, 0, NULL, 0, 0},
+ { HDMI_RX, 0, NULL, 0, 0},
+ { INT_BT_SCO_RX, 0, NULL, 0, 0},
+ { INT_BT_SCO_TX, 0, NULL, 0, 0},
+ { INT_FM_RX, 0, NULL, 0, 0},
+ { INT_FM_TX, 0, NULL, 0, 0},
+ { RT_PROXY_PORT_001_RX, 0, NULL, 0, 0},
+ { RT_PROXY_PORT_001_TX, 0, NULL, 0, 0},
+ { PCM_RX, 0, NULL, 0, 0},
+ { PCM_TX, 0, NULL, 0, 0},
+ { VOICE_PLAYBACK_TX, 0, NULL, 0, 0},
+ { VOICE_RECORD_RX, 0, NULL, 0, 0},
+ { VOICE_RECORD_TX, 0, NULL, 0, 0},
+ { MI2S_RX, 0, NULL, 0, 0},
+ { SECONDARY_I2S_RX, 0, NULL, 0, 0},
+ { SLIMBUS_1_RX, 0, NULL, 0, 0},
+ { SLIMBUS_1_TX, 0, NULL, 0, 0},
+ { SLIMBUS_INVALID, 0, NULL, 0, 0},
+};
+
+
+/* Track ASM playback & capture sessions of DAI */
+static int fe_dai_map[MSM_FRONTEND_DAI_MM_SIZE][2] = {
+ /* MULTIMEDIA1 */
+ {INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA2 */
+ {INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA3 */
+ {INVALID_SESSION, INVALID_SESSION},
+ /* MULTIMEDIA4 */
+ {INVALID_SESSION, INVALID_SESSION},
+};
+
+static void msm_pcm_routing_build_matrix(int fedai_id, int dspst_id,
+ int path_type)
+{
+ int i, port_type;
+ struct route_payload payload;
+
+ payload.num_copps = 0;
+ port_type = (path_type == ADM_PATH_PLAYBACK ?
+ MSM_AFE_PORT_TYPE_RX : MSM_AFE_PORT_TYPE_TX);
+
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if ((afe_get_port_type(msm_bedais[i].port_id) ==
+ port_type) &&
+ msm_bedais[i].active && (test_bit(fedai_id,
+ &msm_bedais[i].fe_sessions)))
+ payload.copp_ids[payload.num_copps++] =
+ msm_bedais[i].port_id;
+ }
+
+ if (payload.num_copps)
+ adm_matrix_map(dspst_id, path_type,
+ payload.num_copps, payload.copp_ids, 0);
+}
+
+void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id, int stream_type)
+{
+ int i, session_type, path_type, port_type;
+ struct route_payload payload;
+ u32 channels;
+
+ if (fedai_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID %d\n", __func__, fedai_id);
+ return;
+ }
+
+ if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) {
+ session_type = SESSION_TYPE_RX;
+ path_type = ADM_PATH_PLAYBACK;
+ port_type = MSM_AFE_PORT_TYPE_RX;
+ } else {
+ session_type = SESSION_TYPE_TX;
+ path_type = ADM_PATH_LIVE_REC;
+ port_type = MSM_AFE_PORT_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ payload.num_copps = 0; /* only RX needs to use payload */
+ fe_dai_map[fedai_id][session_type] = dspst_id;
+ /* re-enable EQ if active */
+ if (eq_data[fedai_id].enable)
+ msm_send_eq_values(fedai_id);
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if ((afe_get_port_type(msm_bedais[i].port_id) ==
+ port_type) && msm_bedais[i].active &&
+ (test_bit(fedai_id,
+ &msm_bedais[i].fe_sessions))) {
+
+ channels = params_channels(msm_bedais[i].hw_params);
+
+ if ((stream_type == SNDRV_PCM_STREAM_PLAYBACK) &&
+ (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[i].port_id,
+ path_type,
+ params_rate(msm_bedais[i].hw_params),
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(msm_bedais[i].port_id,
+ path_type,
+ params_rate(msm_bedais[i].hw_params),
+ params_channels(msm_bedais[i].hw_params),
+ DEFAULT_COPP_TOPOLOGY);
+
+ payload.copp_ids[payload.num_copps++] =
+ msm_bedais[i].port_id;
+ }
+ }
+ if (payload.num_copps)
+ adm_matrix_map(dspst_id, path_type,
+ payload.num_copps, payload.copp_ids, 0);
+
+ mutex_unlock(&routing_lock);
+}
+
+void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type)
+{
+ int i, port_type, session_type;
+
+ if (fedai_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID\n", __func__);
+ return;
+ }
+
+ if (stream_type == SNDRV_PCM_STREAM_PLAYBACK) {
+ port_type = MSM_AFE_PORT_TYPE_RX;
+ session_type = SESSION_TYPE_RX;
+ } else {
+ port_type = MSM_AFE_PORT_TYPE_TX;
+ session_type = SESSION_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if ((afe_get_port_type(msm_bedais[i].port_id) ==
+ port_type) && msm_bedais[i].active &&
+ (test_bit(fedai_id,
+ &msm_bedais[i].fe_sessions)))
+ adm_close(msm_bedais[i].port_id);
+ }
+
+ fe_dai_map[fedai_id][session_type] = INVALID_SESSION;
+
+ mutex_unlock(&routing_lock);
+}
+
+/* Check if FE/BE route is set */
+static bool msm_pcm_routing_route_is_set(u16 be_id, u16 fe_id)
+{
+ bool rc = false;
+
+ if (fe_id > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* recheck FE ID in the mixer control defined in this file */
+ pr_err("%s: bad MM ID\n", __func__);
+ return rc;
+ }
+
+ if (test_bit(fe_id, &msm_bedais[be_id].fe_sessions))
+ rc = true;
+
+ return rc;
+}
+
+static void msm_pcm_routing_process_audio(u16 reg, u16 val, int set)
+{
+ int session_type, path_type;
+ u32 channels;
+
+ pr_debug("%s: reg %x val %x set %x\n", __func__, reg, val, set);
+
+ if (val > MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* recheck FE ID in the mixer control defined in this file */
+ pr_err("%s: bad MM ID\n", __func__);
+ return;
+ }
+
+ if (afe_get_port_type(msm_bedais[reg].port_id) ==
+ MSM_AFE_PORT_TYPE_RX) {
+ session_type = SESSION_TYPE_RX;
+ path_type = ADM_PATH_PLAYBACK;
+ } else {
+ session_type = SESSION_TYPE_TX;
+ path_type = ADM_PATH_LIVE_REC;
+ }
+
+ mutex_lock(&routing_lock);
+
+ if (set) {
+ set_bit(val, &msm_bedais[reg].fe_sessions);
+ if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
+ INVALID_SESSION) {
+
+ channels = params_channels(msm_bedais[reg].hw_params);
+
+ if ((session_type == SESSION_TYPE_RX) && (channels > 2))
+ adm_multi_ch_copp_open(msm_bedais[reg].port_id,
+ path_type,
+ params_rate(msm_bedais[reg].hw_params),
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(msm_bedais[reg].port_id,
+ path_type,
+ params_rate(msm_bedais[reg].hw_params),
+ params_channels(msm_bedais[reg].hw_params),
+ DEFAULT_COPP_TOPOLOGY);
+
+ msm_pcm_routing_build_matrix(val,
+ fe_dai_map[val][session_type], path_type);
+ }
+ } else {
+ clear_bit(val, &msm_bedais[reg].fe_sessions);
+ if (msm_bedais[reg].active && fe_dai_map[val][session_type] !=
+ INVALID_SESSION) {
+ adm_close(msm_bedais[reg].port_id);
+ msm_pcm_routing_build_matrix(val,
+ fe_dai_map[val][session_type], path_type);
+ }
+ }
+ mutex_unlock(&routing_lock);
+}
+
+static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ pr_info("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+
+ if (ucontrol->value.integer.value[0] &&
+ msm_pcm_routing_route_is_set(mc->reg, mc->shift) == false) {
+ msm_pcm_routing_process_audio(mc->reg, mc->shift, 1);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else if (!ucontrol->value.integer.value[0] &&
+ msm_pcm_routing_route_is_set(mc->reg, mc->shift) == true) {
+ msm_pcm_routing_process_audio(mc->reg, mc->shift, 0);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+ pr_info("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 1;
+}
+
+static void msm_pcm_routing_process_voice(u16 reg, u16 val, int set)
+{
+ return;
+}
+
+static int msm_routing_get_voice_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ mutex_lock(&routing_lock);
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ mutex_unlock(&routing_lock);
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_voice_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (ucontrol->value.integer.value[0]) {
+ msm_pcm_routing_process_voice(mc->reg, mc->shift, 1);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else {
+ msm_pcm_routing_process_voice(mc->reg, mc->shift, 0);
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+
+ return 1;
+}
+
+static int msm_routing_get_voice_stub_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ mutex_lock(&routing_lock);
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ mutex_unlock(&routing_lock);
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_voice_stub_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (ucontrol->value.integer.value[0]) {
+ mutex_lock(&routing_lock);
+ set_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions);
+ mutex_unlock(&routing_lock);
+
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else {
+ mutex_lock(&routing_lock);
+ clear_bit(mc->shift, &msm_bedais[mc->reg].fe_sessions);
+ mutex_unlock(&routing_lock);
+
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 1;
+}
+
+static int msm_routing_get_switch_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = fm_switch_enable;
+ pr_debug("%s: FM Switch enable %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+ return 0;
+}
+
+static int msm_routing_put_switch_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+
+ pr_debug("%s: FM Switch enable %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+ if (ucontrol->value.integer.value[0])
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ else
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ fm_switch_enable = ucontrol->value.integer.value[0];
+ return 1;
+}
+
+static int msm_routing_get_port_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ if (test_bit(mc->shift, &msm_bedais[mc->reg].port_sessions))
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg, mc->shift,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_routing_put_port_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pr_debug("%s: reg %x shift %x val %ld\n", __func__, mc->reg,
+ mc->shift, ucontrol->value.integer.value[0]);
+
+ if (ucontrol->value.integer.value[0]) {
+ afe_loopback(1, msm_bedais[mc->reg].port_id,
+ msm_bedais[mc->shift].port_id);
+ set_bit(mc->shift,
+ &msm_bedais[mc->reg].port_sessions);
+ } else {
+ afe_loopback(0, msm_bedais[mc->reg].port_id,
+ msm_bedais[mc->shift].port_id);
+ clear_bit(mc->shift,
+ &msm_bedais[mc->reg].port_sessions);
+ }
+
+ return 1;
+}
+
+static int msm_routing_get_fm_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm_route_fm_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_fm_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ afe_loopback_gain(INT_FM_TX , ucontrol->value.integer.value[0]);
+
+ msm_route_fm_vol_control = ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static int msm_routing_get_lpa_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = msm_route_lpa_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_lpa_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (!lpa_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_lpa_vol_control =
+ ucontrol->value.integer.value[0];
+
+ return 0;
+
+}
+
+static int msm_routing_get_multimedia2_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ ucontrol->value.integer.value[0] = msm_route_multimedia2_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_multimedia2_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (!multi_ch_pcm_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_multimedia2_vol_control =
+ ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static int msm_routing_get_compressed_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+
+ ucontrol->value.integer.value[0] = msm_route_compressed_vol_control;
+ return 0;
+}
+
+static int msm_routing_set_compressed_vol_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ if (!compressed_set_volume(ucontrol->value.integer.value[0]))
+ msm_route_compressed_vol_control =
+ ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static void msm_send_eq_values(int eq_idx)
+{
+ int result;
+ struct audio_client *ac =
+ q6asm_get_audio_client(fe_dai_map[eq_idx][SESSION_TYPE_RX]);
+
+ if (ac == NULL) {
+ pr_err("%s: Could not get audio client for session: %d\n",
+ __func__, fe_dai_map[eq_idx][SESSION_TYPE_RX]);
+ goto done;
+ }
+
+ result = q6asm_equalizer(ac, &eq_data[eq_idx]);
+
+ if (result < 0)
+ pr_err("%s: Call to ASM equalizer failed, returned = %d\n",
+ __func__, result);
+done:
+ return;
+}
+
+static int msm_routing_get_eq_enable_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+
+ ucontrol->value.integer.value[0] = eq_data[eq_idx].enable;
+
+ pr_debug("%s: EQ #%d enable %d\n", __func__,
+ eq_idx, eq_data[eq_idx].enable);
+ return 0;
+}
+
+static int msm_routing_put_eq_enable_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int value = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: EQ #%d enable %d\n", __func__,
+ eq_idx, value);
+ eq_data[eq_idx].enable = value;
+
+ msm_send_eq_values(eq_idx);
+ return 0;
+}
+
+static int msm_routing_get_eq_band_count_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+
+ ucontrol->value.integer.value[0] = eq_data[eq_idx].num_bands;
+
+ pr_debug("%s: EQ #%d bands %d\n", __func__,
+ eq_idx, eq_data[eq_idx].num_bands);
+ return eq_data[eq_idx].num_bands;
+}
+
+static int msm_routing_put_eq_band_count_audio_mixer(
+ struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int value = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: EQ #%d bands %d\n", __func__,
+ eq_idx, value);
+ eq_data[eq_idx].num_bands = value;
+ return 0;
+}
+
+static int msm_routing_get_eq_band_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ ucontrol->value.integer.value[0] =
+ eq_data[eq_idx].eq_bands[band_idx].band_idx;
+ ucontrol->value.integer.value[1] =
+ eq_data[eq_idx].eq_bands[band_idx].filter_type;
+ ucontrol->value.integer.value[2] =
+ eq_data[eq_idx].eq_bands[band_idx].center_freq_hz;
+ ucontrol->value.integer.value[3] =
+ eq_data[eq_idx].eq_bands[band_idx].filter_gain;
+ ucontrol->value.integer.value[4] =
+ eq_data[eq_idx].eq_bands[band_idx].q_factor;
+
+ pr_debug("%s: band_idx = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].band_idx);
+ pr_debug("%s: filter_type = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].filter_type);
+ pr_debug("%s: center_freq_hz = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].center_freq_hz);
+ pr_debug("%s: filter_gain = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].filter_gain);
+ pr_debug("%s: q_factor = %d\n", __func__,
+ eq_data[eq_idx].eq_bands[band_idx].q_factor);
+ return 0;
+}
+
+static int msm_routing_put_eq_band_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int eq_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->reg;
+ int band_idx = ((struct soc_multi_mixer_control *)
+ kcontrol->private_value)->shift;
+
+ eq_data[eq_idx].eq_bands[band_idx].band_idx =
+ ucontrol->value.integer.value[0];
+ eq_data[eq_idx].eq_bands[band_idx].filter_type =
+ ucontrol->value.integer.value[1];
+ eq_data[eq_idx].eq_bands[band_idx].center_freq_hz =
+ ucontrol->value.integer.value[2];
+ eq_data[eq_idx].eq_bands[band_idx].filter_gain =
+ ucontrol->value.integer.value[3];
+ eq_data[eq_idx].eq_bands[band_idx].q_factor =
+ ucontrol->value.integer.value[4];
+ return 0;
+}
+
+static const struct snd_kcontrol_new pri_i2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_PRI_I2S_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_SEC_I2S_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_SLIMBUS_0_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_MI2S_RX ,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+ /* incall music delivery mixer */
+static const struct snd_kcontrol_new incall_music_delivery_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul2_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_PRI_I2S_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new sec_i2s_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_SLIMBUS_0_RX ,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new bt_sco_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_INT_BT_SCO_RX ,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new afe_pcm_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new aux_pcm_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new hdmi_rx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("Voip", MSM_BACKEND_DAI_HDMI_RX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new stub_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_INVALID,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_1_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("Voice Stub", MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new tx_voice_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX_Voice", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX_Voice", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX_Voice",
+ MSM_BACKEND_DAI_INT_BT_SCO_TX, MSM_FRONTEND_DAI_CS_VOICE, 1, 0,
+ msm_routing_get_voice_mixer, msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX_Voice", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_TX_Voice", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new tx_voip_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_TX_Voip", MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX_Voip", MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX_Voip", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AFE_PCM_TX_Voip", MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_TX_Voip", MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_FRONTEND_DAI_VOIP, 1, 0, msm_routing_get_voice_mixer,
+ msm_routing_put_voice_mixer),
+};
+
+static const struct snd_kcontrol_new tx_voice_stub_mixer_controls[] = {
+ SOC_SINGLE_EXT("STUB_TX_HL", MSM_BACKEND_DAI_INVALID,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+ SOC_SINGLE_EXT("SLIM_1_TX", MSM_BACKEND_DAI_SLIMBUS_1_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB, 1, 0, msm_routing_get_voice_stub_mixer,
+ msm_routing_put_voice_stub_mixer),
+};
+
+static const struct snd_kcontrol_new sbus_0_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_INT_FM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_SLIMBUS_0_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new auxpcm_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("AUX_PCM_UL_TX", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_BACKEND_DAI_AUXPCM_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+ SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_BACKEND_DAI_SLIMBUS_0_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new sbus_1_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new bt_sco_rx_port_mixer_controls[] = {
+ SOC_SINGLE_EXT("SLIM_1_TX", MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_TX, 1, 0, msm_routing_get_port_mixer,
+ msm_routing_put_port_mixer),
+};
+
+static const struct snd_kcontrol_new fm_switch_mixer_controls =
+ SOC_SINGLE_EXT("Switch", SND_SOC_NOPM,
+ 0, 1, 0, msm_routing_get_switch_mixer,
+ msm_routing_put_switch_mixer);
+
+static const struct snd_kcontrol_new int_fm_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("Internal FM RX Volume", SND_SOC_NOPM, 0,
+ INT_FM_RX_VOL_GAIN, 0, msm_routing_get_fm_vol_mixer,
+ msm_routing_set_fm_vol_mixer, fm_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new lpa_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("LPA RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_lpa_vol_mixer,
+ msm_routing_set_lpa_vol_mixer, lpa_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new multimedia2_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("HIFI2 RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_multimedia2_vol_mixer,
+ msm_routing_set_multimedia2_vol_mixer, multimedia2_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new compressed_vol_mixer_controls[] = {
+ SOC_SINGLE_EXT_TLV("COMPRESSED RX Volume", SND_SOC_NOPM, 0,
+ INT_RX_VOL_GAIN, 0, msm_routing_get_compressed_vol_mixer,
+ msm_routing_set_compressed_vol_mixer, compressed_rx_vol_gain),
+};
+
+static const struct snd_kcontrol_new eq_enable_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1 EQ Enable", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_eq_enable_mixer,
+ msm_routing_put_eq_enable_mixer),
+ SOC_SINGLE_EXT("MultiMedia2 EQ Enable", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_eq_enable_mixer,
+ msm_routing_put_eq_enable_mixer),
+ SOC_SINGLE_EXT("MultiMedia3 EQ Enable", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_eq_enable_mixer,
+ msm_routing_put_eq_enable_mixer),
+};
+
+static const struct snd_kcontrol_new eq_band_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1 EQ Band Count", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 11, 0,
+ msm_routing_get_eq_band_count_audio_mixer,
+ msm_routing_put_eq_band_count_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2 EQ Band Count", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 11, 0,
+ msm_routing_get_eq_band_count_audio_mixer,
+ msm_routing_put_eq_band_count_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3 EQ Band Count", SND_SOC_NOPM,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 11, 0,
+ msm_routing_get_eq_band_count_audio_mixer,
+ msm_routing_put_eq_band_count_audio_mixer),
+};
+
+static const struct snd_kcontrol_new eq_coeff_mixer_controls[] = {
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band1", EQ_BAND1,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band2", EQ_BAND2,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band3", EQ_BAND3,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band4", EQ_BAND4,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band5", EQ_BAND5,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band6", EQ_BAND6,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band7", EQ_BAND7,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band8", EQ_BAND8,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band9", EQ_BAND9,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band10", EQ_BAND10,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band11", EQ_BAND11,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia1 EQ Band12", EQ_BAND12,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band1", EQ_BAND1,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band2", EQ_BAND2,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band3", EQ_BAND3,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band4", EQ_BAND4,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band5", EQ_BAND5,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band6", EQ_BAND6,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band7", EQ_BAND7,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band8", EQ_BAND8,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band9", EQ_BAND9,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band10", EQ_BAND10,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band11", EQ_BAND11,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia2 EQ Band12", EQ_BAND12,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band1", EQ_BAND1,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band2", EQ_BAND2,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band3", EQ_BAND3,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band4", EQ_BAND4,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band5", EQ_BAND5,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band6", EQ_BAND6,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band7", EQ_BAND7,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band8", EQ_BAND8,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band9", EQ_BAND9,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band10", EQ_BAND10,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band11", EQ_BAND11,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+ SOC_SINGLE_MULTI_EXT("MultiMedia3 EQ Band12", EQ_BAND12,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 255, 0, 5,
+ msm_routing_get_eq_band_audio_mixer,
+ msm_routing_put_eq_band_audio_mixer),
+};
+
+static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
+ /* Frontend AIF */
+ /* Widget name equals to Front-End DAI name<Need confirmation>,
+ * Stream name must contains substring of front-end dai name
+ */
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("VOIP_DL", "VoIP Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("VOIP_UL", "VoIP Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIM0_DL_HL", "SLIMBUS0_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIM0_UL_HL", "SLIMBUS0_HOSTLESS Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("INTFM_DL_HL", "INT_FM_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("INTFM_UL_HL", "INT_FM_HOSTLESS Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("HDMI_DL_HL", "HDMI_HOSTLESS Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AUXPCM_DL_HL", "AUXPCM_HOSTLESS Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AUXPCM_UL_HL", "AUXPCM_HOSTLESS Capture",
+ 0, 0, 0, 0),
+
+ /* Backend AIF */
+ /* Stream name equals to backend dai link stream name
+ */
+ SND_SOC_DAPM_AIF_OUT("PRI_I2S_RX", "Primary I2S Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SEC_I2S_RX", "Secondary I2S Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("HDMI", "HDMI Playback", 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_OUT("MI2S_RX", "MI2S Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("PRI_I2S_TX", "Primary I2S Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_0_TX", "Slimbus Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("INT_BT_SCO_RX", "Internal BT-SCO Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INT_BT_SCO_TX", "Internal BT-SCO Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("INT_FM_RX", "Internal FM Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INT_FM_TX", "Internal FM Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("PCM_RX", "AFE Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("PCM_TX", "AFE Capture",
+ 0, 0, 0 , 0),
+ /* incall */
+ SND_SOC_DAPM_AIF_OUT("VOICE_PLAYBACK_TX", "Voice Farend Playback",
+ 0, 0, 0 , 0),
+ SND_SOC_DAPM_AIF_IN("INCALL_RECORD_TX", "Voice Uplink Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("INCALL_RECORD_RX", "Voice Downlink Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("AUX_PCM_RX", "AUX PCM Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("AUX_PCM_TX", "AUX PCM Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("VOICE_STUB_DL", "VOICE_STUB Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("VOICE_STUB_UL", "VOICE_STUB Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("STUB_RX", "Stub Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("STUB_TX", "Stub Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_1_TX", "Slimbus1 Capture", 0, 0, 0, 0),
+
+ /* Switch Definitions */
+ SND_SOC_DAPM_SWITCH("SLIMBUS_DL_HL", SND_SOC_NOPM, 0, 0,
+ &fm_switch_mixer_controls),
+ /* Mixer definitions */
+ SND_SOC_DAPM_MIXER("PRI_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ pri_i2s_rx_mixer_controls, ARRAY_SIZE(pri_i2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ sec_i2s_rx_mixer_controls, ARRAY_SIZE(sec_i2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_rx_mixer_controls, ARRAY_SIZE(slimbus_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
+ hdmi_mixer_controls, ARRAY_SIZE(hdmi_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ mi2s_rx_mixer_controls, ARRAY_SIZE(mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul2_mixer_controls, ARRAY_SIZE(mmul2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)),
+ /* incall */
+ SND_SOC_DAPM_MIXER("Incall_Music Audio Mixer", SND_SOC_NOPM, 0, 0,
+ incall_music_delivery_mixer_controls,
+ ARRAY_SIZE(incall_music_delivery_mixer_controls)),
+ /* Voice Mixer */
+ SND_SOC_DAPM_MIXER("PRI_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0, pri_rx_voice_mixer_controls,
+ ARRAY_SIZE(pri_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ sec_i2s_rx_voice_mixer_controls,
+ ARRAY_SIZE(sec_i2s_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIM_0_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ slimbus_rx_voice_mixer_controls,
+ ARRAY_SIZE(slimbus_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_BT_SCO_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ bt_sco_rx_voice_mixer_controls,
+ ARRAY_SIZE(bt_sco_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AFE_PCM_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ afe_pcm_rx_voice_mixer_controls,
+ ARRAY_SIZE(afe_pcm_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AUX_PCM_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ aux_pcm_rx_voice_mixer_controls,
+ ARRAY_SIZE(aux_pcm_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("HDMI_RX_Voice Mixer",
+ SND_SOC_NOPM, 0, 0,
+ hdmi_rx_voice_mixer_controls,
+ ARRAY_SIZE(hdmi_rx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Voice_Tx Mixer",
+ SND_SOC_NOPM, 0, 0, tx_voice_mixer_controls,
+ ARRAY_SIZE(tx_voice_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Voip_Tx Mixer",
+ SND_SOC_NOPM, 0, 0, tx_voip_mixer_controls,
+ ARRAY_SIZE(tx_voip_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_BT_SCO_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ int_bt_sco_rx_mixer_controls, ARRAY_SIZE(int_bt_sco_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_FM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ int_fm_rx_mixer_controls, ARRAY_SIZE(int_fm_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AFE_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ afe_pcm_rx_mixer_controls, ARRAY_SIZE(afe_pcm_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Voice Stub Tx Mixer", SND_SOC_NOPM, 0, 0,
+ tx_voice_stub_mixer_controls, ARRAY_SIZE(tx_voice_stub_mixer_controls)),
+ SND_SOC_DAPM_MIXER("STUB_RX Mixer", SND_SOC_NOPM, 0, 0,
+ stub_rx_mixer_controls, ARRAY_SIZE(stub_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_1_RX Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_1_rx_mixer_controls, ARRAY_SIZE(slimbus_1_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, sbus_0_rx_port_mixer_controls,
+ ARRAY_SIZE(sbus_0_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("AUXPCM_RX Port Mixer",
+ SND_SOC_NOPM, 0, 0, auxpcm_rx_port_mixer_controls,
+ ARRAY_SIZE(auxpcm_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_1_RX Port Mixer", SND_SOC_NOPM, 0, 0,
+ sbus_1_rx_port_mixer_controls,
+ ARRAY_SIZE(sbus_1_rx_port_mixer_controls)),
+ SND_SOC_DAPM_MIXER("INTERNAL_BT_SCO_RX Port Mixer", SND_SOC_NOPM, 0, 0,
+ bt_sco_rx_port_mixer_controls,
+ ARRAY_SIZE(bt_sco_rx_port_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"PRI_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"PRI_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"PRI_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"PRI_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PRI_I2S_RX", NULL, "PRI_RX Audio Mixer"},
+
+ {"SEC_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SEC_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SEC_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SEC_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SEC_I2S_RX", NULL, "SEC_RX Audio Mixer"},
+
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"},
+
+ {"HDMI Mixer", "MultiMedia1", "MM_DL1"},
+ {"HDMI Mixer", "MultiMedia2", "MM_DL2"},
+ {"HDMI Mixer", "MultiMedia3", "MM_DL3"},
+ {"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+ {"HDMI", NULL, "HDMI Mixer"},
+
+ /* incall */
+ {"Incall_Music Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"Incall_Music Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"VOICE_PLAYBACK_TX", NULL, "Incall_Music Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "VOC_REC_UL", "INCALL_RECORD_TX"},
+ {"MultiMedia1 Mixer", "VOC_REC_DL", "INCALL_RECORD_RX"},
+ {"MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"MI2S_RX", NULL, "MI2S_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"},
+ {"MultiMedia1 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+ {"MultiMedia1 Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Audio Mixer"},
+
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"INTERNAL_FM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"INT_FM_RX", NULL, "INTERNAL_FM_RX Audio Mixer"},
+
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"AFE_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PCM_RX", NULL, "AFE_PCM_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+
+ {"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"},
+ {"MM_UL1", NULL, "MultiMedia1 Mixer"},
+ {"MultiMedia2 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+ {"MM_UL2", NULL, "MultiMedia2 Mixer"},
+
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"AUX_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"AUX_PCM_RX", NULL, "AUX_PCM_RX Audio Mixer"},
+
+ {"PRI_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"PRI_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"PRI_I2S_RX", NULL, "PRI_RX_Voice Mixer"},
+
+ {"SEC_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"SEC_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"SEC_I2S_RX", NULL, "SEC_RX_Voice Mixer"},
+
+ {"SLIM_0_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"SLIM_0_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"SLIMBUS_0_RX", NULL, "SLIM_0_RX_Voice Mixer"},
+
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX_Voice Mixer"},
+
+ {"AFE_PCM_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"AFE_PCM_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"PCM_RX", NULL, "AFE_PCM_RX_Voice Mixer"},
+
+ {"AUX_PCM_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"AUX_PCM_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"AUX_PCM_RX", NULL, "AUX_PCM_RX_Voice Mixer"},
+
+ {"HDMI_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},
+ {"HDMI_RX_Voice Mixer", "Voip", "VOIP_DL"},
+ {"HDMI", NULL, "HDMI_RX_Voice Mixer"},
+ {"HDMI", NULL, "HDMI_DL_HL"},
+
+ {"Voice_Tx Mixer", "PRI_TX_Voice", "PRI_I2S_TX"},
+ {"Voice_Tx Mixer", "SLIM_0_TX_Voice", "SLIMBUS_0_TX"},
+ {"Voice_Tx Mixer", "INTERNAL_BT_SCO_TX_Voice", "INT_BT_SCO_TX"},
+ {"Voice_Tx Mixer", "AFE_PCM_TX_Voice", "PCM_TX"},
+ {"Voice_Tx Mixer", "AUX_PCM_TX_Voice", "AUX_PCM_TX"},
+ {"CS-VOICE_UL1", NULL, "Voice_Tx Mixer"},
+ {"Voip_Tx Mixer", "PRI_TX_Voip", "PRI_I2S_TX"},
+ {"Voip_Tx Mixer", "SLIM_0_TX_Voip", "SLIMBUS_0_TX"},
+ {"Voip_Tx Mixer", "INTERNAL_BT_SCO_TX_Voip", "INT_BT_SCO_TX"},
+ {"Voip_Tx Mixer", "AFE_PCM_TX_Voip", "PCM_TX"},
+ {"Voip_Tx Mixer", "AUX_PCM_TX_Voip", "AUX_PCM_TX"},
+
+ {"VOIP_UL", NULL, "Voip_Tx Mixer"},
+ {"SLIMBUS_DL_HL", "Switch", "SLIM0_DL_HL"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_DL_HL"},
+ {"SLIM0_UL_HL", NULL, "SLIMBUS_0_TX"},
+ {"INT_FM_RX", NULL, "INTFM_DL_HL"},
+ {"INTFM_UL_HL", NULL, "INT_FM_TX"},
+ {"AUX_PCM_RX", NULL, "AUXPCM_DL_HL"},
+ {"AUXPCM_UL_HL", NULL, "AUX_PCM_TX"},
+ {"SLIMBUS_0_RX Port Mixer", "INTERNAL_FM_TX", "INT_FM_TX"},
+ {"SLIMBUS_0_RX Port Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+ {"SLIMBUS_0_RX Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Port Mixer"},
+
+ {"AUXPCM_RX Port Mixer", "AUX_PCM_UL_TX", "AUX_PCM_TX"},
+ {"AUXPCM_RX Port Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"},
+ {"AUX_PCM_RX", NULL, "AUXPCM_RX Port Mixer"},
+
+ {"Voice Stub Tx Mixer", "STUB_TX_HL", "STUB_TX"},
+ {"Voice Stub Tx Mixer", "SLIM_1_TX", "SLIMBUS_1_TX"},
+ {"Voice Stub Tx Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"VOICE_STUB_UL", NULL, "Voice Stub Tx Mixer"},
+
+ {"STUB_RX Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"STUB_RX", NULL, "STUB_RX Mixer"},
+ {"SLIMBUS_1_RX Mixer", "Voice Stub", "VOICE_STUB_DL"},
+ {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Mixer"},
+ {"INTERNAL_BT_SCO_RX_Voice Mixer", "Voice Stub", "VOICE_STUB_DL"},
+
+ {"SLIMBUS_1_RX Port Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
+ {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Port Mixer"},
+ {"INTERNAL_BT_SCO_RX Port Mixer", "SLIM_1_TX", "SLIMBUS_1_TX"},
+ {"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Port Mixer"},
+};
+
+static int msm_pcm_routing_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->dai_link->be_id;
+
+ if (be_id >= MSM_BACKEND_DAI_MAX) {
+ pr_err("%s: unexpected be_id %d\n", __func__, be_id);
+ return -EINVAL;
+ }
+
+ mutex_lock(&routing_lock);
+ msm_bedais[be_id].hw_params = params;
+ mutex_unlock(&routing_lock);
+ return 0;
+}
+
+static int msm_pcm_routing_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->dai_link->be_id;
+ int i, session_type;
+ struct msm_pcm_routing_bdai_data *bedai;
+
+ if (be_id >= MSM_BACKEND_DAI_MAX) {
+ pr_err("%s: unexpected be_id %d\n", __func__, be_id);
+ return -EINVAL;
+ }
+
+ bedai = &msm_bedais[be_id];
+
+ session_type = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ 0 : 1);
+
+ mutex_lock(&routing_lock);
+
+ for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
+ if (fe_dai_map[i][session_type] != INVALID_SESSION)
+ adm_close(bedai->port_id);
+ }
+
+ bedai->active = 0;
+ bedai->hw_params = NULL;
+
+ mutex_unlock(&routing_lock);
+
+ return 0;
+}
+
+static int msm_pcm_routing_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->dai_link->be_id;
+ int i, path_type, session_type;
+ struct msm_pcm_routing_bdai_data *bedai;
+ u32 channels;
+
+ if (be_id >= MSM_BACKEND_DAI_MAX) {
+ pr_err("%s: unexpected be_id %d\n", __func__, be_id);
+ return -EINVAL;
+ }
+
+
+ bedai = &msm_bedais[be_id];
+
+ if (bedai->hw_params == NULL) {
+ pr_err("%s: HW param is not configured", __func__);
+ return -EINVAL;
+ }
+
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ path_type = ADM_PATH_PLAYBACK;
+ session_type = SESSION_TYPE_RX;
+ } else {
+ path_type = ADM_PATH_LIVE_REC;
+ session_type = SESSION_TYPE_TX;
+ }
+
+ mutex_lock(&routing_lock);
+
+ if (bedai->active == 1)
+ goto done; /* Ignore prepare if back-end already active */
+
+ /* AFE port is not active at this point. However, still
+ * go ahead setting active flag under the notion that
+ * QDSP6 is able to handle ADM starting before AFE port
+ * is started.
+ */
+ bedai->active = 1;
+
+ for_each_set_bit(i, &bedai->fe_sessions, MSM_FRONTEND_DAI_MM_SIZE) {
+ if (fe_dai_map[i][session_type] != INVALID_SESSION) {
+
+ channels = params_channels(bedai->hw_params);
+ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) &&
+ (channels > 2))
+ adm_multi_ch_copp_open(bedai->port_id,
+ path_type,
+ params_rate(bedai->hw_params),
+ channels,
+ DEFAULT_COPP_TOPOLOGY);
+ else
+ adm_open(bedai->port_id,
+ path_type,
+ params_rate(bedai->hw_params),
+ params_channels(bedai->hw_params),
+ DEFAULT_COPP_TOPOLOGY);
+
+ msm_pcm_routing_build_matrix(i,
+ fe_dai_map[i][session_type], path_type);
+ }
+ }
+
+done:
+ mutex_unlock(&routing_lock);
+
+ return 0;
+}
+
+static struct snd_pcm_ops msm_routing_pcm_ops = {
+ .hw_params = msm_pcm_routing_hw_params,
+ .close = msm_pcm_routing_close,
+ .prepare = msm_pcm_routing_prepare,
+};
+
+static unsigned int msm_routing_read(struct snd_soc_platform *platform,
+ unsigned int reg)
+{
+ dev_dbg(platform->dev, "reg %x\n", reg);
+ return 0;
+}
+
+/* Not used but frame seems to require it */
+static int msm_routing_write(struct snd_soc_platform *platform,
+ unsigned int reg, unsigned int val)
+{
+ dev_dbg(platform->dev, "reg %x val %x\n", reg, val);
+ return 0;
+}
+
+/* Not used but frame seems to require it */
+static int msm_routing_probe(struct snd_soc_platform *platform)
+{
+ snd_soc_dapm_new_controls(&platform->dapm, msm_qdsp6_widgets,
+ ARRAY_SIZE(msm_qdsp6_widgets));
+ snd_soc_dapm_add_routes(&platform->dapm, intercon,
+ ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(&platform->dapm);
+
+ snd_soc_add_platform_controls(platform,
+ int_fm_vol_mixer_controls,
+ ARRAY_SIZE(int_fm_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ lpa_vol_mixer_controls,
+ ARRAY_SIZE(lpa_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ eq_enable_mixer_controls,
+ ARRAY_SIZE(eq_enable_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ eq_band_mixer_controls,
+ ARRAY_SIZE(eq_band_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ eq_coeff_mixer_controls,
+ ARRAY_SIZE(eq_coeff_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ multimedia2_vol_mixer_controls,
+ ARRAY_SIZE(multimedia2_vol_mixer_controls));
+
+ snd_soc_add_platform_controls(platform,
+ compressed_vol_mixer_controls,
+ ARRAY_SIZE(compressed_vol_mixer_controls));
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver msm_soc_routing_platform = {
+ .ops = &msm_routing_pcm_ops,
+ .probe = msm_routing_probe,
+ .read = msm_routing_read,
+ .write = msm_routing_write,
+};
+
+static __devinit int msm_routing_pcm_probe(struct platform_device *pdev)
+{
+ dev_dbg(&pdev->dev, "dev name %s\n", dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_routing_platform);
+}
+
+static int msm_routing_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver msm_routing_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-routing",
+ .owner = THIS_MODULE,
+ },
+ .probe = msm_routing_pcm_probe,
+ .remove = __devexit_p(msm_routing_pcm_remove),
+};
+
+int msm_routing_check_backend_enabled(int fedai_id)
+{
+ int i;
+ if (fedai_id >= MSM_FRONTEND_DAI_MM_MAX_ID) {
+ /* bad ID assigned in machine driver */
+ pr_err("%s: bad MM ID\n", __func__);
+ return 0;
+ }
+ for (i = 0; i < MSM_BACKEND_DAI_MAX; i++) {
+ if ((test_bit(fedai_id,
+ &msm_bedais[i].fe_sessions))) {
+ return msm_bedais[i].active;
+ }
+ }
+ return 0;
+}
+
+static int __init msm_soc_routing_platform_init(void)
+{
+ mutex_init(&routing_lock);
+ return platform_driver_register(&msm_routing_pcm_driver);
+}
+module_init(msm_soc_routing_platform_init);
+
+static void __exit msm_soc_routing_platform_exit(void)
+{
+ platform_driver_unregister(&msm_routing_pcm_driver);
+}
+module_exit(msm_soc_routing_platform_exit);
+
+MODULE_DESCRIPTION("MSM routing platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h
new file mode 100644
index 0000000..b971787
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h
@@ -0,0 +1,103 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+#ifndef _MSM_PCM_ROUTING_H
+#define _MSM_PCM_ROUTING_H
+#include <sound/apr_audio-v2.h>
+
+#define LPASS_BE_PRI_I2S_RX "(Backend) PRIMARY_I2S_RX"
+#define LPASS_BE_PRI_I2S_TX "(Backend) PRIMARY_I2S_TX"
+#define LPASS_BE_SLIMBUS_0_RX "(Backend) SLIMBUS_0_RX"
+#define LPASS_BE_SLIMBUS_0_TX "(Backend) SLIMBUS_0_TX"
+#define LPASS_BE_HDMI "(Backend) HDMI"
+#define LPASS_BE_INT_BT_SCO_RX "(Backend) INT_BT_SCO_RX"
+#define LPASS_BE_INT_BT_SCO_TX "(Backend) INT_BT_SCO_TX"
+#define LPASS_BE_INT_FM_RX "(Backend) INT_FM_RX"
+#define LPASS_BE_INT_FM_TX "(Backend) INT_FM_TX"
+#define LPASS_BE_AFE_PCM_RX "(Backend) RT_PROXY_DAI_001_RX"
+#define LPASS_BE_AFE_PCM_TX "(Backend) RT_PROXY_DAI_002_TX"
+#define LPASS_BE_AUXPCM_RX "(Backend) AUX_PCM_RX"
+#define LPASS_BE_AUXPCM_TX "(Backend) AUX_PCM_TX"
+#define LPASS_BE_VOICE_PLAYBACK_TX "(Backend) VOICE_PLAYBACK_TX"
+#define LPASS_BE_INCALL_RECORD_RX "(Backend) INCALL_RECORD_TX"
+#define LPASS_BE_INCALL_RECORD_TX "(Backend) INCALL_RECORD_RX"
+#define LPASS_BE_SEC_I2S_RX "(Backend) SECONDARY_I2S_RX"
+
+#define LPASS_BE_MI2S_RX "(Backend) MI2S_RX"
+#define LPASS_BE_STUB_RX "(Backend) STUB_RX"
+#define LPASS_BE_STUB_TX "(Backend) STUB_TX"
+#define LPASS_BE_SLIMBUS_1_RX "(Backend) SLIMBUS_1_RX"
+#define LPASS_BE_SLIMBUS_1_TX "(Backend) SLIMBUS_1_TX"
+
+/* For multimedia front-ends, asm session is allocated dynamically.
+ * Hence, asm session/multimedia front-end mapping has to be maintained.
+ * Due to this reason, additional multimedia front-end must be placed before
+ * non-multimedia front-ends.
+ */
+
+enum {
+ MSM_FRONTEND_DAI_MULTIMEDIA1 = 0,
+ MSM_FRONTEND_DAI_MULTIMEDIA2,
+ MSM_FRONTEND_DAI_MULTIMEDIA3,
+ MSM_FRONTEND_DAI_MULTIMEDIA4,
+ MSM_FRONTEND_DAI_CS_VOICE,
+ MSM_FRONTEND_DAI_VOIP,
+ MSM_FRONTEND_DAI_AFE_RX,
+ MSM_FRONTEND_DAI_AFE_TX,
+ MSM_FRONTEND_DAI_VOICE_STUB,
+ MSM_FRONTEND_DAI_MAX,
+};
+
+#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA4 + 1)
+#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA4
+
+enum {
+ MSM_BACKEND_DAI_PRI_I2S_RX = 0,
+ MSM_BACKEND_DAI_PRI_I2S_TX,
+ MSM_BACKEND_DAI_SLIMBUS_0_RX,
+ MSM_BACKEND_DAI_SLIMBUS_0_TX,
+ MSM_BACKEND_DAI_HDMI_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_RX,
+ MSM_BACKEND_DAI_INT_BT_SCO_TX,
+ MSM_BACKEND_DAI_INT_FM_RX,
+ MSM_BACKEND_DAI_INT_FM_TX,
+ MSM_BACKEND_DAI_AFE_PCM_RX,
+ MSM_BACKEND_DAI_AFE_PCM_TX,
+ MSM_BACKEND_DAI_AUXPCM_RX,
+ MSM_BACKEND_DAI_AUXPCM_TX,
+ MSM_BACKEND_DAI_VOICE_PLAYBACK_TX,
+ MSM_BACKEND_DAI_INCALL_RECORD_RX,
+ MSM_BACKEND_DAI_INCALL_RECORD_TX,
+ MSM_BACKEND_DAI_MI2S_RX,
+ MSM_BACKEND_DAI_SEC_I2S_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_RX,
+ MSM_BACKEND_DAI_SLIMBUS_1_TX,
+ MSM_BACKEND_DAI_INVALID,
+ MSM_BACKEND_DAI_MAX,
+};
+
+/* dai_id: front-end ID,
+ * dspst_id: DSP audio stream ID
+ * stream_type: playback or capture
+ */
+void msm_pcm_routing_reg_phy_stream(int fedai_id, int dspst_id,
+ int stream_type);
+void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type);
+
+int lpa_set_volume(unsigned volume);
+
+int msm_routing_check_backend_enabled(int fedai_id);
+
+int multi_ch_pcm_set_volume(unsigned volume);
+
+int compressed_set_volume(unsigned volume);
+
+#endif /*_MSM_PCM_H*/
diff --git a/sound/soc/msm/qdsp6v2/q6adm.c b/sound/soc/msm/qdsp6v2/q6adm.c
new file mode 100644
index 0000000..691ca21
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/q6adm.c
@@ -0,0 +1,621 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/sched.h>
+#include <linux/jiffies.h>
+#include <linux/uaccess.h>
+#include <linux/atomic.h>
+
+
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <mach/qdsp6v2/rtac.h>
+
+#include <sound/apr_audio-v2.h>
+#include <mach/qdsp6v2/apr.h>
+#include <sound/q6adm-v2.h>
+#include <sound/q6audio-v2.h>
+
+
+#define TIMEOUT_MS 1000
+
+#define RESET_COPP_ID 99
+#define INVALID_COPP_ID 0xFF
+
+struct adm_ctl {
+ void *apr;
+ atomic_t copp_id[Q6_AFE_MAX_PORTS];
+ atomic_t copp_cnt[Q6_AFE_MAX_PORTS];
+ atomic_t copp_stat[Q6_AFE_MAX_PORTS];
+ u32 mem_map_handle[Q6_AFE_MAX_PORTS];
+ wait_queue_head_t wait[Q6_AFE_MAX_PORTS];
+};
+
+static struct adm_ctl this_adm;
+
+static int32_t adm_callback(struct apr_client_data *data, void *priv)
+{
+ uint32_t *payload;
+ int i, index;
+ payload = data->payload;
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("adm_callback: Reset event is received: %d %d apr[%p]\n",
+ data->reset_event, data->reset_proc,
+ this_adm.apr);
+ if (this_adm.apr) {
+ apr_reset(this_adm.apr);
+ for (i = 0; i < Q6_AFE_MAX_PORTS; i++) {
+ atomic_set(&this_adm.copp_id[i],
+ RESET_COPP_ID);
+ atomic_set(&this_adm.copp_cnt[i], 0);
+ atomic_set(&this_adm.copp_stat[i], 0);
+ }
+ this_adm.apr = NULL;
+ }
+ return 0;
+ }
+
+ pr_debug("%s: code = 0x%x PL#0[%x], PL#1[%x], size = %d\n", __func__,
+ data->opcode, payload[0], payload[1],
+ data->payload_size);
+
+ if (data->payload_size) {
+ index = q6audio_get_port_index(data->token);
+ if (index < 0 || index >= Q6_AFE_MAX_PORTS) {
+ pr_err("%s: invalid port idx %d token %d\n",
+ __func__, index, data->token);
+ return 0;
+ }
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ pr_debug("APR_BASIC_RSP_RESULT\n");
+ switch (payload[0]) {
+ case ADM_CMD_SET_PP_PARAMS_V5:
+ if (rtac_make_adm_callback(
+ payload, data->payload_size))
+ pr_debug("%s: payload[0]: 0x%x\n",
+ __func__, payload[0]);
+ break;
+ case ADM_CMD_DEVICE_CLOSE_V5:
+ case ADM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ case ADM_CMD_SHARED_MEM_MAP_REGIONS:
+ case ADM_CMD_MATRIX_MAP_ROUTINGS_V5:
+ pr_debug("ADM_CMD_MATRIX_MAP_ROUTINGS\n");
+ atomic_set(&this_adm.copp_stat[index], 1);
+ wake_up(&this_adm.wait[index]);
+ break;
+ default:
+ pr_err("%s: Unknown Cmd: 0x%x\n", __func__,
+ payload[0]);
+ break;
+ }
+ return 0;
+ }
+
+ switch (data->opcode) {
+ case ADM_CMDRSP_DEVICE_OPEN_V5: {
+ struct adm_cmd_rsp_device_open_v5 *open =
+ (struct adm_cmd_rsp_device_open_v5 *)data->payload;
+ if (open->copp_id == INVALID_COPP_ID) {
+ pr_err("%s: invalid coppid rxed %d\n",
+ __func__, open->copp_id);
+ atomic_set(&this_adm.copp_stat[index], 1);
+ wake_up(&this_adm.wait[index]);
+ break;
+ }
+ atomic_set(&this_adm.copp_id[index], open->copp_id);
+ atomic_set(&this_adm.copp_stat[index], 1);
+ pr_debug("%s: coppid rxed=%d\n", __func__,
+ open->copp_id);
+ wake_up(&this_adm.wait[index]);
+ }
+ break;
+ case ADM_CMD_GET_PP_PARAMS_V5:
+ pr_debug("%s: ADM_CMD_GET_PP_PARAMS_V5\n", __func__);
+ rtac_make_adm_callback(payload,
+ data->payload_size);
+ break;
+ default:
+ pr_err("%s: Unknown cmd:0x%x\n", __func__,
+ data->opcode);
+ break;
+ }
+ }
+ return 0;
+}
+
+/* TODO: send_adm_cal_block function to be defined
+ when calibration available for 8974 */
+static void send_adm_cal(int port_id, int path)
+{
+ /* function to be defined when calibration available for 8974 */
+ pr_debug("%s\n", __func__);
+}
+
+int adm_open(int port_id, int path, int rate, int channel_mode, int topology)
+{
+ struct adm_cmd_device_open_v5 open;
+ int ret = 0;
+ int index;
+ int tmp_port = q6audio_get_port_id(port_id);
+
+ pr_debug("%s: port %d path:%d rate:%d mode:%d\n", __func__,
+ port_id, path, rate, channel_mode);
+
+ port_id = q6audio_convert_virtual_to_portid(port_id);
+
+ if (q6audio_validate_port(port_id) < 0) {
+ pr_err("%s port idi[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+
+ index = q6audio_get_port_index(port_id);
+ pr_debug("%s: Port ID %d, index %d\n", __func__, port_id, index);
+
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+
+
+ /* Create a COPP if port id are not enabled */
+ if (atomic_read(&this_adm.copp_cnt[index]) == 0) {
+
+ open.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ open.hdr.pkt_size = sizeof(open);
+ open.hdr.src_svc = APR_SVC_ADM;
+ open.hdr.src_domain = APR_DOMAIN_APPS;
+ open.hdr.src_port = tmp_port;
+ open.hdr.dest_svc = APR_SVC_ADM;
+ open.hdr.dest_domain = APR_DOMAIN_ADSP;
+ open.hdr.dest_port = tmp_port;
+ open.hdr.token = port_id;
+ open.hdr.opcode = ADM_CMD_DEVICE_OPEN_V5;
+
+ open.mode_of_operation = path;
+ /* Reserved for future use, need to set this to 0 */
+ open.flags = 0x00;
+ open.endpoint_id_1 = tmp_port;
+ open.endpoint_id_2 = 0xFFFF;
+
+ /* convert path to acdb path */
+ if (path == ADM_PATH_PLAYBACK)
+ open.topology_id = get_adm_rx_topology();
+ else {
+ open.topology_id = get_adm_tx_topology();
+ if ((open.topology_id ==
+ VPM_TX_SM_ECNS_COPP_TOPOLOGY) ||
+ (open.topology_id ==
+ VPM_TX_DM_FLUENCE_COPP_TOPOLOGY))
+ rate = 16000;
+ }
+
+ if (open.topology_id == 0)
+ open.topology_id = topology;
+
+ open.dev_num_channel = channel_mode & 0x00FF;
+ open.bit_width = 16;
+ open.sample_rate = rate;
+ memset(open.dev_channel_mapping, 0, 8);
+
+ if (channel_mode == 1) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FC;
+ } else if (channel_mode == 2) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channel_mode == 6) {
+ open.dev_channel_mapping[0] = PCM_CHANNEL_FL;
+ open.dev_channel_mapping[1] = PCM_CHANNEL_FR;
+ open.dev_channel_mapping[2] = PCM_CHANNEL_LFE;
+ open.dev_channel_mapping[3] = PCM_CHANNEL_FC;
+ open.dev_channel_mapping[4] = PCM_CHANNEL_LB;
+ open.dev_channel_mapping[5] = PCM_CHANNEL_RB;
+ } else {
+ pr_err("%s invalid num_chan %d\n", __func__,
+ channel_mode);
+ return -EINVAL;
+ }
+
+ pr_debug("%s: port_id=%d rate=%d"
+ "topology_id=0x%X\n", __func__, open.endpoint_id_1, \
+ open.sample_rate, open.topology_id);
+
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&open);
+ if (ret < 0) {
+ pr_err("%s:ADM enable for port %d for[%d] failed\n",
+ __func__, tmp_port, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ /* Wait for the callback with copp id */
+ ret = wait_event_timeout(this_adm.wait[index],
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s ADM open failed for port %d"
+ "for [%d]\n", __func__, tmp_port, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ }
+ atomic_inc(&this_adm.copp_cnt[index]);
+ return 0;
+
+fail_cmd:
+
+ return ret;
+}
+
+
+int adm_multi_ch_copp_open(int port_id, int path, int rate, int channel_mode,
+ int topology)
+{
+ int ret = 0;
+
+ ret = adm_open(port_id, path, rate, channel_mode, topology);
+
+ return ret;
+}
+
+int adm_matrix_map(int session_id, int path, int num_copps,
+ unsigned int *port_id, int copp_id)
+{
+ struct adm_cmd_matrix_map_routings_v5 *route;
+ struct adm_session_map_node_v5 *node;
+ uint32_t *copps_list;
+ int cmd_size = 0;
+ int ret = 0, i = 0;
+ void *payload = NULL;
+ void *matrix_map = NULL;
+
+ /* Assumes port_ids have already been validated during adm_open */
+ int index = q6audio_get_port_index(copp_id);
+ if (index < 0 || index >= Q6_AFE_MAX_PORTS) {
+ pr_err("%s: invalid port idx %d token %d\n",
+ __func__, index, copp_id);
+ return 0;
+ }
+ cmd_size = (sizeof(struct adm_cmd_matrix_map_routings_v5) +
+ sizeof(struct adm_session_map_node_v5) +
+ (sizeof(uint32_t) * num_copps));
+ matrix_map = kzalloc(cmd_size, GFP_KERNEL);
+ if (matrix_map == NULL) {
+ pr_err("%s: Mem alloc failed\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ route = (struct adm_cmd_matrix_map_routings_v5 *)matrix_map;
+
+ pr_debug("%s: session 0x%x path:%d num_copps:%d port_id[0] :%d coppid[%d]\n",
+ __func__, session_id, path, num_copps, port_id[0], copp_id);
+
+ route->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ route->hdr.pkt_size = cmd_size;
+ route->hdr.src_svc = 0;
+ route->hdr.src_domain = APR_DOMAIN_APPS;
+ route->hdr.src_port = copp_id;
+ route->hdr.dest_svc = APR_SVC_ADM;
+ route->hdr.dest_domain = APR_DOMAIN_ADSP;
+ route->hdr.dest_port = atomic_read(&this_adm.copp_id[index]);
+ route->hdr.token = copp_id;
+ route->hdr.opcode = ADM_CMD_MATRIX_MAP_ROUTINGS_V5;
+ route->num_sessions = 1;
+
+ switch (path) {
+ case 0x1:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_RX;
+ break;
+ case 0x2:
+ case 0x3:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_TX;
+ break;
+ default:
+ pr_err("%s: Wrong path set[%d]\n", __func__, path);
+ break;
+ }
+ payload = ((u8 *)matrix_map +
+ sizeof(struct adm_cmd_matrix_map_routings_v5));
+ node = (struct adm_session_map_node_v5 *)payload;
+
+ node->session_id = session_id;
+ node->num_copps = num_copps;
+ payload = (u8 *)node + sizeof(struct adm_session_map_node_v5);
+ copps_list = (uint32_t *)payload;
+ for (i = 0; i < num_copps; i++) {
+ int tmp;
+ port_id[i] = q6audio_convert_virtual_to_portid(port_id[i]);
+
+ tmp = q6audio_get_port_index(port_id[i]);
+
+
+ if (tmp >= 0 && tmp < Q6_AFE_MAX_PORTS)
+ copps_list[i] =
+ atomic_read(&this_adm.copp_id[tmp]);
+ pr_debug("%s: port_id[%d]: %d, index: %d act coppid[0x%x]\n",
+ __func__, i, port_id[i], tmp,
+ atomic_read(&this_adm.copp_id[tmp]));
+ }
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)matrix_map);
+ if (ret < 0) {
+ pr_err("%s: ADM routing for port %d failed\n",
+ __func__, port_id[0]);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(this_adm.wait[index],
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: ADM cmd Route failed for port %d\n",
+ __func__, port_id[0]);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ for (i = 0; i < num_copps; i++)
+ send_adm_cal(port_id[i], path);
+
+fail_cmd:
+ kfree(matrix_map);
+ return ret;
+}
+
+int adm_memory_map_regions(int port_id,
+ uint32_t *buf_add, uint32_t mempool_id,
+ uint32_t *bufsz, uint32_t bufcnt)
+{
+ struct avs_cmd_shared_mem_map_regions *mmap_regions = NULL;
+ struct avs_shared_map_region_payload *mregions = NULL;
+ void *mmap_region_cmd = NULL;
+ void *payload = NULL;
+ int ret = 0;
+ int i = 0;
+ int cmd_size = 0;
+ int index = 0;
+
+ pr_debug("%s\n", __func__);
+ if (this_adm.apr == NULL) {
+ this_adm.apr = apr_register("ADSP", "ADM", adm_callback,
+ 0xFFFFFFFF, &this_adm);
+ if (this_adm.apr == NULL) {
+ pr_err("%s: Unable to register ADM\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ rtac_set_adm_handle(this_adm.apr);
+ }
+
+ port_id = q6audio_convert_virtual_to_portid(port_id);
+
+ if (q6audio_validate_port(port_id) < 0) {
+ pr_err("%s port id[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+
+ index = q6audio_get_port_index(port_id);
+
+ cmd_size = sizeof(struct avs_cmd_shared_mem_map_regions)
+ + sizeof(struct avs_shared_map_region_payload)
+ * bufcnt;
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (!mmap_region_cmd) {
+ pr_err("%s: allocate mmap_region_cmd failed\n", __func__);
+ return -ENOMEM;
+ }
+ mmap_regions = (struct avs_cmd_shared_mem_map_regions *)mmap_region_cmd;
+ mmap_regions->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ mmap_regions->hdr.pkt_size = cmd_size;
+ mmap_regions->hdr.src_port = 0;
+ mmap_regions->hdr.dest_port = 0;
+ mmap_regions->hdr.token = 0;
+ mmap_regions->hdr.opcode = ADM_CMD_SHARED_MEM_MAP_REGIONS;
+ mmap_regions->mem_pool_id = ADSP_MEMORY_MAP_EBI_POOL & 0x00ff;
+ mmap_regions->num_regions = bufcnt & 0x00ff;
+ mmap_regions->property_flag = 0x00;
+
+ pr_debug("%s: map_regions->num_regions = %d\n", __func__,
+ mmap_regions->num_regions);
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct avs_cmd_shared_mem_map_regions));
+ mregions = (struct avs_shared_map_region_payload *)payload;
+
+ for (i = 0; i < bufcnt; i++) {
+ mregions->shm_addr_lsw = buf_add[i];
+ mregions->shm_addr_msw = 0x00;
+ mregions->mem_size_bytes = bufsz[i];
+ ++mregions;
+ }
+
+ atomic_set(&this_adm.copp_stat[0], 0);
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *) mmap_region_cmd);
+ if (ret < 0) {
+ pr_err("%s: mmap_regions op[0x%x]rc[%d]\n", __func__,
+ mmap_regions->hdr.opcode, ret);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_adm.wait[index],
+ atomic_read(&this_adm.copp_stat[0]), 5 * HZ);
+ if (!ret) {
+ pr_err("%s: timeout. waited for memory_map\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+fail_cmd:
+ kfree(mmap_region_cmd);
+ return ret;
+}
+
+int adm_memory_unmap_regions(int32_t port_id, uint32_t *buf_add,
+ uint32_t *bufsz, uint32_t bufcnt)
+{
+ struct avs_cmd_shared_mem_unmap_regions unmap_regions;
+ int ret = 0;
+ int cmd_size = 0;
+ int index = 0;
+
+ pr_debug("%s\n", __func__);
+
+ if (this_adm.apr == NULL) {
+ pr_err("%s APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ port_id = q6audio_convert_virtual_to_portid(port_id);
+
+ if (q6audio_validate_port(port_id) < 0) {
+ pr_err("%s port idi[%d] is invalid\n", __func__, port_id);
+ return -ENODEV;
+ }
+
+ index = q6audio_get_port_index(port_id);
+
+ unmap_regions.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ unmap_regions.hdr.pkt_size = cmd_size;
+ unmap_regions.hdr.src_port = 0;
+ unmap_regions.hdr.dest_port = 0;
+ unmap_regions.hdr.token = 0;
+ unmap_regions.hdr.opcode = ADM_CMD_SHARED_MEM_UNMAP_REGIONS;
+ unmap_regions.mem_map_handle = this_adm.mem_map_handle[index];
+ atomic_set(&this_adm.copp_stat[0], 0);
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *) &unmap_regions);
+ if (ret < 0) {
+ pr_err("%s: mmap_regions op[0x%x]rc[%d]\n", __func__,
+ unmap_regions.hdr.opcode, ret);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_adm.wait[index],
+ atomic_read(&this_adm.copp_stat[0]), 5 * HZ);
+ if (!ret) {
+ pr_err("%s: timeout. waited for memory_unmap\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+fail_cmd:
+ return ret;
+}
+
+int adm_get_copp_id(int port_index)
+{
+ pr_debug("%s\n", __func__);
+
+ if (port_index < 0) {
+ pr_err("%s: invalid port_id = %d\n", __func__, port_index);
+ return -EINVAL;
+ }
+
+ return atomic_read(&this_adm.copp_id[port_index]);
+}
+
+int adm_close(int port_id)
+{
+ struct apr_hdr close;
+
+ int ret = 0;
+ int index = 0;
+
+ port_id = q6audio_convert_virtual_to_portid(port_id);
+
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ pr_debug("%s port_id=%d index %d\n", __func__, port_id, index);
+
+ if (!(atomic_read(&this_adm.copp_cnt[index]))) {
+ pr_err("%s: copp count for port[%d]is 0\n", __func__, port_id);
+
+ goto fail_cmd;
+ }
+ atomic_dec(&this_adm.copp_cnt[index]);
+ if (!(atomic_read(&this_adm.copp_cnt[index]))) {
+
+ close.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ close.pkt_size = sizeof(close);
+ close.src_svc = APR_SVC_ADM;
+ close.src_domain = APR_DOMAIN_APPS;
+ close.src_port = port_id;
+ close.dest_svc = APR_SVC_ADM;
+ close.dest_domain = APR_DOMAIN_ADSP;
+ close.dest_port = atomic_read(&this_adm.copp_id[index]);
+ close.token = port_id;
+ close.opcode = ADM_CMD_DEVICE_CLOSE_V5;
+
+ atomic_set(&this_adm.copp_id[index], RESET_COPP_ID);
+ atomic_set(&this_adm.copp_stat[index], 0);
+
+
+ pr_debug("%s:coppid %d portid=%d index=%d coppcnt=%d\n",
+ __func__,
+ atomic_read(&this_adm.copp_id[index]),
+ port_id, index,
+ atomic_read(&this_adm.copp_cnt[index]));
+
+ ret = apr_send_pkt(this_adm.apr, (uint32_t *)&close);
+ if (ret < 0) {
+ pr_err("%s ADM close failed\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_adm.wait[index],
+ atomic_read(&this_adm.copp_stat[index]),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: ADM cmd Route failed for port %d\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rtac_remove_adm_device(port_id);
+ }
+
+fail_cmd:
+ return ret;
+}
+
+static int __init adm_init(void)
+{
+ int i = 0;
+ this_adm.apr = NULL;
+
+ for (i = 0; i < Q6_AFE_MAX_PORTS; i++) {
+ atomic_set(&this_adm.copp_id[i], RESET_COPP_ID);
+ atomic_set(&this_adm.copp_cnt[i], 0);
+ atomic_set(&this_adm.copp_stat[i], 0);
+ init_waitqueue_head(&this_adm.wait[i]);
+ }
+ return 0;
+}
+
+device_initcall(adm_init);
diff --git a/sound/soc/msm/qdsp6v2/q6afe.c b/sound/soc/msm/qdsp6v2/q6afe.c
new file mode 100644
index 0000000..5b30e8e
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/q6afe.c
@@ -0,0 +1,1584 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/slab.h>
+#include <linux/debugfs.h>
+#include <linux/kernel.h>
+#include <linux/kthread.h>
+#include <linux/uaccess.h>
+#include <linux/wait.h>
+#include <linux/jiffies.h>
+#include <linux/sched.h>
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <sound/apr_audio-v2.h>
+#include <sound/q6afe-v2.h>
+
+#include <sound/q6audio-v2.h>
+
+
+struct afe_ctl {
+ void *apr;
+ atomic_t state;
+ atomic_t status;
+ wait_queue_head_t wait[AFE_MAX_PORTS];
+ void (*tx_cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv);
+ void (*rx_cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv);
+ void *tx_private_data;
+ void *rx_private_data;
+};
+
+static struct afe_ctl this_afe;
+
+static struct acdb_cal_block afe_cal_addr[MAX_AUDPROC_TYPES];
+
+#define TIMEOUT_MS 1000
+#define Q6AFE_MAX_VOLUME 0x3FFF
+
+#define SIZEOF_CFG_CMD(y) \
+ (sizeof(struct apr_hdr) + sizeof(u16) + (sizeof(struct y)))
+
+static int32_t afe_callback(struct apr_client_data *data, void *priv)
+{
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("q6afe: reset event = %d %d apr[%p]\n",
+ data->reset_event, data->reset_proc, this_afe.apr);
+ if (this_afe.apr) {
+ apr_reset(this_afe.apr);
+ atomic_set(&this_afe.state, 0);
+ this_afe.apr = NULL;
+ }
+ return 0;
+ }
+ pr_debug("%s:opcode = 0x%x cmd = 0x%x status = 0x%x\n",
+ __func__, data->opcode,
+ ((uint32_t *)(data->payload))[0],
+ ((uint32_t *)(data->payload))[1]);
+ if (data->payload_size) {
+ uint32_t *payload;
+ uint16_t port_id = 0;
+ payload = data->payload;
+ pr_debug("%s:opcode = 0x%x cmd = 0x%x status = 0x%x token=%d\n",
+ __func__, data->opcode,
+ payload[0], payload[1], data->token);
+ /* payload[1] contains the error status for response */
+ if (payload[1] != 0) {
+ atomic_set(&this_afe.status, -1);
+ pr_err("%s: cmd = 0x%x returned error = 0x%x\n",
+ __func__, payload[0], payload[1]);
+ }
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ switch (payload[0]) {
+ case AFE_PORT_CMD_DEVICE_STOP:
+ case AFE_PORT_CMD_DEVICE_START:
+ case AFE_PORT_CMD_SET_PARAM_V2:
+ case AFE_PSEUDOPORT_CMD_START:
+ case AFE_PSEUDOPORT_CMD_STOP:
+ case AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS:
+ case AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS:
+ case AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER:
+ atomic_set(&this_afe.state, 0);
+ wake_up(&this_afe.wait[data->token]);
+ break;
+ case AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER:
+ break;
+ case AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2:
+ port_id = RT_PROXY_PORT_001_TX;
+ break;
+ case AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2:
+ port_id = RT_PROXY_PORT_001_RX;
+ break;
+ default:
+ pr_err("%s:Unknown cmd 0x%x\n", __func__,
+ payload[0]);
+ break;
+ }
+ } else if (data->opcode == AFE_EVENT_RT_PROXY_PORT_STATUS) {
+ port_id = (uint16_t)(0x0000FFFF & payload[0]);
+ }
+ pr_debug("%s:port_id = %x\n", __func__, port_id);
+ switch (port_id) {
+ case RT_PROXY_PORT_001_TX: {
+ if (this_afe.tx_cb) {
+ this_afe.tx_cb(data->opcode, data->token,
+ data->payload,
+ this_afe.tx_private_data);
+ }
+ break;
+ }
+ case RT_PROXY_PORT_001_RX: {
+ if (this_afe.rx_cb) {
+ this_afe.rx_cb(data->opcode, data->token,
+ data->payload,
+ this_afe.rx_private_data);
+ }
+ break;
+ }
+ default:
+ break;
+ }
+ }
+ return 0;
+}
+
+
+int afe_get_port_type(u16 port_id)
+{
+ int ret;
+
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PCM_RX:
+ case SECONDARY_I2S_RX:
+ case MI2S_RX:
+ case HDMI_RX:
+ case SLIMBUS_0_RX:
+ case SLIMBUS_1_RX:
+ case INT_BT_SCO_RX:
+ case INT_BT_A2DP_RX:
+ case INT_FM_RX:
+ case VOICE_PLAYBACK_TX:
+ case RT_PROXY_PORT_001_RX:
+ ret = MSM_AFE_PORT_TYPE_RX;
+ break;
+
+ case PRIMARY_I2S_TX:
+ case PCM_TX:
+ case SECONDARY_I2S_TX:
+ case MI2S_TX:
+ case DIGI_MIC_TX:
+ case VOICE_RECORD_TX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_TX:
+ case INT_FM_TX:
+ case VOICE_RECORD_RX:
+ case INT_BT_SCO_TX:
+ case RT_PROXY_PORT_001_TX:
+ ret = MSM_AFE_PORT_TYPE_TX;
+ break;
+
+ default:
+ pr_err("%s: invalid port id %d\n", __func__, port_id);
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+int afe_sizeof_cfg_cmd(u16 port_id)
+{
+ int ret_size;
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case MI2S_RX:
+ case MI2S_TX:
+ ret_size = SIZEOF_CFG_CMD(afe_param_id_i2s_cfg);
+ break;
+ case HDMI_RX:
+ ret_size =
+ SIZEOF_CFG_CMD(afe_param_id_hdmi_multi_chan_audio_cfg);
+ break;
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ ret_size = SIZEOF_CFG_CMD(afe_param_id_slimbus_cfg);
+ break;
+ case RT_PROXY_PORT_001_RX:
+ case RT_PROXY_PORT_001_TX:
+ ret_size = SIZEOF_CFG_CMD(afe_param_id_rt_proxy_port_cfg);
+ break;
+ case PCM_RX:
+ case PCM_TX:
+ default:
+ ret_size = SIZEOF_CFG_CMD(afe_param_id_pcm_cfg);
+ break;
+ }
+ return ret_size;
+}
+
+int afe_q6_interface_prepare(void)
+{
+ int ret = 0;
+
+ pr_debug("%s:", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ }
+ }
+ return ret;
+}
+static void afe_send_cal_block(int32_t path, u16 port_id)
+{
+ /* To come back */
+}
+
+void afe_send_cal(u16 port_id)
+{
+ pr_debug("%s\n", __func__);
+
+ if (afe_get_port_type(port_id) == MSM_AFE_PORT_TYPE_TX)
+ afe_send_cal_block(TX_CAL, port_id);
+ else if (afe_get_port_type(port_id) == MSM_AFE_PORT_TYPE_RX)
+ afe_send_cal_block(RX_CAL, port_id);
+}
+
+int afe_port_start_nowait(u16 port_id, union afe_port_config *afe_config,
+ u32 rate) /* This function is no blocking */
+{
+ struct afe_port_cmd_device_start start;
+ struct afe_audioif_config_command config;
+ int ret;
+ int cfg_type;
+ int index = 0;
+
+ if (!afe_config) {
+ pr_err("%s: Error, no configuration data\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ pr_err("%s: %d %d\n", __func__, port_id, rate);
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ if ((port_id == RT_PROXY_DAI_001_RX) ||
+ (port_id == RT_PROXY_DAI_002_TX))
+ return -EINVAL;
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+ if (q6audio_validate_port(port_id) < 0) {
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ config.hdr.pkt_size = afe_sizeof_cfg_cmd(port_id);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+
+ config.hdr.token = index;
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ cfg_type = AFE_PARAM_ID_PCM_CONFIG;
+ break;
+ case PCM_RX:
+ case PCM_TX:
+ cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
+ break;
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case MI2S_RX:
+ case MI2S_TX:
+ cfg_type = AFE_PARAM_ID_I2S_CONFIG;
+ break;
+ case HDMI_RX:
+ cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
+ break;
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_3_TX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG;
+ break;
+ default:
+ pr_err("%s: Invalid port id 0x%x\n", __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ config.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+ config.param.port_id = port_id;
+ config.param.payload_size = (afe_sizeof_cfg_cmd(port_id) +
+ sizeof(struct afe_port_param_data_v2));
+ config.param.payload_address_lsw = 0x00;
+ config.param.payload_address_msw = 0x00;
+ config.param.mem_map_handle = 0x00;
+ config.pdata.module_id = AFE_MODULE_AUDIO_DEV_INTERFACE;
+ config.pdata.param_id = cfg_type;
+ config.pdata.param_size = afe_sizeof_cfg_cmd(port_id);
+
+ config.port = *afe_config;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &config);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ /* send AFE cal */
+ afe_send_cal(port_id);
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PORT_CMD_DEVICE_START;
+ start.port_id = port_id;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+
+ if (IS_ERR_VALUE(ret)) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ return 0;
+
+fail_cmd:
+ return ret;
+}
+
+int afe_open(u16 port_id,
+ union afe_port_config *afe_config, int rate)
+{
+ struct afe_port_cmd_device_start start;
+ struct afe_audioif_config_command config;
+ int ret = 0;
+ int cfg_type;
+ int index = 0;
+
+ if (!afe_config) {
+ pr_err("%s: Error, no configuration data\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ pr_err("%s: %d %d\n", __func__, port_id, rate);
+
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ if ((port_id == RT_PROXY_DAI_001_RX) ||
+ (port_id == RT_PROXY_DAI_002_TX))
+ return -EINVAL;
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ if (q6audio_validate_port(port_id) < 0) {
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ config.hdr.pkt_size = sizeof(config);
+ config.hdr.src_port = 0;
+ config.hdr.dest_port = 0;
+ config.hdr.token = index;
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ cfg_type = AFE_PARAM_ID_I2S_CONFIG;
+ break;
+ case PCM_RX:
+ case PCM_TX:
+ cfg_type = AFE_PARAM_ID_PCM_CONFIG;
+ break;
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case MI2S_RX:
+ case MI2S_TX:
+ cfg_type = AFE_PARAM_ID_I2S_CONFIG;
+ break;
+ case HDMI_RX:
+ cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
+ break;
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case SLIMBUS_2_RX:
+ case SLIMBUS_2_TX:
+ case SLIMBUS_3_RX:
+ case SLIMBUS_3_TX:
+ case SLIMBUS_4_RX:
+ case SLIMBUS_4_TX:
+ cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG;
+ break;
+ default:
+ pr_err("%s: Invalid port id 0x%x\n", __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ config.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+ config.param.port_id = q6audio_get_port_id(port_id);
+ config.param.payload_size = sizeof(config) - sizeof(struct apr_hdr)
+ - sizeof(config.param);
+ config.param.payload_address_lsw = 0x00;
+ config.param.payload_address_msw = 0x00;
+ config.param.mem_map_handle = 0x00;
+ config.pdata.module_id = AFE_MODULE_AUDIO_DEV_INTERFACE;
+ config.pdata.param_id = cfg_type;
+ config.pdata.param_size = sizeof(config.port);
+
+ config.port = *afe_config;
+ pr_debug("%s: param PL size=%d iparam_size[%d][%d %d %d %d]"
+ " param_id[%x]\n",
+ __func__, config.param.payload_size, config.pdata.param_size,
+ sizeof(config), sizeof(config.param), sizeof(config.port),
+ sizeof(struct apr_hdr), config.pdata.param_id);
+ atomic_set(&this_afe.state, 1);
+ atomic_set(&this_afe.status, 0);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &config);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d opcode[0x%x]failed\n",
+ __func__, port_id, cfg_type);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ if (atomic_read(&this_afe.status) != 0) {
+ pr_err("%s: config cmd failed\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = index;
+ start.hdr.opcode = AFE_PORT_CMD_DEVICE_START;
+ start.port_id = q6audio_get_port_id(port_id);
+ pr_debug("%s: cmd device start opcode[0x%x] port id[0x%x]\n",
+ __func__, start.hdr.opcode, start.port_id);
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_loopback(u16 enable, u16 rx_port, u16 tx_port)
+{
+ struct afe_loopback_cfg_v1 lb_cmd;
+ int ret = 0;
+ int index = 0;
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ index = q6audio_get_port_index(rx_port);
+ if (q6audio_validate_port(rx_port) < 0)
+ return -EINVAL;
+
+ lb_cmd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(20), APR_PKT_VER);
+ lb_cmd.hdr.pkt_size = APR_PKT_SIZE(APR_HDR_SIZE,
+ sizeof(lb_cmd) - APR_HDR_SIZE);
+ lb_cmd.hdr.src_port = 0;
+ lb_cmd.hdr.dest_port = 0;
+ lb_cmd.hdr.token = 0;
+ lb_cmd.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+ lb_cmd.param.port_id = tx_port;
+ lb_cmd.param.payload_size = (sizeof(lb_cmd) -
+ sizeof(struct apr_hdr) -
+ sizeof(struct afe_port_cmd_set_param_v2));
+ lb_cmd.param.payload_address_lsw = 0x00;
+ lb_cmd.param.payload_address_msw = 0x00;
+ lb_cmd.param.mem_map_handle = 0x00;
+ lb_cmd.pdata.module_id = AFE_MODULE_LOOPBACK;
+ lb_cmd.pdata.param_id = AFE_PARAM_ID_LOOPBACK_CONFIG;
+ lb_cmd.pdata.param_size = lb_cmd.param.payload_size -
+ sizeof(struct afe_port_param_data_v2);
+
+ lb_cmd.dst_port_id = rx_port;
+ lb_cmd.routing_mode = LB_MODE_DEFAULT;
+ lb_cmd.enable = (enable ? 1 : 0);
+ lb_cmd.loopback_cfg_minor_version =
+ AFE_API_VERSION_LOOPBACK_CONFIG;
+ atomic_set(&this_afe.state, 1);
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &lb_cmd);
+ if (ret < 0) {
+ pr_err("%s: AFE loopback failed\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ }
+done:
+ return ret;
+}
+
+int afe_loopback_gain(u16 port_id, u16 volume)
+{
+ struct afe_loopback_gain_per_path_param set_param;
+ int ret = 0;
+ int index = 0;
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ if (q6audio_validate_port(port_id) < 0) {
+
+ pr_err("%s: Failed : Invalid Port id = %d\n", __func__,
+ port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ /* RX ports numbers are even .TX ports numbers are odd. */
+ if (port_id % 2 == 0) {
+ pr_err("%s: Failed : afe loopback gain only for TX ports."
+ " port_id %d\n", __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ pr_debug("%s: %d %hX\n", __func__, port_id, volume);
+
+ set_param.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ set_param.hdr.pkt_size = sizeof(set_param);
+ set_param.hdr.src_port = 0;
+ set_param.hdr.dest_port = 0;
+ set_param.hdr.token = 0;
+ set_param.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+
+ set_param.param.port_id = port_id;
+ set_param.param.payload_size =
+ (sizeof(struct afe_loopback_gain_per_path_param) -
+ sizeof(struct apr_hdr) -
+ sizeof(struct afe_port_cmd_set_param_v2));
+ set_param.param.payload_address_lsw = 0;
+ set_param.param.payload_address_msw = 0;
+ set_param.param.mem_map_handle = 0;
+
+ set_param.pdata.module_id = AFE_MODULE_LOOPBACK;
+ set_param.pdata.param_id = AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH;
+ set_param.pdata.param_size = (set_param.param.payload_size -
+ sizeof(struct afe_port_param_data_v2));
+ set_param.rx_port_id = port_id;
+ set_param.gain = volume;
+
+ set_param.hdr.token = index;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &set_param);
+ if (ret < 0) {
+ pr_err("%s: AFE param set failed for port %d\n",
+ __func__, port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (ret < 0) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_pseudo_port_start_nowait(u16 port_id)
+{
+ struct afe_pseudoport_start_command start;
+ int ret = 0;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE APR is not registered\n", __func__);
+ return -ENODEV;
+ }
+
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PSEUDOPORT_CMD_START;
+ start.port_id = port_id;
+ start.timing = 1;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed %d\n",
+ __func__, port_id, ret);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+int afe_start_pseudo_port(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_start_command start;
+ int index = 0;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ ret = afe_q6_interface_prepare();
+ if (ret != 0)
+ return ret;
+
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = 0;
+ start.hdr.opcode = AFE_PSEUDOPORT_CMD_START;
+ start.port_id = port_id;
+ start.timing = 1;
+
+ start.hdr.token = index;
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &start);
+ if (ret < 0) {
+ pr_err("%s: AFE enable for port %d failed %d\n",
+ __func__, port_id, ret);
+ return -EINVAL;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+int afe_pseudo_port_stop_nowait(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_stop_command stop;
+ int index = 0;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE is already closed\n", __func__);
+ return -EINVAL;
+ }
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PSEUDOPORT_CMD_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ stop.hdr.token = index;
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+ if (ret < 0) {
+ pr_err("%s: AFE close failed %d\n", __func__, ret);
+ return -EINVAL;
+ }
+
+ return 0;
+
+}
+
+int afe_stop_pseudo_port(u16 port_id)
+{
+ int ret = 0;
+ struct afe_pseudoport_stop_command stop;
+ int index = 0;
+
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: AFE is already closed\n", __func__);
+ return -EINVAL;
+ }
+
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PSEUDOPORT_CMD_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ stop.hdr.token = index;
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+ if (ret < 0) {
+ pr_err("%s: AFE close failed %d\n", __func__, ret);
+ return -EINVAL;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*bharath, memory map handle needs to be stored by AFE client */
+int afe_cmd_memory_map(u32 dma_addr_p, u32 dma_buf_sz)
+{
+ int ret = 0;
+ int cmd_size = 0;
+ void *payload = NULL;
+ void *mmap_region_cmd = NULL;
+ struct afe_service_cmd_shared_mem_map_regions *mregion = NULL;
+ struct afe_service_shared_map_region_payload *mregion_pl = NULL;
+ int index = 0;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ cmd_size = sizeof(struct afe_service_cmd_shared_mem_map_regions) \
+ + sizeof(struct afe_service_shared_map_region_payload);
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (!mmap_region_cmd) {
+ pr_err("%s: allocate mmap_region_cmd failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ mregion = (struct afe_service_cmd_shared_mem_map_regions *)
+ mmap_region_cmd;
+ mregion->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion->hdr.pkt_size = sizeof(mregion);
+ mregion->hdr.src_port = 0;
+ mregion->hdr.dest_port = 0;
+ mregion->hdr.token = 0;
+ mregion->hdr.opcode = AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS;
+ mregion->mem_pool_id = ADSP_MEMORY_MAP_EBI_POOL;
+ mregion->num_regions = 1;
+ mregion->property_flag = 0x00;
+ /* Todo */
+ index = mregion->hdr.token = IDX_RSVD_2;
+
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct afe_service_cmd_shared_mem_map_regions));
+
+ mregion_pl = (struct afe_service_shared_map_region_payload *)payload;
+
+ mregion_pl->shm_addr_lsw = dma_addr_p;
+ mregion_pl->shm_addr_msw = 0x00;
+ mregion_pl->mem_size_bytes = dma_buf_sz;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) mmap_region_cmd);
+ if (ret < 0) {
+ pr_err("%s: AFE memory map cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ return 0;
+}
+
+int afe_cmd_memory_map_nowait(int port_id, u32 dma_addr_p, u32 dma_buf_sz)
+{
+ int ret = 0;
+ int cmd_size = 0;
+ void *payload = NULL;
+ void *mmap_region_cmd = NULL;
+ struct afe_service_cmd_shared_mem_map_regions *mregion = NULL;
+ struct afe_service_shared_map_region_payload *mregion_pl = NULL;
+ int index = 0;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ cmd_size = sizeof(struct afe_service_cmd_shared_mem_map_regions)
+ + sizeof(struct afe_service_shared_map_region_payload);
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (!mmap_region_cmd) {
+ pr_err("%s: allocate mmap_region_cmd failed\n", __func__);
+ return -ENOMEM;
+ }
+ mregion = (struct afe_service_cmd_shared_mem_map_regions *)
+ mmap_region_cmd;
+ mregion->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion->hdr.pkt_size = sizeof(mregion);
+ mregion->hdr.src_port = 0;
+ mregion->hdr.dest_port = 0;
+ mregion->hdr.token = 0;
+ mregion->hdr.opcode = AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS;
+ mregion->mem_pool_id = ADSP_MEMORY_MAP_EBI_POOL;
+ mregion->num_regions = 1;
+ mregion->property_flag = 0x00;
+
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct afe_service_cmd_shared_mem_map_regions));
+ mregion_pl = (struct afe_service_shared_map_region_payload *)payload;
+
+ mregion_pl->shm_addr_lsw = dma_addr_p;
+ mregion_pl->shm_addr_msw = 0x00;
+ mregion_pl->mem_size_bytes = dma_buf_sz;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) mmap_region_cmd);
+ if (ret < 0) {
+ pr_err("%s: AFE memory map cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_cmd_memory_unmap(u32 mem_map_handle)
+{
+ int ret = 0;
+ struct afe_service_cmd_shared_mem_unmap_regions mregion;
+ int index = 0;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ mregion.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion.hdr.pkt_size = sizeof(mregion);
+ mregion.hdr.src_port = 0;
+ mregion.hdr.dest_port = 0;
+ mregion.hdr.token = 0;
+ mregion.hdr.opcode = AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS;
+ mregion.mem_map_handle = mem_map_handle;
+
+ /* Todo */
+ index = mregion.hdr.token = IDX_RSVD_2;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &mregion);
+ if (ret < 0) {
+ pr_err("%s: AFE memory unmap cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_cmd_memory_unmap_nowait(u32 mem_map_handle)
+{
+ int ret = 0;
+ struct afe_service_cmd_shared_mem_unmap_regions mregion;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+
+ mregion.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ mregion.hdr.pkt_size = sizeof(mregion);
+ mregion.hdr.src_port = 0;
+ mregion.hdr.dest_port = 0;
+ mregion.hdr.token = 0;
+ mregion.hdr.opcode = AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS;
+ mregion.mem_map_handle = mem_map_handle;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &mregion);
+ if (ret < 0) {
+ pr_err("%s: AFE memory unmap cmd failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ }
+ return 0;
+}
+
+int afe_register_get_events(u16 port_id,
+ void (*cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv),
+ void *private_data)
+{
+ int ret = 0;
+ struct afe_service_cmd_register_rt_port_driver rtproxy;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+ else
+ return -EINVAL;
+
+ if (port_id == RT_PROXY_PORT_001_TX) {
+ this_afe.tx_cb = cb;
+ this_afe.tx_private_data = private_data;
+ } else if (port_id == RT_PROXY_PORT_001_RX) {
+ this_afe.rx_cb = cb;
+ this_afe.rx_private_data = private_data;
+ }
+
+ rtproxy.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ rtproxy.hdr.pkt_size = sizeof(rtproxy);
+ rtproxy.hdr.src_port = 1;
+ rtproxy.hdr.dest_port = 1;
+ rtproxy.hdr.opcode = AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER;
+ rtproxy.port_id = port_id;
+ rtproxy.reserved = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &rtproxy);
+ if (ret < 0) {
+ pr_err("%s: AFE reg. rtproxy_event failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_unregister_get_events(u16 port_id)
+{
+ int ret = 0;
+ struct afe_service_cmd_unregister_rt_port_driver rtproxy;
+ int index = 0;
+
+ pr_debug("%s:\n", __func__);
+
+ if (this_afe.apr == NULL) {
+ this_afe.apr = apr_register("ADSP", "AFE", afe_callback,
+ 0xFFFFFFFF, &this_afe);
+ pr_debug("%s: Register AFE\n", __func__);
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Unable to register AFE\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ }
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ if ((port_id == RT_PROXY_DAI_002_RX) ||
+ (port_id == RT_PROXY_DAI_001_TX))
+ port_id = VIRTUAL_ID_TO_PORTID(port_id);
+ else
+ return -EINVAL;
+
+ rtproxy.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ rtproxy.hdr.pkt_size = sizeof(rtproxy);
+ rtproxy.hdr.src_port = 0;
+ rtproxy.hdr.dest_port = 0;
+ rtproxy.hdr.token = 0;
+ rtproxy.hdr.opcode = AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER;
+ rtproxy.port_id = port_id;
+ rtproxy.reserved = 0;
+
+ rtproxy.hdr.token = index;
+
+ if (port_id == RT_PROXY_PORT_001_TX) {
+ this_afe.tx_cb = NULL;
+ this_afe.tx_private_data = NULL;
+ } else if (port_id == RT_PROXY_PORT_001_RX) {
+ this_afe.rx_cb = NULL;
+ this_afe.rx_private_data = NULL;
+ }
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &rtproxy);
+ if (ret < 0) {
+ pr_err("%s: AFE enable Unreg. rtproxy_event failed %d\n",
+ __func__, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+int afe_rt_proxy_port_write(u32 buf_addr_p, u32 mem_map_handle, int bytes)
+{
+ int ret = 0;
+ struct afe_port_data_cmd_rt_proxy_port_write_v2 afecmd_wr;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s:register to AFE is not done\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ pr_debug("%s: buf_addr_p = 0x%08x bytes = %d\n", __func__,
+ buf_addr_p, bytes);
+
+ afecmd_wr.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ afecmd_wr.hdr.pkt_size = sizeof(afecmd_wr);
+ afecmd_wr.hdr.src_port = 0;
+ afecmd_wr.hdr.dest_port = 0;
+ afecmd_wr.hdr.token = 0;
+ afecmd_wr.hdr.opcode = AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2;
+ afecmd_wr.port_id = RT_PROXY_PORT_001_TX;
+ afecmd_wr.buffer_address_lsw = (uint32_t)buf_addr_p;
+ afecmd_wr.buffer_address_msw = 0x00;
+ afecmd_wr.mem_map_handle = mem_map_handle;
+ afecmd_wr.available_bytes = bytes;
+ afecmd_wr.reserved = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &afecmd_wr);
+ if (ret < 0) {
+ pr_err("%s: AFE rtproxy write to port 0x%x failed %d\n",
+ __func__, afecmd_wr.port_id, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+
+}
+
+int afe_rt_proxy_port_read(u32 buf_addr_p, u32 mem_map_handle, int bytes)
+{
+ int ret = 0;
+ struct afe_port_data_cmd_rt_proxy_port_read_v2 afecmd_rd;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: register to AFE is not done\n", __func__);
+ ret = -ENODEV;
+ return ret;
+ }
+ pr_debug("%s: buf_addr_p = 0x%08x bytes = %d\n", __func__,
+ buf_addr_p, bytes);
+
+ afecmd_rd.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ afecmd_rd.hdr.pkt_size = sizeof(afecmd_rd);
+ afecmd_rd.hdr.src_port = 0;
+ afecmd_rd.hdr.dest_port = 0;
+ afecmd_rd.hdr.token = 0;
+ afecmd_rd.hdr.opcode = AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2;
+ afecmd_rd.port_id = RT_PROXY_PORT_001_RX;
+ afecmd_rd.buffer_address_lsw = (uint32_t)buf_addr_p;
+ afecmd_rd.buffer_address_msw = 0x00;
+ afecmd_rd.available_bytes = bytes;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &afecmd_rd);
+ if (ret < 0) {
+ pr_err("%s: AFE rtproxy read cmd to port 0x%x failed %d\n",
+ __func__, afecmd_rd.port_id, ret);
+ ret = -EINVAL;
+ return ret;
+ }
+ return 0;
+}
+
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_afelb;
+static struct dentry *debugfs_afelb_gain;
+
+static int afe_debug_open(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ pr_info("debug intf %s\n", (char *) file->private_data);
+ return 0;
+}
+
+static int afe_get_parameters(char *buf, long int *param1, int num_of_par)
+{
+ char *token;
+ int base, cnt;
+
+ token = strsep(&buf, " ");
+
+ for (cnt = 0; cnt < num_of_par; cnt++) {
+ if (token != NULL) {
+ if ((token[1] == 'x') || (token[1] == 'X'))
+ base = 16;
+ else
+ base = 10;
+
+ if (strict_strtoul(token, base, ¶m1[cnt]) != 0)
+ return -EINVAL;
+
+ token = strsep(&buf, " ");
+ } else
+ return -EINVAL;
+ }
+ return 0;
+}
+#define AFE_LOOPBACK_ON (1)
+#define AFE_LOOPBACK_OFF (0)
+static ssize_t afe_debug_write(struct file *filp,
+ const char __user *ubuf, size_t cnt, loff_t *ppos)
+{
+ char *lb_str = filp->private_data;
+ char lbuf[32];
+ int rc;
+ unsigned long param[5];
+
+ if (cnt > sizeof(lbuf) - 1)
+ return -EINVAL;
+
+ rc = copy_from_user(lbuf, ubuf, cnt);
+ if (rc)
+ return -EFAULT;
+
+ lbuf[cnt] = '\0';
+
+ if (!strncmp(lb_str, "afe_loopback", 12)) {
+ rc = afe_get_parameters(lbuf, param, 3);
+ if (!rc) {
+ pr_info("%s %lu %lu %lu\n", lb_str, param[0], param[1],
+ param[2]);
+
+ if ((param[0] != AFE_LOOPBACK_ON) && (param[0] !=
+ AFE_LOOPBACK_OFF)) {
+ pr_err("%s: Error, parameter 0 incorrect\n",
+ __func__);
+ rc = -EINVAL;
+ goto afe_error;
+ }
+ if ((q6audio_validate_port(param[1]) < 0) ||
+ (q6audio_validate_port(param[2])) < 0) {
+ pr_err("%s: Error, invalid afe port\n",
+ __func__);
+ }
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Error, AFE not opened\n", __func__);
+ rc = -EINVAL;
+ } else {
+ rc = afe_loopback(param[0], param[1], param[2]);
+ }
+ } else {
+ pr_err("%s: Error, invalid parameters\n", __func__);
+ rc = -EINVAL;
+ }
+
+ } else if (!strncmp(lb_str, "afe_loopback_gain", 17)) {
+ rc = afe_get_parameters(lbuf, param, 2);
+ if (!rc) {
+ pr_info("%s %lu %lu\n", lb_str, param[0], param[1]);
+
+ if (q6audio_validate_port(param[0]) < 0) {
+ pr_err("%s: Error, invalid afe port\n",
+ __func__);
+ rc = -EINVAL;
+ goto afe_error;
+ }
+
+ if (param[1] < 0 || param[1] > 100) {
+ pr_err("%s: Error, volume shoud be 0 to 100"
+ " percentage param = %lu\n",
+ __func__, param[1]);
+ rc = -EINVAL;
+ goto afe_error;
+ }
+
+ param[1] = (Q6AFE_MAX_VOLUME * param[1]) / 100;
+
+ if (this_afe.apr == NULL) {
+ pr_err("%s: Error, AFE not opened\n", __func__);
+ rc = -EINVAL;
+ } else {
+ rc = afe_loopback_gain(param[0], param[1]);
+ }
+ } else {
+ pr_err("%s: Error, invalid parameters\n", __func__);
+ rc = -EINVAL;
+ }
+ }
+
+afe_error:
+ if (rc == 0)
+ rc = cnt;
+ else
+ pr_err("%s: rc = %d\n", __func__, rc);
+
+ return rc;
+}
+
+static const struct file_operations afe_debug_fops = {
+ .open = afe_debug_open,
+ .write = afe_debug_write
+};
+
+static void config_debug_fs_init(void)
+{
+ debugfs_afelb = debugfs_create_file("afe_loopback",
+ S_IFREG | S_IWUGO, NULL, (void *) "afe_loopback",
+ &afe_debug_fops);
+
+ debugfs_afelb_gain = debugfs_create_file("afe_loopback_gain",
+ S_IFREG | S_IWUGO, NULL, (void *) "afe_loopback_gain",
+ &afe_debug_fops);
+}
+static void config_debug_fs_exit(void)
+{
+ if (debugfs_afelb)
+ debugfs_remove(debugfs_afelb);
+ if (debugfs_afelb_gain)
+ debugfs_remove(debugfs_afelb_gain);
+}
+#else
+static void config_debug_fs_init(void)
+{
+ return;
+}
+static void config_debug_fs_exit(void)
+{
+ return;
+}
+#endif
+int afe_sidetone(u16 tx_port_id, u16 rx_port_id, u16 enable, uint16_t gain)
+{
+ struct afe_loopback_cfg_v1 cmd_sidetone;
+ int ret = 0;
+ int index = 0;
+
+ pr_info("%s: tx_port_id:%d rx_port_id:%d enable:%d gain:%d\n", __func__,
+ tx_port_id, rx_port_id, enable, gain);
+ index = q6audio_get_port_index(rx_port_id);
+ if (q6audio_validate_port(rx_port_id) < 0)
+ return -EINVAL;
+
+ cmd_sidetone.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ cmd_sidetone.hdr.pkt_size = sizeof(cmd_sidetone);
+ cmd_sidetone.hdr.src_port = 0;
+ cmd_sidetone.hdr.dest_port = 0;
+ cmd_sidetone.hdr.token = 0;
+ cmd_sidetone.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+ /* should it be rx or tx port id ?? , bharath*/
+ cmd_sidetone.param.port_id = tx_port_id;
+ /* size of data param & payload */
+ cmd_sidetone.param.payload_size = (sizeof(cmd_sidetone) -
+ sizeof(struct apr_hdr) -
+ sizeof(struct afe_port_cmd_set_param_v2));
+ cmd_sidetone.param.payload_address_lsw = 0x00;
+ cmd_sidetone.param.payload_address_msw = 0x00;
+ cmd_sidetone.param.mem_map_handle = 0x00;
+ cmd_sidetone.pdata.module_id = AFE_MODULE_LOOPBACK;
+ cmd_sidetone.pdata.param_id = AFE_PARAM_ID_LOOPBACK_CONFIG;
+ /* size of actual payload only */
+ cmd_sidetone.pdata.param_size = cmd_sidetone.param.payload_size -
+ sizeof(struct afe_port_param_data_v2);
+
+ cmd_sidetone.loopback_cfg_minor_version =
+ AFE_API_VERSION_LOOPBACK_CONFIG;
+ cmd_sidetone.dst_port_id = rx_port_id;
+ cmd_sidetone.routing_mode = LB_MODE_SIDETONE;
+ cmd_sidetone.enable = enable;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &cmd_sidetone);
+ if (ret < 0) {
+ pr_err("%s: AFE sidetone failed for tx_port:%d rx_port:%d\n",
+ __func__, tx_port_id, rx_port_id);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (ret < 0) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return ret;
+}
+
+int afe_port_stop_nowait(int port_id)
+{
+ struct afe_port_cmd_device_stop stop;
+ int ret = 0;
+
+ if (this_afe.apr == NULL) {
+ pr_err("AFE is already closed\n");
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+ port_id = q6audio_convert_virtual_to_portid(port_id);
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = 0;
+ stop.hdr.opcode = AFE_PORT_CMD_DEVICE_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+
+ if (IS_ERR_VALUE(ret)) {
+ pr_err("%s: AFE close failed\n", __func__);
+ ret = -EINVAL;
+ }
+
+fail_cmd:
+ return ret;
+
+}
+
+int afe_close(int port_id)
+{
+ struct afe_port_cmd_device_stop stop;
+ int ret = 0;
+ int index = 0;
+
+
+ if (this_afe.apr == NULL) {
+ pr_err("AFE is already closed\n");
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ pr_debug("%s: port_id=%d\n", __func__, port_id);
+
+ index = q6audio_get_port_index(port_id);
+ if (q6audio_validate_port(port_id) < 0)
+ return -EINVAL;
+
+ port_id = q6audio_convert_virtual_to_portid(port_id);
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = index;
+ stop.hdr.opcode = AFE_PORT_CMD_DEVICE_STOP;
+ stop.port_id = q6audio_get_port_id(port_id);
+ stop.reserved = 0;
+
+ atomic_set(&this_afe.state, 1);
+ ret = apr_send_pkt(this_afe.apr, (uint32_t *) &stop);
+
+ if (ret < 0) {
+ pr_err("%s: AFE close failed\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ ret = wait_event_timeout(this_afe.wait[index],
+ (atomic_read(&this_afe.state) == 0),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ pr_err("%s: wait_event timeout\n", __func__);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+fail_cmd:
+ return ret;
+}
+
+static int __init afe_init(void)
+{
+ int i = 0;
+ atomic_set(&this_afe.state, 0);
+ atomic_set(&this_afe.status, 0);
+ this_afe.apr = NULL;
+ for (i = 0; i < AFE_MAX_PORTS; i++)
+ init_waitqueue_head(&this_afe.wait[i]);
+
+ config_debug_fs_init();
+ return 0;
+}
+
+static void __exit afe_exit(void)
+{
+ int i;
+
+ config_debug_fs_exit();
+ for (i = 0; i < MAX_AUDPROC_TYPES; i++) {
+ if (afe_cal_addr[i].cal_paddr != 0)
+ afe_cmd_memory_unmap_nowait(
+ afe_cal_addr[i].cal_paddr);
+ }
+}
+
+device_initcall(afe_init);
+__exitcall(afe_exit);
diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c
new file mode 100644
index 0000000..f982134
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/q6asm.c
@@ -0,0 +1,3342 @@
+/*
+ * Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ * Author: Brian Swetland <swetland@google.com>
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+#include <linux/fs.h>
+#include <linux/mutex.h>
+#include <linux/wait.h>
+#include <linux/miscdevice.h>
+#include <linux/uaccess.h>
+#include <linux/sched.h>
+#include <linux/dma-mapping.h>
+#include <linux/miscdevice.h>
+#include <linux/delay.h>
+#include <linux/spinlock.h>
+#include <linux/slab.h>
+#include <linux/msm_audio.h>
+#include <linux/android_pmem.h>
+#include <linux/memory_alloc.h>
+#include <linux/debugfs.h>
+#include <linux/time.h>
+#include <linux/atomic.h>
+
+#include <asm/ioctls.h>
+
+#include <mach/memory.h>
+#include <mach/debug_mm.h>
+#include <mach/peripheral-loader.h>
+#include <mach/qdsp6v2/audio_acdb.h>
+#include <mach/qdsp6v2/rtac.h>
+#include <mach/msm_subsystem_map.h>
+
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+
+#define TRUE 0x01
+#define FALSE 0x00
+#define READDONE_IDX_STATUS 0
+#define READDONE_IDX_BUFADD_LSW 1
+#define READDONE_IDX_BUFADD_MSW 2
+#define READDONE_IDX_MEMMAP_HDL 3
+#define READDONE_IDX_SIZE 4
+#define READDONE_IDX_OFFSET 5
+#define READDONE_IDX_LSW_TS 6
+#define READDONE_IDX_MSW_TS 7
+#define READDONE_IDX_FLAGS 8
+#define READDONE_IDX_NUMFRAMES 9
+#define READDONE_IDX_SEQ_ID 10
+
+/* TODO, combine them together */
+static DEFINE_MUTEX(session_lock);
+struct asm_mmap {
+ atomic_t ref_cnt;
+ void *apr;
+};
+
+static struct asm_mmap this_mmap;
+/* session id: 0 reserved */
+static struct audio_client *session[SESSION_MAX+1];
+
+struct asm_buffer_node {
+ struct list_head list;
+ uint32_t buf_addr_lsw;
+ uint32_t mmap_hdl;
+};
+static int32_t q6asm_mmapcallback(struct apr_client_data *data, void *priv);
+static int32_t q6asm_callback(struct apr_client_data *data, void *priv);
+static void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg);
+static void q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg);
+static int q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt);
+static int q6asm_memory_unmap_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt);
+static void q6asm_reset_buf_state(struct audio_client *ac);
+
+
+#ifdef CONFIG_DEBUG_FS
+#define OUT_BUFFER_SIZE 56
+#define IN_BUFFER_SIZE 24
+
+static struct timeval out_cold_tv;
+static struct timeval out_warm_tv;
+static struct timeval out_cont_tv;
+static struct timeval in_cont_tv;
+static long out_enable_flag;
+static long in_enable_flag;
+static struct dentry *out_dentry;
+static struct dentry *in_dentry;
+static int in_cont_index;
+/*This var is used to keep track of first write done for cold output latency */
+static int out_cold_index;
+static char *out_buffer;
+static char *in_buffer;
+static int audio_output_latency_dbgfs_open(struct inode *inode,
+ struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+static ssize_t audio_output_latency_dbgfs_read(struct file *file,
+ char __user *buf, size_t count, loff_t *ppos)
+{
+ snprintf(out_buffer, OUT_BUFFER_SIZE, "%ld,%ld,%ld,%ld,%ld,%ld,",\
+ out_cold_tv.tv_sec, out_cold_tv.tv_usec, out_warm_tv.tv_sec,\
+ out_warm_tv.tv_usec, out_cont_tv.tv_sec, out_cont_tv.tv_usec);
+ return simple_read_from_buffer(buf, OUT_BUFFER_SIZE, ppos,
+ out_buffer, OUT_BUFFER_SIZE);
+}
+static ssize_t audio_output_latency_dbgfs_write(struct file *file,
+ const char __user *buf, size_t count, loff_t *ppos)
+{
+ char *temp;
+
+ if (count > 2*sizeof(char))
+ return -EINVAL;
+ else
+ temp = kmalloc(2*sizeof(char), GFP_KERNEL);
+
+ out_cold_index = 0;
+
+ if (temp) {
+ if (copy_from_user(temp, buf, 2*sizeof(char))) {
+ kfree(temp);
+ return -EFAULT;
+ }
+ if (!strict_strtol(temp, 10, &out_enable_flag)) {
+ kfree(temp);
+ return count;
+ }
+ kfree(temp);
+ }
+ return -EINVAL;
+}
+static const struct file_operations audio_output_latency_debug_fops = {
+ .open = audio_output_latency_dbgfs_open,
+ .read = audio_output_latency_dbgfs_read,
+ .write = audio_output_latency_dbgfs_write
+};
+static int audio_input_latency_dbgfs_open(struct inode *inode,
+ struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+static ssize_t audio_input_latency_dbgfs_read(struct file *file,
+ char __user *buf, size_t count, loff_t *ppos)
+{
+ snprintf(in_buffer, IN_BUFFER_SIZE, "%ld,%ld,",\
+ in_cont_tv.tv_sec, in_cont_tv.tv_usec);
+ return simple_read_from_buffer(buf, IN_BUFFER_SIZE, ppos,
+ in_buffer, IN_BUFFER_SIZE);
+}
+static ssize_t audio_input_latency_dbgfs_write(struct file *file,
+ const char __user *buf, size_t count, loff_t *ppos)
+{
+ char *temp;
+
+ if (count > 2*sizeof(char))
+ return -EINVAL;
+ else
+ temp = kmalloc(2*sizeof(char), GFP_KERNEL);
+ if (temp) {
+ if (copy_from_user(temp, buf, 2*sizeof(char))) {
+ kfree(temp);
+ return -EFAULT;
+ }
+ if (!strict_strtol(temp, 10, &in_enable_flag)) {
+ kfree(temp);
+ return count;
+ }
+ kfree(temp);
+ }
+ return -EINVAL;
+}
+static const struct file_operations audio_input_latency_debug_fops = {
+ .open = audio_input_latency_dbgfs_open,
+ .read = audio_input_latency_dbgfs_read,
+ .write = audio_input_latency_dbgfs_write
+};
+
+static void config_debug_fs_write_cb(void)
+{
+ if (out_enable_flag) {
+ /* For first Write done log the time and reset
+ out_cold_index*/
+ if (out_cold_index != 1) {
+ do_gettimeofday(&out_cold_tv);
+ pr_debug("COLD: apr_send_pkt at %ld"
+ "sec %ld microsec\n",\
+ out_cold_tv.tv_sec,\
+ out_cold_tv.tv_usec);
+ out_cold_index = 1;
+ }
+ pr_debug("out_enable_flag %ld",\
+ out_enable_flag);
+ }
+}
+static void config_debug_fs_read_cb(void)
+{
+ if (in_enable_flag) {
+ /* when in_cont_index == 7, DSP would be
+ * writing into the 8th 512 byte buffer and this
+ * timestamp is tapped here.Once done it then writes
+ * to 9th 512 byte buffer.These two buffers(8th, 9th)
+ * reach the test application in 5th iteration and that
+ * timestamp is tapped at user level. The difference
+ * of these two timestamps gives us the time between
+ * the time at which dsp started filling the sample
+ * required and when it reached the test application.
+ * Hence continuous input latency
+ */
+ if (in_cont_index == 7) {
+ do_gettimeofday(&in_cont_tv);
+ pr_err("In_CONT:previous read buffer done"
+ "at %ld sec %ld microsec\n",\
+ in_cont_tv.tv_sec, in_cont_tv.tv_usec);
+ }
+ in_cont_index++;
+ }
+}
+
+static void config_debug_fs_reset_index(void)
+{
+ in_cont_index = 0;
+}
+
+static void config_debug_fs_run(void)
+{
+ if (out_enable_flag) {
+ do_gettimeofday(&out_cold_tv);
+ pr_debug("COLD: apr_send_pkt at %ld sec %ld microsec\n",\
+ out_cold_tv.tv_sec, out_cold_tv.tv_usec);
+ }
+}
+
+static void config_debug_fs_write(struct audio_buffer *ab)
+{
+ if (out_enable_flag) {
+ char zero_pattern[2] = {0x00, 0x00};
+ /* If First two byte is non zero and last two byte
+ is zero then it is warm output pattern */
+ if ((strncmp(((char *)ab->data), zero_pattern, 2)) &&
+ (!strncmp(((char *)ab->data + 2), zero_pattern, 2))) {
+ do_gettimeofday(&out_warm_tv);
+ pr_debug("WARM:apr_send_pkt at"
+ "%ld sec %ld microsec\n", out_warm_tv.tv_sec,\
+ out_warm_tv.tv_usec);
+ pr_debug("Warm Pattern Matched");
+ }
+ /* If First two byte is zero and last two byte is
+ non zero then it is cont ouput pattern */
+ else if ((!strncmp(((char *)ab->data), zero_pattern, 2))
+ && (strncmp(((char *)ab->data + 2), zero_pattern, 2))) {
+ do_gettimeofday(&out_cont_tv);
+ pr_debug("CONT:apr_send_pkt at"
+ "%ld sec %ld microsec\n", out_cont_tv.tv_sec,\
+ out_cont_tv.tv_usec);
+ pr_debug("Cont Pattern Matched");
+ }
+ }
+}
+static void config_debug_fs_init(void)
+{
+ out_buffer = kmalloc(OUT_BUFFER_SIZE, GFP_KERNEL);
+ out_dentry = debugfs_create_file("audio_out_latency_measurement_node",\
+ S_IFREG | S_IRUGO | S_IWUGO,\
+ NULL, NULL, &audio_output_latency_debug_fops);
+ if (IS_ERR(out_dentry))
+ pr_err("debugfs_create_file failed\n");
+ in_buffer = kmalloc(IN_BUFFER_SIZE, GFP_KERNEL);
+ in_dentry = debugfs_create_file("audio_in_latency_measurement_node",\
+ S_IFREG | S_IRUGO | S_IWUGO,\
+ NULL, NULL, &audio_input_latency_debug_fops);
+ if (IS_ERR(in_dentry))
+ pr_err("debugfs_create_file failed\n");
+}
+#else
+static void config_debug_fs_write(struct audio_buffer *ab)
+{
+ return;
+}
+static void config_debug_fs_run(void)
+{
+ return;
+}
+static void config_debug_fs_reset_index(void)
+{
+ return;
+}
+static void config_debug_fs_read_cb(void)
+{
+ return;
+}
+static void config_debug_fs_write_cb(void)
+{
+ return;
+}
+static void config_debug_fs_init(void)
+{
+ return;
+}
+#endif
+
+
+static int q6asm_session_alloc(struct audio_client *ac)
+{
+ int n;
+ mutex_lock(&session_lock);
+ for (n = 1; n <= SESSION_MAX; n++) {
+ if (!session[n]) {
+ session[n] = ac;
+ mutex_unlock(&session_lock);
+ return n;
+ }
+ }
+ mutex_unlock(&session_lock);
+ return -ENOMEM;
+}
+
+static void q6asm_session_free(struct audio_client *ac)
+{
+ pr_debug("%s: sessionid[%d]\n", __func__, ac->session);
+ rtac_remove_popp_from_adm_devices(ac->session);
+ mutex_lock(&session_lock);
+ session[ac->session] = 0;
+ mutex_unlock(&session_lock);
+ ac->session = 0;
+ return;
+}
+
+int q6asm_audio_client_buf_free(unsigned int dir,
+ struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+ pr_debug("%s: Session id %d\n", __func__, ac->session);
+ mutex_lock(&ac->cmd_lock);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[dir];
+ if (!port->buf) {
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+ }
+ cnt = port->max_buf_cnt - 1;
+
+ if (cnt >= 0) {
+ rc = q6asm_memory_unmap_regions(ac, dir,
+ port->buf[0].size,
+ port->max_buf_cnt);
+ if (rc < 0)
+ pr_err("%s CMD Memory_unmap_regions failed\n",
+ __func__);
+ }
+
+ while (cnt >= 0) {
+ if (port->buf[cnt].data) {
+ ion_unmap_kernel(port->buf[cnt].client,
+ port->buf[cnt].handle);
+ ion_free(port->buf[cnt].client,
+ port->buf[cnt].handle);
+ ion_client_destroy(port->buf[cnt].client);
+ port->buf[cnt].data = NULL;
+ port->buf[cnt].phys = 0;
+ --(port->max_buf_cnt);
+ }
+ --cnt;
+ }
+ kfree(port->buf);
+ port->buf = NULL;
+ }
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+}
+
+int q6asm_audio_client_buf_free_contiguous(unsigned int dir,
+ struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+ pr_debug("%s: Session id %d\n", __func__, ac->session);
+ mutex_lock(&ac->cmd_lock);
+ port = &ac->port[dir];
+ if (!port->buf) {
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+ }
+ cnt = port->max_buf_cnt - 1;
+
+ if (cnt >= 0) {
+ rc = q6asm_memory_unmap(ac, port->buf[0].phys, dir);
+ if (rc < 0)
+ pr_err("%s CMD Memory_unmap_regions failed\n",
+ __func__);
+ }
+
+ if (port->buf[0].data) {
+ ion_unmap_kernel(port->buf[0].client, port->buf[0].handle);
+ ion_free(port->buf[0].client, port->buf[0].handle);
+ ion_client_destroy(port->buf[0].client);
+ pr_debug("%s:data[%p]phys[%p][%p]"
+ ", client[%p] handle[%p]\n",
+ __func__,
+ (void *)port->buf[0].data,
+ (void *)port->buf[0].phys,
+ (void *)&port->buf[0].phys,
+ (void *)port->buf[0].client,
+ (void *)port->buf[0].handle);
+ }
+
+ while (cnt >= 0) {
+ port->buf[cnt].data = NULL;
+ port->buf[cnt].phys = 0;
+ cnt--;
+ }
+ port->max_buf_cnt = 0;
+ kfree(port->buf);
+ port->buf = NULL;
+ mutex_unlock(&ac->cmd_lock);
+ return 0;
+}
+
+int q6asm_mmap_apr_dereg(void)
+{
+ if (atomic_read(&this_mmap.ref_cnt) <= 0) {
+ pr_err("%s: APR Common Port Already Closed\n", __func__);
+ goto done;
+ }
+ atomic_dec(&this_mmap.ref_cnt);
+ if (atomic_read(&this_mmap.ref_cnt) == 0) {
+ apr_deregister(this_mmap.apr);
+ pr_debug("%s:APR De-Register common port\n", __func__);
+ }
+done:
+ return 0;
+}
+
+
+void q6asm_audio_client_free(struct audio_client *ac)
+{
+ int loopcnt;
+ struct audio_port_data *port;
+ if (!ac || !ac->session)
+ return;
+ pr_debug("%s: Session id %d\n", __func__, ac->session);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
+ port = &ac->port[loopcnt];
+ if (!port->buf)
+ continue;
+ pr_debug("%s:loopcnt = %d\n", __func__, loopcnt);
+ q6asm_audio_client_buf_free(loopcnt, ac);
+ }
+ }
+
+ apr_deregister(ac->apr);
+ ac->mmap_apr = NULL;
+ q6asm_session_free(ac);
+ q6asm_mmap_apr_dereg();
+
+ pr_debug("%s: APR De-Register\n", __func__);
+
+/*done:*/
+ kfree(ac);
+ return;
+}
+
+int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode)
+{
+ if (ac == NULL) {
+ pr_err("%s APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ if ((mode == ASYNC_IO_MODE) || (mode == SYNC_IO_MODE)) {
+ ac->io_mode = mode;
+ pr_debug("%s:Set Mode to %d\n", __func__, ac->io_mode);
+ return 0;
+ } else {
+ pr_err("%s:Not an valid IO Mode:%d\n", __func__, ac->io_mode);
+ return -EINVAL;
+ }
+}
+
+void *q6asm_mmap_apr_reg(void)
+{
+ if (atomic_read(&this_mmap.ref_cnt) == 0) {
+ this_mmap.apr = apr_register("ADSP", "ASM", \
+ (apr_fn)q6asm_mmapcallback,\
+ 0x0FFFFFFFF, &this_mmap);
+ if (this_mmap.apr == NULL) {
+ pr_debug("%s Unable to register"
+ "APR ASM common port\n", __func__);
+ goto fail;
+ }
+ }
+ atomic_inc(&this_mmap.ref_cnt);
+ return this_mmap.apr;
+fail:
+ return NULL;
+}
+
+struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv)
+{
+ struct audio_client *ac;
+ int n;
+ int lcnt = 0;
+
+ ac = kzalloc(sizeof(struct audio_client), GFP_KERNEL);
+ if (!ac)
+ return NULL;
+ n = q6asm_session_alloc(ac);
+ if (n <= 0)
+ goto fail_session;
+ ac->session = n;
+ ac->cb = cb;
+ ac->priv = priv;
+ ac->io_mode = SYNC_IO_MODE;
+ ac->apr = apr_register("ADSP", "ASM", \
+ (apr_fn)q6asm_callback,\
+ ((ac->session) << 8 | 0x0001),\
+ ac);
+
+ if (ac->apr == NULL) {
+ pr_err("%s Registration with APR failed\n", __func__);
+ goto fail;
+ }
+ rtac_set_asm_handle(n, ac->apr);
+
+ pr_debug("%s Registering the common port with APR\n", __func__);
+ ac->mmap_apr = q6asm_mmap_apr_reg();
+ if (ac->mmap_apr == NULL)
+ goto fail;
+
+ init_waitqueue_head(&ac->cmd_wait);
+ INIT_LIST_HEAD(&ac->port[0].mem_map_handle);
+ INIT_LIST_HEAD(&ac->port[1].mem_map_handle);
+ pr_debug("%s: mem_map_handle list init'ed\n", __func__);
+ mutex_init(&ac->cmd_lock);
+ for (lcnt = 0; lcnt <= OUT; lcnt++) {
+ mutex_init(&ac->port[lcnt].lock);
+ spin_lock_init(&ac->port[lcnt].dsp_lock);
+ }
+ atomic_set(&ac->cmd_state, 0);
+
+ pr_debug("%s: session[%d]\n", __func__, ac->session);
+
+ return ac;
+fail:
+ q6asm_audio_client_free(ac);
+ return NULL;
+fail_session:
+ kfree(ac);
+ return NULL;
+}
+
+struct audio_client *q6asm_get_audio_client(int session_id)
+{
+ if ((session_id <= 0) || (session_id > SESSION_MAX)) {
+ pr_err("%s: invalid session: %d\n", __func__, session_id);
+ goto err;
+ }
+
+ if (!session[session_id]) {
+ pr_err("%s: session not active: %d\n", __func__, session_id);
+ goto err;
+ }
+
+ return session[session_id];
+err:
+ return NULL;
+}
+
+int q6asm_audio_client_buf_alloc(unsigned int dir,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt)
+{
+ int cnt = 0;
+ int rc = 0;
+ struct audio_buffer *buf;
+ int len;
+
+ if (!(ac) || ((dir != IN) && (dir != OUT)))
+ return -EINVAL;
+
+ pr_debug("%s: session[%d]bufsz[%d]bufcnt[%d]\n", __func__, ac->session,
+ bufsz, bufcnt);
+
+ if (ac->session <= 0 || ac->session > 8)
+ goto fail;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ if (ac->port[dir].buf) {
+ pr_debug("%s: buffer already allocated\n", __func__);
+ return 0;
+ }
+ mutex_lock(&ac->cmd_lock);
+ buf = kzalloc(((sizeof(struct audio_buffer))*bufcnt),
+ GFP_KERNEL);
+
+ if (!buf) {
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ ac->port[dir].buf = buf;
+
+ while (cnt < bufcnt) {
+ if (bufsz > 0) {
+ if (!buf[cnt].data) {
+ buf[cnt].client = msm_ion_client_create
+ (UINT_MAX, "audio_client");
+ if (IS_ERR_OR_NULL((void *)
+ buf[cnt].client)) {
+ pr_err("%s: ION create client"
+ " for AUDIO failed\n",
+ __func__);
+ goto fail;
+ }
+ buf[cnt].handle = ion_alloc
+ (buf[cnt].client, bufsz, SZ_4K,
+ (0x1 << ION_AUDIO_HEAP_ID));
+ if (IS_ERR_OR_NULL((void *)
+ buf[cnt].handle)) {
+ pr_err("%s: ION memory"
+ " allocation for AUDIO failed\n",
+ __func__);
+ goto fail;
+ }
+
+ rc = ion_phys(buf[cnt].client,
+ buf[cnt].handle,
+ (ion_phys_addr_t *)
+ &buf[cnt].phys,
+ (size_t *)&len);
+ if (rc) {
+ pr_err("%s: ION Get Physical"
+ " for AUDIO failed, rc = %d\n",
+ __func__, rc);
+ goto fail;
+ }
+
+ buf[cnt].data = ion_map_kernel
+ (buf[cnt].client, buf[cnt].handle,
+ 0);
+ if (IS_ERR_OR_NULL((void *)
+ buf[cnt].data)) {
+ pr_err("%s: ION memory"
+ " mapping for AUDIO failed\n", __func__);
+ goto fail;
+ }
+ memset((void *)buf[cnt].data, 0, bufsz);
+ buf[cnt].used = 1;
+ buf[cnt].size = bufsz;
+ buf[cnt].actual_size = bufsz;
+ pr_debug("%s data[%p]phys[%p][%p]\n",
+ __func__,
+ (void *)buf[cnt].data,
+ (void *)buf[cnt].phys,
+ (void *)&buf[cnt].phys);
+ cnt++;
+ }
+ }
+ }
+ ac->port[dir].max_buf_cnt = cnt;
+
+ mutex_unlock(&ac->cmd_lock);
+ rc = q6asm_memory_map_regions(ac, dir, bufsz, cnt);
+ if (rc < 0) {
+ pr_err("%s:CMD Memory_map_regions failed\n", __func__);
+ goto fail;
+ }
+ }
+ return 0;
+fail:
+ q6asm_audio_client_buf_free(dir, ac);
+ return -EINVAL;
+}
+
+int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt)
+{
+ int cnt = 0;
+ int rc = 0;
+ struct audio_buffer *buf;
+ int len;
+
+ if (!(ac) || ((dir != IN) && (dir != OUT)))
+ return -EINVAL;
+
+ pr_debug("%s: session[%d]bufsz[%d]bufcnt[%d]\n",
+ __func__, ac->session,
+ bufsz, bufcnt);
+
+ if (ac->session <= 0 || ac->session > 8)
+ goto fail;
+
+ if (ac->port[dir].buf) {
+ pr_debug("%s: buffer already allocated\n", __func__);
+ return 0;
+ }
+ mutex_lock(&ac->cmd_lock);
+ buf = kzalloc(((sizeof(struct audio_buffer))*bufcnt),
+ GFP_KERNEL);
+
+ if (!buf) {
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ ac->port[dir].buf = buf;
+
+ buf[0].client = msm_ion_client_create(UINT_MAX, "audio_client");
+ if (IS_ERR_OR_NULL((void *)buf[0].client)) {
+ pr_err("%s: ION create client for AUDIO failed\n", __func__);
+ goto fail;
+ }
+ buf[0].handle = ion_alloc(buf[0].client, bufsz * bufcnt, SZ_4K,
+ (0x1 << ION_AUDIO_HEAP_ID));
+ if (IS_ERR_OR_NULL((void *) buf[0].handle)) {
+ pr_err("%s: ION memory allocation for AUDIO failed\n",
+ __func__);
+ goto fail;
+ }
+
+ rc = ion_phys(buf[0].client, buf[0].handle,
+ (ion_phys_addr_t *)&buf[0].phys, (size_t *)&len);
+ if (rc) {
+ pr_err("%s: ION Get Physical for AUDIO failed, rc = %d\n",
+ __func__, rc);
+ goto fail;
+ }
+
+ buf[0].data = ion_map_kernel(buf[0].client, buf[0].handle, 0);
+ if (IS_ERR_OR_NULL((void *) buf[0].data)) {
+ pr_err("%s: ION memory mapping for AUDIO failed\n", __func__);
+ goto fail;
+ }
+ memset((void *)buf[0].data, 0, (bufsz * bufcnt));
+ if (!buf[0].data) {
+ pr_err("%s:invalid vaddr,"
+ " iomap failed\n", __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+
+ buf[0].used = dir ^ 1;
+ buf[0].size = bufsz;
+ buf[0].actual_size = bufsz;
+ cnt = 1;
+ while (cnt < bufcnt) {
+ if (bufsz > 0) {
+ buf[cnt].data = buf[0].data + (cnt * bufsz);
+ buf[cnt].phys = buf[0].phys + (cnt * bufsz);
+ if (!buf[cnt].data) {
+ pr_err("%s Buf alloc failed\n",
+ __func__);
+ mutex_unlock(&ac->cmd_lock);
+ goto fail;
+ }
+ buf[cnt].used = dir ^ 1;
+ buf[cnt].size = bufsz;
+ buf[cnt].actual_size = bufsz;
+ pr_debug("%s data[%p]phys[%p][%p]\n", __func__,
+ (void *)buf[cnt].data,
+ (void *)buf[cnt].phys,
+ (void *)&buf[cnt].phys);
+ }
+ cnt++;
+ }
+ ac->port[dir].max_buf_cnt = cnt;
+ mutex_unlock(&ac->cmd_lock);
+ rc = q6asm_memory_map_regions(ac, dir, bufsz, cnt);
+ if (rc < 0) {
+ pr_err("%s:CMD Memory_map_regions failed\n", __func__);
+ goto fail;
+ }
+ return 0;
+fail:
+ q6asm_audio_client_buf_free_contiguous(dir, ac);
+ return -EINVAL;
+}
+
+static int32_t q6asm_mmapcallback(struct apr_client_data *data, void *priv)
+{
+ uint32_t sid = 0;
+ uint32_t dir = 0;
+ uint32_t *payload = data->payload;
+ unsigned long dsp_flags;
+
+ struct audio_client *ac = NULL;
+ struct audio_port_data *port;
+
+ if (!data) {
+ pr_err("%s: Invalid CB\n", __func__);
+ return 0;
+ }
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("%s: Reset event is received: %d %d apr[%p]\n",
+ __func__,
+ data->reset_event,
+ data->reset_proc,
+ this_mmap.apr);
+ apr_reset(this_mmap.apr);
+ atomic_set(&this_mmap.ref_cnt, 0);
+ this_mmap.apr = NULL;
+ return 0;
+ }
+ sid = (data->token >> 8) & 0x0F;
+ ac = q6asm_get_audio_client(sid);
+ pr_debug("%s:ptr0[0x%x]ptr1[0x%x]opcode[0x%x]"
+ "token[0x%x]payload_s[%d] src[%d] dest[%d]sid[%d]dir[%d]\n",
+ __func__, payload[0], payload[1], data->opcode, data->token,
+ data->payload_size, data->src_port, data->dest_port, sid, dir);
+ pr_debug("%s:Payload = [0x%x] status[0x%x]\n",
+ __func__, payload[0], payload[1]);
+
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ switch (payload[0]) {
+ case ASM_CMD_SHARED_MEM_MAP_REGIONS:
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ if (atomic_read(&ac->cmd_state)) {
+ atomic_set(&ac->cmd_state, 0);
+ wake_up(&ac->cmd_wait);
+ }
+ pr_debug("%s:Payload = [0x%x] status[0x%x]\n",
+ __func__, payload[0], payload[1]);
+ break;
+ default:
+ pr_debug("%s:command[0x%x] not expecting rsp\n",
+ __func__, payload[0]);
+ break;
+ }
+ return 0;
+ }
+
+ dir = (data->token & 0x0F);
+ port = &ac->port[dir];
+
+ switch (data->opcode) {
+ case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:{
+ pr_debug("%s:PL#0[0x%x]PL#1 [0x%x] dir=%x s_id=%x\n",
+ __func__, payload[0], payload[1], dir, sid);
+ spin_lock_irqsave(&port->dsp_lock, dsp_flags);
+ if (atomic_read(&ac->cmd_state)) {
+ ac->port[dir].tmp_hdl = payload[0];
+ atomic_set(&ac->cmd_state, 0);
+ wake_up(&ac->cmd_wait);
+ }
+ spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
+ break;
+ }
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:{
+ pr_debug("%s:PL#0[0x%x]PL#1 [0x%x]\n",
+ __func__, payload[0], payload[1]);
+ spin_lock_irqsave(&port->dsp_lock, dsp_flags);
+ if (atomic_read(&ac->cmd_state)) {
+ atomic_set(&ac->cmd_state, 0);
+ wake_up(&ac->cmd_wait);
+ }
+ spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
+
+ break;
+ }
+ default:
+ pr_debug("%s:command[0x%x]success [0x%x]\n",
+ __func__, payload[0], payload[1]);
+ }
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ data->payload, ac->priv);
+ return 0;
+}
+
+
+static int32_t q6asm_callback(struct apr_client_data *data, void *priv)
+{
+ int i = 0;
+ struct audio_client *ac = (struct audio_client *)priv;
+ uint32_t token;
+ unsigned long dsp_flags;
+ uint32_t *payload;
+
+
+ if ((ac == NULL) || (data == NULL)) {
+ pr_err("ac or priv NULL\n");
+ return -EINVAL;
+ }
+ if (ac->session <= 0 || ac->session > 8) {
+ pr_err("%s:Session ID is invalid, session = %d\n", __func__,
+ ac->session);
+ return -EINVAL;
+ }
+
+ payload = data->payload;
+
+ if (data->opcode == RESET_EVENTS) {
+ pr_debug("q6asm_callback: Reset event is received: %d %d apr[%p]\n",
+ data->reset_event, data->reset_proc, ac->apr);
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ (uint32_t *)data->payload, ac->priv);
+ apr_reset(ac->apr);
+ return 0;
+ }
+
+ pr_debug("%s: session[%d]opcode[0x%x]"
+ "token[0x%x]payload_s[%d] src[%d] dest[%d]\n", __func__,
+ ac->session, data->opcode,
+ data->token, data->payload_size, data->src_port,
+ data->dest_port);
+ if ((data->opcode != ASM_DATA_EVENT_RENDERED_EOS) &&
+ (data->opcode != ASM_DATA_EVENT_EOS))
+ pr_debug("%s:Payload = [0x%x] status[0x%x]\n",
+ __func__, payload[0], payload[1]);
+ if (data->opcode == APR_BASIC_RSP_RESULT) {
+ token = data->token;
+ switch (payload[0]) {
+ case ASM_STREAM_CMD_SET_PP_PARAMS_V2:
+ if (rtac_make_asm_callback(ac->session, payload,
+ data->payload_size))
+ break;
+ case ASM_SESSION_CMD_PAUSE:
+ case ASM_DATA_CMD_EOS:
+ case ASM_STREAM_CMD_CLOSE:
+ case ASM_STREAM_CMD_FLUSH:
+ case ASM_SESSION_CMD_RUN_V2:
+ case ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS:
+ case ASM_STREAM_CMD_FLUSH_READBUFS:
+ pr_debug("%s:Payload = [0x%x]\n", __func__, payload[0]);
+ if (token != ac->session) {
+ pr_err("%s:Invalid session[%d] rxed expected[%d]",
+ __func__, token, ac->session);
+ return -EINVAL;
+ }
+ case ASM_STREAM_CMD_OPEN_READ_V2:
+ case ASM_STREAM_CMD_OPEN_WRITE_V2:
+ case ASM_STREAM_CMD_OPEN_READWRITE_V2:
+ case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+ case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+ pr_debug("%s:Payload = [0x%x]stat[0x%x]\n",
+ __func__, payload[0], payload[1]);
+ if (atomic_read(&ac->cmd_state)) {
+ atomic_set(&ac->cmd_state, 0);
+ wake_up(&ac->cmd_wait);
+ }
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ (uint32_t *)data->payload, ac->priv);
+ break;
+ default:
+ pr_debug("%s:command[0x%x] not expecting rsp\n",
+ __func__, payload[0]);
+ break;
+ }
+ return 0;
+ }
+
+ switch (data->opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2:{
+ struct audio_port_data *port = &ac->port[IN];
+ pr_debug("%s: Rxed opcode[0x%x] status[0x%x] token[%d]",
+ __func__, payload[0], payload[1],
+ data->token);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ if (port->buf == NULL) {
+ pr_err("%s: Unexpected Write Done\n",
+ __func__);
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&port->dsp_lock, dsp_flags);
+ if (port->buf[data->token].phys !=
+ payload[0]) {
+ pr_err("Buf expected[%p]rxed[%p]\n",\
+ (void *)port->buf[data->token].phys,\
+ (void *)payload[0]);
+ spin_unlock_irqrestore(&port->dsp_lock,
+ dsp_flags);
+ return -EINVAL;
+ }
+ token = data->token;
+ port->buf[token].used = 1;
+ spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
+
+ config_debug_fs_write_cb();
+
+ for (i = 0; i < port->max_buf_cnt; i++)
+ pr_debug("%d ", port->buf[i].used);
+
+ }
+ break;
+ }
+ case ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2:
+ rtac_make_asm_callback(ac->session, payload,
+ data->payload_size);
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2:{
+
+ struct audio_port_data *port = &ac->port[OUT];
+
+ config_debug_fs_read_cb();
+
+ pr_debug("%s:R-D: status=%d buff_add=%x act_size=%d offset=%d\n",
+ __func__, payload[READDONE_IDX_STATUS],
+ payload[READDONE_IDX_BUFADD_LSW],
+ payload[READDONE_IDX_SIZE],
+ payload[READDONE_IDX_OFFSET]);
+
+ pr_debug("%s:R-D:msw_ts=%d lsw_ts=%d memmap_hdl=%x flags=%d id=%d num=%d\n",
+ __func__, payload[READDONE_IDX_MSW_TS],
+ payload[READDONE_IDX_LSW_TS],
+ payload[READDONE_IDX_MEMMAP_HDL],
+ payload[READDONE_IDX_FLAGS],
+ payload[READDONE_IDX_SEQ_ID],
+ payload[READDONE_IDX_NUMFRAMES]);
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ if (port->buf == NULL) {
+ pr_err("%s: Unexpected Write Done\n", __func__);
+ return -EINVAL;
+ }
+ spin_lock_irqsave(&port->dsp_lock, dsp_flags);
+ token = data->token;
+ port->buf[token].used = 0;
+ if (port->buf[token].phys !=
+ payload[READDONE_IDX_BUFADD_LSW]) {
+ pr_err("Buf expected[%p]rxed[%p]\n",\
+ (void *)port->buf[token].phys,\
+ (void *)payload[READDONE_IDX_BUFADD_LSW]);
+ spin_unlock_irqrestore(&port->dsp_lock,
+ dsp_flags);
+ break;
+ }
+ port->buf[token].actual_size =
+ payload[READDONE_IDX_SIZE];
+ spin_unlock_irqrestore(&port->dsp_lock, dsp_flags);
+ }
+ break;
+ }
+ case ASM_DATA_EVENT_EOS:
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("%s:EOS ACK received: rxed opcode[0x%x]\n",
+ __func__, data->opcode);
+ break;
+ case ASM_SESSION_EVENTX_OVERFLOW:
+ pr_err("ASM_SESSION_EVENTX_OVERFLOW\n");
+ break;
+ case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3:
+ pr_debug("%s: ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3, "
+ "payload[0] = %d, payload[1] = %d, "
+ "payload[2] = %d\n", __func__,
+ payload[0], payload[1], payload[2]);
+ ac->time_stamp = (uint64_t)(((uint64_t)payload[1] << 32) |
+ payload[2]);
+ if (atomic_read(&ac->cmd_state)) {
+ atomic_set(&ac->cmd_state, 0);
+ wake_up(&ac->cmd_wait);
+ }
+ break;
+ case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
+ case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY:
+ pr_debug("%s: ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY, "
+ "payload[0] = %d, payload[1] = %d, "
+ "payload[2] = %d, payload[3] = %d\n", __func__,
+ payload[0], payload[1], payload[2],
+ payload[3]);
+ break;
+ }
+ if (ac->cb)
+ ac->cb(data->opcode, data->token,
+ data->payload, ac->priv);
+
+ return 0;
+}
+
+void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac, uint32_t *size,
+ uint32_t *index)
+{
+ void *data;
+ unsigned char idx;
+ struct audio_port_data *port;
+
+ if (!ac || ((dir != IN) && (dir != OUT)))
+ return NULL;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[dir];
+
+ mutex_lock(&port->lock);
+ idx = port->cpu_buf;
+ if (port->buf == NULL) {
+ pr_debug("%s:Buffer pointer null\n", __func__);
+ mutex_unlock(&port->lock);
+ return NULL;
+ }
+ /* dir 0: used = 0 means buf in use
+ dir 1: used = 1 means buf in use */
+ if (port->buf[idx].used == dir) {
+ /* To make it more robust, we could loop and get the
+ next avail buf, its risky though */
+ pr_debug("%s:Next buf idx[0x%x] not available,"
+ "dir[%d]\n", __func__, idx, dir);
+ mutex_unlock(&port->lock);
+ return NULL;
+ }
+ *size = port->buf[idx].actual_size;
+ *index = port->cpu_buf;
+ data = port->buf[idx].data;
+ pr_debug("%s:session[%d]index[%d] data[%p]size[%d]\n",
+ __func__,
+ ac->session,
+ port->cpu_buf,
+ data, *size);
+ /* By default increase the cpu_buf cnt
+ user accesses this function,increase cpu
+ buf(to avoid another api)*/
+ port->buf[idx].used = dir;
+ port->cpu_buf = ((port->cpu_buf + 1) & (port->max_buf_cnt - 1));
+ mutex_unlock(&port->lock);
+ return data;
+ }
+ return NULL;
+}
+
+void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac,
+ uint32_t *size, uint32_t *index)
+{
+ void *data;
+ unsigned char idx;
+ struct audio_port_data *port;
+
+ if (!ac || ((dir != IN) && (dir != OUT)))
+ return NULL;
+
+ port = &ac->port[dir];
+
+ idx = port->cpu_buf;
+ if (port->buf == NULL) {
+ pr_debug("%s:Buffer pointer null\n", __func__);
+ return NULL;
+ }
+ /*
+ * dir 0: used = 0 means buf in use
+ * dir 1: used = 1 means buf in use
+ */
+ if (port->buf[idx].used == dir) {
+ /*
+ * To make it more robust, we could loop and get the
+ * next avail buf, its risky though
+ */
+ pr_debug("%s:Next buf idx[0x%x] not available,"
+ "dir[%d]\n", __func__, idx, dir);
+ return NULL;
+ }
+ *size = port->buf[idx].actual_size;
+ *index = port->cpu_buf;
+ data = port->buf[idx].data;
+ pr_debug("%s:session[%d]index[%d] data[%p]size[%d]\n",
+ __func__, ac->session, port->cpu_buf,
+ data, *size);
+ /*
+ * By default increase the cpu_buf cnt
+ * user accesses this function,increase cpu
+ * buf(to avoid another api)
+ */
+ port->buf[idx].used = dir;
+ port->cpu_buf = ((port->cpu_buf + 1) & (port->max_buf_cnt - 1));
+ return data;
+}
+
+int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac)
+{
+ int ret = -1;
+ struct audio_port_data *port;
+ uint32_t idx;
+
+ if (!ac || (dir != OUT))
+ return ret;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[dir];
+
+ mutex_lock(&port->lock);
+ idx = port->dsp_buf;
+
+ if (port->buf[idx].used == (dir ^ 1)) {
+ /* To make it more robust, we could loop and get the
+ next avail buf, its risky though */
+ pr_err("Next buf idx[0x%x] not available, dir[%d]\n",
+ idx, dir);
+ mutex_unlock(&port->lock);
+ return ret;
+ }
+ pr_debug("%s: session[%d]dsp_buf=%d cpu_buf=%d\n", __func__,
+ ac->session, port->dsp_buf, port->cpu_buf);
+ ret = ((port->dsp_buf != port->cpu_buf) ? 0 : -1);
+ mutex_unlock(&port->lock);
+ }
+ return ret;
+}
+
+static void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg)
+{
+ pr_debug("%s:pkt_size=%d cmd_flg=%d session=%d\n", __func__, pkt_size,
+ cmd_flg, ac->session);
+ mutex_lock(&ac->cmd_lock);
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(sizeof(struct apr_hdr)),\
+ APR_PKT_VER);
+ hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
+ hdr->src_domain = APR_DOMAIN_APPS;
+ hdr->dest_svc = APR_SVC_ASM;
+ hdr->dest_domain = APR_DOMAIN_ADSP;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ if (cmd_flg) {
+ hdr->token = ac->session;
+ atomic_set(&ac->cmd_state, 1);
+ }
+ hdr->pkt_size = pkt_size;
+ mutex_unlock(&ac->cmd_lock);
+ return;
+}
+
+static void q6asm_add_hdr_async(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, uint32_t cmd_flg)
+{
+ pr_debug("pkt_size = %d, cmd_flg = %d, session = %d\n",
+ pkt_size, cmd_flg, ac->session);
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(sizeof(struct apr_hdr)),\
+ APR_PKT_VER);
+ hdr->src_svc = ((struct apr_svc *)ac->apr)->id;
+ hdr->src_domain = APR_DOMAIN_APPS;
+ hdr->dest_svc = APR_SVC_ASM;
+ hdr->dest_domain = APR_DOMAIN_ADSP;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | 0x01;
+ if (cmd_flg) {
+ hdr->token = ac->session;
+ atomic_set(&ac->cmd_state, 1);
+ }
+ hdr->pkt_size = pkt_size;
+ return;
+}
+
+static void q6asm_add_mmaphdr(struct audio_client *ac, struct apr_hdr *hdr,
+ u32 pkt_size, u32 cmd_flg, u32 token)
+{
+ pr_debug("%s:pkt size=%d cmd_flg=%d\n", __func__, pkt_size, cmd_flg);
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ hdr->src_port = 0;
+ hdr->dest_port = 0;
+ if (cmd_flg) {
+ hdr->token = token;
+ atomic_set(&ac->cmd_state, 1);
+ }
+ hdr->pkt_size = pkt_size;
+ return;
+}
+int q6asm_open_read(struct audio_client *ac,
+ uint32_t format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_read_v2 open;
+
+ uint16_t bits_per_sample = 16;
+
+
+ config_debug_fs_reset_index();
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s:session[%d]", __func__, ac->session);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V2;
+ /* Stream prio : High, provide meta info with encoded frames */
+ open.src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+
+ open.preprocopo_id = get_asm_topology();
+ if (open.preprocopo_id == 0)
+ open.preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT;
+ open.bits_per_sample = bits_per_sample;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open.mode_flags = 0x00;
+ open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.mode_flags = BUFFER_META_ENABLE;
+ open.enc_cfg_id = ASM_MEDIA_FMT_AAC_V2;
+ break;
+ case FORMAT_V13K:
+ open.mode_flags = BUFFER_META_ENABLE;
+ open.enc_cfg_id = ASM_MEDIA_FMT_V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ open.mode_flags = BUFFER_META_ENABLE;
+ open.enc_cfg_id = ASM_MEDIA_FMT_EVRC_FS;
+ break;
+ case FORMAT_AMRNB:
+ open.mode_flags = BUFFER_META_ENABLE ;
+ open.enc_cfg_id = ASM_MEDIA_FMT_AMRNB_FS;
+ break;
+ case FORMAT_AMRWB:
+ open.mode_flags = BUFFER_META_ENABLE ;
+ open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", format);
+ goto fail_cmd;
+ }
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("open failed op[0x%x]rc[%d]\n", \
+ open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout. waited for open read rc[%d]\n", __func__,
+ rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+int q6asm_open_write(struct audio_client *ac, uint32_t format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_write_v2 open;
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] wr_format[0x%x]", __func__, ac->session,
+ format);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V2;
+ open.mode_flags = 0x00;
+ /* source endpoint : matrix */
+ open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+ open.bits_per_sample = 16;
+
+ open.postprocopo_id = get_asm_topology();
+ if (open.postprocopo_id == 0)
+ open.postprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.dec_fmt_id = ASM_MEDIA_FMT_AAC_V2;
+ break;
+ case FORMAT_MPEG4_MULTI_AAC:
+ open.dec_fmt_id = ASM_MEDIA_FMT_DOLBY_AAC;
+ break;
+ case FORMAT_WMA_V9:
+ open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V9_V2;
+ break;
+ case FORMAT_WMA_V10PRO:
+ open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V10PRO_V2;
+ break;
+ case FORMAT_MP3:
+ open.dec_fmt_id = ASM_MEDIA_FMT_MP3;
+ break;
+ default:
+ pr_err("%s: Invalid format[%d]\n", __func__, format);
+ goto fail_cmd;
+ }
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("%s: open failed op[0x%x]rc[%d]\n", \
+ __func__, open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout. waited for open write rc[%d]\n", __func__,
+ rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_open_read_write(struct audio_client *ac,
+ uint32_t rd_format,
+ uint32_t wr_format)
+{
+ int rc = 0x00;
+ struct asm_stream_cmd_open_readwrite_v2 open;
+
+ if ((ac == NULL) || (ac->apr == NULL)) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d]", __func__, ac->session);
+ pr_debug("wr_format[0x%x]rd_format[0x%x]",
+ wr_format, rd_format);
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_READWRITE_V2;
+
+ open.mode_flags = BUFFER_META_ENABLE;
+ open.bits_per_sample = 16;
+ /* source endpoint : matrix */
+ open.postprocopo_id = get_asm_topology();
+ if (open.postprocopo_id == 0)
+ open.postprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT;
+
+ switch (wr_format) {
+ case FORMAT_LINEAR_PCM:
+ open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.dec_fmt_id = ASM_MEDIA_FMT_AAC_V2;
+ break;
+ case FORMAT_MPEG4_MULTI_AAC:
+ open.dec_fmt_id = ASM_MEDIA_FMT_DOLBY_AAC;
+ break;
+ case FORMAT_WMA_V9:
+ open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V9_V2;
+ break;
+ case FORMAT_WMA_V10PRO:
+ open.dec_fmt_id = ASM_MEDIA_FMT_WMA_V10PRO_V2;
+ break;
+ case FORMAT_AMRNB:
+ open.dec_fmt_id = ASM_MEDIA_FMT_AMRNB_FS;
+ break;
+ case FORMAT_AMRWB:
+ open.dec_fmt_id = ASM_MEDIA_FMT_AMRWB_FS;
+ break;
+ case FORMAT_V13K:
+ open.dec_fmt_id = ASM_MEDIA_FMT_V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ open.dec_fmt_id = ASM_MEDIA_FMT_EVRC_FS;
+ break;
+ case FORMAT_EVRCB:
+ open.dec_fmt_id = ASM_MEDIA_FMT_EVRCB_FS;
+ break;
+ case FORMAT_EVRCWB:
+ open.dec_fmt_id = ASM_MEDIA_FMT_EVRCWB_FS;
+ break;
+ case FORMAT_MP3:
+ open.dec_fmt_id = ASM_MEDIA_FMT_MP3;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", wr_format);
+ goto fail_cmd;
+ }
+
+ switch (rd_format) {
+ case FORMAT_LINEAR_PCM:
+ open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ case FORMAT_MPEG4_AAC:
+ open.enc_cfg_id = ASM_MEDIA_FMT_AAC_V2;
+ break;
+ case FORMAT_V13K:
+ open.enc_cfg_id = ASM_MEDIA_FMT_V13K_FS;
+ break;
+ case FORMAT_EVRC:
+ open.enc_cfg_id = ASM_MEDIA_FMT_EVRC_FS;
+ break;
+ case FORMAT_AMRNB:
+ open.enc_cfg_id = ASM_MEDIA_FMT_AMRNB_FS;
+ break;
+ case FORMAT_AMRWB:
+ open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", rd_format);
+ goto fail_cmd;
+ }
+ pr_debug("%s:rdformat[0x%x]wrformat[0x%x]\n", __func__,
+ open.enc_cfg_id, open.dec_fmt_id);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
+ if (rc < 0) {
+ pr_err("open failed op[0x%x]rc[%d]\n", \
+ open.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for open read-write rc[%d]\n", rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts)
+{
+ struct asm_session_cmd_run_v2 run;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s session[%d]", __func__, ac->session);
+ q6asm_add_hdr(ac, &run.hdr, sizeof(run), TRUE);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run.flags = flags;
+ run.time_lsw = lsw_ts;
+ run.time_msw = msw_ts;
+
+ config_debug_fs_run();
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &run);
+ if (rc < 0) {
+ pr_err("Commmand run failed[%d]", rc);
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for run success rc[%d]", rc);
+ goto fail_cmd;
+ }
+
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts)
+{
+ struct asm_session_cmd_run_v2 run;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s:APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("session[%d]", ac->session);
+ q6asm_add_hdr_async(ac, &run.hdr, sizeof(run), TRUE);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run.flags = flags;
+ run.time_lsw = lsw_ts;
+ run.time_msw = msw_ts;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &run);
+ if (rc < 0) {
+ pr_err("%s:Commmand run failed[%d]", __func__, rc);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+
+int q6asm_enc_cfg_blk_aac(struct audio_client *ac,
+ uint32_t frames_per_buf,
+ uint32_t sample_rate, uint32_t channels,
+ uint32_t bit_rate, uint32_t mode, uint32_t format)
+{
+ struct asm_aac_enc_cfg_v2 enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]SR[%d]ch[%d]bitrate[%d]mode[%d]"
+ "format[%d]", __func__, ac->session, frames_per_buf,
+ sample_rate, channels, bit_rate, mode, format);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(struct asm_aac_enc_cfg_v2) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+ enc_cfg.bit_rate = bit_rate;
+ enc_cfg.enc_mode = mode;
+ enc_cfg.aac_fmt_flag = format;
+ enc_cfg.channel_cfg = channels;
+ enc_cfg.sample_rate = sample_rate;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_set_encdec_chan_map(struct audio_client *ac,
+ uint32_t num_channels)
+{
+ /* Todo: */
+ return 0;
+}
+
+int q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_multi_channel_pcm_enc_cfg_v2 enc_cfg;
+ u8 *channel_mapping;
+ u32 frames_per_buf = 0;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d, rate = %d, channels = %d\n", __func__,
+ ac->session, rate, channels);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(enc_cfg) - sizeof(enc_cfg.hdr) -
+ sizeof(enc_cfg.encdec);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.num_channels = channels;
+ enc_cfg.bits_per_sample = 16;
+ enc_cfg.sample_rate = rate;
+ enc_cfg.is_signed = 1;
+ channel_mapping = enc_cfg.channel_mapping; /* ??? PHANI */
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ if (channels == 1) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ } else if (channels == 2) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channels == 6) {
+ channel_mapping[0] = PCM_CHANNEL_FC;
+ channel_mapping[1] = PCM_CHANNEL_FL;
+ channel_mapping[2] = PCM_CHANNEL_FR;
+ channel_mapping[3] = PCM_CHANNEL_LB;
+ channel_mapping[4] = PCM_CHANNEL_RB;
+ channel_mapping[5] = PCM_CHANNEL_LFE;
+ } else {
+ pr_err("%s: ERROR.unsupported num_ch = %u\n", __func__,
+ channels);
+ return -EINVAL;
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd open failed\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout opcode[0x%x] ", enc_cfg.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enable_sbrps(struct audio_client *ac,
+ uint32_t sbr_ps_enable)
+{
+ struct asm_aac_sbr_ps_flag_param sbrps;
+ u32 frames_per_buf = 0;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d\n", __func__, ac->session);
+
+ q6asm_add_hdr(ac, &sbrps.hdr, sizeof(sbrps), TRUE);
+
+ sbrps.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ sbrps.encdec.param_id = ASM_PARAM_ID_AAC_SBR_PS_FLAG;
+ sbrps.encdec.param_size = sizeof(struct asm_aac_sbr_ps_flag_param) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ sbrps.encblk.frames_per_buf = frames_per_buf;
+ sbrps.encblk.enc_cfg_blk_size = sbrps.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ sbrps.sbr_ps_flag = sbr_ps_enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &sbrps);
+ if (rc < 0) {
+ pr_err("Command opcode[0x%x]paramid[0x%x] failed\n",
+ ASM_STREAM_CMD_SET_ENCDEC_PARAM,
+ ASM_PARAM_ID_AAC_SBR_PS_FLAG);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout opcode[0x%x] ", sbrps.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_cfg_dual_mono_aac(struct audio_client *ac,
+ uint16_t sce_left, uint16_t sce_right)
+{
+ struct asm_aac_dual_mono_mapping_param dual_mono;
+ u32 frames_per_buf = 0;
+
+ int rc = 0;
+
+ pr_debug("%s: Session %d, sce_left = %d, sce_right = %d\n",
+ __func__, ac->session, sce_left, sce_right);
+
+ q6asm_add_hdr(ac, &dual_mono.hdr, sizeof(dual_mono), TRUE);
+
+ dual_mono.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ dual_mono.encdec.param_id = ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING;
+ dual_mono.encdec.param_size = sizeof(struct asm_aac_enc_cfg_v2) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ dual_mono.encblk.frames_per_buf = frames_per_buf;
+ dual_mono.encblk.enc_cfg_blk_size = dual_mono.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+ dual_mono.left_channel_sce = sce_left;
+ dual_mono.right_channel_sce = sce_right;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &dual_mono);
+ if (rc < 0) {
+ pr_err("%s:Command opcode[0x%x]paramid[0x%x] failed\n",
+ __func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM,
+ ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout opcode[0x%x]\n", __func__,
+ dual_mono.hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t reduced_rate_level, uint16_t rate_modulation_cmd)
+{
+ struct asm_v13k_enc_cfg enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]min_rate[0x%4x]max_rate[0x%4x]"
+ "reduced_rate_level[0x%4x]rate_modulation_cmd[0x%4x]", __func__,
+ ac->session, frames_per_buf, min_rate, max_rate,
+ reduced_rate_level, rate_modulation_cmd);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(struct asm_v13k_enc_cfg) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.min_rate = min_rate;
+ enc_cfg.max_rate = max_rate;
+ enc_cfg.reduced_rate_cmd = reduced_rate_level;
+ enc_cfg.rate_mod_cmd = rate_modulation_cmd;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for setencdec v13k resp\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t rate_modulation_cmd)
+{
+ struct asm_evrc_enc_cfg enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]min_rate[0x%4x]max_rate[0x%4x]"
+ "rate_modulation_cmd[0x%4x]", __func__, ac->session,
+ frames_per_buf, min_rate, max_rate, rate_modulation_cmd);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(struct asm_evrc_enc_cfg) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.min_rate = min_rate;
+ enc_cfg.max_rate = max_rate;
+ enc_cfg.rate_mod_cmd = rate_modulation_cmd;
+ enc_cfg.reserved = 0;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for encdec evrc\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable)
+{
+ struct asm_amrnb_enc_cfg enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]band_mode[0x%4x]dtx_enable[0x%4x]",
+ __func__, ac->session, frames_per_buf, band_mode, dtx_enable);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(struct asm_amrnb_enc_cfg) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.enc_mode = band_mode;
+ enc_cfg.dtx_mode = dtx_enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for set encdec amrnb\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable)
+{
+ struct asm_amrwb_enc_cfg enc_cfg;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]frames[%d]band_mode[0x%4x]dtx_enable[0x%4x]",
+ __func__, ac->session, frames_per_buf, band_mode, dtx_enable);
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(struct asm_amrwb_enc_cfg) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.enc_mode = band_mode;
+ enc_cfg.dtx_mode = dtx_enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("Comamnd %d failed\n", ASM_STREAM_CMD_SET_ENCDEC_PARAM);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for FORMAT_UPDATE\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+
+int q6asm_media_format_block_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg)
+{
+ return q6asm_media_format_block_multi_aac(ac, cfg);
+}
+
+int q6asm_media_format_block_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
+ u8 *channel_mapping;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session, rate,
+ channels);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+ sizeof(fmt.fmt_blk);
+ fmt.num_channels = channels;
+ fmt.bits_per_sample = 16;
+ fmt.sample_rate = rate;
+ fmt.is_signed = 1;
+
+ channel_mapping = fmt.channel_mapping;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ if (channels == 1) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ } else if (channels == 2) {
+ channel_mapping[0] = PCM_CHANNEL_FL;
+ channel_mapping[1] = PCM_CHANNEL_FR;
+ } else if (channels == 6) {
+ channel_mapping[0] = PCM_CHANNEL_FC;
+ channel_mapping[1] = PCM_CHANNEL_FL;
+ channel_mapping[2] = PCM_CHANNEL_FR;
+ channel_mapping[3] = PCM_CHANNEL_LB;
+ channel_mapping[4] = PCM_CHANNEL_RB;
+ channel_mapping[5] = PCM_CHANNEL_LFE;
+ } else {
+ pr_err("%s: ERROR.unsupported num_ch = %u\n", __func__,
+ channels);
+ return -EINVAL;
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for format update\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_multi_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg)
+{
+ struct asm_aac_fmt_blk_v2 fmt;
+ int rc = 0;
+
+ pr_debug("%s:session[%d]rate[%d]ch[%d]\n", __func__, ac->session,
+ cfg->sample_rate, cfg->ch_cfg);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+ sizeof(fmt.fmt_blk);
+ fmt.aac_fmt_flag = cfg->format;
+ fmt.audio_objype = cfg->aot;
+ /* If zero, PCE is assumed to be available in bitstream*/
+ fmt.total_size_of_PCE_bits = 0;
+ fmt.channel_config = cfg->ch_cfg;
+ fmt.sample_rate = cfg->sample_rate;
+
+ pr_info("%s:format=%x cfg_size=%d aac-cfg=%x aot=%d ch=%d sr=%d\n",
+ __func__, fmt.aac_fmt_flag, fmt.fmt_blk.fmt_blk_size,
+ fmt.aac_fmt_flag,
+ fmt.audio_objype,
+ fmt.channel_config,
+ fmt.sample_rate);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_wma(struct audio_client *ac,
+ void *cfg)
+{
+ struct asm_wmastdv9_fmt_blk_v2 fmt;
+ struct asm_wma_cfg *wma_cfg = (struct asm_wma_cfg *)cfg;
+ int rc = 0;
+
+ pr_debug("session[%d]format_tag[0x%4x] rate[%d] ch[0x%4x] bps[%d],"
+ "balign[0x%4x], bit_sample[0x%4x], ch_msk[%d], enc_opt[0x%4x]\n",
+ ac->session, wma_cfg->format_tag, wma_cfg->sample_rate,
+ wma_cfg->ch_cfg, wma_cfg->avg_bytes_per_sec,
+ wma_cfg->block_align, wma_cfg->valid_bits_per_sample,
+ wma_cfg->ch_mask, wma_cfg->encode_opt);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_MEDIA_FMT_WMA_V9_V2;
+
+ fmt.fmtag = wma_cfg->format_tag;
+ fmt.num_channels = wma_cfg->ch_cfg;
+ fmt.sample_rate = wma_cfg->sample_rate;
+ fmt.avg_bytes_per_sec = wma_cfg->avg_bytes_per_sec;
+ fmt.blk_align = wma_cfg->block_align;
+ fmt.bits_per_sample =
+ wma_cfg->valid_bits_per_sample;
+ fmt.channel_mask = wma_cfg->ch_mask;
+ fmt.enc_options = wma_cfg->encode_opt;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_media_format_block_wmapro(struct audio_client *ac,
+ void *cfg)
+{
+ struct asm_wmaprov10_fmt_blk_v2 fmt;
+ struct asm_wmapro_cfg *wmapro_cfg = (struct asm_wmapro_cfg *)cfg;
+ int rc = 0;
+
+ pr_debug("session[%d]format_tag[0x%4x] rate[%d] ch[0x%4x] bps[%d],"
+ "balign[0x%4x], bit_sample[0x%4x], ch_msk[%d], enc_opt[0x%4x],"
+ "adv_enc_opt[0x%4x], adv_enc_opt2[0x%8x]\n",
+ ac->session, wmapro_cfg->format_tag, wmapro_cfg->sample_rate,
+ wmapro_cfg->ch_cfg, wmapro_cfg->avg_bytes_per_sec,
+ wmapro_cfg->block_align, wmapro_cfg->valid_bits_per_sample,
+ wmapro_cfg->ch_mask, wmapro_cfg->encode_opt,
+ wmapro_cfg->adv_encode_opt, wmapro_cfg->adv_encode_opt2);
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+
+ fmt.hdr.opcode = ASM_MEDIA_FMT_WMA_V10PRO_V2;
+
+ fmt.fmtag = wmapro_cfg->format_tag;
+ fmt.num_channels = wmapro_cfg->ch_cfg;
+ fmt.sample_rate = wmapro_cfg->sample_rate;
+ fmt.avg_bytes_per_sec =
+ wmapro_cfg->avg_bytes_per_sec;
+ fmt.blk_align = wmapro_cfg->block_align;
+ fmt.bits_per_sample = wmapro_cfg->valid_bits_per_sample;
+ fmt.channel_mask = wmapro_cfg->ch_mask;
+ fmt.enc_options = wmapro_cfg->encode_opt;
+ fmt.usAdvancedEncodeOpt = wmapro_cfg->adv_encode_opt;
+ fmt.advanced_enc_options2 = wmapro_cfg->adv_encode_opt2;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s:Comamnd open failed\n", __func__);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s:timeout. waited for FORMAT_UPDATE\n", __func__);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_memory_map(struct audio_client *ac, uint32_t buf_add, int dir,
+ uint32_t bufsz, uint32_t bufcnt)
+{
+ struct avs_cmd_shared_mem_map_regions *mmap_regions = NULL;
+ struct avs_shared_map_region_payload *mregions = NULL;
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ void *mmap_region_cmd = NULL;
+ void *payload = NULL;
+ struct asm_buffer_node *buffer_node = NULL;
+ int rc = 0;
+ int i = 0;
+ int cmd_size = 0;
+
+ if (!ac || ac->apr == NULL || ac->mmap_apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ buffer_node = kmalloc(sizeof(struct asm_buffer_node), GFP_KERNEL);
+ if (!buffer_node)
+ return -ENOMEM;
+ cmd_size = sizeof(struct avs_cmd_shared_mem_map_regions)
+ + sizeof(struct avs_shared_map_region_payload) * bufcnt;
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (mmap_region_cmd == NULL) {
+ pr_err("%s: Mem alloc failed\n", __func__);
+ rc = -EINVAL;
+ return rc;
+ }
+ mmap_regions = (struct avs_cmd_shared_mem_map_regions *)
+ mmap_region_cmd;
+ q6asm_add_mmaphdr(ac, &mmap_regions->hdr, cmd_size,
+ TRUE, ((ac->session << 8) | dir));
+ mmap_regions->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
+ mmap_regions->mem_pool_id = ADSP_MEMORY_MAP_EBI_POOL;
+ mmap_regions->num_regions = bufcnt & 0x00ff;
+ mmap_regions->property_flag = 0x00;
+ pr_debug("map_regions->nregions = %d\n", mmap_regions->num_regions);
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct avs_cmd_shared_mem_map_regions));
+ mregions = (struct avs_shared_map_region_payload *)payload;
+
+ ac->port[dir].tmp_hdl = 0;
+ port = &ac->port[dir];
+ for (i = 0; i < bufcnt; i++) {
+ ab = &port->buf[i];
+ mregions->shm_addr_lsw = ab->phys;
+ /* Using only 32 bit address */
+ mregions->shm_addr_msw = 0;
+ mregions->mem_size_bytes = ab->size;
+ ++mregions;
+ }
+
+ rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) mmap_region_cmd);
+ if (rc < 0) {
+ pr_err("mmap op[0x%x]rc[%d]\n",
+ mmap_regions->hdr.opcode, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0 &&
+ ac->port[dir].tmp_hdl), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_map\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ buffer_node->buf_addr_lsw = buf_add;
+ buffer_node->mmap_hdl = ac->port[dir].tmp_hdl;
+ list_add_tail(&buffer_node->list, &ac->port[dir].mem_map_handle);
+ ac->port[dir].tmp_hdl = 0;
+ rc = 0;
+
+fail_cmd:
+ kfree(mmap_region_cmd);
+ return rc;
+}
+
+int q6asm_memory_unmap(struct audio_client *ac, uint32_t buf_add, int dir)
+{
+ struct avs_cmd_shared_mem_unmap_regions mem_unmap;
+ struct asm_buffer_node *buf_node = NULL;
+ struct list_head *ptr, *next;
+
+ int rc = 0;
+
+ if (!ac || ac->apr == NULL || this_mmap.apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ q6asm_add_mmaphdr(ac, &mem_unmap.hdr,
+ sizeof(struct avs_cmd_shared_mem_unmap_regions),
+ TRUE, ((ac->session << 8) | dir));
+
+ mem_unmap.hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
+ list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == buf_add) {
+ pr_info("%s: Found the element\n", __func__);
+ mem_unmap.mem_map_handle = buf_node->mmap_hdl;
+ break;
+ }
+ }
+ pr_debug("%s: mem_unmap-mem_map_handle: 0x%x",
+ __func__, mem_unmap.mem_map_handle);
+ rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) &mem_unmap);
+ if (rc < 0) {
+ pr_err("mem_unmap op[0x%x]rc[%d]\n",
+ mem_unmap.hdr.opcode, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5 * HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_map\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == buf_add) {
+ list_del(&buf_node->list);
+ kfree(buf_node);
+ }
+ }
+
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+
+static int q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt)
+{
+ struct avs_cmd_shared_mem_map_regions *mmap_regions = NULL;
+ struct avs_shared_map_region_payload *mregions = NULL;
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ void *mmap_region_cmd = NULL;
+ void *payload = NULL;
+ struct asm_buffer_node *buffer_node = NULL;
+ int rc = 0;
+ int i = 0;
+ int cmd_size = 0;
+
+ if (!ac || ac->apr == NULL || ac->mmap_apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ cmd_size = sizeof(struct avs_cmd_shared_mem_map_regions)
+ + (sizeof(struct avs_shared_map_region_payload));
+
+ buffer_node = kzalloc(sizeof(struct asm_buffer_node) * bufcnt,
+ GFP_KERNEL);
+
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if ((mmap_region_cmd == NULL) || (buffer_node == NULL)) {
+ pr_err("%s: Mem alloc failed\n", __func__);
+ rc = -EINVAL;
+ return rc;
+ }
+ mmap_regions = (struct avs_cmd_shared_mem_map_regions *)
+ mmap_region_cmd;
+ q6asm_add_mmaphdr(ac, &mmap_regions->hdr, cmd_size, TRUE,
+ ((ac->session << 8) | dir));
+ pr_debug("mmap_region=0x%p token=0x%x\n",
+ mmap_regions, ((ac->session << 8) | dir));
+
+ mmap_regions->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
+ mmap_regions->mem_pool_id = ADSP_MEMORY_MAP_EBI_POOL;
+ mmap_regions->num_regions = 1; /*bufcnt & 0x00ff; */
+ mmap_regions->property_flag = 0x00;
+ pr_debug("map_regions->nregions = %d\n", mmap_regions->num_regions);
+ payload = ((u8 *) mmap_region_cmd +
+ sizeof(struct avs_cmd_shared_mem_map_regions));
+ mregions = (struct avs_shared_map_region_payload *)payload;
+
+ ac->port[dir].tmp_hdl = 0;
+ port = &ac->port[dir];
+ ab = &port->buf[0];
+ mregions->shm_addr_lsw = ab->phys;
+ /* Using only 32 bit address */
+ mregions->shm_addr_msw = 0;
+ mregions->mem_size_bytes = (bufsz * bufcnt);
+
+ rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) mmap_region_cmd);
+ if (rc < 0) {
+ pr_err("mmap_regions op[0x%x]rc[%d]\n",
+ mmap_regions->hdr.opcode, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0)
+ , 5*HZ);
+ /*ac->port[dir].tmp_hdl), 5*HZ);*/
+ if (!rc) {
+ pr_err("timeout. waited for memory_map\n");
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ mutex_lock(&ac->cmd_lock);
+
+ for (i = 0; i < bufcnt; i++) {
+ ab = &port->buf[i];
+ buffer_node[i].buf_addr_lsw = ab->phys;
+ buffer_node[i].mmap_hdl = ac->port[dir].tmp_hdl;
+ list_add_tail(&buffer_node[i].list,
+ &ac->port[dir].mem_map_handle);
+ pr_debug("%s: i=%d, bufadd[i] = 0x%x, maphdl[i] = 0x%x\n",
+ __func__, i, buffer_node[i].buf_addr_lsw,
+ buffer_node[i].mmap_hdl);
+ }
+ ac->port[dir].tmp_hdl = 0;
+ mutex_unlock(&ac->cmd_lock);
+ rc = 0;
+ pr_debug("%s: exit\n", __func__);
+fail_cmd:
+ kfree(mmap_region_cmd);
+ return rc;
+}
+
+static int q6asm_memory_unmap_regions(struct audio_client *ac, int dir,
+ uint32_t bufsz, uint32_t bufcnt)
+{
+ struct avs_cmd_shared_mem_unmap_regions mem_unmap;
+ struct audio_port_data *port = NULL;
+ struct asm_buffer_node *buf_node = NULL;
+ struct list_head *ptr, *next;
+ uint32_t buf_add;
+ int rc = 0;
+ int cmd_size = 0;
+
+ if (!ac || ac->apr == NULL || ac->mmap_apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: Session[%d]\n", __func__, ac->session);
+
+ cmd_size = sizeof(struct avs_cmd_shared_mem_unmap_regions);
+ q6asm_add_mmaphdr(ac, &mem_unmap.hdr, cmd_size,
+ TRUE, ((ac->session << 8) | dir));
+ port = &ac->port[dir];
+ buf_add = (uint32_t)port->buf->phys;
+ mem_unmap.hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
+ list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == buf_add) {
+ pr_debug("%s: Found the element\n", __func__);
+ mem_unmap.mem_map_handle = buf_node->mmap_hdl;
+ break;
+ }
+ }
+
+ pr_debug("%s: mem_unmap-mem_map_handle: 0x%x",
+ __func__, mem_unmap.mem_map_handle);
+ rc = apr_send_pkt(ac->mmap_apr, (uint32_t *) &mem_unmap);
+ if (rc < 0) {
+ pr_err("mmap_regions op[0x%x]rc[%d]\n",
+ mem_unmap.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for memory_unmap\n");
+ goto fail_cmd;
+ }
+ list_for_each_safe(ptr, next, &ac->port[dir].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == buf_add) {
+ list_del(&buf_node->list);
+ kfree(buf_node);
+ }
+ }
+ rc = 0;
+
+fail_cmd:
+ return rc;
+}
+
+int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain)
+{
+ struct asm_volume_ctrl_lr_chan_gain lrgain;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_volume_ctrl_lr_chan_gain);
+ q6asm_add_hdr_async(ac, &lrgain.hdr, sz, TRUE);
+ lrgain.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
+ lrgain.param.data_payload_addr_lsw = 0;
+ lrgain.param.data_payload_addr_msw = 0;
+ lrgain.param.mem_map_handle = 0;
+ lrgain.param.data_payload_size = sizeof(lrgain) -
+ sizeof(lrgain.hdr) - sizeof(lrgain.param);
+ lrgain.data.module_id = ASM_MODULE_ID_VOL_CTRL;
+ lrgain.data.param_id = ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN;
+ lrgain.data.param_size = lrgain.param.data_payload_size -
+ sizeof(lrgain.data);
+ lrgain.data.reserved = 0;
+ lrgain.l_chan_gain = left_gain;
+ lrgain.r_chan_gain = right_gain;
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &lrgain);
+ if (rc < 0) {
+ pr_err("%s: set-params send failed paramid[0x%x]\n", __func__,
+ lrgain.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
+ lrgain.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_set_mute(struct audio_client *ac, int muteflag)
+{
+ struct asm_volume_ctrl_mute_config mute;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_volume_ctrl_mute_config);
+ q6asm_add_hdr_async(ac, &mute.hdr, sz, TRUE);
+ mute.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
+ mute.param.data_payload_addr_lsw = 0;
+ mute.param.data_payload_addr_msw = 0;
+ mute.param.mem_map_handle = 0;
+ mute.param.data_payload_size = sizeof(mute) -
+ sizeof(mute.hdr) - sizeof(mute.param);
+ mute.data.module_id = ASM_MODULE_ID_VOL_CTRL;
+ mute.data.param_id = ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG;
+ mute.data.param_size = mute.param.data_payload_size - sizeof(mute.data);
+ mute.data.reserved = 0;
+ mute.mute_flag = muteflag;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &mute);
+ if (rc < 0) {
+ pr_err("%s: set-params send failed paramid[0x%x]\n", __func__,
+ mute.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
+ mute.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_set_volume(struct audio_client *ac, int volume)
+{
+ struct asm_volume_ctrl_master_gain vol;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_volume_ctrl_master_gain);
+ q6asm_add_hdr_async(ac, &vol.hdr, sz, TRUE);
+ vol.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
+ vol.param.data_payload_addr_lsw = 0;
+ vol.param.data_payload_addr_msw = 0;
+
+
+ vol.param.mem_map_handle = 0;
+ vol.param.data_payload_size = sizeof(vol) -
+ sizeof(vol.hdr) - sizeof(vol.param);
+ vol.data.module_id = ASM_MODULE_ID_VOL_CTRL;
+ vol.data.param_id = ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN;
+ vol.data.param_size = vol.param.data_payload_size - sizeof(vol.data);
+ vol.data.reserved = 0;
+ vol.master_gain = volume;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &vol);
+ if (rc < 0) {
+ pr_err("%s: set-params send failed paramid[0x%x]\n", __func__,
+ vol.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
+ vol.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+int q6asm_set_softpause(struct audio_client *ac,
+ struct asm_softpause_params *pause_param)
+{
+ struct asm_soft_pause_params softpause;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_soft_pause_params);
+ q6asm_add_hdr_async(ac, &softpause.hdr, sz, TRUE);
+ softpause.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
+
+ softpause.param.data_payload_addr_lsw = 0;
+ softpause.param.data_payload_addr_msw = 0;
+ softpause.param.mem_map_handle = 0;
+ softpause.param.data_payload_size = sizeof(softpause) -
+ sizeof(softpause.hdr) - sizeof(softpause.param);
+ softpause.data.module_id = ASM_MODULE_ID_VOL_CTRL;
+ softpause.data.param_id = ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS;
+ softpause.data.param_size = softpause.param.data_payload_size -
+ sizeof(softpause.data);
+ softpause.data.reserved = 0;
+ softpause.enable_flag = pause_param->enable;
+ softpause.period = pause_param->period;
+ softpause.step = pause_param->step;
+ softpause.ramping_curve = pause_param->rampingcurve;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &softpause);
+ if (rc < 0) {
+ pr_err("%s: set-params send failed paramid[0x%x]\n", __func__,
+ softpause.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
+ softpause.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_set_softvolume(struct audio_client *ac,
+ struct asm_softvolume_params *softvol_param)
+{
+ struct asm_soft_step_volume_params softvol;
+ int sz = 0;
+ int rc = 0;
+
+ sz = sizeof(struct asm_soft_step_volume_params);
+ q6asm_add_hdr_async(ac, &softvol.hdr, sz, TRUE);
+ softvol.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
+ softvol.param.data_payload_addr_lsw = 0;
+ softvol.param.data_payload_addr_msw = 0;
+ softvol.param.mem_map_handle = 0;
+ softvol.param.data_payload_size = sizeof(softvol) -
+ sizeof(softvol.hdr) - sizeof(softvol.param);
+ softvol.data.module_id = ASM_MODULE_ID_VOL_CTRL;
+ softvol.data.param_id = ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS;
+ softvol.data.param_size = softvol.param.data_payload_size -
+ sizeof(softvol.data);
+ softvol.data.reserved = 0;
+ softvol.period = softvol_param->period;
+ softvol.step = softvol_param->step;
+ softvol.ramping_curve = softvol_param->rampingcurve;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &softvol);
+ if (rc < 0) {
+ pr_err("%s: set-params send failed paramid[0x%x]\n", __func__,
+ softvol.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
+ softvol.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_equalizer(struct audio_client *ac, void *eq_p)
+{
+ struct asm_eq_params eq;
+ struct msm_audio_eq_stream_config *eq_params = NULL;
+ int i = 0;
+ int sz = 0;
+ int rc = 0;
+
+ if (eq_p == NULL) {
+ pr_err("%s[%d]: Invalid Eq param\n", __func__, ac->session);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ sz = sizeof(struct asm_eq_params);
+ eq_params = (struct msm_audio_eq_stream_config *) eq_p;
+ q6asm_add_hdr(ac, &eq.hdr, sz, TRUE);
+
+ eq.hdr.opcode = ASM_STREAM_CMD_SET_PP_PARAMS_V2;
+ eq.param.data_payload_addr_lsw = 0;
+ eq.param.data_payload_addr_msw = 0;
+ eq.param.mem_map_handle = 0;
+ eq.param.data_payload_size = sizeof(eq) -
+ sizeof(eq.hdr) - sizeof(eq.param);
+ eq.data.module_id = ASM_MODULE_ID_EQUALIZER;
+ eq.data.param_id = ASM_PARAM_ID_EQUALIZER_PARAMETERS;
+ eq.data.param_size = eq.param.data_payload_size - sizeof(eq.data);
+ eq.enable_flag = eq_params->enable;
+ eq.num_bands = eq_params->num_bands;
+
+ pr_debug("%s: enable:%d numbands:%d\n", __func__, eq_params->enable,
+ eq_params->num_bands);
+ for (i = 0; i < eq_params->num_bands; i++) {
+ eq.eq_bands[i].band_idx =
+ eq_params->eq_bands[i].band_idx;
+ eq.eq_bands[i].filterype =
+ eq_params->eq_bands[i].filter_type;
+ eq.eq_bands[i].center_freq_hz =
+ eq_params->eq_bands[i].center_freq_hz;
+ eq.eq_bands[i].filter_gain =
+ eq_params->eq_bands[i].filter_gain;
+ eq.eq_bands[i].q_factor =
+ eq_params->eq_bands[i].q_factor;
+ pr_debug("%s: filter_type:%u bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].filter_type, i);
+ pr_debug("%s: center_freq_hz:%u bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].center_freq_hz, i);
+ pr_debug("%s: filter_gain:%d bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].filter_gain, i);
+ pr_debug("%s: q_factor:%d bandnum:%d\n", __func__,
+ eq_params->eq_bands[i].q_factor, i);
+ }
+ rc = apr_send_pkt(ac->apr, (uint32_t *)&eq);
+ if (rc < 0) {
+ pr_err("%s: set-params send failed paramid[0x%x]\n", __func__,
+ eq.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout, set-params paramid[0x%x]\n", __func__,
+ eq.data.param_id);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = 0;
+fail_cmd:
+ return rc;
+}
+
+int q6asm_read(struct audio_client *ac)
+{
+ struct asm_data_cmd_read_v2 read;
+ struct asm_buffer_node *buf_node = NULL;
+ struct list_head *ptr, *next;
+ struct audio_buffer *ab;
+ int dsp_buf;
+ struct audio_port_data *port;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[OUT];
+
+ q6asm_add_hdr(ac, &read.hdr, sizeof(read), FALSE);
+
+ mutex_lock(&port->lock);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ pr_debug("%s:session[%d]dsp-buf[%d][%p]cpu_buf[%d][%p]\n",
+ __func__,
+ ac->session,
+ dsp_buf,
+ (void *)port->buf[dsp_buf].data,
+ port->cpu_buf,
+ (void *)port->buf[port->cpu_buf].phys);
+
+ read.hdr.opcode = ASM_DATA_CMD_READ_V2;
+ read.buf_addr_lsw = ab->phys;
+ read.buf_addr_msw = 0;
+
+ list_for_each_safe(ptr, next, &ac->port[OUT].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == (uint32_t) ab->phys)
+ read.mem_map_handle = buf_node->mmap_hdl;
+ }
+ pr_debug("memory_map handle in q6asm_read: [%0x]:",
+ read.mem_map_handle);
+ read.buf_size = ab->size;
+ read.seq_id = port->dsp_buf;
+ read.hdr.token = port->dsp_buf;
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+ mutex_unlock(&port->lock);
+ pr_debug("%s:buf add[0x%x] token[%d] uid[%d]\n", __func__,
+ read.buf_addr_lsw,
+ read.hdr.token,
+ read.seq_id);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
+ if (rc < 0) {
+ pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_read_nolock(struct audio_client *ac)
+{
+ struct asm_data_cmd_read_v2 read;
+ struct asm_buffer_node *buf_node = NULL;
+ struct list_head *ptr, *next;
+ struct audio_buffer *ab;
+ int dsp_buf;
+ struct audio_port_data *port;
+ int rc;
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[OUT];
+
+ q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
+
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ pr_debug("%s:session[%d]dsp-buf[%d][%p]cpu_buf[%d][%p]\n",
+ __func__,
+ ac->session,
+ dsp_buf,
+ (void *)port->buf[dsp_buf].data,
+ port->cpu_buf,
+ (void *)port->buf[port->cpu_buf].phys);
+
+ read.hdr.opcode = ASM_DATA_CMD_READ_V2;
+ read.buf_addr_lsw = ab->phys;
+ read.buf_addr_msw = 0;
+ read.buf_size = ab->size;
+ read.seq_id = port->dsp_buf;
+ read.hdr.token = port->dsp_buf;
+
+ list_for_each_safe(ptr, next, &ac->port[OUT].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == (uint32_t)ab->phys) {
+ read.mem_map_handle = buf_node->mmap_hdl;
+ break;
+ }
+ }
+
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+ pr_debug("%s:buf add[0x%x] token[%d] uid[%d]\n", __func__,
+ read.buf_addr_lsw,
+ read.hdr.token,
+ read.seq_id);
+ pr_debug("q6asm_read_nolock mem-map handle is %x",
+ read.mem_map_handle);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
+ if (rc < 0) {
+ pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_async_write(struct audio_client *ac,
+ struct audio_aio_write_param *param)
+{
+ int rc = 0;
+ struct asm_data_cmd_write_v2 write;
+ struct asm_buffer_node *buf_node = NULL;
+ struct list_head *ptr, *next;
+ struct audio_buffer *ab;
+ struct audio_port_data *port;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ q6asm_add_hdr_async(ac, &write.hdr, sizeof(write), FALSE);
+
+ port = &ac->port[IN];
+ ab = &port->buf[port->dsp_buf];
+
+ /* Pass physical address as token for AIO scheme */
+ write.hdr.token = param->uid;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
+ write.buf_addr_lsw = param->paddr;
+ write.buf_addr_msw = 0x00;
+ write.buf_size = param->len;
+ write.timestamp_msw = param->msw_ts;
+ write.timestamp_lsw = param->lsw_ts;
+ pr_debug("%s: token[0x%x], buf_addr_lsw[0x%x], buf_size[0x%x],"
+ "ts_msw[0x%x], ts_lsw[0x%x]\n",
+ __func__, write.hdr.token, write.buf_addr_lsw,
+ write.buf_size, write.timestamp_msw,
+ write.timestamp_lsw);
+ /* Use 0xFF00 for disabling timestamps */
+ if (param->flags == 0xFF00)
+ write.flags = (0x00000000 | (param->flags & 0x800000FF));
+ else
+ write.flags = (0x80000000 | param->flags);
+
+ write.seq_id = param->uid;
+ list_for_each_safe(ptr, next, &ac->port[IN].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == (uint32_t)write.buf_addr_lsw) {
+ write.mem_map_handle = buf_node->mmap_hdl;
+ pr_debug("%s:buf_node->mmap_hdl = 0x%x,"
+ "write.mem_map_handle = 0x%x\n",
+ __func__,
+ buf_node->mmap_hdl,
+ (uint32_t)write.mem_map_handle);
+ break;
+ }
+ }
+
+ pr_debug("%s: session[%d] bufadd[0x%x]len[0x%x],"
+ "mem_map_handle[0x%x]\n", __func__, ac->session,
+ write.buf_addr_lsw, write.buf_size, write.mem_map_handle);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
+ if (rc < 0) {
+ pr_debug("[%s] write op[0x%x]rc[%d]\n", __func__,
+ write.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_async_read(struct audio_client *ac,
+ struct audio_aio_read_param *param)
+{
+ int rc = 0;
+ struct asm_data_cmd_read_v2 read;
+ struct asm_buffer_node *buf_node = NULL;
+ struct list_head *ptr, *next;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s: APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ q6asm_add_hdr_async(ac, &read.hdr, sizeof(read), FALSE);
+
+ /* Pass physical address as token for AIO scheme */
+ read.hdr.token = param->paddr;
+ read.hdr.opcode = ASM_DATA_CMD_READ_V2;
+ read.buf_addr_lsw = param->paddr;
+ read.buf_addr_msw = 0;
+ read.buf_size = param->len;
+ read.seq_id = param->uid;
+
+ list_for_each_safe(ptr, next, &ac->port[IN].mem_map_handle) {
+ buf_node = list_entry(ptr, struct asm_buffer_node,
+ list);
+ if (buf_node->buf_addr_lsw == param->paddr)
+ read.mem_map_handle = buf_node->mmap_hdl;
+ }
+
+ pr_debug("%s: session[%d] bufadd[0x%x]len[0x%x]", __func__, ac->session,
+ read.buf_addr_lsw, read.buf_size);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &read);
+ if (rc < 0) {
+ pr_debug("[%s] read op[0x%x]rc[%d]\n", __func__,
+ read.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags)
+{
+ int rc = 0;
+ struct asm_data_cmd_write_v2 write;
+ struct asm_buffer_node *buf_node = NULL;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int dsp_buf = 0;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] len=%d", __func__, ac->session, len);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[IN];
+
+ q6asm_add_hdr(ac, &write.hdr, sizeof(write),
+ FALSE);
+ mutex_lock(&port->lock);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
+ write.buf_addr_lsw = ab->phys;
+ write.buf_addr_msw = 0;
+ write.buf_size = len;
+ write.seq_id = port->dsp_buf;
+ write.timestamp_lsw = lsw_ts;
+ write.timestamp_msw = msw_ts;
+ /* Use 0xFF00 for disabling timestamps */
+ if (flags == 0xFF00)
+ write.flags = (0x00000000 | (flags & 0x800000FF));
+ else
+ write.flags = (0x80000000 | flags);
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+ buf_node = list_first_entry(&ac->port[IN].mem_map_handle,
+ struct asm_buffer_node,
+ list);
+ write.mem_map_handle = buf_node->mmap_hdl;
+
+ pr_debug("%s:ab->phys[0x%x]bufadd[0x%x]"
+ "token[0x%x]buf_id[0x%x]buf_size[0x%x]mmaphdl[0x%x]"
+ , __func__,
+ ab->phys,
+ write.buf_addr_lsw,
+ write.hdr.token,
+ write.seq_id,
+ write.buf_size,
+ write.mem_map_handle);
+ mutex_unlock(&port->lock);
+
+ config_debug_fs_write(ab);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
+ if (rc < 0) {
+ pr_err("write op[0x%x]rc[%d]\n", write.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ pr_debug("%s: WRITE SUCCESS\n", __func__);
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags)
+{
+ int rc = 0;
+ struct asm_data_cmd_write_v2 write;
+ struct asm_buffer_node *buf_node = NULL;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int dsp_buf = 0;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s: session[%d] len=%d", __func__, ac->session, len);
+ if (ac->io_mode == SYNC_IO_MODE) {
+ port = &ac->port[IN];
+
+ q6asm_add_hdr_async(ac, &write.hdr, sizeof(write),
+ FALSE);
+
+ dsp_buf = port->dsp_buf;
+ ab = &port->buf[dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
+ write.buf_addr_lsw = ab->phys;
+ write.buf_addr_msw = 0;
+ write.buf_size = len;
+ write.seq_id = port->dsp_buf;
+ write.timestamp_lsw = lsw_ts;
+ write.timestamp_msw = msw_ts;
+ buf_node = list_first_entry(&ac->port[IN].mem_map_handle,
+ struct asm_buffer_node,
+ list);
+ write.mem_map_handle = buf_node->mmap_hdl;
+ /* Use 0xFF00 for disabling timestamps */
+ if (flags == 0xFF00)
+ write.flags = (0x00000000 | (flags & 0x800000FF));
+ else
+ write.flags = (0x80000000 | flags);
+ port->dsp_buf = (port->dsp_buf + 1) & (port->max_buf_cnt - 1);
+
+ pr_err("%s:ab->phys[0x%x]bufadd[0x%x]token[0x%x]"
+ "buf_id[0x%x]buf_size[0x%x]mmaphdl[0x%x]"
+ , __func__,
+ ab->phys,
+ write.buf_addr_lsw,
+ write.hdr.token,
+ write.seq_id,
+ write.buf_size,
+ write.mem_map_handle);
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &write);
+ if (rc < 0) {
+ pr_err("write op[0x%x]rc[%d]\n", write.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ pr_debug("%s: WRITE SUCCESS\n", __func__);
+ return 0;
+ }
+fail_cmd:
+ return -EINVAL;
+}
+
+uint64_t q6asm_get_session_time(struct audio_client *ac)
+{
+ struct apr_hdr hdr;
+ int rc;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), TRUE);
+ hdr.opcode = ASM_SESSION_CMD_GET_SESSIONTIME_V3;
+ atomic_set(&ac->cmd_state, 1);
+
+ pr_debug("%s: session[%d]opcode[0x%x]\n", __func__,
+ ac->session,
+ hdr.opcode);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
+ if (rc < 0) {
+ pr_err("Commmand 0x%x failed\n", hdr.opcode);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout in getting session time from DSP\n",
+ __func__);
+ goto fail_cmd;
+ }
+ return ac->time_stamp;
+
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_cmd(struct audio_client *ac, int cmd)
+{
+ struct apr_hdr hdr;
+ int rc;
+ atomic_t *state;
+ int cnt = 0;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), TRUE);
+ switch (cmd) {
+ case CMD_PAUSE:
+ pr_debug("%s:CMD_PAUSE\n", __func__);
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ state = &ac->cmd_state;
+ break;
+ case CMD_FLUSH:
+ pr_debug("%s:CMD_FLUSH\n", __func__);
+ hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ state = &ac->cmd_state;
+ break;
+ case CMD_OUT_FLUSH:
+ pr_debug("%s:CMD_OUT_FLUSH\n", __func__);
+ hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ state = &ac->cmd_state;
+ break;
+ case CMD_EOS:
+ pr_debug("%s:CMD_EOS\n", __func__);
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ atomic_set(&ac->cmd_state, 0);
+ state = &ac->cmd_state;
+ break;
+ case CMD_CLOSE:
+ pr_debug("%s:CMD_CLOSE\n", __func__);
+ hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ state = &ac->cmd_state;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", cmd);
+ goto fail_cmd;
+ }
+ pr_debug("%s:session[%d]opcode[0x%x] ", __func__,
+ ac->session,
+ hdr.opcode);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
+ if (rc < 0) {
+ pr_err("Commmand 0x%x failed\n", hdr.opcode);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait, (atomic_read(state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for response opcode[0x%x]\n",
+ hdr.opcode);
+ goto fail_cmd;
+ }
+ if (cmd == CMD_FLUSH)
+ q6asm_reset_buf_state(ac);
+ if (cmd == CMD_CLOSE) {
+ /* check if DSP return all buffers */
+ if (ac->port[IN].buf) {
+ for (cnt = 0; cnt < ac->port[IN].max_buf_cnt;
+ cnt++) {
+ if (ac->port[IN].buf[cnt].used == IN) {
+ pr_debug("Write Buf[%d] not returned\n",
+ cnt);
+ }
+ }
+ }
+ if (ac->port[OUT].buf) {
+ for (cnt = 0; cnt < ac->port[OUT].max_buf_cnt; cnt++) {
+ if (ac->port[OUT].buf[cnt].used == OUT) {
+ pr_debug("Read Buf[%d] not returned\n",
+ cnt);
+ }
+ }
+ }
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
+{
+ struct apr_hdr hdr;
+ int rc;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("%s:APR handle NULL\n", __func__);
+ return -EINVAL;
+ }
+ q6asm_add_hdr_async(ac, &hdr, sizeof(hdr), TRUE);
+ switch (cmd) {
+ case CMD_PAUSE:
+ pr_debug("%s:CMD_PAUSE\n", __func__);
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_EOS:
+ pr_debug("%s:CMD_EOS\n", __func__);
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ break;
+ default:
+ pr_err("%s:Invalid format[%d]\n", __func__, cmd);
+ goto fail_cmd;
+ }
+ pr_debug("%s:session[%d]opcode[0x%x] ", __func__,
+ ac->session,
+ hdr.opcode);
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &hdr);
+ if (rc < 0) {
+ pr_err("%s:Commmand 0x%x failed\n", __func__, hdr.opcode);
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+ int cnt = 0;
+ int loopcnt = 0;
+ struct audio_port_data *port = NULL;
+
+ if (ac->io_mode == SYNC_IO_MODE) {
+ mutex_lock(&ac->cmd_lock);
+ for (loopcnt = 0; loopcnt <= OUT; loopcnt++) {
+ port = &ac->port[loopcnt];
+ cnt = port->max_buf_cnt - 1;
+ port->dsp_buf = 0;
+ port->cpu_buf = 0;
+ while (cnt >= 0) {
+ if (!port->buf)
+ continue;
+ port->buf[cnt].used = 1;
+ cnt--;
+ }
+ }
+ mutex_unlock(&ac->cmd_lock);
+ }
+}
+
+int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable)
+{
+ struct asm_session_cmd_regx_overflow tx_overflow;
+ int rc;
+
+ if (!ac || ac->apr == NULL) {
+ pr_err("APR handle NULL\n");
+ return -EINVAL;
+ }
+ pr_debug("%s:session[%d]enable[%d]\n", __func__,
+ ac->session, enable);
+ q6asm_add_hdr(ac, &tx_overflow.hdr, sizeof(tx_overflow), TRUE);
+
+ tx_overflow.hdr.opcode = \
+ ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS;
+ /* tx overflow event: enable */
+ tx_overflow.enable_flag = enable;
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &tx_overflow);
+ if (rc < 0) {
+ pr_err("tx overflow op[0x%x]rc[%d]\n", \
+ tx_overflow.hdr.opcode, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) == 0), 5*HZ);
+ if (!rc) {
+ pr_err("timeout. waited for tx overflow\n");
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return -EINVAL;
+}
+
+int q6asm_get_apr_service_id(int session_id)
+{
+ pr_debug("%s\n", __func__);
+
+ if (session_id < 0 || session_id > SESSION_MAX) {
+ pr_err("%s: invalid session_id = %d\n", __func__, session_id);
+ return -EINVAL;
+ }
+
+ return ((struct apr_svc *)session[session_id]->apr)->id;
+}
+
+
+static int __init q6asm_init(void)
+{
+ pr_debug("%s\n", __func__);
+ memset(session, 0, sizeof(session));
+
+ config_debug_fs_init();
+
+ return 0;
+}
+
+device_initcall(q6asm_init);
diff --git a/sound/soc/msm/qdsp6v2/q6audio-v2.c b/sound/soc/msm/qdsp6v2/q6audio-v2.c
new file mode 100644
index 0000000..8c524fa
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/q6audio-v2.c
@@ -0,0 +1,151 @@
+/* Copyright (c) 2012, Code Aurora Forum. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/sched.h>
+#include <linux/jiffies.h>
+#include <linux/uaccess.h>
+#include <linux/atomic.h>
+#include <sound/q6afe-v2.h>
+#include <sound/q6audio-v2.h>
+
+int q6audio_get_port_index(u16 port_id)
+{
+ switch (port_id) {
+ case PRIMARY_I2S_RX: return IDX_PRIMARY_I2S_RX;
+ case PRIMARY_I2S_TX: return IDX_PRIMARY_I2S_TX;
+ case PCM_RX: return IDX_PCM_RX;
+ case PCM_TX: return IDX_PCM_TX;
+ case SECONDARY_I2S_RX: return IDX_SECONDARY_I2S_RX;
+ case SECONDARY_I2S_TX: return IDX_SECONDARY_I2S_TX;
+ case MI2S_RX: return IDX_MI2S_RX;
+ case MI2S_TX: return IDX_MI2S_TX;
+ case HDMI_RX: return IDX_HDMI_RX;
+ case RSVD_2: return IDX_RSVD_2;
+ case RSVD_3: return IDX_RSVD_3;
+ case DIGI_MIC_TX: return IDX_DIGI_MIC_TX;
+ case VOICE_RECORD_RX: return IDX_VOICE_RECORD_RX;
+ case VOICE_RECORD_TX: return IDX_VOICE_RECORD_TX;
+ case VOICE_PLAYBACK_TX: return IDX_VOICE_PLAYBACK_TX;
+ case SLIMBUS_0_RX: return IDX_SLIMBUS_0_RX;
+ case SLIMBUS_0_TX: return IDX_SLIMBUS_0_TX;
+ case SLIMBUS_1_RX: return IDX_SLIMBUS_1_RX;
+ case SLIMBUS_1_TX: return IDX_SLIMBUS_1_TX;
+ case INT_BT_SCO_RX: return IDX_INT_BT_SCO_RX;
+ case INT_BT_SCO_TX: return IDX_INT_BT_SCO_TX;
+ case INT_BT_A2DP_RX: return IDX_INT_BT_A2DP_RX;
+ case INT_FM_RX: return IDX_INT_FM_RX;
+ case INT_FM_TX: return IDX_INT_FM_TX;
+ case RT_PROXY_PORT_001_RX: return IDX_RT_PROXY_PORT_001_RX;
+ case RT_PROXY_PORT_001_TX: return IDX_RT_PROXY_PORT_001_TX;
+
+ default: return -EINVAL;
+ }
+}
+
+int q6audio_get_port_id(u16 port_id)
+{
+ switch (port_id) {
+ case PRIMARY_I2S_RX: return AFE_PORT_ID_PRIMARY_MI2S_RX;
+ case PRIMARY_I2S_TX: return AFE_PORT_ID_PRIMARY_MI2S_TX;
+ case PCM_RX: return AFE_PORT_ID_PRIMARY_PCM_RX;
+ case PCM_TX: return AFE_PORT_ID_PRIMARY_PCM_TX;
+ case SECONDARY_I2S_RX: return AFE_PORT_ID_SECONDARY_MI2S_RX;
+ case SECONDARY_I2S_TX: return AFE_PORT_ID_SECONDARY_MI2S_TX;
+ case MI2S_RX: return AFE_PORT_ID_PRIMARY_MI2S_RX;
+ case MI2S_TX: return AFE_PORT_ID_PRIMARY_MI2S_TX;
+ case HDMI_RX: return AFE_PORT_ID_MULTICHAN_HDMI_RX;
+ case RSVD_2: return IDX_RSVD_2;
+ case RSVD_3: return IDX_RSVD_3;
+ case DIGI_MIC_TX: return AFE_PORT_ID_DIGITAL_MIC_TX;
+ case VOICE_RECORD_RX: return AFE_PORT_ID_VOICE_RECORD_RX;
+ case VOICE_RECORD_TX: return AFE_PORT_ID_VOICE_RECORD_TX;
+ case VOICE_PLAYBACK_TX: return AFE_PORT_ID_VOICE_PLAYBACK_TX;
+ case SLIMBUS_0_RX: return AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX;
+ case SLIMBUS_0_TX: return AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX;
+ case SLIMBUS_1_RX: return AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX;
+ case SLIMBUS_1_TX: return AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX;
+ case INT_BT_SCO_RX: return AFE_PORT_ID_INTERNAL_BT_SCO_RX;
+ case INT_BT_SCO_TX: return AFE_PORT_ID_INTERNAL_BT_SCO_TX;
+ case INT_BT_A2DP_RX: return AFE_PORT_ID_INTERNAL_BT_A2DP_RX;
+ case INT_FM_RX: return AFE_PORT_ID_INTERNAL_FM_RX;
+ case INT_FM_TX: return AFE_PORT_ID_INTERNAL_FM_TX;
+ case RT_PROXY_PORT_001_RX: return AFE_PORT_ID_RT_PROXY_PORT_001_RX;
+ case RT_PROXY_PORT_001_TX: return AFE_PORT_ID_RT_PROXY_PORT_001_TX;
+
+ default: return -EINVAL;
+ }
+}
+int q6audio_convert_virtual_to_portid(u16 port_id)
+{
+ int ret;
+
+ /* if port_id is virtual, convert to physical..
+ * if port_id is already physical, return physical
+ */
+ if (q6audio_validate_port(port_id) < 0) {
+ if (port_id == RT_PROXY_DAI_001_RX ||
+ port_id == RT_PROXY_DAI_001_TX ||
+ port_id == RT_PROXY_DAI_002_RX ||
+ port_id == RT_PROXY_DAI_002_TX)
+ ret = VIRTUAL_ID_TO_PORTID(port_id);
+ else
+ ret = -EINVAL;
+ } else
+ ret = port_id;
+
+ return ret;
+}
+
+int q6audio_validate_port(u16 port_id)
+{
+ int ret;
+
+ switch (port_id) {
+ case PRIMARY_I2S_RX:
+ case PRIMARY_I2S_TX:
+ case PCM_RX:
+ case PCM_TX:
+ case SECONDARY_I2S_RX:
+ case SECONDARY_I2S_TX:
+ case MI2S_RX:
+ case MI2S_TX:
+ case HDMI_RX:
+ case RSVD_2:
+ case RSVD_3:
+ case DIGI_MIC_TX:
+ case VOICE_RECORD_RX:
+ case VOICE_RECORD_TX:
+ case VOICE_PLAYBACK_TX:
+ case SLIMBUS_0_RX:
+ case SLIMBUS_0_TX:
+ case SLIMBUS_1_RX:
+ case SLIMBUS_1_TX:
+ case INT_BT_SCO_RX:
+ case INT_BT_SCO_TX:
+ case INT_BT_A2DP_RX:
+ case INT_FM_RX:
+ case INT_FM_TX:
+ case RT_PROXY_PORT_001_RX:
+ case RT_PROXY_PORT_001_TX:
+ {
+ ret = 0;
+ break;
+ }
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
diff --git a/sound/soc/omap/abe/Makefile b/sound/soc/omap/abe/Makefile
new file mode 100644
index 0000000..0d5649b
--- /dev/null
+++ b/sound/soc/omap/abe/Makefile
@@ -0,0 +1,14 @@
+snd-soc-abe-hal-objs += abe_main.o \
+ abe_core.o \
+ abe_gain.o \
+ abe_port.o \
+ abe_aess.o \
+ abe_dbg.o \
+ abe_dat.o \
+ abe_ini.o \
+ abe_irq.o \
+ abe_seq.o \
+ abe_asrc.o \
+ port_mgr.o \
+
+obj-$(CONFIG_SND_OMAP_SOC_ABE_DSP) += snd-soc-abe-hal.o
diff --git a/sound/soc/omap/abe/abe.h b/sound/soc/omap/abe/abe.h
new file mode 100644
index 0000000..c465764
--- /dev/null
+++ b/sound/soc/omap/abe/abe.h
@@ -0,0 +1,159 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_H_
+#define _ABE_H_
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+
+#include "abe_def.h"
+#include "abe_define.h"
+#include "abe_fw.h"
+#include "abe_ext.h"
+#include "abe_dbg.h"
+
+/*
+ * BASIC TYPES
+ */
+#define MAX_UINT8 ((((1L << 7) - 1) << 1) + 1)
+#define MAX_UINT16 ((((1L << 15) - 1) << 1) + 1)
+#define MAX_UINT32 ((((1L << 31) - 1) << 1) + 1)
+
+#define s8 char
+#define u8 unsigned char
+#define s16 short
+#define u16 unsigned short
+#define s32 int
+#define u32 unsigned int
+
+struct omap_abe_equ {
+ /* type of filter */
+ u32 equ_type;
+ /* filter length */
+ u32 equ_length;
+ union {
+ /* parameters are the direct and recursive coefficients in */
+ /* Q6.26 integer fixed-point format. */
+ s32 type1[NBEQ1];
+ struct {
+ /* center frequency of the band [Hz] */
+ s32 freq[NBEQ2];
+ /* gain of each band. [dB] */
+ s32 gain[NBEQ2];
+ /* Q factor of this band [dB] */
+ s32 q[NBEQ2];
+ } type2;
+ } coef;
+ s32 equ_param3;
+};
+
+struct omap_abe {
+ void __iomem *io_base[5];
+ u32 firmware_version_number;
+ u16 MultiFrame[PROCESSING_SLOTS][TASKS_IN_SLOT];
+ u32 compensated_mixer_gain;
+ u8 muted_gains_indicator[MAX_NBGAIN_CMEM];
+ u32 desired_gains_decibel[MAX_NBGAIN_CMEM];
+ u32 muted_gains_decibel[MAX_NBGAIN_CMEM];
+ u32 desired_gains_linear[MAX_NBGAIN_CMEM];
+ u32 desired_ramp_delay_ms[MAX_NBGAIN_CMEM];
+ struct mutex mutex;
+ u32 warm_boot;
+
+ u32 irq_dbg_read_ptr;
+
+ struct omap_abe_dbg dbg;
+};
+
+extern struct omap_abe *abe;
+
+void omap_abe_dbg_log(struct omap_abe *abe, u32 x, u32 y, u32 z, u32 t);
+void omap_abe_dbg_error(struct omap_abe *abe, int level, int error);
+int omap_abe_set_opp_processing(struct omap_abe *abe, u32 opp);
+int omap_abe_connect_debug_trace(struct omap_abe *abe,
+ struct omap_abe_dma *dma2);
+
+int omap_abe_use_compensated_gain(struct omap_abe *abe, int on_off);
+int omap_abe_write_equalizer(struct omap_abe *abe,
+ u32 id, struct omap_abe_equ *param);
+
+int omap_abe_disable_gain(struct omap_abe *abe, u32 id, u32 p);
+int omap_abe_enable_gain(struct omap_abe *abe, u32 id, u32 p);
+int omap_abe_mute_gain(struct omap_abe *abe, u32 id, u32 p);
+int omap_abe_unmute_gain(struct omap_abe *abe, u32 id, u32 p);
+
+int omap_abe_write_gain(struct omap_abe *abe,
+ u32 id, s32 f_g, u32 ramp, u32 p);
+int omap_abe_write_mixer(struct omap_abe *abe,
+ u32 id, s32 f_g, u32 f_ramp, u32 p);
+int omap_abe_read_gain(struct omap_abe *abe,
+ u32 id, u32 *f_g, u32 p);
+int omap_abe_read_mixer(struct omap_abe *abe,
+ u32 id, u32 *f_g, u32 p);
+
+/*
+ * MACROS
+ */
+#define _log(x, y, z, t) { if (x & abe->dbg.mask) omap_abe_dbg_log(abe, x, y, z, t); }
+
+#endif/* _ABE_H_ */
diff --git a/sound/soc/omap/abe/abe_aess.c b/sound/soc/omap/abe/abe_aess.c
new file mode 100644
index 0000000..eb35b58
--- /dev/null
+++ b/sound/soc/omap/abe/abe_aess.c
@@ -0,0 +1,191 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_dbg.h"
+#include "abe.h"
+#include "abe_mem.h"
+#include "abe_aess.h"
+
+/**
+ * omap_abe_hw_configuration
+ *
+ */
+void omap_abe_hw_configuration(struct omap_abe *abe)
+{
+ /* enables the DMAreq from AESS AESS_DMAENABLE_SET = 255 */
+ omap_abe_reg_writel(abe, AESS_DMAENABLE_SET, DMA_ENABLE_ALL);
+ /* enables the MCU IRQ from AESS to Cortex A9 */
+ omap_abe_reg_writel(abe, AESS_MCU_IRQENABLE_SET, INT_SET);
+}
+
+/**
+ * omap_abe_clear_irq - clear ABE interrupt
+ * @abe: Pointer on abe handle
+ *
+ * This subroutine is call to clear MCU Irq
+ */
+int omap_abe_clear_irq(struct omap_abe *abe)
+{
+ omap_abe_reg_writel(abe, ABE_MCU_IRQSTATUS, INT_CLR);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_clear_irq);
+
+/**
+ * abe_write_event_generator - Selects event generator source
+ * @abe: Pointer on abe handle
+ * @e: Event Generation Counter, McPDM, DMIC or default.
+ *
+ * Loads the AESS event generator hardware source.
+ * Loads the firmware parameters accordingly.
+ * Indicates to the FW which data stream is the most important to preserve
+ * in case all the streams are asynchronous.
+ * If the parameter is "default", then HAL decides which Event source
+ * is the best appropriate based on the opened ports.
+ *
+ * When neither the DMIC and the McPDM are activated, the AE will have
+ * its EVENT generator programmed with the EVENT_COUNTER.
+ * The event counter will be tuned in order to deliver a pulse frequency higher
+ * than 96 kHz.
+ * The DPLL output at 100% OPP is MCLK = (32768kHz x6000) = 196.608kHz
+ * The ratio is (MCLK/96000)+(1<<1) = 2050
+ * (1<<1) in order to have the same speed at 50% and 100% OPP
+ * (only 15 MSB bits are used at OPP50%)
+ */
+int omap_abe_write_event_generator(struct omap_abe *abe, u32 e)
+{
+ u32 event, selection;
+ u32 counter = EVENT_GENERATOR_COUNTER_DEFAULT;
+
+ _log(ABE_ID_WRITE_EVENT_GENERATOR, e, 0, 0);
+
+ switch (e) {
+ case EVENT_TIMER:
+ selection = EVENT_SOURCE_COUNTER;
+ event = 0;
+ break;
+ case EVENT_44100:
+ selection = EVENT_SOURCE_COUNTER;
+ event = 0;
+ counter = EVENT_GENERATOR_COUNTER_44100;
+ break;
+ default:
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API, ABE_BLOCK_COPY_ERR);
+ }
+ omap_abe_reg_writel(abe, EVENT_GENERATOR_COUNTER, counter);
+ omap_abe_reg_writel(abe, EVENT_SOURCE_SELECTION, selection);
+ omap_abe_reg_writel(abe, EVENT_GENERATOR_START, EVENT_GENERATOR_ON);
+ omap_abe_reg_writel(abe, AUDIO_ENGINE_SCHEDULER, event);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_write_event_generator);
+
+/**
+ * omap_abe_start_event_generator - Starts event generator source
+ * @abe: Pointer on abe handle
+ *
+ * Start the event genrator of AESS. No more event will be send to AESS engine.
+ * Upper layer must wait 1/96kHz to be sure that engine reaches
+ * the IDLE instruction.
+ */
+int omap_abe_start_event_generator(struct omap_abe *abe)
+{
+ /* Start the event Generator */
+ omap_abe_reg_writel(abe, EVENT_GENERATOR_START, 1);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_start_event_generator);
+
+/**
+ * omap_abe_stop_event_generator - Stops event generator source
+ * @abe: Pointer on abe handle
+ *
+ * Stop the event genrator of AESS. No more event will be send to AESS engine.
+ * Upper layer must wait 1/96kHz to be sure that engine reaches
+ * the IDLE instruction.
+ */
+int omap_abe_stop_event_generator(struct omap_abe *abe)
+{
+ /* Stop the event Generator */
+ omap_abe_reg_writel(abe, EVENT_GENERATOR_START, 0);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_stop_event_generator);
+
+/**
+ * omap_abe_disable_irq - disable MCU/DSP ABE interrupt
+ * @abe: Pointer on abe handle
+ *
+ * This subroutine is disabling ABE MCU/DSP Irq
+ */
+int omap_abe_disable_irq(struct omap_abe *abe)
+{
+ /* disables the DMAreq from AESS AESS_DMAENABLE_CLR = 127
+ * DMA_Req7 will still be enabled as it is used for ABE trace */
+ omap_abe_reg_writel(abe, AESS_DMAENABLE_CLR, 0x7F);
+ /* disables the MCU IRQ from AESS to Cortex A9 */
+ omap_abe_reg_writel(abe, AESS_MCU_IRQENABLE_CLR, 0x01);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_disable_irq);
diff --git a/sound/soc/omap/abe/abe_aess.h b/sound/soc/omap/abe/abe_aess.h
new file mode 100644
index 0000000..70c54f8e
--- /dev/null
+++ b/sound/soc/omap/abe/abe_aess.h
@@ -0,0 +1,113 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_AESS_H_
+#define _ABE_AESS_H_
+
+#define AESS_REVISION 0x00
+#define AESS_MCU_IRQSTATUS 0x28
+#define AESS_MCU_IRQENABLE_SET 0x3C
+#define AESS_MCU_IRQENABLE_CLR 0x40
+#define AESS_DMAENABLE_SET 0x60
+#define AESS_DMAENABLE_CLR 0x64
+#define EVENT_GENERATOR_COUNTER 0x68
+#define EVENT_GENERATOR_START 0x6C
+#define EVENT_SOURCE_SELECTION 0x70
+#define AUDIO_ENGINE_SCHEDULER 0x74
+
+/*
+ * AESS_MCU_IRQSTATUS bit field
+ */
+#define INT_CLEAR 0x01
+
+/*
+ * AESS_MCU_IRQENABLE_SET bit field
+ */
+#define INT_SET 0x01
+
+/*
+ * AESS_MCU_IRQENABLE_CLR bit field
+ */
+#define INT_CLR 0x01
+
+/*
+ * AESS_DMAENABLE_SET bit fields
+ */
+#define DMA_ENABLE_ALL 0xFF
+
+/*
+ * AESS_DMAENABLE_CLR bit fields
+ */
+#define DMA_DISABLE_ALL 0xFF
+
+/*
+ * EVENT_GENERATOR_COUNTER COUNTER_VALUE bit field
+ */
+/* PLL output/desired sampling rate = (32768 * 6000)/96000 */
+#define EVENT_GENERATOR_COUNTER_DEFAULT (2048-1)
+/* PLL output/desired sampling rate = (32768 * 6000)/88200 */
+#define EVENT_GENERATOR_COUNTER_44100 (2228-1)
+
+
+int omap_abe_start_event_generator(struct omap_abe *abe);
+int omap_abe_stop_event_generator(struct omap_abe *abe);
+int omap_abe_write_event_generator(struct omap_abe *abe, u32 e);
+
+void omap_abe_hw_configuration(struct omap_abe *abe);
+
+#endif/* _ABE_AESS_H_ */
diff --git a/sound/soc/omap/abe/abe_api.h b/sound/soc/omap/abe/abe_api.h
new file mode 100644
index 0000000..430aa5b
--- /dev/null
+++ b/sound/soc/omap/abe/abe_api.h
@@ -0,0 +1,540 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_API_H_
+#define _ABE_API_H_
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_dm_addr.h"
+#include "abe_dbg.h"
+
+#define ABE_TASK_ID(ID) (OMAP_ABE_D_TASKSLIST_ADDR + sizeof(ABE_STask)*(ID))
+
+#define TASK_ASRC_VX_DL_SLT 0
+#define TASK_ASRC_VX_DL_IDX 3
+#define TASK_VX_DL_SLT 1
+#define TASK_VX_DL_IDX 3
+#define TASK_DL2Mixer_SLT 1
+#define TASK_DL2Mixer_IDX 6
+#define TASK_DL1Mixer_SLT 2
+#define TASK_DL1Mixer_IDX 0
+#define TASK_VX_UL_SLT 12
+#define TASK_VX_UL_IDX 5
+#define TASK_BT_DL_48_8_SLT 14
+#define TASK_BT_DL_48_8_IDX 4
+#define TASK_ASRC_BT_UL_SLT 15
+#define TASK_ASRC_BT_UL_IDX 6
+#define TASK_ASRC_VX_UL_SLT 16
+#define TASK_ASRC_VX_UL_IDX 2
+#define TASK_BT_UL_8_48_SLT 17
+#define TASK_BT_UL_8_48_IDX 2
+#define TASK_IO_MM_DL_SLT 18
+#define TASK_IO_MM_DL_IDX 0
+#define TASK_ASRC_BT_DL_SLT 18
+#define TASK_ASRC_BT_DL_IDX 6
+
+
+struct omap_abe {
+ void __iomem *io_base[5];
+ u32 firmware_version_number;
+ u16 MultiFrame[PROCESSING_SLOTS][TASKS_IN_SLOT];
+ u32 compensated_mixer_gain;
+ u8 muted_gains_indicator[MAX_NBGAIN_CMEM];
+ u32 desired_gains_decibel[MAX_NBGAIN_CMEM];
+ u32 muted_gains_decibel[MAX_NBGAIN_CMEM];
+ u32 desired_gains_linear[MAX_NBGAIN_CMEM];
+ u32 desired_ramp_delay_ms[MAX_NBGAIN_CMEM];
+ struct mutex mutex;
+ u32 warm_boot;
+
+ u32 irq_dbg_read_ptr;
+ u32 dbg_param;
+
+ struct omap_abe_dbg dbg;
+};
+
+/**
+ * abe_reset_hal - reset the ABE/HAL
+ * @rdev: regulator source
+ * @constraints: constraints to apply
+ *
+ * Operations : reset the HAL by reloading the static variables and
+ * default AESS registers.
+ * Called after a PRCM cold-start reset of ABE
+ */
+abehal_status abe_reset_hal(void);
+/**
+ * abe_load_fw_param - Load ABE Firmware memories
+ * @PMEM: Pointer of Program memory data
+ * @PMEM_SIZE: Size of PMEM data
+ * @CMEM: Pointer of Coeffients memory data
+ * @CMEM_SIZE: Size of CMEM data
+ * @SMEM: Pointer of Sample memory data
+ * @SMEM_SIZE: Size of SMEM data
+ * @DMEM: Pointer of Data memory data
+ * @DMEM_SIZE: Size of DMEM data
+ *
+ */
+abehal_status abe_load_fw_param(u32 *FW);
+/**
+ * abe_irq_processing - Process ABE interrupt
+ *
+ * This subroutine is call upon reception of "MA_IRQ_99 ABE_MPU_IRQ" Audio
+ * back-end interrupt. This subroutine will check the ATC Hrdware, the
+ * IRQ_FIFO from the AE and act accordingly. Some IRQ source are originated
+ * for the delivery of "end of time sequenced tasks" notifications, some are
+ * originated from the Ping-Pong protocols, some are generated from
+ * the embedded debugger when the firmware stops on programmable break-points,
+ * etc ...
+ */
+abehal_status abe_irq_processing(void);
+/**
+ * abe_irq_clear - clear ABE interrupt
+ *
+ * This subroutine is call to clear MCU Irq
+ */
+abehal_status abe_clear_irq(void);
+/**
+ * abe_disable_irq - disable MCU/DSP ABE interrupt
+ *
+ * This subroutine is disabling ABE MCU/DSP Irq
+ */
+abehal_status abe_disable_irq(void);
+/*
+ * abe_check_activity - check all ports are closed
+ */
+u32 abe_check_activity(void);
+/**
+ * abe_wakeup - Wakeup ABE
+ *
+ * Wakeup ABE in case of retention
+ */
+abehal_status abe_wakeup(void);
+/**
+ * abe_start_event_generator - Stops event generator source
+ *
+ * Start the event genrator of AESS. No more event will be send to AESS engine.
+ * Upper layer must wait 1/96kHz to be sure that engine reaches
+ * the IDLE instruction.
+ */
+abehal_status abe_start_event_generator(void);
+/**
+ * abe_stop_event_generator - Stops event generator source
+ *
+ * Stop the event genrator of AESS. No more event will be send to AESS engine.
+ * Upper layer must wait 1/96kHz to be sure that engine reaches
+ * the IDLE instruction.
+ */
+abehal_status abe_stop_event_generator(void);
+
+/**
+ * abe_write_event_generator - Selects event generator source
+ * @e: Event Generation Counter, McPDM, DMIC or default.
+ *
+ * Loads the AESS event generator hardware source.
+ * Loads the firmware parameters accordingly.
+ * Indicates to the FW which data stream is the most important to preserve
+ * in case all the streams are asynchronous.
+ * If the parameter is "default", then HAL decides which Event source
+ * is the best appropriate based on the opened ports.
+ *
+ * When neither the DMIC and the McPDM are activated, the AE will have
+ * its EVENT generator programmed with the EVENT_COUNTER.
+ * The event counter will be tuned in order to deliver a pulse frequency higher
+ * than 96 kHz.
+ * The DPLL output at 100% OPP is MCLK = (32768kHz x6000) = 196.608kHz
+ * The ratio is (MCLK/96000)+(1<<1) = 2050
+ * (1<<1) in order to have the same speed at 50% and 100% OPP
+ * (only 15 MSB bits are used at OPP50%)
+ */
+abehal_status abe_write_event_generator(u32 e);
+/**
+ * abe_set_opp_processing - Set OPP mode for ABE Firmware
+ * @opp: OOPP mode
+ *
+ * New processing network and OPP:
+ * 0: Ultra Lowest power consumption audio player (no post-processing, no mixer)
+ * 1: OPP 25% (simple multimedia features, including low-power player)
+ * 2: OPP 50% (multimedia and voice calls)
+ * 3: OPP100% (EANC, multimedia complex use-cases)
+ *
+ * Rearranges the FW task network to the corresponding OPP list of features.
+ * The corresponding AE ports are supposed to be set/reset accordingly before
+ * this switch.
+ *
+ */
+abehal_status abe_set_opp_processing(u32 opp);
+/**
+ * abe_set_ping_pong_buffer
+ * @port: ABE port ID
+ * @n_bytes: Size of Ping/Pong buffer
+ *
+ * Updates the next ping-pong buffer with "size" bytes copied from the
+ * host processor. This API notifies the FW that the data transfer is done.
+ */
+abehal_status abe_set_ping_pong_buffer(u32 port, u32 n_bytes);
+/**
+ * abe_read_next_ping_pong_buffer
+ * @port: ABE portID
+ * @p: Next buffer address (pointer)
+ * @n: Next buffer size (pointer)
+ *
+ * Tell the next base address of the next ping_pong Buffer and its size
+ */
+abehal_status abe_read_next_ping_pong_buffer(u32 port, u32 *p, u32 *n);
+/**
+ * abe_init_ping_pong_buffer
+ * @id: ABE port ID
+ * @size_bytes:size of the ping pong
+ * @n_buffers:number of buffers (2 = ping/pong)
+ * @p:returned address of the ping-pong list of base address (byte offset
+ from DMEM start)
+ *
+ * Computes the base address of the ping_pong buffers
+ */
+abehal_status abe_init_ping_pong_buffer(u32 id, u32 size_bytes, u32 n_buffers,
+ u32 *p);
+/**
+ * abe_read_offset_from_ping_buffer
+ * @id: ABE port ID
+ * @n: returned address of the offset
+ * from the ping buffer start address (in samples)
+ *
+ * Computes the current firmware ping pong read pointer location,
+ * expressed in samples, as the offset from the start address of ping buffer.
+ */
+abehal_status abe_read_offset_from_ping_buffer(u32 id, u32 *n);
+/**
+ * abe_plug_subroutine
+ * @id: returned sequence index after plugging a new subroutine
+ * @f: subroutine address to be inserted
+ * @n: number of parameters of this subroutine
+ * @params: pointer on parameters
+ *
+ * register a list of subroutines for call-back purpose
+ */
+abehal_status abe_plug_subroutine(u32 *id, abe_subroutine2 f, u32 n,
+ u32 *params);
+/**
+ * abe_set_sequence_time_accuracy
+ * @fast: fast counter
+ * @slow: slow counter
+ *
+ */
+abehal_status abe_set_sequence_time_accuracy(u32 fast, u32 slow);
+/**
+ * abe_reset_port
+ * @id: ABE port ID
+ *
+ * stop the port activity and reload default parameters on the associated
+ * processing features.
+ * Clears the internal AE buffers.
+ */
+abehal_status abe_reset_port(u32 id);
+/**
+ * abe_read_remaining_data
+ * @id: ABE port_ID
+ * @n: size pointer to the remaining number of 32bits words
+ *
+ * computes the remaining amount of data in the buffer.
+ */
+abehal_status abe_read_remaining_data(u32 port, u32 *n);
+/**
+ * abe_disable_data_transfer
+ * @id: ABE port id
+ *
+ * disables the ATC descriptor and stop IO/port activities
+ * disable the IO task (@f = 0)
+ * clear ATC DMEM buffer, ATC enabled
+ */
+abehal_status abe_disable_data_transfer(u32 id);
+/**
+ * abe_enable_data_transfer
+ * @ip: ABE port id
+ *
+ * enables the ATC descriptor
+ * reset ATC pointers
+ * enable the IO task (@f <> 0)
+ */
+abehal_status abe_enable_data_transfer(u32 id);
+/**
+ * abe_set_dmic_filter
+ * @d: DMIC decimation ratio : 16/25/32/40
+ *
+ * Loads in CMEM a specific list of coefficients depending on the DMIC sampling
+ * frequency (2.4MHz or 3.84MHz). This table compensates the DMIC decimator
+ * roll-off at 20kHz.
+ * The default table is loaded with the DMIC 2.4MHz recommended configuration.
+ */
+abehal_status abe_set_dmic_filter(u32 d);
+/**
+ * abe_connect_cbpr_dmareq_port
+ * @id: port name
+ * @f: desired data format
+ * @d: desired dma_request line (0..7)
+ * @a: returned pointer to the base address of the CBPr register and number of
+ * samples to exchange during a DMA_request.
+ *
+ * enables the data echange between a DMA and the ABE through the
+ * CBPr registers of AESS.
+ */
+abehal_status abe_connect_cbpr_dmareq_port(u32 id, abe_data_format_t *f, u32 d,
+ abe_dma_t *returned_dma_t);
+/**
+ * abe_connect_irq_ping_pong_port
+ * @id: port name
+ * @f: desired data format
+ * @I: index of the call-back subroutine to call
+ * @s: half-buffer (ping) size
+ * @p: returned base address of the first (ping) buffer)
+ *
+ * enables the data echanges between a direct access to the DMEM
+ * memory of ABE using cache flush. On each IRQ activation a subroutine
+ * registered with "abe_plug_subroutine" will be called. This subroutine
+ * will generate an amount of samples, send them to DMEM memory and call
+ * "abe_set_ping_pong_buffer" to notify the new amount of samples in the
+ * pong buffer.
+ */
+abehal_status abe_connect_irq_ping_pong_port(u32 id, abe_data_format_t *f,
+ u32 subroutine_id, u32 size,
+ u32 *sink, u32 dsp_mcu_flag);
+/**
+ * abe_connect_serial_port()
+ * @id: port name
+ * @f: data format
+ * @i: peripheral ID (McBSP #1, #2, #3)
+ *
+ * Operations : enables the data echanges between a McBSP and an ATC buffer in
+ * DMEM. This API is used connect 48kHz McBSP streams to MM_DL and 8/16kHz
+ * voice streams to VX_UL, VX_DL, BT_VX_UL, BT_VX_DL. It abstracts the
+ * abe_write_port API.
+ */
+abehal_status abe_connect_serial_port(u32 id, abe_data_format_t *f,
+ u32 mcbsp_id);
+/**
+ * abe_read_port_address
+ * @dma: output pointer to the DMA iteration and data destination pointer
+ *
+ * This API returns the address of the DMA register used on this audio port.
+ * Depending on the protocol being used, adds the base address offset L3
+ * (DMA) or MPU (ARM)
+ */
+abehal_status abe_read_port_address(u32 port, abe_dma_t *dma2);
+/**
+ * abe_write_equalizer
+ * @id: name of the equalizer
+ * @param : equalizer coefficients
+ *
+ * Load the coefficients in CMEM.
+ */
+abehal_status abe_write_equalizer(u32 id, abe_equ_t *param);
+/**
+ * abe_write_asrc
+ * @id: name of the port
+ * @param: drift value to compensate [ppm]
+ *
+ * Load the drift variables to the FW memory. This API can be called only
+ * when the corresponding port has been already opened and the ASRC has
+ * been correctly initialized with API abe_init_asrc_... If this API is
+ * used such that the drift has been changed from positive to negative drift
+ * or vice versa, there will be click in the output signal. Loading the drift
+ * value with zero disables the feature.
+ */
+abehal_status abe_write_asrc(u32 port, s32 dppm);
+/**
+ * abe_write_aps
+ * @id: name of the aps filter
+ * @param: table of filter coefficients
+ *
+ * Load the filters and thresholds coefficients in FW memory. This AP
+ * can be called when the corresponding APS is not activated. After
+ * reloading the firmware the default coefficients corresponds to "no APS
+ * activated".
+ * Loading all the coefficients value with zero disables the feature.
+ */
+abehal_status abe_write_aps(u32 id, struct abe_aps_t *param);
+/**
+ * abe_write_mixer
+ * @id: name of the mixer
+ * @param: list of input gains of the mixer
+ * @p: list of port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's gain
+ * in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+abehal_status abe_write_gain(u32 id, s32 f_g, u32 ramp, u32 p);
+abehal_status abe_use_compensated_gain(u32 on_off);
+abehal_status abe_enable_gain(u32 id, u32 p);
+abehal_status abe_disable_gain(u32 id, u32 p);
+abehal_status abe_mute_gain(u32 id, u32 p);
+abehal_status abe_unmute_gain(u32 id, u32 p);
+/**
+ * abe_write_mixer
+ * @id: name of the mixer
+ * @param: input gains and delay ramp of the mixer
+ * @p: port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's
+ * gain in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+abehal_status abe_write_mixer(u32 id, s32 f_g, u32 f_ramp, u32 p);
+/**
+ * abe_read_gain
+ * @id: name of the mixer
+ * @param: list of input gains of the mixer
+ * @p: list of port corresponding to the above gains
+ *
+ */
+abehal_status abe_read_gain(u32 id, u32 *f_g, u32 p);
+/**
+ * abe_read_mixer
+ * @id: name of the mixer
+ * @param: gains of the mixer
+ * @p: port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's
+ * gain in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+abehal_status abe_read_mixer(u32 id, u32 *f_g, u32 p);
+/**
+ * abe_mono_mixer
+ * id: name of the mixer (MIXDL1 or MIXDL2)
+ * on_off: enable\disable flag
+ *
+ * This API Programs DL1Mixer or DL2Mixer to output mono data
+ * on both left and right data paths.
+ */
+abehal_status abe_mono_mixer(u32 id, u32 on_off);
+/**
+ * abe_set_router_configuration
+ * @Id: name of the router
+ * @Conf: id of the configuration
+ * @param: list of output index of the route
+ *
+ * The uplink router takes its input from DMIC (6 samples), AMIC (2 samples)
+ * and PORT1/2 (2 stereo ports). Each sample will be individually stored in
+ * an intermediate table of 10 elements. The intermediate table is used to
+ * route the samples to three directions : REC1 mixer, 2 EANC DMIC source of
+ * filtering and MM recording audio path.
+ */
+abehal_status abe_set_router_configuration(u32 id, u32 k, u32 *param);
+/**
+ * ABE_READ_DEBUG_TRACE
+ *
+ * Parameters :
+ * @data: data destination pointer
+ * @n : max number of read data
+ *
+ * Operations :
+ * Reads the AE circular data pointer that holds pairs of debug data +
+ * timestamps, and stores the pairs, via linear addressing, to the parameter
+ * pointer.
+ * Stops the copy when the max parameter is reached or when the FIFO is empty.
+ *
+ * Return value :
+ * None.
+ */
+abehal_status abe_read_debug_trace(u32 *data, u32 *n);
+/**
+ * abe_connect_debug_trace
+ * @dma2:pointer to the DMEM trace buffer
+ *
+ * returns the address and size of the real-time debug trace buffer,
+ * the content of which will vary from one firmware release to an other
+ */
+abehal_status abe_connect_debug_trace(abe_dma_t *dma2);
+/**
+ * abe_set_debug_trace
+ * @debug: debug ID from a list to be defined
+ *
+ * load a mask which filters the debug trace to dedicated types of data
+ */
+abehal_status abe_set_debug_trace(abe_dbg_t debug);
+/**
+ * abe_init_mem - Allocate Kernel space memory map for ABE
+ *
+ * Memory map of ABE memory space for PMEM/DMEM/SMEM/DMEM
+ */
+void abe_init_mem(void __iomem **_io_base);
+
+/**
+ * abe_write_pdmdl_offset - write the desired offset on the DL1/DL2 paths
+ *
+ * Parameters:
+ * path: 1 for the DL1 ABE path, 2 for the DL2 ABE path
+ * offset_left: integer value that will be added on all PDM left samples
+ * offset_right: integer value that will be added on all PDM right samples
+ *
+ */
+void abe_write_pdmdl_offset(u32 path, u32 offset_left, u32 offset_right);
+
+#endif/* _ABE_API_H_ */
diff --git a/sound/soc/omap/abe/abe_asrc.c b/sound/soc/omap/abe/abe_asrc.c
new file mode 100644
index 0000000..4a52235
--- /dev/null
+++ b/sound/soc/omap/abe/abe_asrc.c
@@ -0,0 +1,1231 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_legacy.h"
+#include "abe_dbg.h"
+
+#include "abe_typedef.h"
+#include "abe_initxxx_labels.h"
+#include "abe_dbg.h"
+#include "abe_mem.h"
+#include "abe_sm_addr.h"
+#include "abe_cm_addr.h"
+
+/**
+ * abe_write_fifo
+ * @mem_bank: currently only ABE_DMEM supported
+ * @addr: FIFO descriptor address ( descriptor fields : READ ptr, WRITE ptr,
+ * FIFO START_ADDR, FIFO END_ADDR)
+ * @data: data to write to FIFO
+ * @number: number of 32-bit words to write to DMEM FIFO
+ *
+ * write DMEM FIFO and update FIFO descriptor,
+ * it is assumed that FIFO descriptor is located in DMEM
+ */
+void abe_write_fifo(u32 memory_bank, u32 descr_addr, u32 *data, u32 nb_data32)
+{
+ u32 fifo_addr[4];
+ u32 i;
+ /* read FIFO descriptor from DMEM */
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, descr_addr,
+ &fifo_addr[0], 4 * sizeof(u32));
+ /* WRITE ptr < FIFO start address */
+ if (fifo_addr[1] < fifo_addr[2])
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_DBG,
+ ABE_FW_FIFO_WRITE_PTR_ERR);
+ /* WRITE ptr > FIFO end address */
+ if (fifo_addr[1] > fifo_addr[3])
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_DBG,
+ ABE_FW_FIFO_WRITE_PTR_ERR);
+ switch (memory_bank) {
+ case ABE_DMEM:
+ for (i = 0; i < nb_data32; i++) {
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (s32) fifo_addr[1], (u32 *) (data + i),
+ 4);
+ /* increment WRITE pointer */
+ fifo_addr[1] = fifo_addr[1] + 4;
+ if (fifo_addr[1] > fifo_addr[3])
+ fifo_addr[1] = fifo_addr[2];
+ if (fifo_addr[1] == fifo_addr[0])
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_DBG,
+ ABE_FW_FIFO_WRITE_PTR_ERR);
+ }
+ /* update WRITE pointer in DMEM */
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, descr_addr +
+ sizeof(u32), &fifo_addr[1], 4);
+ break;
+ default:
+ break;
+ }
+}
+
+/**
+ * abe_write_asrc
+ * @id: name of the port
+ * @param: drift value to compensate [ppm]
+ *
+ * Load the drift variables to the FW memory. This API can be called only
+ * when the corresponding port has been already opened and the ASRC has
+ * been correctly initialized with API abe_init_asrc_... If this API is
+ * used such that the drift has been changed from positive to negative drift
+ * or vice versa, there will be click in the output signal. Loading the drift
+ * value with zero disables the feature.
+ */
+abehal_status abe_write_asrc(u32 port, s32 dppm)
+{
+ s32 dtempvalue, adppm, drift_sign, drift_sign_addr, alpha_params_addr;
+ s32 alpha_params[3];
+ _log(ABE_ID_WRITE_ASRC, port, dppm, dppm >> 8);
+ /*
+ * x = ppm
+ *
+ * - 1000000/x must be multiple of 16
+ * - deltaalpha = round(2^20*x*16/1000000)=round(2^18/5^6*x) on 22 bits.
+ * then shifted by 2bits
+ * - minusdeltaalpha
+ * - oneminusepsilon = 1-deltaalpha/2.
+ *
+ * ppm = 250
+ * - 1000000/250=4000
+ * - deltaalpha = 4194.3 ~ 4195 => 0x00418c
+ */
+ /* examples for -6250 ppm */
+ /* atempvalue32[1] = -1; d_driftsign */
+ /* atempvalue32[3] = 0x00066668; d_deltaalpha */
+ /* atempvalue32[4] = 0xfff99998; d_minusdeltaalpha */
+ /* atempvalue32[5] = 0x003ccccc; d_oneminusepsilon */
+ /* example for 100 ppm */
+ /* atempvalue32[1] = 1;* d_driftsign */
+ /* atempvalue32[3] = 0x00001a38; d_deltaalpha */
+ /* atempvalue32[4] = 0xffffe5c8; d_minusdeltaalpha */
+ /* atempvalue32[5] = 0x003ccccc; d_oneminusepsilon */
+ /* compute new value for the ppm */
+ if (dppm >= 0) {
+ /* d_driftsign */
+ drift_sign = 1;
+ adppm = dppm;
+ } else {
+ /* d_driftsign */
+ drift_sign = -1;
+ adppm = (-1 * dppm);
+ }
+ if (dppm == 0) {
+ /* delta_alpha */
+ alpha_params[0] = 0;
+ /* minusdelta_alpha */
+ alpha_params[1] = 0;
+ /* one_minusepsilon */
+ alpha_params[2] = 0x003ffff0;
+ } else {
+ dtempvalue = (adppm << 4) + adppm - ((adppm * 3481L) / 15625L);
+ /* delta_alpha */
+ alpha_params[0] = dtempvalue << 2;
+ /* minusdelta_alpha */
+ alpha_params[1] = (-dtempvalue) << 2;
+ /* one_minusepsilon */
+ alpha_params[2] = (0x00100000 - (dtempvalue / 2)) << 2;
+ }
+ switch (port) {
+ /* asynchronous sample-rate-converter for the uplink voice path */
+ case OMAP_ABE_VX_DL_PORT:
+ drift_sign_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (1 * sizeof(s32));
+ alpha_params_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (3 * sizeof(s32));
+ break;
+ /* asynchronous sample-rate-converter for the downlink voice path */
+ case OMAP_ABE_VX_UL_PORT:
+ drift_sign_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (1 * sizeof(s32));
+ alpha_params_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (3 * sizeof(s32));
+ break;
+ /* asynchronous sample-rate-converter for the BT_UL path */
+ case OMAP_ABE_BT_VX_UL_PORT:
+ drift_sign_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (1 * sizeof(s32));
+ alpha_params_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (3 * sizeof(s32));
+ break;
+ /* asynchronous sample-rate-converter for the BT_DL path */
+ case OMAP_ABE_BT_VX_DL_PORT:
+ drift_sign_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (1 * sizeof(s32));
+ alpha_params_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (3 * sizeof(s32));
+ break;
+ default:
+ /* asynchronous sample-rate-converter for the MM_EXT_IN path */
+ case OMAP_ABE_MM_EXT_IN_PORT:
+ drift_sign_addr = OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (1 * sizeof(s32));
+ alpha_params_addr =
+ OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (3 * sizeof(s32));
+ break;
+ }
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, drift_sign_addr,
+ (u32 *) &drift_sign, 4);
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, alpha_params_addr,
+ (u32 *) &alpha_params[0], 12);
+ return 0;
+}
+EXPORT_SYMBOL(abe_write_asrc);
+/**
+ * abe_init_asrc_vx_dl
+ *
+ * Initialize the following ASRC VX_DL parameters :
+ * 1. DriftSign = D_AsrcVars[1] = 1 or -1
+ * 2. Subblock = D_AsrcVars[2] = 0
+ * 3. DeltaAlpha = D_AsrcVars[3] =
+ * (round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 4. MinusDeltaAlpha = D_AsrcVars[4] =
+ * (-round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 5. OneMinusEpsilon = D_AsrcVars[5] = 1 - DeltaAlpha/2
+ * 6. AlphaCurrent = 0x000020 (CMEM), initial value of Alpha parameter
+ * 7. BetaCurrent = 0x3fffe0 (CMEM), initial value of Beta parameter
+ * AlphaCurrent + BetaCurrent = 1 (=0x400000 in CMEM = 2^20 << 2)
+ * 8. drift_ASRC = 0 & drift_io = 0
+ * 9. SMEM for ASRC_DL_VX_Coefs pointer
+ * 10. CMEM for ASRC_DL_VX_Coefs pointer
+ * ASRC_DL_VX_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ * C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1
+ * 11. SMEM for XinASRC_DL_VX pointer
+ * 12. CMEM for XinASRC_DL_VX pointer
+ * XinASRC_DL_VX = S_XinASRC_DL_VX_ADDR/S_XinASRC_DL_VX_sizeof/0/1/0/0/0/0
+ * 13. SMEM for IO_VX_DL_ASRC pointer
+ * 14. CMEM for IO_VX_DL_ASRC pointer
+ * IO_VX_DL_ASRC =
+ * S_XinASRC_DL_VX_ADDR/S_XinASRC_DL_VX_sizeof/
+ * ASRC_DL_VX_FIR_L+ASRC_margin/1/0/0/0/0
+ */
+void abe_init_asrc_vx_dl(s32 dppm)
+{
+ s32 el[45];
+ s32 temp0, temp1, adppm, dtemp, mem_tag, mem_addr;
+ u32 i = 0;
+ u32 n_fifo_el = 42;
+ temp0 = 0;
+ temp1 = 1;
+ /* 1. DriftSign = D_AsrcVars[1] = 1 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (1 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm >= 0) {
+ el[i + 1] = 1;
+ adppm = dppm;
+ } else {
+ el[i + 1] = -1;
+ adppm = (-1 * dppm);
+ }
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ dtemp = (adppm << 4) + adppm - ((adppm * 3481L) / 15625L);
+ /* 2. Subblock = D_AsrcVars[2] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (2 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 3. DeltaAlpha = D_AsrcVars[3] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (3 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = dtemp << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 4. MinusDeltaAlpha = D_AsrcVars[4] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (4 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = (-dtemp) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /*5. OneMinusEpsilon = D_AsrcVars[5] = 0x00400000 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_DL_VX_ADDR + (5 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0x00400000;
+ else
+ el[i + 1] = (0x00100000 - (dtemp / 2)) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 6. AlphaCurrent = 0x000020 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_ALPHACURRENT_DL_VX_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x00000020;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 7. BetaCurrent = 0x3fffe0 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_BETACURRENT_DL_VX_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x003fffe0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 8. drift_ASRC = 0 & drift_io = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_VX_DL_PORT * sizeof(struct ABE_SIODescriptor))
+ + drift_asrc_;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 9. SMEM for ASRC_DL_VX_Coefs pointer */
+ /* ASRC_DL_VX_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_SMEM;
+ mem_addr = ASRC_DL_VX_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ if (dppm == 0) {
+ el[i + 1] = OMAP_ABE_C_COEFASRC16_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC16_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC15_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC15_VX_SIZE >> 2);
+ } else {
+ el[i + 1] = OMAP_ABE_C_COEFASRC1_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC1_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC2_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC2_VX_SIZE >> 2);
+ }
+ i = i + 3;
+ /* 10. CMEM for ASRC_DL_VX_Coefs pointer */
+ /* ASRC_DL_VX_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_CMEM;
+ mem_addr = ASRC_DL_VX_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp1;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 11. SMEM for XinASRC_DL_VX pointer */
+ /* XinASRC_DL_VX =
+ S_XinASRC_DL_VX_ADDR/S_XinASRC_DL_VX_sizeof/0/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = XinASRC_DL_VX_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_DL_VX_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_DL_VX_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 12. CMEM for XinASRC_DL_VX pointer */
+ /* XinASRC_DL_VX =
+ S_XinASRC_DL_VX_ADDR/S_XinASRC_DL_VX_sizeof/0/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = XinASRC_DL_VX_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 13. SMEM for IO_VX_DL_ASRC pointer */
+ /* IO_VX_DL_ASRC = S_XinASRC_DL_VX_ADDR/S_XinASRC_DL_VX_sizeof/
+ ASRC_DL_VX_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = IO_VX_DL_ASRC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_DL_VX_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_DL_VX_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 14. CMEM for IO_VX_DL_ASRC pointer */
+ /* IO_VX_DL_ASRC = S_XinASRC_DL_VX_ADDR/S_XinASRC_DL_VX_sizeof/
+ ASRC_DL_VX_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = IO_VX_DL_ASRC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_DL_VX_FIR_L + ASRC_margin) << 16) + (temp1 << 12)
+ + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ abe_write_fifo(ABE_DMEM, OMAP_ABE_D_FWMEMINITDESCR_ADDR, (u32 *) &el[0],
+ n_fifo_el);
+}
+/**
+ * abe_init_asrc_vx_ul
+ *
+ * Initialize the following ASRC VX_UL parameters :
+ * 1. DriftSign = D_AsrcVars[1] = 1 or -1
+ * 2. Subblock = D_AsrcVars[2] = 0
+ * 3. DeltaAlpha = D_AsrcVars[3] =
+ * (round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 4. MinusDeltaAlpha = D_AsrcVars[4] =
+ * (-round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 5. OneMinusEpsilon = D_AsrcVars[5] = 1 - DeltaAlpha/2
+ * 6. AlphaCurrent = 0x000020 (CMEM), initial value of Alpha parameter
+ * 7. BetaCurrent = 0x3fffe0 (CMEM), initial value of Beta parameter
+ * AlphaCurrent + BetaCurrent = 1 (=0x400000 in CMEM = 2^20 << 2)
+ * 8. drift_ASRC = 0 & drift_io = 0
+ * 9. SMEM for ASRC_UL_VX_Coefs pointer
+ * 10. CMEM for ASRC_UL_VX_Coefs pointer
+ * ASRC_UL_VX_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ * C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1
+ * 11. SMEM for XinASRC_UL_VX pointer
+ * 12. CMEM for XinASRC_UL_VX pointer
+ * XinASRC_UL_VX = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/0/1/0/0/0/0
+ * 13. SMEM for UL_48_8_DEC pointer
+ * 14. CMEM for UL_48_8_DEC pointer
+ * UL_48_8_DEC = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/
+ * ASRC_UL_VX_FIR_L+ASRC_margin/1/0/0/0/0
+ * 15. SMEM for UL_48_16_DEC pointer
+ * 16. CMEM for UL_48_16_DEC pointer
+ * UL_48_16_DEC = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/
+ * ASRC_UL_VX_FIR_L+ASRC_margin/1/0/0/0/0
+ */
+void abe_init_asrc_vx_ul(s32 dppm)
+{
+ s32 el[51];
+ s32 temp0, temp1, adppm, dtemp, mem_tag, mem_addr;
+ u32 i = 0;
+ u32 n_fifo_el = 48;
+ temp0 = 0;
+ temp1 = 1;
+ /* 1. DriftSign = D_AsrcVars[1] = 1 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (1 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm >= 0) {
+ el[i + 1] = 1;
+ adppm = dppm;
+ } else {
+ el[i + 1] = -1;
+ adppm = (-1 * dppm);
+ }
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ dtemp = (adppm << 4) + adppm - ((adppm * 3481L) / 15625L);
+ /* 2. Subblock = D_AsrcVars[2] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (2 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 3. DeltaAlpha = D_AsrcVars[3] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (3 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = dtemp << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 4. MinusDeltaAlpha = D_AsrcVars[4] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (4 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = (-dtemp) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 5. OneMinusEpsilon = D_AsrcVars[5] = 0x00400000 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_UL_VX_ADDR + (5 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0x00400000;
+ else
+ el[i + 1] = (0x00100000 - (dtemp / 2)) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 6. AlphaCurrent = 0x000020 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_ALPHACURRENT_UL_VX_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x00000020;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 7. BetaCurrent = 0x3fffe0 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_BETACURRENT_UL_VX_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x003fffe0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 8. drift_ASRC = 0 & drift_io = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_VX_UL_PORT * sizeof(struct ABE_SIODescriptor))
+ + drift_asrc_;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 9. SMEM for ASRC_UL_VX_Coefs pointer */
+ /* ASRC_UL_VX_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_SMEM;
+ mem_addr = ASRC_UL_VX_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ if (dppm == 0) {
+ el[i + 1] = OMAP_ABE_C_COEFASRC16_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC16_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC15_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC15_VX_SIZE >> 2);
+ } else {
+ el[i + 1] = OMAP_ABE_C_COEFASRC1_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC1_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC2_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC2_VX_SIZE >> 2);
+ }
+ i = i + 3;
+ /* 10. CMEM for ASRC_UL_VX_Coefs pointer */
+ /* ASRC_UL_VX_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_CMEM;
+ mem_addr = ASRC_UL_VX_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp1;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 11. SMEM for XinASRC_UL_VX pointer */
+ /* XinASRC_UL_VX = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/0/1/
+ 0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = XinASRC_UL_VX_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_UL_VX_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_UL_VX_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 12. CMEM for XinASRC_UL_VX pointer */
+ /* XinASRC_UL_VX = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/0/1/
+ 0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = XinASRC_UL_VX_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 13. SMEM for UL_48_8_DEC pointer */
+ /* UL_48_8_DEC = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/
+ ASRC_UL_VX_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = UL_48_8_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_UL_VX_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_UL_VX_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 14. CMEM for UL_48_8_DEC pointer */
+ /* UL_48_8_DEC = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/
+ ASRC_UL_VX_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = UL_48_8_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_UL_VX_FIR_L + ASRC_margin) << 16) + (temp1 << 12)
+ + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 15. SMEM for UL_48_16_DEC pointer */
+ /* UL_48_16_DEC = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/
+ ASRC_UL_VX_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = UL_48_16_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_UL_VX_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_UL_VX_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 16. CMEM for UL_48_16_DEC pointer */
+ /* UL_48_16_DEC = S_XinASRC_UL_VX_ADDR/S_XinASRC_UL_VX_sizeof/
+ ASRC_UL_VX_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = UL_48_16_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_UL_VX_FIR_L + ASRC_margin) << 16) + (temp1 << 12)
+ + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ abe_write_fifo(ABE_DMEM, OMAP_ABE_D_FWMEMINITDESCR_ADDR, (u32 *) &el[0],
+ n_fifo_el);
+}
+/**
+ * abe_init_asrc_mm_ext_in
+ *
+ * Initialize the following ASRC MM_EXT_IN parameters :
+ * 1. DriftSign = D_AsrcVars[1] = 1 or -1
+ * 2. Subblock = D_AsrcVars[2] = 0
+ * 3. DeltaAlpha = D_AsrcVars[3] =
+ * (round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 4. MinusDeltaAlpha = D_AsrcVars[4] =
+ * (-round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 5. OneMinusEpsilon = D_AsrcVars[5] = 1 - DeltaAlpha/2
+ * 6. AlphaCurrent = 0x000020 (CMEM), initial value of Alpha parameter
+ * 7. BetaCurrent = 0x3fffe0 (CMEM), initial value of Beta parameter
+ * AlphaCurrent + BetaCurrent = 1 (=0x400000 in CMEM = 2^20 << 2)
+ * 8. drift_ASRC = 0 & drift_io = 0
+ * 9. SMEM for ASRC_MM_EXT_IN_Coefs pointer
+ * 10. CMEM for ASRC_MM_EXT_IN_Coefs pointer
+ * ASRC_MM_EXT_IN_Coefs = C_CoefASRC16_MM_ADDR/C_CoefASRC16_MM_sizeof/0/1/
+ * C_CoefASRC15_MM_ADDR/C_CoefASRC15_MM_sizeof/0/1
+ * 11. SMEM for XinASRC_MM_EXT_IN pointer
+ * 12. CMEM for XinASRC_MM_EXT_IN pointer
+ * XinASRC_MM_EXT_IN = S_XinASRC_MM_EXT_IN_ADDR/S_XinASRC_MM_EXT_IN_sizeof/0/1/
+ * 0/0/0/0
+ * 13. SMEM for IO_MM_EXT_IN_ASRC pointer
+ * 14. CMEM for IO_MM_EXT_IN_ASRC pointer
+ * IO_MM_EXT_IN_ASRC = S_XinASRC_MM_EXT_IN_ADDR/S_XinASRC_MM_EXT_IN_sizeof/
+ * ASRC_MM_EXT_IN_FIR_L+ASRC_margin+ASRC_N_48k/1/0/0/0/0
+ */
+void abe_init_asrc_mm_ext_in(s32 dppm)
+{
+ s32 el[45];
+ s32 temp0, temp1, adppm, dtemp, mem_tag, mem_addr;
+ u32 i = 0;
+ u32 n_fifo_el = 42;
+ temp0 = 0;
+ temp1 = 1;
+ /* 1. DriftSign = D_AsrcVars[1] = 1 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (1 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm >= 0) {
+ el[i + 1] = 1;
+ adppm = dppm;
+ } else {
+ el[i + 1] = -1;
+ adppm = (-1 * dppm);
+ }
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ dtemp = (adppm << 4) + adppm - ((adppm * 3481L) / 15625L);
+ /* 2. Subblock = D_AsrcVars[2] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (2 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 3. DeltaAlpha = D_AsrcVars[3] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (3 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = dtemp << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 4. MinusDeltaAlpha = D_AsrcVars[4] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (4 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = (-dtemp) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 5. OneMinusEpsilon = D_AsrcVars[5] = 0x00400000 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR + (5 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0x00400000;
+ else
+ el[i + 1] = (0x00100000 - (dtemp / 2)) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 6. AlphaCurrent = 0x000020 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_ALPHACURRENT_MM_EXT_IN_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x00000020;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 7. BetaCurrent = 0x3fffe0 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_BETACURRENT_MM_EXT_IN_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x003fffe0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 8. drift_ASRC = 0 & drift_io = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_MM_EXT_IN_PORT * sizeof(struct ABE_SIODescriptor))
+ + drift_asrc_;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 9. SMEM for ASRC_MM_EXT_IN_Coefs pointer */
+ /* ASRC_MM_EXT_IN_Coefs = C_CoefASRC16_MM_ADDR/C_CoefASRC16_MM_sizeof/
+ 0/1/C_CoefASRC15_MM_ADDR/C_CoefASRC15_MM_sizeof/0/1 */
+ mem_tag = ABE_SMEM;
+ mem_addr = ASRC_MM_EXT_IN_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ if (dppm == 0) {
+ el[i + 1] = OMAP_ABE_C_COEFASRC16_MM_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC16_MM_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC15_MM_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC15_MM_SIZE >> 2);
+ } else {
+ el[i + 1] = OMAP_ABE_C_COEFASRC1_MM_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC1_MM_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC2_MM_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC2_MM_SIZE >> 2);
+ }
+ i = i + 3;
+ /*10. CMEM for ASRC_MM_EXT_IN_Coefs pointer */
+ /* ASRC_MM_EXT_IN_Coefs = C_CoefASRC16_MM_ADDR/C_CoefASRC16_MM_sizeof/
+ 0/1/C_CoefASRC15_MM_ADDR/C_CoefASRC15_MM_sizeof/0/1 */
+ mem_tag = ABE_CMEM;
+ mem_addr = ASRC_MM_EXT_IN_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp1;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 11. SMEM for XinASRC_MM_EXT_IN pointer */
+ /* XinASRC_MM_EXT_IN = S_XinASRC_MM_EXT_IN_ADDR/
+ S_XinASRC_MM_EXT_IN_sizeof/0/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = XinASRC_MM_EXT_IN_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_MM_EXT_IN_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_MM_EXT_IN_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 12. CMEM for XinASRC_MM_EXT_IN pointer */
+ /* XinASRC_MM_EXT_IN = S_XinASRC_MM_EXT_IN_ADDR/
+ S_XinASRC_MM_EXT_IN_sizeof/0/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = XinASRC_MM_EXT_IN_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 13. SMEM for IO_MM_EXT_IN_ASRC pointer */
+ /* IO_MM_EXT_IN_ASRC =
+ S_XinASRC_MM_EXT_IN_ADDR/S_XinASRC_MM_EXT_IN_sizeof/
+ ASRC_MM_EXT_IN_FIR_L+ASRC_margin+ASRC_N_48k/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = IO_MM_EXT_IN_ASRC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_MM_EXT_IN_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_MM_EXT_IN_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 14. CMEM for IO_MM_EXT_IN_ASRC pointer */
+ /* IO_MM_EXT_IN_ASRC =
+ S_XinASRC_MM_EXT_IN_ADDR/S_XinASRC_MM_EXT_IN_sizeof/
+ ASRC_MM_EXT_IN_FIR_L+ASRC_margin+ASRC_N_48k/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = IO_MM_EXT_IN_ASRC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_MM_EXT_IN_FIR_L + ASRC_margin + ASRC_N_48k) << 16) +
+ (temp1 << 12) + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ abe_write_fifo(ABE_DMEM, OMAP_ABE_D_FWMEMINITDESCR_ADDR, (u32 *) &el[0],
+ n_fifo_el);
+}
+/**
+ * abe_init_asrc_bt_ul
+ *
+ * Initialize the following ASRC BT_UL parameters :
+ * 1. DriftSign = D_AsrcVars[1] = 1 or -1
+ * 2. Subblock = D_AsrcVars[2] = 0
+ * 3. DeltaAlpha = D_AsrcVars[3] =
+ * (round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 4. MinusDeltaAlpha = D_AsrcVars[4] =
+ * (-round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 5. OneMinusEpsilon = D_AsrcVars[5] = 1 - DeltaAlpha/2
+ * 6. AlphaCurrent = 0x000020 (CMEM), initial value of Alpha parameter
+ * 7. BetaCurrent = 0x3fffe0 (CMEM), initial value of Beta parameter
+ * AlphaCurrent + BetaCurrent = 1 (=0x400000 in CMEM = 2^20 << 2)
+ * 8. drift_ASRC = 0 & drift_io = 0
+ * 9. SMEM for ASRC_BT_UL_Coefs pointer
+ * 10. CMEM for ASRC_BT_UL_Coefs pointer
+ * ASRC_BT_UL_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ * C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1
+ * 11. SMEM for XinASRC_BT_UL pointer
+ * 12. CMEM for XinASRC_BT_UL pointer
+ * XinASRC_BT_UL = S_XinASRC_BT_UL_ADDR/S_XinASRC_BT_UL_sizeof/0/1/0/0/0/0
+ * 13. SMEM for IO_BT_UL_ASRC pointer
+ * 14. CMEM for IO_BT_UL_ASRC pointer
+ * IO_BT_UL_ASRC = S_XinASRC_BT_UL_ADDR/S_XinASRC_BT_UL_sizeof/
+ * ASRC_BT_UL_FIR_L+ASRC_margin/1/0/0/0/0
+ */
+void abe_init_asrc_bt_ul(s32 dppm)
+{
+ s32 el[45];
+ s32 temp0, temp1, adppm, dtemp, mem_tag, mem_addr;
+ u32 i = 0;
+ u32 n_fifo_el = 42;
+ temp0 = 0;
+ temp1 = 1;
+ /* 1. DriftSign = D_AsrcVars[1] = 1 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (1 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm >= 0) {
+ el[i + 1] = 1;
+ adppm = dppm;
+ } else {
+ el[i + 1] = -1;
+ adppm = (-1 * dppm);
+ }
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ dtemp = (adppm << 4) + adppm - ((adppm * 3481L) / 15625L);
+ /* 2. Subblock = D_AsrcVars[2] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (2 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 3. DeltaAlpha = D_AsrcVars[3] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (3 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = dtemp << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 4. MinusDeltaAlpha = D_AsrcVars[4] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (4 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = (-dtemp) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /*5. OneMinusEpsilon = D_AsrcVars[5] = 0x00400000 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_UL_ADDR + (5 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0x00400000;
+ else
+ el[i + 1] = (0x00100000 - (dtemp / 2)) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 6. AlphaCurrent = 0x000020 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_ALPHACURRENT_BT_UL_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x00000020;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 7. BetaCurrent = 0x3fffe0 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_BETACURRENT_BT_UL_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x003fffe0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 8. drift_ASRC = 0 & drift_io = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_BT_VX_UL_PORT * sizeof(struct ABE_SIODescriptor))
+ + drift_asrc_;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 9. SMEM for ASRC_BT_UL_Coefs pointer */
+ /* ASRC_BT_UL_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_SMEM;
+ mem_addr = ASRC_BT_UL_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ if (dppm == 0) {
+ el[i + 1] = OMAP_ABE_C_COEFASRC16_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC16_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC15_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC15_VX_SIZE >> 2);
+ } else {
+ el[i + 1] = OMAP_ABE_C_COEFASRC1_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC1_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC2_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC2_VX_SIZE >> 2);
+ }
+ i = i + 3;
+ /* 10. CMEM for ASRC_BT_UL_Coefs pointer */
+ /* ASRC_BT_UL_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_CMEM;
+ mem_addr = ASRC_BT_UL_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp1;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 11. SMEM for XinASRC_BT_UL pointer */
+ /* XinASRC_BT_UL = S_XinASRC_BT_UL_ADDR/S_XinASRC_BT_UL_sizeof/0/1/
+ 0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = XinASRC_BT_UL_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_BT_UL_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_BT_UL_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 12. CMEM for XinASRC_BT_UL pointer */
+ /* XinASRC_BT_UL = S_XinASRC_BT_UL_ADDR/S_XinASRC_BT_UL_sizeof/0/1/
+ 0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = XinASRC_BT_UL_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 13. SMEM for IO_BT_UL_ASRC pointer */
+ /* IO_BT_UL_ASRC = S_XinASRC_BT_UL_ADDR/S_XinASRC_BT_UL_sizeof/
+ ASRC_BT_UL_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = IO_BT_UL_ASRC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_BT_UL_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_BT_UL_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 14. CMEM for IO_BT_UL_ASRC pointer */
+ /* IO_BT_UL_ASRC = S_XinASRC_BT_UL_ADDR/S_XinASRC_BT_UL_sizeof/
+ ASRC_BT_UL_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = IO_BT_UL_ASRC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_BT_UL_FIR_L + ASRC_margin) << 16) + (temp1 << 12)
+ + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ abe_write_fifo(ABE_DMEM, OMAP_ABE_D_FWMEMINITDESCR_ADDR, (u32 *) &el[0],
+ n_fifo_el);
+}
+/**
+ * abe_init_asrc_bt_dl
+ *
+ * Initialize the following ASRC BT_DL parameters :
+ * 1. DriftSign = D_AsrcVars[1] = 1 or -1
+ * 2. Subblock = D_AsrcVars[2] = 0
+ * 3. DeltaAlpha = D_AsrcVars[3] =
+ * (round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 4. MinusDeltaAlpha = D_AsrcVars[4] =
+ * (-round(nb_phases * drift[ppm] * 10^-6 * 2^20)) << 2
+ * 5. OneMinusEpsilon = D_AsrcVars[5] = 1 - DeltaAlpha/2
+ * 6. AlphaCurrent = 0x000020 (CMEM), initial value of Alpha parameter
+ * 7. BetaCurrent = 0x3fffe0 (CMEM), initial value of Beta parameter
+ * AlphaCurrent + BetaCurrent = 1 (=0x400000 in CMEM = 2^20 << 2)
+ * 8. drift_ASRC = 0 & drift_io = 0
+ * 9. SMEM for ASRC_BT_DL_Coefs pointer
+ * 10. CMEM for ASRC_BT_DL_Coefs pointer
+ * ASRC_BT_DL_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ * C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1
+ * 11. SMEM for XinASRC_BT_DL pointer
+ * 12. CMEM for XinASRC_BT_DL pointer
+ * XinASRC_BT_DL = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/0/1/0/0/0/0
+ * 13. SMEM for DL_48_8_DEC pointer
+ * 14. CMEM for DL_48_8_DEC pointer
+ * DL_48_8_DEC = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/
+ * ASRC_BT_DL_FIR_L+ASRC_margin/1/0/0/0/0
+ * 15. SMEM for DL_48_16_DEC pointer
+ * 16. CMEM for DL_48_16_DEC pointer
+ * DL_48_16_DEC = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/
+ * ASRC_BT_DL_FIR_L+ASRC_margin/1/0/0/0/0
+ */
+void abe_init_asrc_bt_dl(s32 dppm)
+{
+ s32 el[51];
+ s32 temp0, temp1, adppm, dtemp, mem_tag, mem_addr;
+ u32 i = 0;
+ u32 n_fifo_el = 48;
+ temp0 = 0;
+ temp1 = 1;
+ /* 1. DriftSign = D_AsrcVars[1] = 1 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (1 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm >= 0) {
+ el[i + 1] = 1;
+ adppm = dppm;
+ } else {
+ el[i + 1] = -1;
+ adppm = (-1 * dppm);
+ }
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ dtemp = (adppm << 4) + adppm - ((adppm * 3481L) / 15625L);
+ /* 2. Subblock = D_AsrcVars[2] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (2 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 3. DeltaAlpha = D_AsrcVars[3] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (3 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = dtemp << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 4. MinusDeltaAlpha = D_AsrcVars[4] = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (4 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0;
+ else
+ el[i + 1] = (-dtemp) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 5. OneMinusEpsilon = D_AsrcVars[5] = 0x00400000 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_ASRCVARS_BT_DL_ADDR + (5 * sizeof(s32));
+ el[i] = (mem_tag << 16) + mem_addr;
+ if (dppm == 0)
+ el[i + 1] = 0x00400000;
+ else
+ el[i + 1] = (0x00100000 - (dtemp / 2)) << 2;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 6. AlphaCurrent = 0x000020 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_ALPHACURRENT_BT_DL_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x00000020;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 7. BetaCurrent = 0x3fffe0 (CMEM) */
+ mem_tag = ABE_CMEM;
+ mem_addr = OMAP_ABE_C_BETACURRENT_BT_DL_ADDR;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = 0x003fffe0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 8. drift_ASRC = 0 & drift_io = 0 */
+ mem_tag = ABE_DMEM;
+ mem_addr = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_BT_VX_DL_PORT * sizeof(struct ABE_SIODescriptor))
+ + drift_asrc_;
+ el[i] = (mem_tag << 16) + mem_addr;
+ el[i + 1] = temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 9. SMEM for ASRC_BT_DL_Coefs pointer */
+ /* ASRC_BT_DL_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_SMEM;
+ mem_addr = ASRC_BT_DL_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ if (dppm == 0) {
+ el[i + 1] = OMAP_ABE_C_COEFASRC16_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC16_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC15_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC15_VX_SIZE >> 2);
+ } else {
+ el[i + 1] = OMAP_ABE_C_COEFASRC1_VX_ADDR >> 2;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_C_COEFASRC1_VX_SIZE >> 2);
+ el[i + 2] = OMAP_ABE_C_COEFASRC2_VX_ADDR >> 2;
+ el[i + 2] = (el[i + 2] << 8) + (OMAP_ABE_C_COEFASRC2_VX_SIZE >> 2);
+ }
+ i = i + 3;
+ /* 10. CMEM for ASRC_BT_DL_Coefs pointer */
+ /* ASRC_BT_DL_Coefs = C_CoefASRC16_VX_ADDR/C_CoefASRC16_VX_sizeof/0/1/
+ C_CoefASRC15_VX_ADDR/C_CoefASRC15_VX_sizeof/0/1 */
+ mem_tag = ABE_CMEM;
+ mem_addr = ASRC_BT_DL_Coefs_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp1;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 11. SMEM for XinASRC_BT_DL pointer */
+ /* XinASRC_BT_DL =
+ S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/0/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = XinASRC_BT_DL_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_BT_DL_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_BT_DL_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 12. CMEM for XinASRC_BT_DL pointer */
+ /* XinASRC_BT_DL =
+ S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/0/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = XinASRC_BT_DL_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = (temp0 << 16) + (temp1 << 12) + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 13. SMEM for DL_48_8_DEC pointer */
+ /* DL_48_8_DEC = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/
+ ASRC_BT_DL_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = DL_48_8_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_BT_DL_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_BT_DL_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 14. CMEM for DL_48_8_DEC pointer */
+ /* DL_48_8_DEC = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/
+ ASRC_BT_DL_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = DL_48_8_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_BT_DL_FIR_L + ASRC_margin) << 16) + (temp1 << 12)
+ + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 15. SMEM for DL_48_16_DEC pointer */
+ /* DL_48_16_DEC = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/
+ ASRC_BT_DL_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_SMEM;
+ mem_addr = DL_48_16_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ el[i + 1] = OMAP_ABE_S_XINASRC_BT_DL_ADDR >> 3;
+ el[i + 1] = (el[i + 1] << 8) + (OMAP_ABE_S_XINASRC_BT_DL_SIZE >> 3);
+ el[i + 2] = temp0;
+ i = i + 3;
+ /* 16. CMEM for DL_48_16_DEC pointer */
+ /* DL_48_16_DEC = S_XinASRC_BT_DL_ADDR/S_XinASRC_BT_DL_sizeof/
+ ASRC_BT_DL_FIR_L+ASRC_margin/1/0/0/0/0 */
+ mem_tag = ABE_CMEM;
+ mem_addr = DL_48_16_DEC_labelID;
+ el[i] = (mem_tag << 16) + (mem_addr << 2);
+ /* el[i+1] = iam1<<16 + inc1<<12 + iam2<<4 + inc2 */
+ el[i + 1] = ((ASRC_BT_DL_FIR_L + ASRC_margin) << 16) + (temp1 << 12)
+ + (temp0 << 4) + temp0;
+ /* dummy field */
+ el[i + 2] = temp0;
+ abe_write_fifo(ABE_DMEM, OMAP_ABE_D_FWMEMINITDESCR_ADDR, (u32 *) &el[0],
+ n_fifo_el);
+}
diff --git a/sound/soc/omap/abe/abe_cm_addr.h b/sound/soc/omap/abe/abe_cm_addr.h
new file mode 100644
index 0000000..e5c97f3
--- /dev/null
+++ b/sound/soc/omap/abe/abe_cm_addr.h
@@ -0,0 +1,217 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#define OMAP_ABE_INIT_CM_ADDR 0x0
+#define OMAP_ABE_INIT_CM_SIZE 0x640
+#define OMAP_ABE_C_DATA_LSB_2_ADDR 0x640
+#define OMAP_ABE_C_DATA_LSB_2_SIZE 0x4
+#define OMAP_ABE_C_1_ALPHA_ADDR 0x644
+#define OMAP_ABE_C_1_ALPHA_SIZE 0x48
+#define OMAP_ABE_C_ALPHA_ADDR 0x68C
+#define OMAP_ABE_C_ALPHA_SIZE 0x48
+#define OMAP_ABE_C_GAINSWRAMP_ADDR 0x6D4
+#define OMAP_ABE_C_GAINSWRAMP_SIZE 0x38
+#define OMAP_ABE_C_GAINS_DL1M_ADDR 0x70C
+#define OMAP_ABE_C_GAINS_DL1M_SIZE 0x10
+#define OMAP_ABE_C_GAINS_DL2M_ADDR 0x71C
+#define OMAP_ABE_C_GAINS_DL2M_SIZE 0x10
+#define OMAP_ABE_C_GAINS_ECHOM_ADDR 0x72C
+#define OMAP_ABE_C_GAINS_ECHOM_SIZE 0x8
+#define OMAP_ABE_C_GAINS_SDTM_ADDR 0x734
+#define OMAP_ABE_C_GAINS_SDTM_SIZE 0x8
+#define OMAP_ABE_C_GAINS_VXRECM_ADDR 0x73C
+#define OMAP_ABE_C_GAINS_VXRECM_SIZE 0x10
+#define OMAP_ABE_C_GAINS_ULM_ADDR 0x74C
+#define OMAP_ABE_C_GAINS_ULM_SIZE 0x10
+#define OMAP_ABE_C_GAINS_BTUL_ADDR 0x75C
+#define OMAP_ABE_C_GAINS_BTUL_SIZE 0x8
+#define OMAP_ABE_C_SDT_COEFS_ADDR 0x764
+#define OMAP_ABE_C_SDT_COEFS_SIZE 0x24
+#define OMAP_ABE_C_COEFASRC1_VX_ADDR 0x788
+#define OMAP_ABE_C_COEFASRC1_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC2_VX_ADDR 0x7D4
+#define OMAP_ABE_C_COEFASRC2_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC3_VX_ADDR 0x820
+#define OMAP_ABE_C_COEFASRC3_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC4_VX_ADDR 0x86C
+#define OMAP_ABE_C_COEFASRC4_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC5_VX_ADDR 0x8B8
+#define OMAP_ABE_C_COEFASRC5_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC6_VX_ADDR 0x904
+#define OMAP_ABE_C_COEFASRC6_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC7_VX_ADDR 0x950
+#define OMAP_ABE_C_COEFASRC7_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC8_VX_ADDR 0x99C
+#define OMAP_ABE_C_COEFASRC8_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC9_VX_ADDR 0x9E8
+#define OMAP_ABE_C_COEFASRC9_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC10_VX_ADDR 0xA34
+#define OMAP_ABE_C_COEFASRC10_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC11_VX_ADDR 0xA80
+#define OMAP_ABE_C_COEFASRC11_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC12_VX_ADDR 0xACC
+#define OMAP_ABE_C_COEFASRC12_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC13_VX_ADDR 0xB18
+#define OMAP_ABE_C_COEFASRC13_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC14_VX_ADDR 0xB64
+#define OMAP_ABE_C_COEFASRC14_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC15_VX_ADDR 0xBB0
+#define OMAP_ABE_C_COEFASRC15_VX_SIZE 0x4C
+#define OMAP_ABE_C_COEFASRC16_VX_ADDR 0xBFC
+#define OMAP_ABE_C_COEFASRC16_VX_SIZE 0x4C
+#define OMAP_ABE_C_ALPHACURRENT_UL_VX_ADDR 0xC48
+#define OMAP_ABE_C_ALPHACURRENT_UL_VX_SIZE 0x4
+#define OMAP_ABE_C_BETACURRENT_UL_VX_ADDR 0xC4C
+#define OMAP_ABE_C_BETACURRENT_UL_VX_SIZE 0x4
+#define OMAP_ABE_C_ALPHACURRENT_DL_VX_ADDR 0xC50
+#define OMAP_ABE_C_ALPHACURRENT_DL_VX_SIZE 0x4
+#define OMAP_ABE_C_BETACURRENT_DL_VX_ADDR 0xC54
+#define OMAP_ABE_C_BETACURRENT_DL_VX_SIZE 0x4
+#define OMAP_ABE_C_COEFASRC1_MM_ADDR 0xC58
+#define OMAP_ABE_C_COEFASRC1_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC2_MM_ADDR 0xCA0
+#define OMAP_ABE_C_COEFASRC2_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC3_MM_ADDR 0xCE8
+#define OMAP_ABE_C_COEFASRC3_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC4_MM_ADDR 0xD30
+#define OMAP_ABE_C_COEFASRC4_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC5_MM_ADDR 0xD78
+#define OMAP_ABE_C_COEFASRC5_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC6_MM_ADDR 0xDC0
+#define OMAP_ABE_C_COEFASRC6_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC7_MM_ADDR 0xE08
+#define OMAP_ABE_C_COEFASRC7_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC8_MM_ADDR 0xE50
+#define OMAP_ABE_C_COEFASRC8_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC9_MM_ADDR 0xE98
+#define OMAP_ABE_C_COEFASRC9_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC10_MM_ADDR 0xEE0
+#define OMAP_ABE_C_COEFASRC10_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC11_MM_ADDR 0xF28
+#define OMAP_ABE_C_COEFASRC11_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC12_MM_ADDR 0xF70
+#define OMAP_ABE_C_COEFASRC12_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC13_MM_ADDR 0xFB8
+#define OMAP_ABE_C_COEFASRC13_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC14_MM_ADDR 0x1000
+#define OMAP_ABE_C_COEFASRC14_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC15_MM_ADDR 0x1048
+#define OMAP_ABE_C_COEFASRC15_MM_SIZE 0x48
+#define OMAP_ABE_C_COEFASRC16_MM_ADDR 0x1090
+#define OMAP_ABE_C_COEFASRC16_MM_SIZE 0x48
+#define OMAP_ABE_C_ALPHACURRENT_MM_EXT_IN_ADDR 0x10D8
+#define OMAP_ABE_C_ALPHACURRENT_MM_EXT_IN_SIZE 0x4
+#define OMAP_ABE_C_BETACURRENT_MM_EXT_IN_ADDR 0x10DC
+#define OMAP_ABE_C_BETACURRENT_MM_EXT_IN_SIZE 0x4
+#define OMAP_ABE_C_DL2_L_COEFS_ADDR 0x10E0
+#define OMAP_ABE_C_DL2_L_COEFS_SIZE 0x64
+#define OMAP_ABE_C_DL2_R_COEFS_ADDR 0x1144
+#define OMAP_ABE_C_DL2_R_COEFS_SIZE 0x64
+#define OMAP_ABE_C_DL1_COEFS_ADDR 0x11A8
+#define OMAP_ABE_C_DL1_COEFS_SIZE 0x64
+#define OMAP_ABE_C_SRC_3_LP_COEFS_ADDR 0x120C
+#define OMAP_ABE_C_SRC_3_LP_COEFS_SIZE 0x34
+#define OMAP_ABE_C_SRC_3_LP_GAIN_COEFS_ADDR 0x1240
+#define OMAP_ABE_C_SRC_3_LP_GAIN_COEFS_SIZE 0x34
+#define OMAP_ABE_C_SRC_3_HP_COEFS_ADDR 0x1274
+#define OMAP_ABE_C_SRC_3_HP_COEFS_SIZE 0x14
+#define OMAP_ABE_C_SRC_6_LP_COEFS_ADDR 0x1288
+#define OMAP_ABE_C_SRC_6_LP_COEFS_SIZE 0x34
+#define OMAP_ABE_C_SRC_6_LP_GAIN_COEFS_ADDR 0x12BC
+#define OMAP_ABE_C_SRC_6_LP_GAIN_COEFS_SIZE 0x34
+#define OMAP_ABE_C_SRC_6_HP_COEFS_ADDR 0x12F0
+#define OMAP_ABE_C_SRC_6_HP_COEFS_SIZE 0x1C
+#define OMAP_ABE_C_ALPHACURRENT_ECHO_REF_ADDR 0x130C
+#define OMAP_ABE_C_ALPHACURRENT_ECHO_REF_SIZE 0x4
+#define OMAP_ABE_C_BETACURRENT_ECHO_REF_ADDR 0x1310
+#define OMAP_ABE_C_BETACURRENT_ECHO_REF_SIZE 0x4
+#define OMAP_ABE_C_VIBRA2_CONSTS_ADDR 0x1314
+#define OMAP_ABE_C_VIBRA2_CONSTS_SIZE 0x10
+#define OMAP_ABE_C_VIBRA1_COEFFS_ADDR 0x1324
+#define OMAP_ABE_C_VIBRA1_COEFFS_SIZE 0x2C
+#define OMAP_ABE_C_48_96_LP_COEFS_ADDR 0x1350
+#define OMAP_ABE_C_48_96_LP_COEFS_SIZE 0x3C
+#define OMAP_ABE_C_96_48_AMIC_COEFS_ADDR 0x138C
+#define OMAP_ABE_C_96_48_AMIC_COEFS_SIZE 0x4C
+#define OMAP_ABE_C_96_48_DMIC_COEFS_ADDR 0x13D8
+#define OMAP_ABE_C_96_48_DMIC_COEFS_SIZE 0x4C
+#define OMAP_ABE_C_INPUT_SCALE_ADDR 0x1424
+#define OMAP_ABE_C_INPUT_SCALE_SIZE 0x4
+#define OMAP_ABE_C_OUTPUT_SCALE_ADDR 0x1428
+#define OMAP_ABE_C_OUTPUT_SCALE_SIZE 0x4
+#define OMAP_ABE_C_MUTE_SCALING_ADDR 0x142C
+#define OMAP_ABE_C_MUTE_SCALING_SIZE 0x4
+#define OMAP_ABE_C_GAINS_0DB_ADDR 0x1430
+#define OMAP_ABE_C_GAINS_0DB_SIZE 0x8
+#define OMAP_ABE_C_ALPHACURRENT_BT_DL_ADDR 0x1438
+#define OMAP_ABE_C_ALPHACURRENT_BT_DL_SIZE 0x4
+#define OMAP_ABE_C_BETACURRENT_BT_DL_ADDR 0x143C
+#define OMAP_ABE_C_BETACURRENT_BT_DL_SIZE 0x4
+#define OMAP_ABE_C_ALPHACURRENT_BT_UL_ADDR 0x1440
+#define OMAP_ABE_C_ALPHACURRENT_BT_UL_SIZE 0x4
+#define OMAP_ABE_C_BETACURRENT_BT_UL_ADDR 0x1444
+#define OMAP_ABE_C_BETACURRENT_BT_UL_SIZE 0x4
+#define OMAP_ABE_C_SRC_FIR6_LP_GAIN_COEFS_ADDR 0x1448
+#define OMAP_ABE_C_SRC_FIR6_LP_GAIN_COEFS_SIZE 0x2A0
+#define OMAP_ABE_C_SRC_44P1_COEFS_ADDR 0x16E8
+#define OMAP_ABE_C_SRC_44P1_COEFS_SIZE 0x480
+#define OMAP_ABE_C_SRC_MM_DL_44P1_STEP_ADDR 0x1B68
+#define OMAP_ABE_C_SRC_MM_DL_44P1_STEP_SIZE 0x8
+#define OMAP_ABE_C_SRC_TONES_44P1_STEP_ADDR 0x1B70
+#define OMAP_ABE_C_SRC_TONES_44P1_STEP_SIZE 0x8
+#define OMAP_ABE_C_SRC_44P1_MULFAC2_ADDR 0x1B78
+#define OMAP_ABE_C_SRC_44P1_MULFAC2_SIZE 0x8
diff --git a/sound/soc/omap/abe/abe_core.c b/sound/soc/omap/abe/abe_core.c
new file mode 100644
index 0000000..cdfbcea
--- /dev/null
+++ b/sound/soc/omap/abe/abe_core.c
@@ -0,0 +1,607 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_dbg.h"
+#include "abe.h"
+#include "abe_gain.h"
+#include "abe_aess.h"
+#include "abe_port.h"
+#include "abe_mem.h"
+#include "abe_taskid.h"
+
+#define OMAP_ABE_IRQ_FIFO_MASK ((OMAP_ABE_D_MCUIRQFIFO_SIZE >> 2) - 1)
+
+void abe_init_asrc_vx_dl(s32 dppm);
+void abe_init_asrc_vx_ul(s32 dppm);
+void abe_init_asrc_mm_ext_in(s32 dppm);
+void abe_init_asrc_bt_ul(s32 dppm);
+void abe_init_asrc_bt_dl(s32 dppm);
+
+void abe_irq_aps(u32 aps_info);
+void abe_irq_ping_pong(void);
+void abe_irq_check_for_sequences(u32 seq_info);
+extern u32 abe_size_pingpong;
+extern u32 abe_base_address_pingpong[];
+
+void abe_add_subroutine(u32 *id, abe_subroutine2 f, u32 nparam, u32 *params);
+
+
+/**
+ * abe_omap_abe_reset_hal - reset the ABE/HAL
+ * @abe: Pointer on abe handle
+ *
+ * Operations : reset the ABE by reloading the static variables and
+ * default AESS registers.
+ * Called after a PRCM cold-start reset of ABE
+ */
+int omap_abe_reset_hal(struct omap_abe *abe)
+{
+ u32 i;
+
+ omap_abe_dbg_reset(&abe->dbg);
+
+ _log(ABE_ID_RESET_HAL, 0, 0, 0);
+
+ /* IRQ & DBG circular read pointer in DMEM */
+ abe->irq_dbg_read_ptr = 0;
+
+ /* default = disable the mixer's adaptive gain control */
+ omap_abe_use_compensated_gain(abe, 0);
+
+ /* reset the default gain values */
+ for (i = 0; i < MAX_NBGAIN_CMEM; i++) {
+ abe->muted_gains_indicator[i] = 0;
+ abe->desired_gains_decibel[i] = (u32) GAIN_MUTE;
+ abe->desired_gains_linear[i] = 0;
+ abe->desired_ramp_delay_ms[i] = 0;
+ abe->muted_gains_decibel[i] = (u32) GAIN_TOOLOW;
+ }
+ omap_abe_hw_configuration(abe);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_reset_hal);
+
+/**
+ * omap_abe_wakeup - Wakeup ABE
+ * @abe: Pointer on abe handle
+ *
+ * Wakeup ABE in case of retention
+ */
+int omap_abe_wakeup(struct omap_abe *abe)
+{
+ /* Restart event generator */
+ omap_abe_write_event_generator(abe, EVENT_TIMER);
+
+ /* reconfigure DMA Req and MCU Irq visibility */
+ omap_abe_hw_configuration(abe);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_wakeup);
+
+/**
+ * abe_monitoring
+ *
+ * checks the internal status of ABE and HAL
+ */
+void abe_monitoring(void)
+{
+}
+
+/**
+ * omap_abe_irq_processing - Process ABE interrupt
+ * @abe: Pointer on abe handle
+ *
+ * This subroutine is call upon reception of "MA_IRQ_99 ABE_MPU_IRQ" Audio
+ * back-end interrupt. This subroutine will check the ATC Hrdware, the
+ * IRQ_FIFO from the AE and act accordingly. Some IRQ source are originated
+ * for the delivery of "end of time sequenced tasks" notifications, some are
+ * originated from the Ping-Pong protocols, some are generated from
+ * the embedded debugger when the firmware stops on programmable break-points,
+ * etc ...
+ */
+int omap_abe_irq_processing(struct omap_abe *abe)
+{
+ u32 abe_irq_dbg_write_ptr, i, cmem_src, sm_cm;
+ abe_irq_data_t IRQ_data;
+
+ _log(ABE_ID_IRQ_PROCESSING, 0, 0, 0);
+
+ /* extract the write pointer index from CMEM memory (INITPTR format) */
+ /* CMEM address of the write pointer in bytes */
+ cmem_src = MCU_IRQ_FIFO_ptr_labelID << 2;
+ omap_abe_mem_read(abe, OMAP_ABE_CMEM, cmem_src,
+ &sm_cm, sizeof(abe_irq_dbg_write_ptr));
+ /* AESS left-pointer index located on MSBs */
+ abe_irq_dbg_write_ptr = sm_cm >> 16;
+ abe_irq_dbg_write_ptr &= 0xFF;
+ /* loop on the IRQ FIFO content */
+ for (i = 0; i < OMAP_ABE_D_MCUIRQFIFO_SIZE; i++) {
+ /* stop when the FIFO is empty */
+ if (abe_irq_dbg_write_ptr == abe->irq_dbg_read_ptr)
+ break;
+ /* read the IRQ/DBG FIFO */
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ (OMAP_ABE_D_MCUIRQFIFO_ADDR +
+ (abe->irq_dbg_read_ptr << 2)),
+ (u32 *) &IRQ_data, sizeof(IRQ_data));
+ abe->irq_dbg_read_ptr = (abe->irq_dbg_read_ptr + 1) & OMAP_ABE_IRQ_FIFO_MASK;
+ /* select the source of the interrupt */
+ switch (IRQ_data.tag) {
+ case IRQtag_APS:
+ _log(ABE_ID_IRQ_PROCESSING, IRQ_data.data, 0, 1);
+ abe_irq_aps(IRQ_data.data);
+ break;
+ case IRQtag_PP:
+ _log(ABE_ID_IRQ_PROCESSING, 0, 0, 2);
+ abe_irq_ping_pong();
+ break;
+ case IRQtag_COUNT:
+ _log(ABE_ID_IRQ_PROCESSING, IRQ_data.data, 0, 3);
+ abe_irq_check_for_sequences(IRQ_data.data);
+ break;
+ default:
+ break;
+ }
+ }
+ abe_monitoring();
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_irq_processing);
+
+/**
+ * oamp_abe_set_ping_pong_buffer
+ * @abe: Pointer on abe handle
+ * @port: ABE port ID
+ * @n_bytes: Size of Ping/Pong buffer
+ *
+ * Updates the next ping-pong buffer with "size" bytes copied from the
+ * host processor. This API notifies the FW that the data transfer is done.
+ */
+int omap_abe_set_ping_pong_buffer(struct omap_abe *abe, u32 port, u32 n_bytes)
+{
+ u32 sio_pp_desc_address, struct_offset, n_samples, datasize,
+ base_and_size, *src;
+ struct ABE_SPingPongDescriptor desc_pp;
+
+ _log(ABE_ID_SET_PING_PONG_BUFFER, port, n_bytes, n_bytes >> 8);
+
+ /* ping_pong is only supported on MM_DL */
+ if (port != OMAP_ABE_MM_DL_PORT) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+ /* translates the number of bytes in samples */
+ /* data size in DMEM words */
+ datasize = omap_abe_dma_port_iter_factor((struct omap_abe_data_format *)&((abe_port[port]).format));
+ /* data size in bytes */
+ datasize = datasize << 2;
+ n_samples = n_bytes / datasize;
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, OMAP_ABE_D_PINGPONGDESC_ADDR,
+ (u32 *) &desc_pp, sizeof(desc_pp));
+ /*
+ * read the port SIO descriptor and extract the current pointer
+ * address after reading the counter
+ */
+ if ((desc_pp.counter & 0x1) == 0) {
+ struct_offset = (u32) &(desc_pp.nextbuff0_BaseAddr) -
+ (u32) &(desc_pp);
+ base_and_size = desc_pp.nextbuff0_BaseAddr;
+ } else {
+ struct_offset = (u32) &(desc_pp.nextbuff1_BaseAddr) -
+ (u32) &(desc_pp);
+ base_and_size = desc_pp.nextbuff1_BaseAddr;
+ }
+ base_and_size = (base_and_size & 0xFFFFL) + (n_samples << 16);
+ sio_pp_desc_address = OMAP_ABE_D_PINGPONGDESC_ADDR + struct_offset;
+ src = &base_and_size;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, sio_pp_desc_address,
+ (u32 *) &base_and_size, sizeof(u32));
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_set_ping_pong_buffer);
+
+/**
+ * omap_abe_read_next_ping_pong_buffer
+ * @abe: Pointer on abe handle
+ * @port: ABE portID
+ * @p: Next buffer address (pointer)
+ * @n: Next buffer size (pointer)
+ *
+ * Tell the next base address of the next ping_pong Buffer and its size
+ */
+int omap_abe_read_next_ping_pong_buffer(struct omap_abe *abe, u32 port, u32 *p, u32 *n)
+{
+ u32 sio_pp_desc_address;
+ struct ABE_SPingPongDescriptor desc_pp;
+
+ _log(ABE_ID_READ_NEXT_PING_PONG_BUFFER, port, 0, 0);
+
+ /* ping_pong is only supported on MM_DL */
+ if (port != OMAP_ABE_MM_DL_PORT) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+ /* read the port SIO descriptor and extract the current pointer
+ address after reading the counter */
+ sio_pp_desc_address = OMAP_ABE_D_PINGPONGDESC_ADDR;
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, sio_pp_desc_address,
+ (u32 *) &desc_pp, sizeof(struct ABE_SPingPongDescriptor));
+ if ((desc_pp.counter & 0x1) == 0) {
+ _log(ABE_ID_READ_NEXT_PING_PONG_BUFFER, port, 0, 0);
+ *p = desc_pp.nextbuff0_BaseAddr;
+ } else {
+ _log(ABE_ID_READ_NEXT_PING_PONG_BUFFER, port, 1, 0);
+ *p = desc_pp.nextbuff1_BaseAddr;
+ }
+ /* translates the number of samples in bytes */
+ *n = abe_size_pingpong;
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_read_next_ping_pong_buffer);
+
+/**
+ * omap_abe_init_ping_pong_buffer
+ * @abe: Pointer on abe handle
+ * @id: ABE port ID
+ * @size_bytes:size of the ping pong
+ * @n_buffers:number of buffers (2 = ping/pong)
+ * @p:returned address of the ping-pong list of base addresses
+ * (byte offset from DMEM start)
+ *
+ * Computes the base address of the ping_pong buffers
+ */
+int omap_abe_init_ping_pong_buffer(struct omap_abe *abe,
+ u32 id, u32 size_bytes, u32 n_buffers,
+ u32 *p)
+{
+ u32 i, dmem_addr;
+
+ _log(ABE_ID_INIT_PING_PONG_BUFFER, id, size_bytes, n_buffers);
+
+ /* ping_pong is supported in 2 buffers configuration right now but FW
+ is ready for ping/pong/pung/pang... */
+ if (id != OMAP_ABE_MM_DL_PORT || n_buffers > MAX_PINGPONG_BUFFERS) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+ for (i = 0; i < n_buffers; i++) {
+ dmem_addr = OMAP_ABE_D_PING_ADDR + (i * size_bytes);
+ /* base addresses of the ping pong buffers in U8 unit */
+ abe_base_address_pingpong[i] = dmem_addr;
+ }
+ /* global data */
+ abe_size_pingpong = size_bytes;
+ *p = (u32) OMAP_ABE_D_PING_ADDR;
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_init_ping_pong_buffer);
+
+/**
+ * omap_abe_read_offset_from_ping_buffer
+ * @abe: Pointer on abe handle
+ * @id: ABE port ID
+ * @n: returned address of the offset
+ * from the ping buffer start address (in samples)
+ *
+ * Computes the current firmware ping pong read pointer location,
+ * expressed in samples, as the offset from the start address of ping buffer.
+ */
+int omap_abe_read_offset_from_ping_buffer(struct omap_abe *abe,
+ u32 id, u32 *n)
+{
+ u32 sio_pp_desc_address;
+ struct ABE_SPingPongDescriptor desc_pp;
+
+ /* ping_pong is only supported on MM_DL */
+ if (OMAP_ABE_MM_DL_PORT != id) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ } else {
+ /* read the port SIO ping pong descriptor */
+ sio_pp_desc_address = OMAP_ABE_D_PINGPONGDESC_ADDR;
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ sio_pp_desc_address, (u32 *) &desc_pp,
+ sizeof(struct ABE_SPingPongDescriptor));
+ /* extract the current ping pong buffer read pointer based on
+ the value of the counter */
+ if ((desc_pp.counter & 0x1) == 0) {
+ /* the next is buffer0, hence the current is buffer1 */
+ switch (abe_port[OMAP_ABE_MM_DL_PORT].format.samp_format) {
+ case MONO_MSB:
+ case MONO_RSHIFTED_16:
+ case STEREO_16_16:
+ *n = abe_size_pingpong / 4 +
+ desc_pp.nextbuff1_Samples -
+ desc_pp.workbuff_Samples;
+ break;
+ case STEREO_MSB:
+ case STEREO_RSHIFTED_16:
+ *n = abe_size_pingpong / 8 +
+ desc_pp.nextbuff1_Samples -
+ desc_pp.workbuff_Samples;
+ break;
+ default:
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ break;
+ }
+ } else {
+ /* the next is buffer1, hence the current is buffer0 */
+ *n = desc_pp.nextbuff0_Samples -
+ desc_pp.workbuff_Samples;
+ }
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_read_offset_from_ping_buffer);
+
+/**
+ * abe_set_router_configuration
+ * @Id: name of the router
+ * @Conf: id of the configuration
+ * @param: list of output index of the route
+ *
+ * The uplink router takes its input from DMIC (6 samples), AMIC (2 samples)
+ * and PORT1/2 (2 stereo ports). Each sample will be individually stored in
+ * an intermediate table of 10 elements.
+ *
+ * Example of router table parameter for voice uplink with phoenix microphones
+ *
+ * indexes 0 .. 9 = MM_UL description (digital MICs and MMEXTIN)
+ * DMIC1_L_labelID, DMIC1_R_labelID, DMIC2_L_labelID, DMIC2_R_labelID,
+ * MM_EXT_IN_L_labelID, MM_EXT_IN_R_labelID, ZERO_labelID, ZERO_labelID,
+ * ZERO_labelID, ZERO_labelID,
+ * indexes 10 .. 11 = MM_UL2 description (recording on DMIC3)
+ * DMIC3_L_labelID, DMIC3_R_labelID,
+ * indexes 12 .. 13 = VX_UL description (VXUL based on PDMUL data)
+ * AMIC_L_labelID, AMIC_R_labelID,
+ * indexes 14 .. 15 = RESERVED (NULL)
+ * ZERO_labelID, ZERO_labelID,
+ */
+int omap_abe_set_router_configuration(struct omap_abe *abe,
+ u32 id, u32 k, u32 *param)
+{
+ _log(ABE_ID_SET_ROUTER_CONFIGURATION, id, (u32) param, (u32) param >> 8);
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_AUPLINKROUTING_ADDR,
+ param, OMAP_ABE_D_AUPLINKROUTING_SIZE);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_set_router_configuration);
+
+/**
+ * abe_set_opp_processing - Set OPP mode for ABE Firmware
+ * @opp: OOPP mode
+ *
+ * New processing network and OPP:
+ * 0: Ultra Lowest power consumption audio player (no post-processing, no mixer)
+ * 1: OPP 25% (simple multimedia features, including low-power player)
+ * 2: OPP 50% (multimedia and voice calls)
+ * 3: OPP100% ( multimedia complex use-cases)
+ *
+ * Rearranges the FW task network to the corresponding OPP list of features.
+ * The corresponding AE ports are supposed to be set/reset accordingly before
+ * this switch.
+ *
+ */
+int omap_abe_set_opp_processing(struct omap_abe *abe, u32 opp)
+{
+ u32 dOppMode32, sio_desc_address;
+ struct ABE_SIODescriptor sio_desc;
+
+ _log(ABE_ID_SET_OPP_PROCESSING, opp, 0, 0);
+
+ switch (opp) {
+ case ABE_OPP25:
+ /* OPP25% */
+ dOppMode32 = DOPPMODE32_OPP25;
+ break;
+ case ABE_OPP50:
+ /* OPP50% */
+ dOppMode32 = DOPPMODE32_OPP50;
+ break;
+ default:
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_BLOCK_COPY_ERR);
+ case ABE_OPP100:
+ /* OPP100% */
+ dOppMode32 = DOPPMODE32_OPP100;
+ break;
+ }
+ /* Write Multiframe inside DMEM */
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MAXTASKBYTESINSLOT_ADDR, &dOppMode32, sizeof(u32));
+
+ sio_desc_address = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_MM_EXT_IN_PORT *
+ sizeof(struct ABE_SIODescriptor));
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, sio_desc_address,
+ (u32 *) &sio_desc, sizeof(sio_desc));
+ if (dOppMode32 == DOPPMODE32_OPP100) {
+ /* ASRC input buffer, size 40 */
+ sio_desc.smem_addr1 = smem_mm_ext_in_opp100;
+ /* Init MM_EXT_IN ASRC and enable its adaptation */
+ abe_init_asrc_mm_ext_in(250);
+ } else
+ /* at OPP 50 or without ASRC */
+ sio_desc.smem_addr1 = smem_mm_ext_in_opp50;
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, sio_desc_address,
+ (u32 *) &sio_desc, sizeof(sio_desc));
+
+ sio_desc_address = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_BT_VX_UL_PORT *
+ sizeof(struct ABE_SIODescriptor));
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, sio_desc_address,
+ (u32 *) &sio_desc, sizeof(sio_desc));
+
+ if (abe_port[OMAP_ABE_BT_VX_UL_PORT].format.f == 8000) {
+ if (dOppMode32 == DOPPMODE32_OPP100)
+ /* ASRC input buffer, size 40 */
+ sio_desc.smem_addr1 = smem_bt_vx_ul_opp100;
+ else
+ /* at OPP 50 without ASRC */
+ sio_desc.smem_addr1 = BT_UL_8k_labelID;
+ } else {
+ if (dOppMode32 == DOPPMODE32_OPP100)
+ /* ASRC input buffer, size 40 */
+ sio_desc.smem_addr1 = smem_bt_vx_ul_opp100;
+ else
+ /* at OPP 50 without ASRC */
+ sio_desc.smem_addr1 = BT_UL_16k_labelID;
+ }
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, sio_desc_address,
+ (u32 *) &sio_desc, sizeof(sio_desc));
+
+ sio_desc_address = OMAP_ABE_D_IODESCR_ADDR + (OMAP_ABE_BT_VX_DL_PORT *
+ sizeof(struct ABE_SIODescriptor));
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, sio_desc_address,
+ (u32 *) &sio_desc, sizeof(sio_desc));
+
+#define ABE_TASK_ID(ID) (OMAP_ABE_D_TASKSLIST_ADDR + sizeof(ABE_STask)*(ID))
+#define TASK_BT_DL_48_8_SLT 14
+#define TASK_BT_DL_48_8_IDX 4
+ if (abe_port[OMAP_ABE_BT_VX_DL_PORT].format.f == 8000) {
+ if (dOppMode32 == DOPPMODE32_OPP100) {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_8_OPP100);
+ sio_desc.smem_addr1 = BT_DL_8k_opp100_labelID;
+ } else {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_8);
+ sio_desc.smem_addr1 = BT_DL_8k_labelID;
+ }
+ } else {
+ if (dOppMode32 == DOPPMODE32_OPP100) {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_16_OPP100);
+ sio_desc.smem_addr1 = BT_DL_16k_opp100_labelID;
+ } else {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_16);
+ sio_desc.smem_addr1 = BT_DL_16k_labelID;
+ }
+ }
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_MULTIFRAME_ADDR,
+ (u32 *) abe->MultiFrame, sizeof(abe->MultiFrame));
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, sio_desc_address,
+ (u32 *) &sio_desc, sizeof(sio_desc));
+
+ if (dOppMode32 == DOPPMODE32_OPP100) {
+ /* Init BT_VX_UL ASRC and enable its adaptation */
+ abe_init_asrc_bt_ul(250);
+ /* Init BT_VX_DL ASRC and enable its adaptation */
+ abe_init_asrc_bt_dl(-250);
+ }
+ return 0;
+
+}
+EXPORT_SYMBOL(omap_abe_set_opp_processing);
+
+/**
+ * omap_abe_check_activity - Check if some ABE activity.
+ *
+ * Check if any ABE ports are running.
+ * return 1: still activity on ABE
+ * return 0: no more activity on ABE. Event generator can be stopped
+ *
+ */
+int omap_abe_check_activity(struct omap_abe *abe)
+{
+ int i, ret = 0;
+
+ for (i = 0; i < (LAST_PORT_ID - 1); i++) {
+ if (abe_port[abe_port_priority[i]].status ==
+ OMAP_ABE_PORT_ACTIVITY_RUNNING)
+ break;
+ }
+ if (i < (LAST_PORT_ID - 1))
+ ret = 1;
+ return ret;
+}
+EXPORT_SYMBOL(omap_abe_check_activity);
+
+/**
+ * abe_plug_subroutine
+ * @id: returned sequence index after plugging a new subroutine
+ * @f: subroutine address to be inserted
+ * @n: number of parameters of this subroutine
+ * @params: pointer on parameters
+ *
+ * register a list of subroutines for call-back purpose
+ */
+abehal_status abe_plug_subroutine(u32 *id, abe_subroutine2 f, u32 n,
+ u32 *params)
+{
+ _log(ABE_ID_PLUG_SUBROUTINE, (u32) (*id), (u32) f, n);
+
+ abe_add_subroutine(id, (abe_subroutine2) f, n, (u32 *) params);
+ return 0;
+}
+EXPORT_SYMBOL(abe_plug_subroutine);
diff --git a/sound/soc/omap/abe/abe_dat.c b/sound/soc/omap/abe/abe_dat.c
new file mode 100644
index 0000000..62ec791
--- /dev/null
+++ b/sound/soc/omap/abe/abe_dat.c
@@ -0,0 +1,458 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include "abe_legacy.h"
+
+struct omap_abe *abe;
+
+/*
+ * HAL/FW ports status / format / sampling / protocol(call_back) / features
+ * / gain / name
+ */
+abe_port_t abe_port[LAST_PORT_ID]; /* list of ABE ports */
+const abe_port_t abe_port_init[LAST_PORT_ID] = {
+ /* Status Data Format Drift Call-Back Protocol+selector desc_addr;
+ buf_addr; buf_size; iter; irq_addr irq_data DMA_T $Features
+ reseted at start Port Name for the debug trace */
+ /* DMIC */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {96000, SIX_MSB},
+ NODRIFT, NOCALLBACK, 0, (DMIC_ITER/6),
+ {
+ SNK_P, DMIC_PORT_PROT,
+ {{dmem_dmic, dmem_dmic_size, DMIC_ITER} }
+ },
+ {0, 0},
+ {EQDMIC, 0}, "DMIC"},
+ /* PDM_UL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {96000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_amic, (MCPDM_UL_ITER/2),
+ {
+ SNK_P, MCPDMUL_PORT_PROT,
+ {{dmem_amic, dmem_amic_size, MCPDM_UL_ITER} }
+ },
+ {0, 0},
+ {EQAMIC, 0}, "PDM_UL"},
+ /* BT_VX_UL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {8000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_bt_vx_ul_opp50, 1,
+ {
+ SNK_P, SERIAL_PORT_PROT, {{
+ (MCBSP1_DMA_TX*ATC_SIZE),
+ dmem_bt_vx_ul,
+ dmem_bt_vx_ul_size,
+ (1*SCHED_LOOP_8kHz)
+ } }
+ },
+ {0, 0}, {0}, "BT_VX_UL"},
+ /* MM_UL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_mm_ul, 1,
+ {
+ SRC_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX3*ATC_SIZE),
+ dmem_mm_ul, dmem_mm_ul_size,
+ (10*SCHED_LOOP_48kHz),
+ ABE_DMASTATUS_RAW, (1 << 3)
+ } }
+ },
+ {CIRCULAR_BUFFER_PERIPHERAL_R__3, 120},
+ {UPROUTE, 0}, "MM_UL"},
+ /* MM_UL2 */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_mm_ul2, 1,
+ {
+ SRC_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX4*ATC_SIZE),
+ dmem_mm_ul2, dmem_mm_ul2_size,
+ (2*SCHED_LOOP_48kHz),
+ ABE_DMASTATUS_RAW, (1 << 4)
+ } }
+ },
+ {CIRCULAR_BUFFER_PERIPHERAL_R__4, 24},
+ {UPROUTE, 0}, "MM_UL2"},
+ /* VX_UL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {8000, MONO_MSB},
+ NODRIFT, NOCALLBACK, smem_vx_ul, 1,
+ {
+ SRC_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX2*ATC_SIZE),
+ dmem_vx_ul, dmem_vx_ul_size,
+ (1*SCHED_LOOP_8kHz),
+ ABE_DMASTATUS_RAW, (1 << 2)
+ } }
+ }, {
+ CIRCULAR_BUFFER_PERIPHERAL_R__2, 2},
+ {ASRC2, 0}, "VX_UL"},
+ /* MM_DL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_mm_dl, 1,
+ {
+ SNK_P, PINGPONG_PORT_PROT, {{
+ (CBPr_DMA_RTX0*ATC_SIZE),
+ dmem_mm_dl, dmem_mm_dl_size,
+ (2*SCHED_LOOP_48kHz),
+ ABE_DMASTATUS_RAW, (1 << 0)
+ } }
+ },
+ {CIRCULAR_BUFFER_PERIPHERAL_R__0, 24},
+ {ASRC3, 0}, "MM_DL"},
+ /* VX_DL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {8000, MONO_MSB},
+ NODRIFT, NOCALLBACK, smem_vx_dl, 1,
+ {
+ SNK_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX1*ATC_SIZE),
+ dmem_vx_dl, dmem_vx_dl_size,
+ (1*SCHED_LOOP_8kHz),
+ ABE_DMASTATUS_RAW, (1 << 1)
+ } }
+ },
+ {CIRCULAR_BUFFER_PERIPHERAL_R__1, 2},
+ {ASRC1, 0}, "VX_DL"},
+ /* TONES_DL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_tones_dl, 1,
+ {
+ SNK_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX5*ATC_SIZE),
+ dmem_tones_dl,
+ dmem_tones_dl_size,
+ (2*SCHED_LOOP_48kHz),
+ ABE_DMASTATUS_RAW, (1 << 5)
+ } }
+ },
+ {CIRCULAR_BUFFER_PERIPHERAL_R__5, 24},
+ {0}, "TONES_DL"},
+ /* VIB_DL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {24000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_vib, 1,
+ {
+ SNK_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX6*ATC_SIZE),
+ dmem_vib_dl, dmem_vib_dl_size,
+ (2*SCHED_LOOP_24kHz),
+ ABE_DMASTATUS_RAW, (1 << 6)
+ } }
+ },
+ {CIRCULAR_BUFFER_PERIPHERAL_R__6, 12},
+ {0}, "VIB_DL"},
+ /* BT_VX_DL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {8000, MONO_MSB},
+ NODRIFT, NOCALLBACK, smem_bt_vx_dl_opp50, 1,
+ {
+ SRC_P, SERIAL_PORT_PROT, {{
+ (MCBSP1_DMA_RX*ATC_SIZE),
+ dmem_bt_vx_dl,
+ dmem_bt_vx_dl_size,
+ (1*SCHED_LOOP_8kHz),
+ } }
+ },
+ {0, 0}, {0}, "BT_VX_DL"},
+ /* PDM_DL */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {96000, SIX_MSB},
+ NODRIFT, NOCALLBACK, 0, (MCPDM_DL_ITER/6),
+ {SRC_P, MCPDMDL_PORT_PROT, {{dmem_mcpdm,
+ dmem_mcpdm_size} } },
+ {0, 0},
+ {MIXDL1, EQ1, APS1, MIXDL2, EQ2L, EQ2R, APS2L, APS2R, 0},
+ "PDM_DL"},
+ /* MM_EXT_OUT */
+ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_mm_ext_out, 1,
+ {
+ SRC_P, SERIAL_PORT_PROT, {{
+ (MCBSP1_DMA_TX*ATC_SIZE),
+ dmem_mm_ext_out, dmem_mm_ext_out_size,
+ (2*SCHED_LOOP_48kHz)
+ } }
+ }, {0, 0}, {0}, "MM_EXT_OUT"},
+ /* MM_EXT_IN */
+ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, smem_mm_ext_in_opp100, 1,
+ {
+ SNK_P, SERIAL_PORT_PROT, {{
+ (MCBSP1_DMA_RX*ATC_SIZE),
+ dmem_mm_ext_in, dmem_mm_ext_in_size,
+ (2*SCHED_LOOP_48kHz)
+ } }
+ },
+ {0, 0}, {0}, "MM_EXT_IN"},
+ /* PCM3_TX */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, 0, 1,
+ {
+ SRC_P, TDM_SERIAL_PORT_PROT, {{
+ (MCBSP3_DMA_TX *
+ ATC_SIZE),
+ dmem_mm_ext_out,
+ dmem_mm_ext_out_size,
+ (2*SCHED_LOOP_48kHz)
+ } }
+ },
+ {0, 0}, {0}, "TDM_OUT"},
+ /* PCM3_RX */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, STEREO_MSB},
+ NODRIFT, NOCALLBACK, 0, 1,
+ {
+ SRC_P, TDM_SERIAL_PORT_PROT, {{
+ (MCBSP3_DMA_RX *
+ ATC_SIZE),
+ dmem_mm_ext_in,
+ dmem_mm_ext_in_size,
+ (2*SCHED_LOOP_48kHz)
+ } }
+ },
+ {0, 0}, {0}, "TDM_IN"},
+ /* SCHD_DBG_PORT */ {
+ OMAP_ABE_PORT_ACTIVITY_IDLE, {48000, MONO_MSB},
+ NODRIFT, NOCALLBACK, 0, 1,
+ {
+ SRC_P, DMAREQ_PORT_PROT, {{
+ (CBPr_DMA_RTX7 *
+ ATC_SIZE),
+ dmem_mm_trace,
+ dmem_mm_trace_size,
+ (2*SCHED_LOOP_48kHz),
+ ABE_DMASTATUS_RAW,
+ (1 << 4)
+ } }
+ }, {CIRCULAR_BUFFER_PERIPHERAL_R__7, 24},
+ {FEAT_SEQ, FEAT_CTL, FEAT_GAINS, 0}, "SCHD_DBG"},
+};
+/*
+ * AESS/ATC destination and source address translation (except McASPs)
+ * from the original 64bits words address
+ */
+const u32 abe_atc_dstid[ABE_ATC_DESC_SIZE >> 3] = {
+ /* DMA_0 DMIC PDM_DL PDM_UL McB1TX McB1RX McB2TX McB2RX 0 .. 7 */
+ 0, 0, 12, 0, 1, 0, 2, 0,
+ /* McB3TX McB3RX SLIMT0 SLIMT1 SLIMT2 SLIMT3 SLIMT4 SLIMT5 8 .. 15 */
+ 3, 0, 4, 5, 6, 7, 8, 9,
+ /* SLIMT6 SLIMT7 SLIMR0 SLIMR1 SLIMR2 SLIMR3 SLIMR4 SLIMR5 16 .. 23 */
+ 10, 11, 0, 0, 0, 0, 0, 0,
+ /* SLIMR6 SLIMR7 McASP1X ----- ----- McASP1R ----- ----- 24 .. 31 */
+ 0, 0, 14, 0, 0, 0, 0, 0,
+ /* CBPrT0 CBPrT1 CBPrT2 CBPrT3 CBPrT4 CBPrT5 CBPrT6 CBPrT7 32 .. 39 */
+ 63, 63, 63, 63, 63, 63, 63, 63,
+ /* CBP_T0 CBP_T1 CBP_T2 CBP_T3 CBP_T4 CBP_T5 CBP_T6 CBP_T7 40 .. 47 */
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ /* CBP_T8 CBP_T9 CBP_T10 CBP_T11 CBP_T12 CBP_T13 CBP_T14
+ CBP_T15 48 .. 63 */
+ 0, 0, 0, 0, 0, 0, 0, 0,
+};
+const u32 abe_atc_srcid[ABE_ATC_DESC_SIZE >> 3] = {
+ /* DMA_0 DMIC PDM_DL PDM_UL McB1TX McB1RX McB2TX McB2RX 0 .. 7 */
+ 0, 12, 0, 13, 0, 1, 0, 2,
+ /* McB3TX McB3RX SLIMT0 SLIMT1 SLIMT2 SLIMT3 SLIMT4 SLIMT5 8 .. 15 */
+ 0, 3, 0, 0, 0, 0, 0, 0,
+ /* SLIMT6 SLIMT7 SLIMR0 SLIMR1 SLIMR2 SLIMR3 SLIMR4 SLIMR5 16 .. 23 */
+ 0, 0, 4, 5, 6, 7, 8, 9,
+ /* SLIMR6 SLIMR7 McASP1X ----- ----- McASP1R ----- ----- 24 .. 31 */
+ 10, 11, 0, 0, 0, 14, 0, 0,
+ /* CBPrT0 CBPrT1 CBPrT2 CBPrT3 CBPrT4 CBPrT5 CBPrT6 CBPrT7 32 .. 39 */
+ 63, 63, 63, 63, 63, 63, 63, 63,
+ /* CBP_T0 CBP_T1 CBP_T2 CBP_T3 CBP_T4 CBP_T5 CBP_T6 CBP_T7 40 .. 47 */
+ 0, 0, 0, 0, 0, 0, 0, 0,
+ /* CBP_T8 CBP_T9 CBP_T10 CBP_T11 CBP_T12 CBP_T13 CBP_T14
+ CBP_T15 48 .. 63 */
+ 0, 0, 0, 0, 0, 0, 0, 0,
+};
+/*
+ * preset default routing configurations
+ * This is given as implementation EXAMPLES
+ * the programmer uses "abe_set_router_configuration" with its own tables
+ */
+const abe_router_t abe_router_ul_table_preset[NBROUTE_CONFIG][NBROUTE_UL] = {
+ /* VOICE UPLINK WITH PHOENIX MICROPHONES - UPROUTE_CONFIG_AMIC */
+ {
+ /* 0 .. 9 = MM_UL */
+ DMIC1_L_labelID, DMIC1_R_labelID, DMIC2_L_labelID, DMIC2_R_labelID,
+ MM_EXT_IN_L_labelID, MM_EXT_IN_R_labelID, AMIC_L_labelID,
+ AMIC_L_labelID,
+ ZERO_labelID, ZERO_labelID,
+ /* 10 .. 11 = MM_UL2 */
+ AMIC_L_labelID, AMIC_L_labelID,
+ /* 12 .. 13 = VX_UL */
+ AMIC_L_labelID, AMIC_R_labelID,
+ /* 14 .. 15 = RESERVED */
+ ZERO_labelID, ZERO_labelID,
+ },
+ /* VOICE UPLINK WITH THE FIRST DMIC PAIR - UPROUTE_CONFIG_DMIC1 */
+ {
+ /* 0 .. 9 = MM_UL */
+ DMIC2_L_labelID, DMIC2_R_labelID, DMIC3_L_labelID, DMIC3_R_labelID,
+ DMIC1_L_labelID, DMIC1_R_labelID, ZERO_labelID, ZERO_labelID,
+ ZERO_labelID, ZERO_labelID,
+ /* 10 .. 11 = MM_UL2 */
+ DMIC1_L_labelID, DMIC1_R_labelID,
+ /* 12 .. 13 = VX_UL */
+ DMIC1_L_labelID, DMIC1_R_labelID,
+ /* 14 .. 15 = RESERVED */
+ ZERO_labelID, ZERO_labelID,
+ },
+ /* VOICE UPLINK WITH THE SECOND DMIC PAIR - UPROUTE_CONFIG_DMIC2 */
+ {
+ /* 0 .. 9 = MM_UL */
+ DMIC3_L_labelID, DMIC3_R_labelID, DMIC1_L_labelID, DMIC1_R_labelID,
+ DMIC2_L_labelID, DMIC2_R_labelID, ZERO_labelID, ZERO_labelID,
+ ZERO_labelID, ZERO_labelID,
+ /* 10 .. 11 = MM_UL2 */
+ DMIC2_L_labelID, DMIC2_R_labelID,
+ /* 12 .. 13 = VX_UL */
+ DMIC2_L_labelID, DMIC2_R_labelID,
+ /* 14 .. 15 = RESERVED */
+ ZERO_labelID, ZERO_labelID,
+ },
+ /* VOICE UPLINK WITH THE LAST DMIC PAIR - UPROUTE_CONFIG_DMIC3 */
+ {
+ /* 0 .. 9 = MM_UL */
+ AMIC_L_labelID, AMIC_R_labelID, DMIC2_L_labelID, DMIC2_R_labelID,
+ DMIC3_L_labelID, DMIC3_R_labelID, ZERO_labelID, ZERO_labelID,
+ ZERO_labelID, ZERO_labelID,
+ /* 10 .. 11 = MM_UL2 */
+ DMIC3_L_labelID, DMIC3_R_labelID,
+ /* 12 .. 13 = VX_UL */
+ DMIC3_L_labelID, DMIC3_R_labelID,
+ /* 14 .. 15 = RESERVED */
+ ZERO_labelID, ZERO_labelID,
+ },
+ /* VOICE UPLINK WITH THE BT - UPROUTE_CONFIG_BT */
+ {
+ /* 0 .. 9 = MM_UL */
+ BT_UL_L_labelID, BT_UL_R_labelID, DMIC2_L_labelID, DMIC2_R_labelID,
+ DMIC3_L_labelID, DMIC3_R_labelID, DMIC1_L_labelID, DMIC1_R_labelID,
+ ZERO_labelID, ZERO_labelID,
+ /* 10 .. 11 = MM_UL2 */
+ AMIC_L_labelID, AMIC_R_labelID,
+ /* 12 .. 13 = VX_UL */
+ BT_UL_L_labelID, BT_UL_R_labelID,
+ /* 14 .. 15 = RESERVED */
+ ZERO_labelID, ZERO_labelID,
+ },
+ /* VOICE UPLINK WITH THE BT - UPROUTE_ECHO_MMUL2 */
+ {
+ /* 0 .. 9 = MM_UL */
+ MM_EXT_IN_L_labelID, MM_EXT_IN_R_labelID, BT_UL_L_labelID,
+ BT_UL_R_labelID, AMIC_L_labelID, AMIC_R_labelID,
+ ZERO_labelID, ZERO_labelID, ZERO_labelID, ZERO_labelID,
+ /* 10 .. 11 = MM_UL2 */
+ EchoRef_L_labelID, EchoRef_R_labelID,
+ /* 12 .. 13 = VX_UL */
+ AMIC_L_labelID, AMIC_L_labelID,
+ /* 14 .. 15 = RESERVED */
+ ZERO_labelID, ZERO_labelID,
+ },
+};
+/* all default routing configurations */
+abe_router_t abe_router_ul_table[NBROUTE_CONFIG_MAX][NBROUTE_UL];
+
+const abe_sequence_t seq_null = {
+ NOMASK, {CL_M1, 0, {0, 0, 0, 0}, 0}, {CL_M1, 0, {0, 0, 0, 0}, 0}
+};
+/* table of new subroutines called in the sequence */
+abe_subroutine2 abe_all_subsubroutine[MAXNBSUBROUTINE];
+/* number of parameters per calls */
+u32 abe_all_subsubroutine_nparam[MAXNBSUBROUTINE];
+/* index of the subroutine */
+u32 abe_subroutine_id[MAXNBSUBROUTINE];
+/* paramters of the subroutine (if any) */
+u32 *abe_all_subroutine_params[MAXNBSUBROUTINE];
+u32 abe_subroutine_write_pointer;
+/* table of all sequences */
+abe_sequence_t abe_all_sequence[MAXNBSEQUENCE];
+u32 abe_sequence_write_pointer;
+/* current number of pending sequences (avoids to look in the table) */
+u32 abe_nb_pending_sequences;
+/* pending sequences due to ressource collision */
+u32 abe_pending_sequences[MAXNBSEQUENCE];
+/* mask of unsharable ressources among other sequences */
+u32 abe_global_sequence_mask;
+/* table of active sequences */
+abe_seq_t abe_active_sequence[MAXACTIVESEQUENCE][MAXSEQUENCESTEPS];
+/* index of the plugged subroutine doing ping-pong cache-flush DMEM accesses */
+u32 abe_irq_pingpong_player_id;
+EXPORT_SYMBOL(abe_irq_pingpong_player_id);
+/* index of the plugged subroutine doing acoustics protection adaptation */
+u32 abe_irq_aps_adaptation_id;
+/* base addresses of the ping pong buffers in bytes addresses */
+u32 abe_base_address_pingpong[MAX_PINGPONG_BUFFERS];
+/* size of each ping/pong buffers */
+u32 abe_size_pingpong;
+/* number of ping/pong buffer being used */
+u32 abe_nb_pingpong;
+/*
+ * MAIN PORT SELECTION
+ */
+const u32 abe_port_priority[LAST_PORT_ID - 1] = {
+ OMAP_ABE_PDM_DL_PORT,
+ OMAP_ABE_PDM_UL_PORT,
+ OMAP_ABE_MM_EXT_OUT_PORT,
+ OMAP_ABE_MM_EXT_IN_PORT,
+ OMAP_ABE_DMIC_PORT,
+ OMAP_ABE_MM_UL_PORT,
+ OMAP_ABE_MM_UL2_PORT,
+ OMAP_ABE_MM_DL_PORT,
+ OMAP_ABE_TONES_DL_PORT,
+ OMAP_ABE_VX_UL_PORT,
+ OMAP_ABE_VX_DL_PORT,
+ OMAP_ABE_BT_VX_DL_PORT,
+ OMAP_ABE_BT_VX_UL_PORT,
+ OMAP_ABE_VIB_DL_PORT,
+};
diff --git a/sound/soc/omap/abe/abe_dbg.c b/sound/soc/omap/abe/abe_dbg.c
new file mode 100644
index 0000000..d1b160f
--- /dev/null
+++ b/sound/soc/omap/abe/abe_dbg.c
@@ -0,0 +1,201 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_dbg.h"
+#include "abe.h"
+#include "abe_mem.h"
+
+/**
+ * omap_abe_dbg_reset
+ * @dbg: Pointer on abe debug handle
+ *
+ * Called in order to reset Audio Back End debug global data.
+ * This ensures that ABE debug trace pointer is reset correctly.
+ */
+int omap_abe_dbg_reset(struct omap_abe_dbg *dbg)
+{
+ dbg->activity_log_write_pointer = 0;
+ dbg->mask = 0;
+
+ return 0;
+}
+
+/**
+ * omap_abe_connect_debug_trace
+ * @abe: Pointer on abe handle
+ * @dma2:pointer to the DMEM trace buffer
+ *
+ * returns the address and size of the real-time debug trace buffer,
+ * the content of which will vary from one firmware release to another
+ */
+int omap_abe_connect_debug_trace(struct omap_abe *abe,
+ struct omap_abe_dma *dma2)
+{
+ _log(ABE_ID_CONNECT_DEBUG_TRACE, 0, 0, 0);
+
+ /* return tohe base address of the ping buffer in L3 and L4 spaces */
+ (*dma2).data = (void *)(OMAP_ABE_D_DEBUG_FIFO_ADDR +
+ ABE_DEFAULT_BASE_ADDRESS_L3 + ABE_DMEM_BASE_OFFSET_MPU);
+ (*dma2).l3_dmem = (void *)(OMAP_ABE_D_DEBUG_FIFO_ADDR +
+ ABE_DEFAULT_BASE_ADDRESS_L3 + ABE_DMEM_BASE_OFFSET_MPU);
+ (*dma2).l4_dmem = (void *)(OMAP_ABE_D_DEBUG_FIFO_ADDR +
+ ABE_DEFAULT_BASE_ADDRESS_L4 + ABE_DMEM_BASE_OFFSET_MPU);
+ (*dma2).iter = (OMAP_ABE_D_DEBUG_FIFO_SIZE + OMAP_ABE_D_DEBUG_FIFO_HAL_SIZE)>>2;
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_connect_debug_trace);
+
+/**
+ * omap_abe_set_debug_trace
+ * @dbg: Pointer on abe debug handle
+ * @debug: debug log level
+ *
+ * Set the debug level for ABE trace. This level allows to manage the number
+ * of information put inside the ABE trace buffer. This buffer can contains
+ * both AESS firmware and MPU traces.
+ */
+int omap_abe_set_debug_trace(struct omap_abe_dbg *dbg, int debug)
+{
+ _log(ABE_ID_SET_DEBUG_TRACE, 0, 0, 0);
+
+ dbg->mask = debug;
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_set_debug_trace);
+
+/**
+ * omap_abe_dbg_log - Log ABE trace inside circular buffer
+ * @x: data to be logged
+ * @y: data to be logged
+ * @z: data to be logged
+ * @t: data to be logged
+ * Parameter :
+ *
+ * abe_dbg_activity_log : global circular buffer holding the data
+ * abe_dbg_activity_log_write_pointer : circular write pointer
+ *
+ * saves data in the log file
+ */
+void omap_abe_dbg_log(struct omap_abe *abe, u32 x, u32 y, u32 z, u32 t)
+{
+ u32 time_stamp, data;
+ struct omap_abe_dbg *dbg = &abe->dbg;
+
+ if (dbg->activity_log_write_pointer >=
+ (OMAP_ABE_D_DEBUG_HAL_TASK_SIZE - 2))
+ dbg->activity_log_write_pointer = 0;
+
+ /* copy in DMEM trace buffer and CortexA9 local buffer and a small 7
+ words circular buffer of the DMA trace ending with 0x55555555
+ (tag for last word) */
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, OMAP_ABE_D_LOOPCOUNTER_ADDR,
+ (u32 *) &time_stamp, sizeof(time_stamp));
+ dbg->activity_log[dbg->activity_log_write_pointer] = time_stamp;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_DEBUG_HAL_TASK_ADDR +
+ (dbg->activity_log_write_pointer << 2),
+ (u32 *) &time_stamp, sizeof(time_stamp));
+ dbg->activity_log_write_pointer++;
+
+ data = ((x & MAX_UINT8) << 24) | ((y & MAX_UINT8) << 16) |
+ ((z & MAX_UINT8) << 8)
+ | (t & MAX_UINT8);
+ dbg->activity_log[dbg->activity_log_write_pointer] = data;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_DEBUG_HAL_TASK_ADDR +
+ (dbg->activity_log_write_pointer << 2),
+ (u32 *) &data, sizeof(data));
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_DEBUG_FIFO_HAL_ADDR +
+ ((dbg->activity_log_write_pointer << 2) &
+ (OMAP_ABE_D_DEBUG_FIFO_HAL_SIZE - 1)), (u32 *) &data,
+ sizeof(data));
+
+ data = ABE_DBG_MAGIC_NUMBER;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_DEBUG_FIFO_HAL_ADDR +
+ (((dbg->activity_log_write_pointer + 1) << 2) &
+ (OMAP_ABE_D_DEBUG_FIFO_HAL_SIZE - 1)),
+ (u32 *) &data, sizeof(data));
+ dbg->activity_log_write_pointer++;
+
+ if (dbg->activity_log_write_pointer >= OMAP_ABE_D_DEBUG_HAL_TASK_SIZE)
+ dbg->activity_log_write_pointer = 0;
+}
+
+/**
+ * omap_abe_dbg_error_log - Log ABE error
+ * @abe: Pointer on abe handle
+ * @level: level of error
+ * @error: error ID to log
+ *
+ * Log the ABE errors.
+ */
+void omap_abe_dbg_error(struct omap_abe *abe, int level, int error)
+{
+ omap_abe_dbg_log(abe, error, MAX_UINT8, MAX_UINT8, MAX_UINT8);
+}
diff --git a/sound/soc/omap/abe/abe_dbg.h b/sound/soc/omap/abe/abe_dbg.h
new file mode 100644
index 0000000..2cdced9
--- /dev/null
+++ b/sound/soc/omap/abe/abe_dbg.h
@@ -0,0 +1,206 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_DBG_H_
+#define _ABE_DBG_H_
+
+#include "abe_typ.h"
+#include "abe_dm_addr.h"
+
+/*
+ * Debug trace format
+ * TIME 2 bytes from ABE : 4kHz period of the FW scheduler
+ * SUBID 1 byte : HAL API index
+ * From 0 to 16 bytes : parameters of the subroutine
+ * on every 32 dumps a tag is pushed on the debug trace : 0x55555555
+ */
+#define dbg_bitfield_offset 8
+#define dbg_api_calls 0
+#define dbg_mapi (1L << (dbg_api_calls + dbg_bitfield_offset))
+#define dbg_external_data_access 1
+#define dbg_mdata (1L << (dbg_external_data_access + dbg_bitfield_offset))
+#define dbg_err_codes 2
+#define dbg_merr (1L << (dbg_api_calls + dbg_bitfield_offset))
+#define ABE_DBG_MAGIC_NUMBER 0x55555555
+/*
+ * IDs used for traces
+ */
+#define ABE_ID_RESET_HAL (1 + dbg_mapi)
+#define ABE_ID_LOAD_FW (2 + dbg_mapi)
+#define ABE_ID_DEFAULT_CONFIGURATION (3 + dbg_mapi)
+#define ABE_ID_IRQ_PROCESSING (4 + dbg_mapi)
+#define ABE_ID_EVENT_GENERATOR_SWITCH (5 + dbg_mapi)
+#define ABE_ID_READ_HARDWARE_CONFIGURATION (6 + dbg_mapi)
+#define ABE_ID_READ_LOWEST_OPP (7 + dbg_mapi)
+#define ABE_ID_WRITE_GAIN (8 + dbg_mapi)
+#define ABE_ID_SET_ASRC_DRIFT_CONTROL (9 + dbg_mapi)
+#define ABE_ID_PLUG_SUBROUTINE (10 + dbg_mapi)
+#define ABE_ID_UNPLUG_SUBROUTINE (11 + dbg_mapi)
+#define ABE_ID_PLUG_SEQUENCE (12 + dbg_mapi)
+#define ABE_ID_LAUNCH_SEQUENCE (13 + dbg_mapi)
+#define ABE_ID_LAUNCH_SEQUENCE_param (14 + dbg_mapi)
+#define ABE_ID_CONNECT_IRQ_PING_PONG_PORT (15 + dbg_mapi)
+#define ABE_ID_READ_ANALOG_GAIN_DL (16 + dbg_mapi)
+#define ABE_ID_READ_ANALOG_GAIN_UL (17 + dbg_mapi)
+#define ABE_ID_ENABLE_DYN_UL_GAIN (18 + dbg_mapi)
+#define ABE_ID_DISABLE_DYN_UL_GAIN (19 + dbg_mapi)
+#define ABE_ID_ENABLE_DYN_EXTENSION (20 + dbg_mapi)
+#define ABE_ID_DISABLE_DYN_EXTENSION (21 + dbg_mapi)
+#define ABE_ID_NOTIFY_ANALOG_GAIN_CHANGED (22 + dbg_mapi)
+#define ABE_ID_RESET_PORT (23 + dbg_mapi)
+#define ABE_ID_READ_REMAINING_DATA (24 + dbg_mapi)
+#define ABE_ID_DISABLE_DATA_TRANSFER (25 + dbg_mapi)
+#define ABE_ID_ENABLE_DATA_TRANSFER (26 + dbg_mapi)
+#define ABE_ID_READ_GLOBAL_COUNTER (27 + dbg_mapi)
+#define ABE_ID_SET_DMIC_FILTER (28 + dbg_mapi)
+#define ABE_ID_SET_OPP_PROCESSING (29 + dbg_mapi)
+#define ABE_ID_SET_PING_PONG_BUFFER (30 + dbg_mapi)
+#define ABE_ID_READ_PORT_ADDRESS (31 + dbg_mapi)
+#define ABE_ID_LOAD_FW_param (32 + dbg_mapi)
+#define ABE_ID_WRITE_HEADSET_OFFSET (33 + dbg_mapi)
+#define ABE_ID_READ_GAIN_RANGES (34 + dbg_mapi)
+#define ABE_ID_WRITE_EQUALIZER (35 + dbg_mapi)
+#define ABE_ID_WRITE_ASRC (36 + dbg_mapi)
+#define ABE_ID_WRITE_APS (37 + dbg_mapi)
+#define ABE_ID_WRITE_MIXER (38 + dbg_mapi)
+#define ABE_ID_WRITE_EANC (39 + dbg_mapi)
+#define ABE_ID_WRITE_ROUTER (40 + dbg_mapi)
+#define ABE_ID_READ_PORT_GAIN (41 + dbg_mapi)
+#define ABE_ID_ASRC (42 + dbg_mapi)
+#define ABE_ID_READ_APS (43 + dbg_mapi)
+#define ABE_ID_READ_APS_energy (44 + dbg_mapi)
+#define ABE_ID_READ_MIXER (45 + dbg_mapi)
+#define ABE_READ_EANC (46 + dbg_mapi)
+#define ABE_ID_READ_ROUTER (47 + dbg_mapi)
+#define ABE_ID_READ_DEBUG_TRACE (48 + dbg_mapi)
+#define ABE_ID_SET_SEQUENCE_TIME_ACCURACY (49 + dbg_mapi)
+#define ABE_ID_SET_DEBUG_PINS (50 + dbg_mapi)
+#define ABE_ID_SELECT_MAIN_PORT (51 + dbg_mapi)
+#define ABE_ID_WRITE_EVENT_GENERATOR (52 + dbg_mapi)
+#define ABE_ID_READ_USE_CASE_OPP (53 + dbg_mapi)
+#define ABE_ID_SELECT_DATA_SOURCE (54 + dbg_mapi)
+#define ABE_ID_READ_NEXT_PING_PONG_BUFFER (55 + dbg_mapi)
+#define ABE_ID_INIT_PING_PONG_BUFFER (56 + dbg_mapi)
+#define ABE_ID_CONNECT_CBPR_DMAREQ_PORT (57 + dbg_mapi)
+#define ABE_ID_CONNECT_DMAREQ_PORT (58 + dbg_mapi)
+#define ABE_ID_CONNECT_DMAREQ_PING_PONG_PORT (59 + dbg_mapi)
+#define ABE_ID_CONNECT_SERIAL_PORT (60 + dbg_mapi)
+#define ABE_ID_CONNECT_SLIMBUS_PORT (61 + dbg_mapi)
+#define ABE_ID_READ_GAIN (62 + dbg_mapi)
+#define ABE_ID_SET_ROUTER_CONFIGURATION (63 + dbg_mapi)
+#define ABE_ID_CONNECT_DEBUG_TRACE (64 + dbg_mapi)
+#define ABE_ID_SET_DEBUG_TRACE (65 + dbg_mapi)
+#define ABE_ID_REMOTE_DEBUGGER_INTERFACE (66 + dbg_mapi)
+#define ABE_ID_ENABLE_TEST_PATTERN (67 + dbg_mapi)
+#define ABE_ID_CONNECT_TDM_PORT (68 + dbg_mapi)
+/*
+ * IDs used for error codes
+ */
+#define NOERR 0
+#define ABE_SET_MEMORY_CONFIG_ERR (1 + dbg_merr)
+#define ABE_BLOCK_COPY_ERR (2 + dbg_merr)
+#define ABE_SEQTOOLONG (3 + dbg_merr)
+#define ABE_BADSAMPFORMAT (4 + dbg_merr)
+#define ABE_SET_ATC_ABE_BLOCK_COPY_ERR MEMORY_CONFIG_ERR (5 + dbg_merr)
+#define ABE_PROTOCOL_ERROR (6 + dbg_merr)
+#define ABE_PARAMETER_ERROR (7 + dbg_merr)
+/* port programmed while still running */
+#define ABE_PORT_REPROGRAMMING (8 + dbg_merr)
+#define ABE_READ_USE_CASE_OPP_ERR (9 + dbg_merr)
+#define ABE_PARAMETER_OVERFLOW (10 + dbg_merr)
+#define ABE_FW_FIFO_WRITE_PTR_ERR (11 + dbg_merr)
+
+/*
+ * IDs used for error codes
+ */
+#define OMAP_ABE_ERR_LIB (1 << 1)
+#define OMAP_ABE_ERR_API (1 << 2)
+#define OMAP_ABE_ERR_INI (1 << 3)
+#define OMAP_ABE_ERR_SEQ (1 << 4)
+#define OMAP_ABE_ERR_DBG (1 << 5)
+#define OMAP_ABE_ERR_EXT (1 << 6)
+
+struct omap_abe_dbg {
+ /* Debug Data */
+ u32 activity_log[OMAP_ABE_D_DEBUG_HAL_TASK_SIZE];
+ u32 activity_log_write_pointer;
+ u32 mask;
+};
+
+struct omap_abe_dma {
+ /* OCP L3 pointer to the first address of the */
+ void *data;
+ /* destination buffer (either DMA or Ping-Pong read/write pointers). */
+ /* address L3 when addressing the DMEM buffer instead of CBPr */
+ void *l3_dmem;
+ /* address L3 translated to L4 the ARM memory space */
+ void *l4_dmem;
+ /* number of iterations for the DMA data moves. */
+ u32 iter;
+};
+
+/**
+ * omap_abe_dbg_reset
+ * @dbg: Pointer on abe debug handle
+ *
+ * Called in order to reset Audio Back End debug global data.
+ * This ensures that ABE debug trace pointer is reset correctly.
+ */
+int omap_abe_dbg_reset(struct omap_abe_dbg *dbg);
+
+#endif /* _ABE_DBG_H_ */
diff --git a/sound/soc/omap/abe/abe_def.h b/sound/soc/omap/abe/abe_def.h
new file mode 100644
index 0000000..ac1d263
--- /dev/null
+++ b/sound/soc/omap/abe/abe_def.h
@@ -0,0 +1,307 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_DEF_H_
+#define _ABE_DEF_H_
+/*
+ * HARDWARE AND PERIPHERAL DEFINITIONS
+ */
+/* MM_DL */
+#define ABE_CBPR0_IDX 0
+/* VX_DL */
+#define ABE_CBPR1_IDX 1
+/* VX_UL */
+#define ABE_CBPR2_IDX 2
+/* MM_UL */
+#define ABE_CBPR3_IDX 3
+/* MM_UL2 */
+#define ABE_CBPR4_IDX 4
+/* TONES */
+#define ABE_CBPR5_IDX 5
+/* VIB */
+#define ABE_CBPR6_IDX 6
+/* DEBUG/CTL */
+#define ABE_CBPR7_IDX 7
+#define CIRCULAR_BUFFER_PERIPHERAL_R__0 (0x100 + ABE_CBPR0_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__1 (0x100 + ABE_CBPR1_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__2 (0x100 + ABE_CBPR2_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__3 (0x100 + ABE_CBPR3_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__4 (0x100 + ABE_CBPR4_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__5 (0x100 + ABE_CBPR5_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__6 (0x100 + ABE_CBPR6_IDX*4)
+#define CIRCULAR_BUFFER_PERIPHERAL_R__7 (0x100 + ABE_CBPR7_IDX*4)
+#define PING_PONG_WITH_MCU_IRQ 1
+#define PING_PONG_WITH_DSP_IRQ 2
+/* ID used for LIB memory copy subroutines */
+#define COPY_FROM_ABE_TO_HOST 1
+#define COPY_FROM_HOST_TO_ABE 2
+/*
+ * INTERNAL DEFINITIONS
+ */
+#define ABE_FIRMWARE_MAX_SIZE 26629
+/* 24 Q6.26 coefficients */
+#define NBEQ1 25
+/* 2x12 Q6.26 coefficients */
+#define NBEQ2 13
+/* TBD APS first set of parameters */
+#define NBAPS1 10
+/* TBD APS second set of parameters */
+#define NBAPS2 10
+/* Mixer used for sending tones to the uplink voice path */
+#define NBMIX_AUDIO_UL 2
+/* Main downlink mixer */
+#define NBMIX_DL1 4
+/* Handsfree downlink mixer */
+#define NBMIX_DL2 4
+/* Side-tone mixer */
+#define NBMIX_SDT 2
+/* Echo reference mixer */
+#define NBMIX_ECHO 2
+/* Voice record mixer */
+#define NBMIX_VXREC 4
+/* unsigned version of (-1) */
+#define CC_M1 0xFF
+#define CS_M1 0xFFFF
+#define CL_M1 0xFFFFFFFFL
+/*
+ Mixer ID Input port ID Comments
+ DL1_MIXER 0 MMDL path
+ 1 MMUL2 path
+ 2 VXDL path
+ 3 TONES path
+ SDT_MIXER 0 Uplink path
+ 1 Downlink path
+ ECHO_MIXER 0 DL1_MIXER path
+ 1 DL2_MIXER path
+ AUDUL_MIXER 0 TONES_DL path
+ 1 Uplink path
+ 2 MM_DL path
+ VXREC_MIXER 0 TONES_DL path
+ 1 VX_DL path
+ 2 MM_DL path
+ 3 VX_UL path
+*/
+#define MIX_VXUL_INPUT_MM_DL 0
+#define MIX_VXUL_INPUT_TONES 1
+#define MIX_VXUL_INPUT_VX_UL 2
+#define MIX_VXUL_INPUT_VX_DL 3
+#define MIX_DL1_INPUT_MM_DL 0
+#define MIX_DL1_INPUT_MM_UL2 1
+#define MIX_DL1_INPUT_VX_DL 2
+#define MIX_DL1_INPUT_TONES 3
+#define MIX_DL2_INPUT_MM_DL 0
+#define MIX_DL2_INPUT_MM_UL2 1
+#define MIX_DL2_INPUT_VX_DL 2
+#define MIX_DL2_INPUT_TONES 3
+#define MIX_SDT_INPUT_UP_MIXER 0
+#define MIX_SDT_INPUT_DL1_MIXER 1
+#define MIX_AUDUL_INPUT_MM_DL 0
+#define MIX_AUDUL_INPUT_TONES 1
+#define MIX_AUDUL_INPUT_UPLINK 2
+#define MIX_AUDUL_INPUT_VX_DL 3
+#define MIX_VXREC_INPUT_MM_DL 0
+#define MIX_VXREC_INPUT_TONES 1
+#define MIX_VXREC_INPUT_VX_UL 2
+#define MIX_VXREC_INPUT_VX_DL 3
+#define MIX_ECHO_DL1 0
+#define MIX_ECHO_DL2 1
+/* nb of samples to route */
+#define NBROUTE_UL 16
+/* 10 routing tables max */
+#define NBROUTE_CONFIG_MAX 10
+/* 5 pre-computed routing tables */
+#define NBROUTE_CONFIG 6
+/* AMIC on VX_UL */
+#define UPROUTE_CONFIG_AMIC 0
+/* DMIC first pair on VX_UL */
+#define UPROUTE_CONFIG_DMIC1 1
+/* DMIC second pair on VX_UL */
+#define UPROUTE_CONFIG_DMIC2 2
+/* DMIC last pair on VX_UL */
+#define UPROUTE_CONFIG_DMIC3 3
+/* BT_UL on VX_UL */
+#define UPROUTE_CONFIG_BT 4
+/* ECHO_REF on MM_UL2 */
+#define UPROUTE_ECHO_MMUL2 5
+/* call-back indexes */
+#define MAXCALLBACK 100
+/* subroutines */
+#define MAXNBSUBROUTINE 100
+/* time controlled sequenced */
+#define MAXNBSEQUENCE 20
+/* maximum simultaneous active sequences */
+#define MAXACTIVESEQUENCE 20
+/* max number of steps in the sequences */
+#define MAXSEQUENCESTEPS 2
+/* max number of feature associated to a port */
+#define MAXFEATUREPORT 12
+#define SUB_0_PARAM 0
+/* number of parameters per sequence calls */
+#define SUB_1_PARAM 1
+#define SUB_2_PARAM 2
+#define SUB_3_PARAM 3
+#define SUB_4_PARAM 4
+/* active sequence mask = 0 means the line is free */
+#define FREE_LINE 0
+/* no ask for collision protection */
+#define NOMASK (1 << 0)
+/* do not allow a PDM OFF during the execution of this sequence */
+#define MASK_PDM_OFF (1 << 1)
+/* do not allow a PDM ON during the execution of this sequence */
+#define MASK_PDM_ON (1 << 2)
+/* explicit name of the feature */
+#define NBCHARFEATURENAME 16
+/* explicit name of the port */
+#define NBCHARPORTNAME 16
+/* sink / input port from Host point of view (or AESS for DMIC/McPDM/.. */
+#define SNK_P ABE_ATC_DIRECTION_IN
+/* source / ouptut port */
+#define SRC_P ABE_ATC_DIRECTION_OUT
+/* no ASRC applied */
+#define NODRIFT 0
+/* for abe_set_asrc_drift_control */
+#define FORCED_DRIFT_CONTROL 1
+/* for abe_set_asrc_drift_control */
+#define ADPATIVE_DRIFT_CONTROL 2
+/* number of task/slot depending on the OPP value */
+#define DOPPMODE32_OPP100 (0x00000010)
+#define DOPPMODE32_OPP50 (0x0000000C)
+#define DOPPMODE32_OPP25 (0x0000004)
+/*
+ * ABE CONST AREA FOR PARAMETERS TRANSLATION
+ */
+#define GAIN_MAXIMUM 3000L
+#define GAIN_24dB 2400L
+#define GAIN_18dB 1800L
+#define GAIN_12dB 1200L
+#define GAIN_6dB 600L
+/* default gain = 1 */
+#define GAIN_0dB 0L
+#define GAIN_M6dB -600L
+#define GAIN_M12dB -1200L
+#define GAIN_M18dB -1800L
+#define GAIN_M24dB -2400L
+#define GAIN_M30dB -3000L
+#define GAIN_M40dB -4000L
+#define GAIN_M50dB -5000L
+/* muted gain = -120 decibels */
+#define MUTE_GAIN -12000L
+#define GAIN_TOOLOW -13000L
+#define GAIN_MUTE MUTE_GAIN
+#define RAMP_MINLENGTH 0L
+/* ramp_t is in milli- seconds */
+#define RAMP_0MS 0L
+#define RAMP_1MS 1L
+#define RAMP_2MS 2L
+#define RAMP_5MS 5L
+#define RAMP_10MS 10L
+#define RAMP_20MS 20L
+#define RAMP_50MS 50L
+#define RAMP_100MS 100L
+#define RAMP_200MS 200L
+#define RAMP_500MS 500L
+#define RAMP_1000MS 1000L
+#define RAMP_MAXLENGTH 10000L
+/* for abe_translate_gain_format */
+#define LINABE_TO_DECIBELS 1
+#define DECIBELS_TO_LINABE 2
+/* for abe_translate_ramp_format */
+#define IIRABE_TO_MICROS 1
+#define MICROS_TO_IIABE 2
+/*
+ * ABE CONST AREA FOR PERIPHERAL TUNING
+ */
+/* port idled IDLE_P */
+#define OMAP_ABE_PORT_ACTIVITY_IDLE 1
+/* port initialized, ready to be activated */
+#define OMAP_ABE_PORT_INITIALIZED 3
+/* port activated RUN_P */
+#define OMAP_ABE_PORT_ACTIVITY_RUNNING 2
+#define NOCALLBACK 0
+#define NOPARAMETER 0
+/* number of ATC access upon AMIC DMArequests, all the FIFOs are enabled */
+#define MCPDM_UL_ITER 4
+/* All the McPDM FIFOs are enabled simultaneously */
+#define MCPDM_DL_ITER 24
+/* All the DMIC FIFOs are enabled simultaneously */
+#define DMIC_ITER 12
+/* TBD later if needed */
+#define MAX_PINGPONG_BUFFERS 2
+/*
+ * Indexes to the subroutines
+ */
+#define SUB_WRITE_MIXER 1
+#define SUB_WRITE_PORT_GAIN 2
+/* OLD WAY */
+#define c_feat_init_eq 1
+#define c_feat_read_eq1 2
+#define c_write_eq1 3
+#define c_feat_read_eq2 4
+#define c_write_eq2 5
+#define c_feat_read_eq3 6
+#define c_write_eq3 7
+/* max number of gain to be controlled by HAL */
+#define MAX_NBGAIN_CMEM 36
+/*
+ * MACROS
+ */
+#define maximum(a, b) (((a) < (b)) ? (b) : (a))
+#define minimum(a, b) (((a) > (b)) ? (b) : (a))
+#define absolute(a) (((a) > 0) ? (a) : ((-1)*(a)))
+#define HAL_VERSIONS 9
+#endif/* _ABE_DEF_H_ */
diff --git a/sound/soc/omap/abe/abe_define.h b/sound/soc/omap/abe/abe_define.h
new file mode 100644
index 0000000..41b700a
--- /dev/null
+++ b/sound/soc/omap/abe/abe_define.h
@@ -0,0 +1,120 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#ifndef _ABE_DEFINE_H_
+#define _ABE_DEFINE_H_
+
+#define ATC_DESCRIPTOR_NUMBER 64
+#define PROCESSING_SLOTS 25
+#define TASK_POOL_LENGTH 136
+#define MCU_IRQ 0x24
+#define MCU_IRQ_SHIFT2 0x90
+#define DMA_REQ_SHIFT2 0x210
+#define DSP_IRQ 0x4c
+#define IRQtag_APS 0x000a
+#define IRQtag_COUNT 0x000c
+#define IRQtag_PP 0x000d
+#define DMAreq_7 0x0080
+#define IRQ_FIFO_LENGTH 16
+#define SDT_EQ_ORDER 4
+#define DL_EQ_ORDER 12
+#define MIC_FILTER_ORDER 4
+#define GAINS_WITH_RAMP1 14
+#define GAINS_WITH_RAMP2 22
+#define GAINS_WITH_RAMP_TOTAL 36
+#define ASRC_MEMLENGTH 40
+#define ASRC_UL_VX_FIR_L 19
+#define ASRC_DL_VX_FIR_L 19
+#define ASRC_MM_EXT_IN_FIR_L 18
+#define ASRC_margin 2
+#define ASRC_N_8k 2
+#define ASRC_N_16k 4
+#define ASRC_N_48k 12
+#define VIBRA_N 5
+#define VIBRA1_IIR_MEMSIZE 11
+#define SAMP_LOOP_96K 24
+#define SAMP_LOOP_48K 12
+#define SAMP_LOOP_48KM1 11
+#define SAMP_LOOP_48KM2 10
+#define SAMP_LOOP_16K 4
+#define SAMP_LOOP_8K 2
+#define INPUT_SCALE_SHIFTM2 5156
+#define SATURATION 8420
+#define SATURATION_7FFF 8416
+#define OUTPUT_SCALE_SHIFTM2 5160
+#define NTAPS_SRC_44P1 24
+#define NTAPS_SRC_44P1_M4 96
+#define NTAPS_SRC_44P1_THR 48
+#define NTAPS_SRC_44P1_THRM4 192
+#define DRIFT_COUNTER_44P1M1 443
+#define NB_OF_PHASES_SRC44P1 12
+#define NB_OF_PHASES_SRC44P1M1 11
+#define SRC44P1_BUFFER_SIZE 96
+#define SRC44P1_BUFFER_SIZE_M4 384
+#define SRC44P1_INIT_RPTR 60
+#define MUTE_SCALING 5164
+#define ABE_PMEM 1
+#define ABE_CMEM 2
+#define ABE_SMEM 3
+#define ABE_DMEM 4
+#define ABE_ATC 5
+#define ASRC_BT_UL_FIR_L 19
+#define ASRC_BT_DL_FIR_L 19
+#define SRC44P1_COEF_ADDR 1466
+#define NTAPS_P_SRC_44P1_M4 144
+
+#endif /* _ABE_DEFINE_H_ */
diff --git a/sound/soc/omap/abe/abe_dm_addr.h b/sound/soc/omap/abe/abe_dm_addr.h
new file mode 100644
index 0000000..a9c67a1
--- /dev/null
+++ b/sound/soc/omap/abe/abe_dm_addr.h
@@ -0,0 +1,229 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#define OMAP_ABE_D_ATCDESCRIPTORS_ADDR 0x0
+#define OMAP_ABE_D_ATCDESCRIPTORS_SIZE 0x200
+#define OMAP_ABE_STACK_ADDR 0x200
+#define OMAP_ABE_STACK_SIZE 0x70
+#define OMAP_ABE_D_VERSION_ADDR 0x270
+#define OMAP_ABE_D_VERSION_SIZE 0x4
+#define OMAP_ABE_D_IODESCR_ADDR 0x274
+#define OMAP_ABE_D_IODESCR_SIZE 0x280
+#define OMAP_ABE_D_ZERO_ADDR 0x4F4
+#define OMAP_ABE_D_ZERO_SIZE 0x4
+#define OMAP_ABE_DBG_TRACE1_ADDR 0x4F8
+#define OMAP_ABE_DBG_TRACE1_SIZE 0x1
+#define OMAP_ABE_DBG_TRACE2_ADDR 0x4F9
+#define OMAP_ABE_DBG_TRACE2_SIZE 0x1
+#define OMAP_ABE_DBG_TRACE3_ADDR 0x4FA
+#define OMAP_ABE_DBG_TRACE3_SIZE 0x1
+#define OMAP_ABE_D_MULTIFRAME_ADDR 0x4FC
+#define OMAP_ABE_D_MULTIFRAME_SIZE 0x190
+#define OMAP_ABE_D_IDLETASK_ADDR 0x68C
+#define OMAP_ABE_D_IDLETASK_SIZE 0x2
+#define OMAP_ABE_D_TYPELENGTHCHECK_ADDR 0x68E
+#define OMAP_ABE_D_TYPELENGTHCHECK_SIZE 0x2
+#define OMAP_ABE_D_MAXTASKBYTESINSLOT_ADDR 0x690
+#define OMAP_ABE_D_MAXTASKBYTESINSLOT_SIZE 0x2
+#define OMAP_ABE_D_REWINDTASKBYTES_ADDR 0x692
+#define OMAP_ABE_D_REWINDTASKBYTES_SIZE 0x2
+#define OMAP_ABE_D_PCURRENTTASK_ADDR 0x694
+#define OMAP_ABE_D_PCURRENTTASK_SIZE 0x2
+#define OMAP_ABE_D_PFASTLOOPBACK_ADDR 0x696
+#define OMAP_ABE_D_PFASTLOOPBACK_SIZE 0x2
+#define OMAP_ABE_D_PNEXTFASTLOOPBACK_ADDR 0x698
+#define OMAP_ABE_D_PNEXTFASTLOOPBACK_SIZE 0x4
+#define OMAP_ABE_D_PPCURRENTTASK_ADDR 0x69C
+#define OMAP_ABE_D_PPCURRENTTASK_SIZE 0x2
+#define OMAP_ABE_D_SLOTCOUNTER_ADDR 0x6A0
+#define OMAP_ABE_D_SLOTCOUNTER_SIZE 0x2
+#define OMAP_ABE_D_LOOPCOUNTER_ADDR 0x6A4
+#define OMAP_ABE_D_LOOPCOUNTER_SIZE 0x4
+#define OMAP_ABE_D_REWINDFLAG_ADDR 0x6A8
+#define OMAP_ABE_D_REWINDFLAG_SIZE 0x2
+#define OMAP_ABE_D_SLOT23_CTRL_ADDR 0x6AC
+#define OMAP_ABE_D_SLOT23_CTRL_SIZE 0x4
+#define OMAP_ABE_D_MCUIRQFIFO_ADDR 0x6B0
+#define OMAP_ABE_D_MCUIRQFIFO_SIZE 0x40
+#define OMAP_ABE_D_PINGPONGDESC_ADDR 0x6F0
+#define OMAP_ABE_D_PINGPONGDESC_SIZE 0x18
+#define OMAP_ABE_D_PP_MCU_IRQ_ADDR 0x708
+#define OMAP_ABE_D_PP_MCU_IRQ_SIZE 0x2
+#define OMAP_ABE_D_SRC44P1_MMDL_STRUCT_ADDR 0x70C
+#define OMAP_ABE_D_SRC44P1_MMDL_STRUCT_SIZE 0x12
+#define OMAP_ABE_D_SRC44P1_TONES_STRUCT_ADDR 0x720
+#define OMAP_ABE_D_SRC44P1_TONES_STRUCT_SIZE 0x12
+#define OMAP_ABE_D_CTRLPORTFIFO_ADDR 0x740
+#define OMAP_ABE_D_CTRLPORTFIFO_SIZE 0x10
+#define OMAP_ABE_D_IDLE_STATE_ADDR 0x750
+#define OMAP_ABE_D_IDLE_STATE_SIZE 0x4
+#define OMAP_ABE_D_STOP_REQUEST_ADDR 0x754
+#define OMAP_ABE_D_STOP_REQUEST_SIZE 0x4
+#define OMAP_ABE_D_REF0_ADDR 0x758
+#define OMAP_ABE_D_REF0_SIZE 0x2
+#define OMAP_ABE_D_DEBUGREGISTER_ADDR 0x75C
+#define OMAP_ABE_D_DEBUGREGISTER_SIZE 0x8C
+#define OMAP_ABE_D_GCOUNT_ADDR 0x7E8
+#define OMAP_ABE_D_GCOUNT_SIZE 0x2
+#define OMAP_ABE_D_FASTCOUNTER_ADDR 0x7EC
+#define OMAP_ABE_D_FASTCOUNTER_SIZE 0x4
+#define OMAP_ABE_D_SLOWCOUNTER_ADDR 0x7F0
+#define OMAP_ABE_D_SLOWCOUNTER_SIZE 0x4
+#define OMAP_ABE_D_AUPLINKROUTING_ADDR 0x7F4
+#define OMAP_ABE_D_AUPLINKROUTING_SIZE 0x20
+#define OMAP_ABE_D_VIRTAUDIOLOOP_ADDR 0x814
+#define OMAP_ABE_D_VIRTAUDIOLOOP_SIZE 0x4
+#define OMAP_ABE_D_ASRCVARS_DL_VX_ADDR 0x818
+#define OMAP_ABE_D_ASRCVARS_DL_VX_SIZE 0x20
+#define OMAP_ABE_D_ASRCVARS_UL_VX_ADDR 0x838
+#define OMAP_ABE_D_ASRCVARS_UL_VX_SIZE 0x20
+#define OMAP_ABE_D_COEFADDRESSES_VX_ADDR 0x858
+#define OMAP_ABE_D_COEFADDRESSES_VX_SIZE 0x20
+#define OMAP_ABE_D_ASRCVARS_MM_EXT_IN_ADDR 0x878
+#define OMAP_ABE_D_ASRCVARS_MM_EXT_IN_SIZE 0x20
+#define OMAP_ABE_D_COEFADDRESSES_MM_ADDR 0x898
+#define OMAP_ABE_D_COEFADDRESSES_MM_SIZE 0x20
+#define OMAP_ABE_D_TRACEBUFADR_ADDR 0x8B8
+#define OMAP_ABE_D_TRACEBUFADR_SIZE 0x2
+#define OMAP_ABE_D_TRACEBUFOFFSET_ADDR 0x8BA
+#define OMAP_ABE_D_TRACEBUFOFFSET_SIZE 0x2
+#define OMAP_ABE_D_TRACEBUFLENGTH_ADDR 0x8BC
+#define OMAP_ABE_D_TRACEBUFLENGTH_SIZE 0x2
+#define OMAP_ABE_D_PEMPTY_ADDR 0x8C0
+#define OMAP_ABE_D_PEMPTY_SIZE 0x54
+#define OMAP_ABE_D_ECHO_REF_48_16_WRAP_ADDR 0x914
+#define OMAP_ABE_D_ECHO_REF_48_16_WRAP_SIZE 0x8
+#define OMAP_ABE_D_ECHO_REF_48_8_WRAP_ADDR 0x91C
+#define OMAP_ABE_D_ECHO_REF_48_8_WRAP_SIZE 0x8
+#define OMAP_ABE_D_BT_UL_16_48_WRAP_ADDR 0x924
+#define OMAP_ABE_D_BT_UL_16_48_WRAP_SIZE 0x8
+#define OMAP_ABE_D_BT_UL_8_48_WRAP_ADDR 0x92C
+#define OMAP_ABE_D_BT_UL_8_48_WRAP_SIZE 0x8
+#define OMAP_ABE_D_BT_DL_48_16_WRAP_ADDR 0x934
+#define OMAP_ABE_D_BT_DL_48_16_WRAP_SIZE 0x8
+#define OMAP_ABE_D_BT_DL_48_8_WRAP_ADDR 0x93C
+#define OMAP_ABE_D_BT_DL_48_8_WRAP_SIZE 0x8
+#define OMAP_ABE_D_VX_DL_16_48_WRAP_ADDR 0x944
+#define OMAP_ABE_D_VX_DL_16_48_WRAP_SIZE 0x8
+#define OMAP_ABE_D_VX_DL_8_48_WRAP_ADDR 0x94C
+#define OMAP_ABE_D_VX_DL_8_48_WRAP_SIZE 0x8
+#define OMAP_ABE_D_VX_UL_48_16_WRAP_ADDR 0x954
+#define OMAP_ABE_D_VX_UL_48_16_WRAP_SIZE 0x8
+#define OMAP_ABE_D_VX_UL_48_8_WRAP_ADDR 0x95C
+#define OMAP_ABE_D_VX_UL_48_8_WRAP_SIZE 0x8
+#define OMAP_ABE_D_ASRCVARS_BT_UL_ADDR 0x964
+#define OMAP_ABE_D_ASRCVARS_BT_UL_SIZE 0x20
+#define OMAP_ABE_D_ASRCVARS_BT_DL_ADDR 0x984
+#define OMAP_ABE_D_ASRCVARS_BT_DL_SIZE 0x20
+#define OMAP_ABE_D_BT_DL_48_8_OPP100_WRAP_ADDR 0x9A4
+#define OMAP_ABE_D_BT_DL_48_8_OPP100_WRAP_SIZE 0x8
+#define OMAP_ABE_D_BT_DL_48_16_OPP100_WRAP_ADDR 0x9AC
+#define OMAP_ABE_D_BT_DL_48_16_OPP100_WRAP_SIZE 0x8
+#define OMAP_ABE_D_VX_DL_8_48_FIR_WRAP_ADDR 0x9B4
+#define OMAP_ABE_D_VX_DL_8_48_FIR_WRAP_SIZE 0x8
+#define OMAP_ABE_D_BT_UL_8_48_FIR_WRAP_ADDR 0x9BC
+#define OMAP_ABE_D_BT_UL_8_48_FIR_WRAP_SIZE 0x8
+#define OMAP_ABE_D_TASKSLIST_ADDR 0x9C4
+#define OMAP_ABE_D_TASKSLIST_SIZE 0x880
+#define OMAP_ABE_D_HW_TEST_ADDR 0x1244
+#define OMAP_ABE_D_HW_TEST_SIZE 0x28
+#define OMAP_ABE_D_TRACEBUFADR_HAL_ADDR 0x126C
+#define OMAP_ABE_D_TRACEBUFADR_HAL_SIZE 0x4
+#define OMAP_ABE_D_DEBUG_FW_TASK_ADDR 0x1400
+#define OMAP_ABE_D_DEBUG_FW_TASK_SIZE 0x100
+#define OMAP_ABE_D_DEBUG_FIFO_ADDR 0x1500
+#define OMAP_ABE_D_DEBUG_FIFO_SIZE 0x60
+#define OMAP_ABE_D_DEBUG_FIFO_HAL_ADDR 0x1560
+#define OMAP_ABE_D_DEBUG_FIFO_HAL_SIZE 0x20
+#define OMAP_ABE_D_FWMEMINIT_ADDR 0x1580
+#define OMAP_ABE_D_FWMEMINIT_SIZE 0x3C0
+#define OMAP_ABE_D_FWMEMINITDESCR_ADDR 0x1940
+#define OMAP_ABE_D_FWMEMINITDESCR_SIZE 0x10
+#define OMAP_ABE_D_BT_DL_FIFO_ADDR 0x1C00
+#define OMAP_ABE_D_BT_DL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_BT_UL_FIFO_ADDR 0x1E00
+#define OMAP_ABE_D_BT_UL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_MM_EXT_OUT_FIFO_ADDR 0x2000
+#define OMAP_ABE_D_MM_EXT_OUT_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_MM_EXT_IN_FIFO_ADDR 0x2200
+#define OMAP_ABE_D_MM_EXT_IN_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_MM_UL2_FIFO_ADDR 0x2400
+#define OMAP_ABE_D_MM_UL2_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_DMIC_UL_FIFO_ADDR 0x2600
+#define OMAP_ABE_D_DMIC_UL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_MM_UL_FIFO_ADDR 0x2800
+#define OMAP_ABE_D_MM_UL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_MM_DL_FIFO_ADDR 0x2A00
+#define OMAP_ABE_D_MM_DL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_TONES_DL_FIFO_ADDR 0x2C00
+#define OMAP_ABE_D_TONES_DL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_VIB_DL_FIFO_ADDR 0x2E00
+#define OMAP_ABE_D_VIB_DL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_DEBUG_HAL_TASK_ADDR 0x3000
+#define OMAP_ABE_D_DEBUG_HAL_TASK_SIZE 0x800
+#define OMAP_ABE_D_MCPDM_DL_FIFO_ADDR 0x3800
+#define OMAP_ABE_D_MCPDM_DL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_MCPDM_UL_FIFO_ADDR 0x3A00
+#define OMAP_ABE_D_MCPDM_UL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_VX_UL_FIFO_ADDR 0x3C00
+#define OMAP_ABE_D_VX_UL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_VX_DL_FIFO_ADDR 0x3E00
+#define OMAP_ABE_D_VX_DL_FIFO_SIZE 0x1E0
+#define OMAP_ABE_D_PING_ADDR 0x4000
+#define OMAP_ABE_D_PING_SIZE 0x6000
+#define OMAP_ABE_D_PONG_ADDR 0xA000
+#define OMAP_ABE_D_PONG_SIZE 0x6000
diff --git a/sound/soc/omap/abe/abe_ext.h b/sound/soc/omap/abe/abe_ext.h
new file mode 100644
index 0000000..8fb1aac
--- /dev/null
+++ b/sound/soc/omap/abe/abe_ext.h
@@ -0,0 +1,242 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_EXT_H_
+#define _ABE_EXT_H_
+
+/*
+ * OS DEPENDENT MMU CONFIGURATION
+ */
+#define ABE_PMEM_BASE_OFFSET_MPU 0xe0000
+#define ABE_CMEM_BASE_OFFSET_MPU 0xa0000
+#define ABE_SMEM_BASE_OFFSET_MPU 0xc0000
+#define ABE_DMEM_BASE_OFFSET_MPU 0x80000
+#define ABE_ATC_BASE_OFFSET_MPU 0xf1000
+/* default base address for io_base */
+#define ABE_DEFAULT_BASE_ADDRESS_L3 0x49000000L
+#define ABE_DEFAULT_BASE_ADDRESS_L4 0x40100000L
+#define ABE_DEFAULT_BASE_ADDRESS_DEFAULT ABE_DEFAULT_BASE_ADDRESS_L3
+/*
+ * HARDWARE AND PERIPHERAL DEFINITIONS
+ */
+/* PMEM SIZE in bytes (1024 words of 64 bits: : #32bits words x 4)*/
+#define ABE_PMEM_SIZE 8192
+/* CMEM SIZE in bytes (2048 coeff : #32bits words x 4)*/
+#define ABE_CMEM_SIZE 8192
+/* SMEM SIZE in bytes (3072 stereo samples : #32bits words x 4)*/
+#define ABE_SMEM_SIZE 24576
+/* DMEM SIZE in bytes */
+#define ABE_DMEM_SIZE 65536L
+/* ATC REGISTERS SIZE in bytes */
+#define ABE_ATC_DESC_SIZE 512
+/* holds the MCU Irq signal */
+#define ABE_MCU_IRQSTATUS_RAW 0x24
+/* status : clear the IRQ */
+#define ABE_MCU_IRQSTATUS 0x28
+/* holds the DSP Irq signal */
+#define ABE_DSP_IRQSTATUS_RAW 0x4C
+/* holds the DMA req lines to the sDMA */
+#define ABE_DMASTATUS_RAW 0x84
+#define EVENT_GENERATOR_COUNTER 0x68
+/* PLL output/desired sampling rate = (32768 * 6000)/96000 */
+#define EVENT_GENERATOR_COUNTER_DEFAULT (2048-1)
+/* PLL output/desired sampling rate = (32768 * 6000)/88200 */
+#define EVENT_GENERATOR_COUNTER_44100 (2228-1)
+/* start / stop the EVENT generator */
+#define EVENT_GENERATOR_START 0x6C
+#define EVENT_GENERATOR_ON 1
+#define EVENT_GENERATOR_OFF 0
+/* selection of the EVENT generator source */
+#define EVENT_SOURCE_SELECTION 0x70
+#define EVENT_SOURCE_DMA 0
+#define EVENT_SOURCE_COUNTER 1
+/* selection of the ABE DMA req line from ATC */
+#define AUDIO_ENGINE_SCHEDULER 0x74
+#define ABE_ATC_DMIC_DMA_REQ 1
+#define ABE_ATC_MCPDMDL_DMA_REQ 2
+#define ABE_ATC_MCPDMUL_DMA_REQ 3
+/* Direction=0 means input from ABE point of view */
+#define ABE_ATC_DIRECTION_IN 0
+/* Direction=1 means output from ABE point of view */
+#define ABE_ATC_DIRECTION_OUT 1
+/*
+ * DMA requests
+ */
+/*Internal connection doesn't connect at ABE boundary */
+#define External_DMA_0 0
+/*Transmit request digital microphone */
+#define DMIC_DMA_REQ 1
+/*Multichannel PDM downlink */
+#define McPDM_DMA_DL 2
+/*Multichannel PDM uplink */
+#define McPDM_DMA_UP 3
+/*MCBSP module 1 - transmit request */
+#define MCBSP1_DMA_TX 4
+/*MCBSP module 1 - receive request */
+#define MCBSP1_DMA_RX 5
+/*MCBSP module 2 - transmit request */
+#define MCBSP2_DMA_TX 6
+/*MCBSP module 2 - receive request */
+#define MCBSP2_DMA_RX 7
+/*MCBSP module 3 - transmit request */
+#define MCBSP3_DMA_TX 8
+/*MCBSP module 3 - receive request */
+#define MCBSP3_DMA_RX 9
+/*SLIMBUS module 1 - transmit request channel 0 */
+#define SLIMBUS1_DMA_TX0 10
+/*SLIMBUS module 1 - transmit request channel 1 */
+#define SLIMBUS1_DMA_TX1 11
+/*SLIMBUS module 1 - transmit request channel 2 */
+#define SLIMBUS1_DMA_TX2 12
+/*SLIMBUS module 1 - transmit request channel 3 */
+#define SLIMBUS1_DMA_TX3 13
+/*SLIMBUS module 1 - transmit request channel 4 */
+#define SLIMBUS1_DMA_TX4 14
+/*SLIMBUS module 1 - transmit request channel 5 */
+#define SLIMBUS1_DMA_TX5 15
+/*SLIMBUS module 1 - transmit request channel 6 */
+#define SLIMBUS1_DMA_TX6 16
+/*SLIMBUS module 1 - transmit request channel 7 */
+#define SLIMBUS1_DMA_TX7 17
+/*SLIMBUS module 1 - receive request channel 0 */
+#define SLIMBUS1_DMA_RX0 18
+/*SLIMBUS module 1 - receive request channel 1 */
+#define SLIMBUS1_DMA_RX1 19
+/*SLIMBUS module 1 - receive request channel 2 */
+#define SLIMBUS1_DMA_RX2 20
+/*SLIMBUS module 1 - receive request channel 3 */
+#define SLIMBUS1_DMA_RX3 21
+/*SLIMBUS module 1 - receive request channel 4 */
+#define SLIMBUS1_DMA_RX4 22
+/*SLIMBUS module 1 - receive request channel 5 */
+#define SLIMBUS1_DMA_RX5 23
+/*SLIMBUS module 1 - receive request channel 6 */
+#define SLIMBUS1_DMA_RX6 24
+/*SLIMBUS module 1 - receive request channel 7 */
+#define SLIMBUS1_DMA_RX7 25
+/*McASP - Data transmit DMA request line */
+#define McASP1_AXEVT 26
+/*McASP - Data receive DMA request line */
+#define McASP1_AREVT 29
+/*DUMMY FIFO @@@ */
+#define _DUMMY_FIFO_ 30
+/*DMA of the Circular buffer peripheral 0 */
+#define CBPr_DMA_RTX0 32
+/*DMA of the Circular buffer peripheral 1 */
+#define CBPr_DMA_RTX1 33
+/*DMA of the Circular buffer peripheral 2 */
+#define CBPr_DMA_RTX2 34
+/*DMA of the Circular buffer peripheral 3 */
+#define CBPr_DMA_RTX3 35
+/*DMA of the Circular buffer peripheral 4 */
+#define CBPr_DMA_RTX4 36
+/*DMA of the Circular buffer peripheral 5 */
+#define CBPr_DMA_RTX5 37
+/*DMA of the Circular buffer peripheral 6 */
+#define CBPr_DMA_RTX6 38
+/*DMA of the Circular buffer peripheral 7 */
+#define CBPr_DMA_RTX7 39
+/*
+ * ATC DESCRIPTORS - DESTINATIONS
+ */
+#define DEST_DMEM_access 0x00
+#define DEST_MCBSP1_ TX 0x01
+#define DEST_MCBSP2_ TX 0x02
+#define DEST_MCBSP3_TX 0x03
+#define DEST_SLIMBUS1_TX0 0x04
+#define DEST_SLIMBUS1_TX1 0x05
+#define DEST_SLIMBUS1_TX2 0x06
+#define DEST_SLIMBUS1_TX3 0x07
+#define DEST_SLIMBUS1_TX4 0x08
+#define DEST_SLIMBUS1_TX5 0x09
+#define DEST_SLIMBUS1_TX6 0x0A
+#define DEST_SLIMBUS1_TX7 0x0B
+#define DEST_MCPDM_DL 0x0C
+#define DEST_MCASP_TX0 0x0D
+#define DEST_MCASP_TX1 0x0E
+#define DEST_MCASP_TX2 0x0F
+#define DEST_MCASP_TX3 0x10
+#define DEST_EXTPORT0 0x11
+#define DEST_EXTPORT1 0x12
+#define DEST_EXTPORT2 0x13
+#define DEST_EXTPORT3 0x14
+#define DEST_MCPDM_ON 0x15
+#define DEST_CBP_CBPr 0x3F
+/*
+ * ATC DESCRIPTORS - SOURCES
+ */
+#define SRC_DMEM_access 0x0
+#define SRC_MCBSP1_ RX 0x01
+#define SRC_MCBSP2_RX 0x02
+#define SRC_MCBSP3_RX 0x03
+#define SRC_SLIMBUS1_RX0 0x04
+#define SRC_SLIMBUS1_RX1 0x05
+#define SRC_SLIMBUS1_RX2 0x06
+#define SRC_SLIMBUS1_RX3 0x07
+#define SRC_SLIMBUS1_RX4 0x08
+#define SRC_SLIMBUS1_RX5 0x09
+#define SRC_SLIMBUS1_RX6 0x0A
+#define SRC_SLIMBUS1_RX7 0x0B
+#define SRC_DMIC_UP 0x0C
+#define SRC_MCPDM_UP 0x0D
+#define SRC_MCASP_RX0 0x0E
+#define SRC_MCASP_RX1 0x0F
+#define SRC_MCASP_RX2 0x10
+#define SRC_MCASP_RX3 0x11
+#define SRC_CBP_CBPr 0x3F
+#endif/* _ABE_EXT_H_ */
diff --git a/sound/soc/omap/abe/abe_firmware.c b/sound/soc/omap/abe/abe_firmware.c
new file mode 100644
index 0000000..ce70be3
--- /dev/null
+++ b/sound/soc/omap/abe/abe_firmware.c
@@ -0,0 +1,25886 @@
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diff --git a/sound/soc/omap/abe/abe_functionsid.h b/sound/soc/omap/abe/abe_functionsid.h
new file mode 100644
index 0000000..db21988
--- /dev/null
+++ b/sound/soc/omap/abe/abe_functionsid.h
@@ -0,0 +1,117 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#ifndef _ABE_FUNCTIONSID_H_
+#define _ABE_FUNCTIONSID_H_
+/*
+ * TASK function ID definitions
+ */
+#define C_ABE_FW_FUNCTION_IIR 0
+#define C_ABE_FW_FUNCTION_monoToStereoPack 1
+#define C_ABE_FW_FUNCTION_stereoToMonoSplit 2
+#define C_ABE_FW_FUNCTION_decimator 3
+#define C_ABE_FW_FUNCTION_OS0Fill 4
+#define C_ABE_FW_FUNCTION_mixer2 5
+#define C_ABE_FW_FUNCTION_mixer4 6
+#define C_ABE_FW_FUNCTION_mixer4_dual_mono 7
+#define C_ABE_FW_FUNCTION_inplaceGain 8
+#define C_ABE_FW_FUNCTION_StreamRouting 9
+#define C_ABE_FW_FUNCTION_gainConverge 10
+#define C_ABE_FW_FUNCTION_dualIir 11
+#define C_ABE_FW_FUNCTION_IO_DL_pp 12
+#define C_ABE_FW_FUNCTION_IO_generic 13
+#define C_ABE_FW_FUNCTION_irq_fifo_debug 14
+#define C_ABE_FW_FUNCTION_synchronize_pointers 15
+#define C_ABE_FW_FUNCTION_VIBRA2 16
+#define C_ABE_FW_FUNCTION_VIBRA1 17
+#define C_ABE_FW_FUNCTION_IIR_SRC_MIC 18
+#define C_ABE_FW_FUNCTION_wrappers 19
+#define C_ABE_FW_FUNCTION_ASRC_DL_wrapper 20
+#define C_ABE_FW_FUNCTION_ASRC_UL_wrapper 21
+#define C_ABE_FW_FUNCTION_mem_init 22
+#define C_ABE_FW_FUNCTION_debug_vx_asrc 23
+#define C_ABE_FW_FUNCTION_IIR_SRC2 24
+#define C_ABE_FW_FUNCTION_ASRC_DL_wrapper_sibling 25
+#define C_ABE_FW_FUNCTION_ASRC_UL_wrapper_sibling 26
+#define C_ABE_FW_FUNCTION_FIR6 27
+#define C_ABE_FW_FUNCTION_SRC44P1 28
+#define C_ABE_FW_FUNCTION_SRC44P1_1211 29
+/*
+ * COPY function ID definitions
+ */
+#define NULL_COPY_CFPID 0
+#define S2D_STEREO_16_16_CFPID 1
+#define S2D_MONO_MSB_CFPID 2
+#define S2D_STEREO_MSB_CFPID 3
+#define S2D_STEREO_RSHIFTED_16_CFPID 4
+#define S2D_MONO_RSHIFTED_16_CFPID 5
+#define D2S_STEREO_16_16_CFPID 6
+#define D2S_MONO_MSB_CFPID 7
+#define D2S_MONO_RSHIFTED_16_CFPID 8
+#define D2S_STEREO_RSHIFTED_16_CFPID 9
+#define D2S_STEREO_MSB_CFPID 10
+#define COPY_DMIC_CFPID 11
+#define COPY_MCPDM_DL_CFPID 12
+#define COPY_MM_UL_CFPID 13
+#define SPLIT_SMEM_CFPID 14
+#define MERGE_SMEM_CFPID 15
+#define SPLIT_TDM_CFPID 16
+#define MERGE_TDM_CFPID 17
+#define ROUTE_MM_UL_CFPID 18
+#define IO_IP_CFPID 19
+#define COPY_UNDERFLOW_CFPID 20
+#endif /* _ABE_FUNCTIONSID_H_ */
diff --git a/sound/soc/omap/abe/abe_fw.h b/sound/soc/omap/abe/abe_fw.h
new file mode 100644
index 0000000..a29dbb3
--- /dev/null
+++ b/sound/soc/omap/abe/abe_fw.h
@@ -0,0 +1,214 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_FW_H_
+#define _ABE_FW_H_
+
+#include "abe_cm_addr.h"
+#include "abe_sm_addr.h"
+#include "abe_dm_addr.h"
+#include "abe_typedef.h"
+/*
+ * GLOBAL DEFINITION
+ */
+/* one scheduler loop = 4kHz = 12 samples at 48kHz */
+#define FW_SCHED_LOOP_FREQ 4000
+/* one scheduler loop = 4kHz = 12 samples at 48kHz */
+#define FW_SCHED_LOOP_FREQ_DIV1000 (FW_SCHED_LOOP_FREQ/1000)
+#define EVENT_FREQUENCY 96000
+#define SLOTS_IN_SCHED_LOOP (96000/FW_SCHED_LOOP_FREQ)
+#define SCHED_LOOP_8kHz (8000/FW_SCHED_LOOP_FREQ)
+#define SCHED_LOOP_16kHz (16000/FW_SCHED_LOOP_FREQ)
+#define SCHED_LOOP_24kHz (24000/FW_SCHED_LOOP_FREQ)
+#define SCHED_LOOP_48kHz (48000/FW_SCHED_LOOP_FREQ)
+#define TASKS_IN_SLOT 8
+/*
+ * DMEM AREA - SCHEDULER
+ */
+#define dmem_mm_trace OMAP_ABE_D_DEBUG_FIFO_ADDR
+#define dmem_mm_trace_size ((OMAP_ABE_D_DEBUG_FIFO_SIZE)/4)
+#define ATC_SIZE 8 /* 8 bytes per descriptors */
+struct omap_abe_atc_desc {
+ unsigned rdpt:7; /* first 32bits word of the descriptor */
+ unsigned reserved0:1;
+ unsigned cbsize:7;
+ unsigned irqdest:1;
+ unsigned cberr:1;
+ unsigned reserved1:5;
+ unsigned cbdir:1;
+ unsigned nw:1;
+ unsigned wrpt:7;
+ unsigned reserved2:1;
+ unsigned badd:12; /* second 32bits word of the descriptor */
+ unsigned iter:7; /* iteration field overlaps 16-bit boundary */
+ unsigned srcid:6;
+ unsigned destid:6;
+ unsigned desen:1;
+};
+/*
+ * Infinite counter incremented on each sheduler periods (~250 us)
+ * uint16 dmem_debug_time_stamp
+ */
+#define dmem_debug_time_stamp OMAP_ABE_D_LOOPCOUNTER_ADDR
+/*
+ * ATC BUFFERS + IO TASKS SMEM buffers
+ */
+#define dmem_dmic OMAP_ABE_D_DMIC_UL_FIFO_ADDR
+#define dmem_dmic_size (OMAP_ABE_D_DMIC_UL_FIFO_SIZE/4)
+#define dmem_amic OMAP_ABE_D_MCPDM_UL_FIFO_ADDR
+#define dmem_amic_size (OMAP_ABE_D_MCPDM_UL_FIFO_SIZE/4)
+#define smem_amic AMIC_96_labelID
+#define dmem_mcpdm OMAP_ABE_D_MCPDM_DL_FIFO_ADDR
+#define dmem_mcpdm_size (OMAP_ABE_D_MCPDM_DL_FIFO_SIZE/4)
+#define dmem_mm_ul OMAP_ABE_D_MM_UL_FIFO_ADDR
+#define dmem_mm_ul_size (OMAP_ABE_D_MM_UL_FIFO_SIZE/4)
+/* managed directly by the router */
+#define smem_mm_ul MM_UL_labelID
+#define dmem_mm_ul2 OMAP_ABE_D_MM_UL2_FIFO_ADDR
+#define dmem_mm_ul2_size (OMAP_ABE_D_MM_UL2_FIFO_SIZE/4)
+/* managed directly by the router */
+#define smem_mm_ul2 MM_UL2_labelID
+#define dmem_mm_dl OMAP_ABE_D_MM_DL_FIFO_ADDR
+#define dmem_mm_dl_size (OMAP_ABE_D_MM_DL_FIFO_SIZE/4)
+#define smem_mm_dl MM_DL_labelID
+#define dmem_vx_dl OMAP_ABE_D_VX_DL_FIFO_ADDR
+#define dmem_vx_dl_size (OMAP_ABE_D_VX_DL_FIFO_SIZE/4)
+#define smem_vx_dl IO_VX_DL_ASRC_labelID /* Voice_16k_DL_labelID */
+#define dmem_vx_ul OMAP_ABE_D_VX_UL_FIFO_ADDR
+#define dmem_vx_ul_size (OMAP_ABE_D_VX_UL_FIFO_SIZE/4)
+#define smem_vx_ul Voice_8k_UL_labelID
+#define dmem_tones_dl OMAP_ABE_D_TONES_DL_FIFO_ADDR
+#define dmem_tones_dl_size (OMAP_ABE_D_TONES_DL_FIFO_SIZE/4)
+#define smem_tones_dl Tones_labelID
+#define dmem_vib_dl OMAP_ABE_D_VIB_DL_FIFO_ADDR
+#define dmem_vib_dl_size (OMAP_ABE_D_VIB_DL_FIFO_SIZE/4)
+#define smem_vib IO_VIBRA_DL_labelID
+#define dmem_mm_ext_out OMAP_ABE_D_MM_EXT_OUT_FIFO_ADDR
+#define dmem_mm_ext_out_size (OMAP_ABE_D_MM_EXT_OUT_FIFO_SIZE/4)
+#define smem_mm_ext_out DL1_GAIN_out_labelID
+#define dmem_mm_ext_in OMAP_ABE_D_MM_EXT_IN_FIFO_ADDR
+#define dmem_mm_ext_in_size (OMAP_ABE_D_MM_EXT_IN_FIFO_SIZE/4)
+/*IO_MM_EXT_IN_ASRC_labelID ASRC input buffer, size 40 */
+#define smem_mm_ext_in_opp100 IO_MM_EXT_IN_ASRC_labelID
+/* at OPP 50 without ASRC */
+#define smem_mm_ext_in_opp50 MM_EXT_IN_labelID
+#define dmem_bt_vx_dl OMAP_ABE_D_BT_DL_FIFO_ADDR
+#define dmem_bt_vx_dl_size (OMAP_ABE_D_BT_DL_FIFO_SIZE/4)
+#define smem_bt_vx_dl_opp50 BT_DL_8k_labelID
+/*BT_DL_8k_opp100_labelID ASRC output buffer, size 40 */
+#define smem_bt_vx_dl_opp100 BT_DL_8k_opp100_labelID
+#define dmem_bt_vx_ul OMAP_ABE_D_BT_UL_FIFO_ADDR
+#define dmem_bt_vx_ul_size (OMAP_ABE_D_BT_UL_FIFO_SIZE/4)
+#define smem_bt_vx_ul_opp50 BT_UL_8k_labelID
+/*IO_BT_UL_ASRC_labelID ASRC input buffer, size 40 */
+#define smem_bt_vx_ul_opp100 IO_BT_UL_ASRC_labelID
+/*
+ * SMEM AREA
+ */
+/*
+ * GAIN SMEM on PORT
+ * int32 smem_G0 [18] : desired gain on the ports
+ * format of G0 = 6 bits left shifted desired gain in linear 24bits format
+ * int24 stereo G0 [18] = G0
+ * int24 stereo GI [18] current value of the gain in the same format of G0
+ * List of smoothed gains :
+ * 6 DMIC 0 1 2 3 4 5
+ * 2 AMIC L R
+ * 4 PORT1/2_RX L R
+ * 2 MM_EXT L R
+ * 2 MM_VX_DL L R
+ * 2 IHF L R
+ * ---------------
+ * 18 = TOTAL
+ */
+/*
+ * COEFFICIENTS AREA
+ */
+/*
+ * delay coefficients used in the IIR-1 filters
+ * int24 cmem_gain_delay_iir1[9 x 2] (a, (1-a))
+ *
+ * 3 for 6 DMIC 0 1 2 3 4 5
+ * 1 for 2 AMIC L R
+ * 2 for 4 PORT1/2_RX L R
+ * 1 for 2 MM_EXT L R
+ * 1 for 2 MM_VX_DL L R
+ * 1 for 2 IHF L R
+ */
+/*
+ * gain controls
+ */
+#define GAIN_LEFT_OFFSET 0
+#define GAIN_RIGHT_OFFSET 1
+/* stereo gains */
+#define dmic1_gains_offset 0
+#define dmic2_gains_offset 2
+#define dmic3_gains_offset 4
+#define amic_gains_offset 6
+#define dl1_gains_offset 8
+#define dl2_gains_offset 10
+#define splitters_gains_offset 12
+#define mixer_dl1_offset 14
+#define mixer_dl2_offset 18
+#define mixer_echo_offset 22
+#define mixer_sdt_offset 24
+#define mixer_vxrec_offset 26
+#define mixer_audul_offset 30
+#define btul_gains_offset 34
+
+#endif/* _ABE_FW_H_ */
diff --git a/sound/soc/omap/abe/abe_gain.c b/sound/soc/omap/abe/abe_gain.c
new file mode 100644
index 0000000..9c148da
--- /dev/null
+++ b/sound/soc/omap/abe/abe_gain.c
@@ -0,0 +1,790 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_dbg.h"
+#include "abe.h"
+#include "abe_gain.h"
+#include "abe_mem.h"
+
+/*
+ * ABE CONST AREA FOR PARAMETERS TRANSLATION
+ */
+#define min_mdb (-12000)
+#define max_mdb (3000)
+#define sizeof_db2lin_table (1 + ((max_mdb - min_mdb)/100))
+
+const u32 abe_db2lin_table[sizeof_db2lin_table] = {
+ 0x00000000, /* SMEM coding of -120 dB */
+ 0x00000000, /* SMEM coding of -119 dB */
+ 0x00000000, /* SMEM coding of -118 dB */
+ 0x00000000, /* SMEM coding of -117 dB */
+ 0x00000000, /* SMEM coding of -116 dB */
+ 0x00000000, /* SMEM coding of -115 dB */
+ 0x00000000, /* SMEM coding of -114 dB */
+ 0x00000000, /* SMEM coding of -113 dB */
+ 0x00000000, /* SMEM coding of -112 dB */
+ 0x00000000, /* SMEM coding of -111 dB */
+ 0x00000000, /* SMEM coding of -110 dB */
+ 0x00000000, /* SMEM coding of -109 dB */
+ 0x00000001, /* SMEM coding of -108 dB */
+ 0x00000001, /* SMEM coding of -107 dB */
+ 0x00000001, /* SMEM coding of -106 dB */
+ 0x00000001, /* SMEM coding of -105 dB */
+ 0x00000001, /* SMEM coding of -104 dB */
+ 0x00000001, /* SMEM coding of -103 dB */
+ 0x00000002, /* SMEM coding of -102 dB */
+ 0x00000002, /* SMEM coding of -101 dB */
+ 0x00000002, /* SMEM coding of -100 dB */
+ 0x00000002, /* SMEM coding of -99 dB */
+ 0x00000003, /* SMEM coding of -98 dB */
+ 0x00000003, /* SMEM coding of -97 dB */
+ 0x00000004, /* SMEM coding of -96 dB */
+ 0x00000004, /* SMEM coding of -95 dB */
+ 0x00000005, /* SMEM coding of -94 dB */
+ 0x00000005, /* SMEM coding of -93 dB */
+ 0x00000006, /* SMEM coding of -92 dB */
+ 0x00000007, /* SMEM coding of -91 dB */
+ 0x00000008, /* SMEM coding of -90 dB */
+ 0x00000009, /* SMEM coding of -89 dB */
+ 0x0000000A, /* SMEM coding of -88 dB */
+ 0x0000000B, /* SMEM coding of -87 dB */
+ 0x0000000D, /* SMEM coding of -86 dB */
+ 0x0000000E, /* SMEM coding of -85 dB */
+ 0x00000010, /* SMEM coding of -84 dB */
+ 0x00000012, /* SMEM coding of -83 dB */
+ 0x00000014, /* SMEM coding of -82 dB */
+ 0x00000017, /* SMEM coding of -81 dB */
+ 0x0000001A, /* SMEM coding of -80 dB */
+ 0x0000001D, /* SMEM coding of -79 dB */
+ 0x00000021, /* SMEM coding of -78 dB */
+ 0x00000025, /* SMEM coding of -77 dB */
+ 0x00000029, /* SMEM coding of -76 dB */
+ 0x0000002E, /* SMEM coding of -75 dB */
+ 0x00000034, /* SMEM coding of -74 dB */
+ 0x0000003A, /* SMEM coding of -73 dB */
+ 0x00000041, /* SMEM coding of -72 dB */
+ 0x00000049, /* SMEM coding of -71 dB */
+ 0x00000052, /* SMEM coding of -70 dB */
+ 0x0000005D, /* SMEM coding of -69 dB */
+ 0x00000068, /* SMEM coding of -68 dB */
+ 0x00000075, /* SMEM coding of -67 dB */
+ 0x00000083, /* SMEM coding of -66 dB */
+ 0x00000093, /* SMEM coding of -65 dB */
+ 0x000000A5, /* SMEM coding of -64 dB */
+ 0x000000B9, /* SMEM coding of -63 dB */
+ 0x000000D0, /* SMEM coding of -62 dB */
+ 0x000000E9, /* SMEM coding of -61 dB */
+ 0x00000106, /* SMEM coding of -60 dB */
+ 0x00000126, /* SMEM coding of -59 dB */
+ 0x0000014A, /* SMEM coding of -58 dB */
+ 0x00000172, /* SMEM coding of -57 dB */
+ 0x0000019F, /* SMEM coding of -56 dB */
+ 0x000001D2, /* SMEM coding of -55 dB */
+ 0x0000020B, /* SMEM coding of -54 dB */
+ 0x0000024A, /* SMEM coding of -53 dB */
+ 0x00000292, /* SMEM coding of -52 dB */
+ 0x000002E2, /* SMEM coding of -51 dB */
+ 0x0000033C, /* SMEM coding of -50 dB */
+ 0x000003A2, /* SMEM coding of -49 dB */
+ 0x00000413, /* SMEM coding of -48 dB */
+ 0x00000492, /* SMEM coding of -47 dB */
+ 0x00000521, /* SMEM coding of -46 dB */
+ 0x000005C2, /* SMEM coding of -45 dB */
+ 0x00000676, /* SMEM coding of -44 dB */
+ 0x0000073F, /* SMEM coding of -43 dB */
+ 0x00000822, /* SMEM coding of -42 dB */
+ 0x00000920, /* SMEM coding of -41 dB */
+ 0x00000A3D, /* SMEM coding of -40 dB */
+ 0x00000B7D, /* SMEM coding of -39 dB */
+ 0x00000CE4, /* SMEM coding of -38 dB */
+ 0x00000E76, /* SMEM coding of -37 dB */
+ 0x0000103A, /* SMEM coding of -36 dB */
+ 0x00001235, /* SMEM coding of -35 dB */
+ 0x0000146E, /* SMEM coding of -34 dB */
+ 0x000016EC, /* SMEM coding of -33 dB */
+ 0x000019B8, /* SMEM coding of -32 dB */
+ 0x00001CDC, /* SMEM coding of -31 dB */
+ 0x00002061, /* SMEM coding of -30 dB */
+ 0x00002455, /* SMEM coding of -29 dB */
+ 0x000028C4, /* SMEM coding of -28 dB */
+ 0x00002DBD, /* SMEM coding of -27 dB */
+ 0x00003352, /* SMEM coding of -26 dB */
+ 0x00003995, /* SMEM coding of -25 dB */
+ 0x0000409C, /* SMEM coding of -24 dB */
+ 0x0000487E, /* SMEM coding of -23 dB */
+ 0x00005156, /* SMEM coding of -22 dB */
+ 0x00005B43, /* SMEM coding of -21 dB */
+ 0x00006666, /* SMEM coding of -20 dB */
+ 0x000072E5, /* SMEM coding of -19 dB */
+ 0x000080E9, /* SMEM coding of -18 dB */
+ 0x000090A4, /* SMEM coding of -17 dB */
+ 0x0000A24B, /* SMEM coding of -16 dB */
+ 0x0000B618, /* SMEM coding of -15 dB */
+ 0x0000CC50, /* SMEM coding of -14 dB */
+ 0x0000E53E, /* SMEM coding of -13 dB */
+ 0x00010137, /* SMEM coding of -12 dB */
+ 0x0001209A, /* SMEM coding of -11 dB */
+ 0x000143D1, /* SMEM coding of -10 dB */
+ 0x00016B54, /* SMEM coding of -9 dB */
+ 0x000197A9, /* SMEM coding of -8 dB */
+ 0x0001C967, /* SMEM coding of -7 dB */
+ 0x00020137, /* SMEM coding of -6 dB */
+ 0x00023FD6, /* SMEM coding of -5 dB */
+ 0x00028619, /* SMEM coding of -4 dB */
+ 0x0002D4EF, /* SMEM coding of -3 dB */
+ 0x00032D64, /* SMEM coding of -2 dB */
+ 0x000390A4, /* SMEM coding of -1 dB */
+ 0x00040000, /* SMEM coding of 0 dB */
+ 0x00047CF2, /* SMEM coding of 1 dB */
+ 0x00050923, /* SMEM coding of 2 dB */
+ 0x0005A670, /* SMEM coding of 3 dB */
+ 0x000656EE, /* SMEM coding of 4 dB */
+ 0x00071CF5, /* SMEM coding of 5 dB */
+ 0x0007FB26, /* SMEM coding of 6 dB */
+ 0x0008F473, /* SMEM coding of 7 dB */
+ 0x000A0C2B, /* SMEM coding of 8 dB */
+ 0x000B4606, /* SMEM coding of 9 dB */
+ 0x000CA62C, /* SMEM coding of 10 dB */
+ 0x000E314A, /* SMEM coding of 11 dB */
+ 0x000FEC9E, /* SMEM coding of 12 dB */
+ 0x0011DE0A, /* SMEM coding of 13 dB */
+ 0x00140C28, /* SMEM coding of 14 dB */
+ 0x00167E60, /* SMEM coding of 15 dB */
+ 0x00193D00, /* SMEM coding of 16 dB */
+ 0x001C515D, /* SMEM coding of 17 dB */
+ 0x001FC5EB, /* SMEM coding of 18 dB */
+ 0x0023A668, /* SMEM coding of 19 dB */
+ 0x00280000, /* SMEM coding of 20 dB */
+ 0x002CE178, /* SMEM coding of 21 dB */
+ 0x00325B65, /* SMEM coding of 22 dB */
+ 0x00388062, /* SMEM coding of 23 dB */
+ 0x003F654E, /* SMEM coding of 24 dB */
+ 0x00472194, /* SMEM coding of 25 dB */
+ 0x004FCF7C, /* SMEM coding of 26 dB */
+ 0x00598C81, /* SMEM coding of 27 dB */
+ 0x006479B7, /* SMEM coding of 28 dB */
+ 0x0070BC3D, /* SMEM coding of 29 dB */
+ 0x007E7DB9, /* SMEM coding of 30 dB */
+};
+
+const u32 abe_1_alpha_iir[64] = {
+ 0x040002, 0x040002, 0x040002, 0x040002, /* 0 */
+ 0x50E955, 0x48CA65, 0x40E321, 0x72BE78, /* 1 [ms] */
+ 0x64BA68, 0x57DF14, 0x4C3D60, 0x41D690, /* 2 */
+ 0x38A084, 0x308974, 0x297B00, 0x235C7C, /* 4 */
+ 0x1E14B0, 0x198AF0, 0x15A800, 0x125660, /* 8 */
+ 0x0F82A0, 0x0D1B5C, 0x0B113C, 0x0956CC, /* 16 */
+ 0x07E054, 0x06A3B8, 0x059844, 0x04B680, /* 32 */
+ 0x03F80C, 0x035774, 0x02D018, 0x025E0C, /* 64 */
+ 0x7F8057, 0x6B482F, 0x5A4297, 0x4BEECB, /* 128 */
+ 0x3FE00B, 0x35BAA7, 0x2D3143, 0x2602AF, /* 256 */
+ 0x1FF803, 0x1AE2FB, 0x169C9F, 0x13042B, /* 512 */
+ 0x0FFE03, 0x0D72E7, 0x0B4F4F, 0x0982CB, /* 1.024 [s] */
+ 0x07FF83, 0x06B9CF, 0x05A7E7, 0x04C193, /* 2.048 */
+ 0x03FFE3, 0x035CFF, 0x02D403, 0x0260D7, /* 4.096 */
+ 0x01FFFB, 0x01AE87, 0x016A07, 0x01306F, /* 8.192 */
+ 0x00FFFF, 0x00D743, 0x00B503, 0x009837,
+};
+
+const u32 abe_alpha_iir[64] = {
+ 0x000000, 0x000000, 0x000000, 0x000000, /* 0 */
+ 0x5E2D58, 0x6E6B3C, 0x7E39C0, 0x46A0C5, /* 1 [ms] */
+ 0x4DA2CD, 0x541079, 0x59E151, 0x5F14B9, /* 2 */
+ 0x63AFC1, 0x67BB45, 0x6B4281, 0x6E51C1, /* 4 */
+ 0x70F5A9, 0x733A89, 0x752C01, 0x76D4D1, /* 8 */
+ 0x783EB1, 0x797251, 0x7A7761, 0x7B549D, /* 16 */
+ 0x7C0FD5, 0x7CAE25, 0x7D33DD, 0x7DA4C1, /* 32 */
+ 0x7E03FD, 0x7E5449, 0x7E97F5, 0x7ED0F9, /* 64 */
+ 0x7F0101, 0x7F2971, 0x7F4B7D, 0x7F6825, /* 128 */
+ 0x7F8041, 0x7F948D, 0x7FA59D, 0x7FB3FD, /* 256 */
+ 0x7FC011, 0x7FCA3D, 0x7FD2C9, 0x7FD9F9, /* 512 */
+ 0x7FE005, 0x7FE51D, 0x7FE961, 0x7FECFD, /* 1.024 [s] */
+ 0x7FF001, 0x7FF28D, 0x7FF4B1, 0x7FF67D, /* 2.048 */
+ 0x7FF801, 0x7FF949, 0x7FFA59, 0x7FFB41, /* 4.096 */
+ 0x7FFC01, 0x7FFCA5, 0x7FFD2D, 0x7FFDA1, /* 8.192 */
+ 0x7FFE01, 0x7FFE51, 0x7FFE95, 0x7FFED1,
+};
+
+/**
+ * abe_int_2_float
+ * returns a mantissa on 16 bits and the exponent
+ * 0x4000.0000 leads to M=0x4000 X=15
+ * 0x0004.0000 leads to M=0x4000 X=4
+ * 0x0000.0001 leads to M=0x4000 X=-14
+ *
+ */
+void abe_int_2_float16(u32 data, u32 *mantissa, u32 *exp)
+{
+ u32 i;
+ *exp = 0;
+ *mantissa = 0;
+ for (i = 0; i < 32; i++) {
+ if ((1 << i) > data)
+ break;
+ }
+ *exp = i - 15;
+ *mantissa = (*exp > 0) ? data >> (*exp) : data << (*exp);
+}
+
+/**
+ * abe_use_compensated_gain
+ * @on_off:
+ *
+ * Selects the automatic Mixer's gain management
+ * on_off = 1 allows the "abe_write_gain" to adjust the overall
+ * gains of the mixer to be tuned not to create saturation
+ */
+int omap_abe_use_compensated_gain(struct omap_abe *abe, int on_off)
+{
+ abe->compensated_mixer_gain = on_off;
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_use_compensated_gain);
+
+/**
+ * omap_abe_gain_offset
+ * returns the offset to firmware data structures
+ *
+ */
+void omap_abe_gain_offset(struct omap_abe *abe, u32 id, u32 *mixer_offset)
+{
+ switch (id) {
+ default:
+ case GAINS_DMIC1:
+ *mixer_offset = dmic1_gains_offset;
+ break;
+ case GAINS_DMIC2:
+ *mixer_offset = dmic2_gains_offset;
+ break;
+ case GAINS_DMIC3:
+ *mixer_offset = dmic3_gains_offset;
+ break;
+ case GAINS_AMIC:
+ *mixer_offset = amic_gains_offset;
+ break;
+ case GAINS_DL1:
+ *mixer_offset = dl1_gains_offset;
+ break;
+ case GAINS_DL2:
+ *mixer_offset = dl2_gains_offset;
+ break;
+ case GAINS_SPLIT:
+ *mixer_offset = splitters_gains_offset;
+ break;
+ case MIXDL1:
+ *mixer_offset = mixer_dl1_offset;
+ break;
+ case MIXDL2:
+ *mixer_offset = mixer_dl2_offset;
+ break;
+ case MIXECHO:
+ *mixer_offset = mixer_echo_offset;
+ break;
+ case MIXSDT:
+ *mixer_offset = mixer_sdt_offset;
+ break;
+ case MIXVXREC:
+ *mixer_offset = mixer_vxrec_offset;
+ break;
+ case MIXAUDUL:
+ *mixer_offset = mixer_audul_offset;
+ break;
+ case GAINS_BTUL:
+ *mixer_offset = btul_gains_offset;
+ break;
+ }
+}
+
+/**
+ * oamp_abe_write_equalizer
+ * @abe: Pointer on abe handle
+ * @id: name of the equalizer
+ * @param : equalizer coefficients
+ *
+ * Load the coefficients in CMEM.
+ */
+int omap_abe_write_equalizer(struct omap_abe *abe,
+ u32 id, struct omap_abe_equ *param)
+{
+ u32 eq_offset, length, *src, eq_mem, eq_mem_len;
+ _log(ABE_ID_WRITE_EQUALIZER, id, 0, 0);
+ switch (id) {
+ default:
+ case EQ1:
+ eq_offset = OMAP_ABE_C_DL1_COEFS_ADDR;
+ eq_mem = OMAP_ABE_S_DL1_M_EQ_DATA_ADDR;
+ eq_mem_len = OMAP_ABE_S_DL1_M_EQ_DATA_SIZE;
+ break;
+ case EQ2L:
+ eq_offset = OMAP_ABE_C_DL2_L_COEFS_ADDR;
+ eq_mem = OMAP_ABE_S_DL2_M_LR_EQ_DATA_ADDR;
+ eq_mem_len = OMAP_ABE_S_DL2_M_LR_EQ_DATA_SIZE;
+ break;
+ case EQ2R:
+ eq_offset = OMAP_ABE_C_DL2_R_COEFS_ADDR;
+ eq_mem = OMAP_ABE_S_DL2_M_LR_EQ_DATA_ADDR;
+ eq_mem_len = OMAP_ABE_S_DL2_M_LR_EQ_DATA_SIZE;
+ break;
+ case EQSDT:
+ eq_offset = OMAP_ABE_C_SDT_COEFS_ADDR;
+ eq_mem = OMAP_ABE_S_SDT_F_DATA_ADDR;
+ eq_mem_len = OMAP_ABE_S_SDT_F_DATA_SIZE;
+ break;
+ case EQAMIC:
+ eq_offset = OMAP_ABE_C_96_48_AMIC_COEFS_ADDR;
+ eq_mem = OMAP_ABE_S_AMIC_96_48_DATA_ADDR;
+ eq_mem_len = OMAP_ABE_S_AMIC_96_48_DATA_SIZE;
+ break;
+ case EQDMIC:
+ eq_offset = OMAP_ABE_C_96_48_DMIC_COEFS_ADDR;
+ eq_mem = OMAP_ABE_S_DMIC0_96_48_DATA_ADDR;
+ eq_mem_len = OMAP_ABE_S_DMIC0_96_48_DATA_SIZE;
+ /* three DMIC are clear at the same time DMIC0 DMIC1 DMIC2 */
+ eq_mem_len *= 3;
+ break;
+ }
+ /* reset SMEM buffers before the coefficients are loaded */
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM, eq_mem, eq_mem_len);
+
+ length = param->equ_length;
+ src = (u32 *) ((param->coef).type1);
+ /* translate in bytes */
+ length <<= 2;
+ omap_abe_mem_write(abe, OMAP_ABE_CMEM, eq_offset, src, length);
+
+ /* reset SMEM buffers after the coefficients are loaded */
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM, eq_mem, eq_mem_len);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_write_equalizer);
+
+/**
+ * omap_abe_disable_gain
+ * @abe: Pointer on abe handle
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+int omap_abe_disable_gain(struct omap_abe *abe, u32 id, u32 p)
+{
+ u32 mixer_offset, f_g, ramp;
+ omap_abe_gain_offset(abe, id, &mixer_offset);
+ /* save the input parameters for mute/unmute */
+ ramp = abe->desired_ramp_delay_ms[mixer_offset + p];
+ f_g = GAIN_MUTE;
+ if (!(abe->muted_gains_indicator[mixer_offset + p] &
+ OMAP_ABE_GAIN_DISABLED)) {
+ /* Check if we are in mute */
+ if (!(abe->muted_gains_indicator[mixer_offset + p] &
+ OMAP_ABE_GAIN_MUTED)) {
+ abe->muted_gains_decibel[mixer_offset + p] =
+ abe->desired_gains_decibel[mixer_offset + p];
+ /* mute the gain */
+ omap_abe_write_gain(abe, id, f_g, ramp, p);
+ }
+ abe->muted_gains_indicator[mixer_offset + p] |=
+ OMAP_ABE_GAIN_DISABLED;
+ }
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_disable_gain);
+
+/**
+ * omap_abe_enable_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+int omap_abe_enable_gain(struct omap_abe *abe, u32 id, u32 p)
+{
+ u32 mixer_offset, f_g, ramp;
+ omap_abe_gain_offset(abe, id, &mixer_offset);
+ if ((abe->muted_gains_indicator[mixer_offset + p] &
+ OMAP_ABE_GAIN_DISABLED)) {
+ /* restore the input parameters for mute/unmute */
+ f_g = abe->muted_gains_decibel[mixer_offset + p];
+ ramp = abe->desired_ramp_delay_ms[mixer_offset + p];
+ abe->muted_gains_indicator[mixer_offset + p] &=
+ ~OMAP_ABE_GAIN_DISABLED;
+ /* unmute the gain */
+ omap_abe_write_gain(abe, id, f_g, ramp, p);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_enable_gain);
+/**
+ * omap_abe_mute_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+int omap_abe_mute_gain(struct omap_abe *abe, u32 id, u32 p)
+{
+ u32 mixer_offset, f_g, ramp;
+ omap_abe_gain_offset(abe, id, &mixer_offset);
+ /* save the input parameters for mute/unmute */
+ ramp = abe->desired_ramp_delay_ms[mixer_offset + p];
+ f_g = GAIN_MUTE;
+ if (!abe->muted_gains_indicator[mixer_offset + p]) {
+ abe->muted_gains_decibel[mixer_offset + p] =
+ abe->desired_gains_decibel[mixer_offset + p];
+ /* mute the gain */
+ omap_abe_write_gain(abe, id, f_g, ramp, p);
+ }
+ abe->muted_gains_indicator[mixer_offset + p] |= OMAP_ABE_GAIN_MUTED;
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_mute_gain);
+/**
+ * omap_abe_unmute_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+int omap_abe_unmute_gain(struct omap_abe *abe, u32 id, u32 p)
+{
+ u32 mixer_offset, f_g, ramp;
+ omap_abe_gain_offset(abe, id, &mixer_offset);
+ if ((abe->muted_gains_indicator[mixer_offset + p] &
+ OMAP_ABE_GAIN_MUTED)) {
+ /* restore the input parameters for mute/unmute */
+ f_g = abe->muted_gains_decibel[mixer_offset + p];
+ ramp = abe->desired_ramp_delay_ms[mixer_offset + p];
+ abe->muted_gains_indicator[mixer_offset + p] &=
+ ~OMAP_ABE_GAIN_MUTED;
+ /* unmute the gain */
+ omap_abe_write_gain(abe, id, f_g, ramp, p);
+ }
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_unmute_gain);
+
+/**
+ * omap_abe_write_gain
+ * @id: gain name or mixer name
+ * @f_g: list of input gains of the mixer
+ * @ramp: gain ramp speed factor
+ * @p: list of ports corresponding to the above gains
+ *
+ * Loads the gain coefficients to FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's gain
+ * in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+int omap_abe_write_gain(struct omap_abe *abe,
+ u32 id, s32 f_g, u32 ramp, u32 p)
+{
+ u32 lin_g, sum_g, mixer_target, mixer_offset, i, mean_gain, mean_exp;
+ u32 new_gain_linear[4];
+ s32 gain_index;
+ u32 alpha, beta;
+ u32 ramp_index;
+
+ _log(ABE_ID_WRITE_GAIN, id, f_g, p);
+ gain_index = ((f_g - min_mdb) / 100);
+ gain_index = maximum(gain_index, 0);
+ gain_index = minimum(gain_index, sizeof_db2lin_table);
+ lin_g = abe_db2lin_table[gain_index];
+ omap_abe_gain_offset(abe, id, &mixer_offset);
+ /* save the input parameters for mute/unmute */
+ abe->desired_gains_linear[mixer_offset + p] = lin_g;
+ abe->desired_gains_decibel[mixer_offset + p] = f_g;
+ abe->desired_ramp_delay_ms[mixer_offset + p] = ramp;
+ /* SMEM address in bytes */
+ mixer_target = OMAP_ABE_S_GTARGET1_ADDR;
+ mixer_target += (mixer_offset<<2);
+ mixer_target += (p<<2);
+
+ if (abe->compensated_mixer_gain) {
+ switch (id) {
+ case MIXDL1:
+ case MIXDL2:
+ case MIXVXREC:
+ case MIXAUDUL:
+ /* compute the sum of the gain of the mixer */
+ for (sum_g = i = 0; i < 4; i++)
+ sum_g += abe->desired_gains_linear[mixer_offset +
+ i];
+ /* lets avoid a division by 0 */
+ if (sum_g == 0)
+ break;
+ /* if the sum is OK with less than 1, then
+ do not weight the gains */
+ if (sum_g < 0x00040000) { /* REMOVE HARD CONST */
+ /* recompute all gains from original
+ desired values */
+ sum_g = 0x00040000;
+ }
+ /* translate it in Q16 format for the later division */
+ abe_int_2_float16(sum_g, &mean_gain, &mean_exp);
+ mean_exp = 10 - mean_exp;
+ for (i = 0; i < 4; i++) {
+ /* new gain = desired gain divided by sum of gains */
+ new_gain_linear[i] =
+ (abe->desired_gains_linear
+ [mixer_offset + i]
+ << 8) / mean_gain;
+ new_gain_linear[i] = (mean_exp > 0) ?
+ new_gain_linear[i] << mean_exp :
+ new_gain_linear[i] >> mean_exp;
+ }
+ /* load the whole adpated S_G_Target SMEM MIXER table */
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM,
+ mixer_target - (p << 2),
+ new_gain_linear, (4 * sizeof(lin_g)));
+ break;
+ default:
+ /* load the S_G_Target SMEM table */
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM,
+ mixer_target,
+ (u32 *) &lin_g, sizeof(lin_g));
+ break;
+ }
+ } else {
+ if (!abe->muted_gains_indicator[mixer_offset + p])
+ /* load the S_G_Target SMEM table */
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM,
+ mixer_target, (u32 *) &lin_g,
+ sizeof(lin_g));
+ else
+ /* update muted gain with new value */
+ abe->muted_gains_decibel[mixer_offset + p] = f_g;
+ }
+ ramp = maximum(minimum(RAMP_MAXLENGTH, ramp), RAMP_MINLENGTH);
+ /* ramp data should be interpolated in the table instead */
+ ramp_index = 3;
+ if ((RAMP_2MS <= ramp) && (ramp < RAMP_5MS))
+ ramp_index = 8;
+ if ((RAMP_5MS <= ramp) && (ramp < RAMP_50MS))
+ ramp_index = 24;
+ if ((RAMP_50MS <= ramp) && (ramp < RAMP_500MS))
+ ramp_index = 36;
+ if (ramp > RAMP_500MS)
+ ramp_index = 48;
+ beta = abe_alpha_iir[ramp_index];
+ alpha = abe_1_alpha_iir[ramp_index];
+ /* CMEM bytes address */
+ mixer_target = OMAP_ABE_C_1_ALPHA_ADDR;
+ /* a pair of gains is updated once in the firmware */
+ mixer_target += ((p + mixer_offset) >> 1) << 2;
+ /* load the ramp delay data */
+ omap_abe_mem_write(abe, OMAP_ABE_CMEM, mixer_target,
+ (u32 *) &alpha, sizeof(alpha));
+ /* CMEM bytes address */
+ mixer_target = OMAP_ABE_C_ALPHA_ADDR;
+ /* a pair of gains is updated once in the firmware */
+ mixer_target += ((p + mixer_offset) >> 1) << 2;
+ omap_abe_mem_write(abe, OMAP_ABE_CMEM, mixer_target,
+ (u32 *) &beta, sizeof(beta));
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_write_gain);
+/**
+ * omap_abe_write_mixer
+ * @id: name of the mixer
+ * @param: input gains and delay ramp of the mixer
+ * @p: port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's
+ * gain in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+int omap_abe_write_mixer(struct omap_abe *abe,
+ u32 id, s32 f_g, u32 f_ramp, u32 p)
+{
+ _log(ABE_ID_WRITE_MIXER, id, f_ramp, p);
+ omap_abe_write_gain(abe, id, f_g, f_ramp, p);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_write_mixer);
+
+/**
+ * omap_abe_read_gain
+ * @id: name of the mixer
+ * @param: list of input gains of the mixer
+ * @p: list of port corresponding to the above gains
+ *
+ */
+int omap_abe_read_gain(struct omap_abe *abe,
+ u32 id, u32 *f_g, u32 p)
+{
+ u32 mixer_target, mixer_offset, i;
+ _log(ABE_ID_READ_GAIN, id, (u32) f_g, p);
+ omap_abe_gain_offset(abe, id, &mixer_offset);
+ /* SMEM bytes address */
+ mixer_target = OMAP_ABE_S_GTARGET1_ADDR;
+ mixer_target += (mixer_offset<<2);
+ mixer_target += (p<<2);
+ if (!abe->muted_gains_indicator[mixer_offset + p]) {
+ /* load the S_G_Target SMEM table */
+ omap_abe_mem_read(abe, OMAP_ABE_SMEM, mixer_target,
+ (u32 *) f_g, sizeof(*f_g));
+ for (i = 0; i < sizeof_db2lin_table; i++) {
+ if (abe_db2lin_table[i] == *f_g)
+ goto found;
+ }
+ *f_g = 0;
+ return -1;
+ found:
+ *f_g = (i * 100) + min_mdb;
+ } else {
+ /* update muted gain with new value */
+ *f_g = abe->muted_gains_decibel[mixer_offset + p];
+ }
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_read_gain);
+
+/**
+ * abe_read_mixer
+ * @id: name of the mixer
+ * @param: gains of the mixer
+ * @p: port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's
+ * gain in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+int omap_abe_read_mixer(struct omap_abe *abe,
+ u32 id, u32 *f_g, u32 p)
+{
+ _log(ABE_ID_READ_MIXER, id, 0, p);
+ omap_abe_read_gain(abe, id, f_g, p);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_read_mixer);
+
+/**
+ * abe_reset_gain_mixer
+ * @id: name of the mixer
+ * @p: list of port corresponding to the above gains
+ *
+ * restart the working gain value of the mixers when a port is enabled
+ */
+void omap_abe_reset_gain_mixer(struct omap_abe *abe, u32 id, u32 p)
+{
+ u32 lin_g, mixer_target, mixer_offset;
+ switch (id) {
+ default:
+ case GAINS_DMIC1:
+ mixer_offset = dmic1_gains_offset;
+ break;
+ case GAINS_DMIC2:
+ mixer_offset = dmic2_gains_offset;
+ break;
+ case GAINS_DMIC3:
+ mixer_offset = dmic3_gains_offset;
+ break;
+ case GAINS_AMIC:
+ mixer_offset = amic_gains_offset;
+ break;
+ case GAINS_DL1:
+ mixer_offset = dl1_gains_offset;
+ break;
+ case GAINS_DL2:
+ mixer_offset = dl2_gains_offset;
+ break;
+ case GAINS_SPLIT:
+ mixer_offset = splitters_gains_offset;
+ break;
+ case MIXDL1:
+ mixer_offset = mixer_dl1_offset;
+ break;
+ case MIXDL2:
+ mixer_offset = mixer_dl2_offset;
+ break;
+ case MIXECHO:
+ mixer_offset = mixer_echo_offset;
+ break;
+ case MIXSDT:
+ mixer_offset = mixer_sdt_offset;
+ break;
+ case MIXVXREC:
+ mixer_offset = mixer_vxrec_offset;
+ break;
+ case MIXAUDUL:
+ mixer_offset = mixer_audul_offset;
+ break;
+ case GAINS_BTUL:
+ mixer_offset = btul_gains_offset;
+ break;
+ }
+ /* SMEM bytes address for the CURRENT gain values */
+ mixer_target = OMAP_ABE_S_GCURRENT_ADDR;
+ mixer_target += (mixer_offset<<2);
+ mixer_target += (p<<2);
+ lin_g = 0;
+ /* load the S_G_Target SMEM table */
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, mixer_target,
+ (u32 *) &lin_g, sizeof(lin_g));
+}
diff --git a/sound/soc/omap/abe/abe_gain.h b/sound/soc/omap/abe/abe_gain.h
new file mode 100644
index 0000000..f332837
--- /dev/null
+++ b/sound/soc/omap/abe/abe_gain.h
@@ -0,0 +1,111 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_GAIN_H_
+#define _ABE_GAIN_H_
+
+#include "abe_typ.h"
+#include "abe_dm_addr.h"
+#include "abe_sm_addr.h"
+#include "abe_cm_addr.h"
+
+#define OMAP_ABE_GAIN_MUTED (0x0001<<0)
+#define OMAP_ABE_GAIN_DISABLED (0x0001<<1)
+
+#define OMAP_ABE_GAIN_DMIC1_LEFT 0
+#define OMAP_ABE_GAIN_DMIC1_RIGTH 1
+#define OMAP_ABE_GAIN_DMIC2_LEFT 2
+#define OMAP_ABE_GAIN_DMIC2_RIGTH 3
+#define OMAP_ABE_GAIN_DMIC3_LEFT 4
+#define OMAP_ABE_GAIN_DMIC3_RIGTH 5
+#define OMAP_ABE_GAIN_AMIC_LEFT 6
+#define OMAP_ABE_GAIN_AMIC_RIGTH 7
+#define OMAP_ABE_GAIN_DL1_LEFT 8
+#define OMAP_ABE_GAIN_DL1_RIGTH 9
+#define OMAP_ABE_GAIN_DL2_LEFT 10
+#define OMAP_ABE_GAIN_DL2_RIGTH 11
+#define OMAP_ABE_GAIN_SPLIT_LEFT 12
+#define OMAP_ABE_GAIN_SPLIT_RIGTH 13
+#define OMAP_ABE_MIXDL1_MM_DL 14
+#define OMAP_ABE_MIXDL1_MM_UL2 15
+#define OMAP_ABE_MIXDL1_VX_DL 16
+#define OMAP_ABE_MIXDL1_TONES 17
+#define OMAP_ABE_MIXDL2_MM_DL 18
+#define OMAP_ABE_MIXDL2_MM_UL2 19
+#define OMAP_ABE_MIXDL2_VX_DL 20
+#define OMAP_ABE_MIXDL2_TONES 21
+#define OMAP_ABE_MIXECHO_DL1 22
+#define OMAP_ABE_MIXECHO_DL2 23
+#define OMAP_ABE_MIXSDT_UL 24
+#define OMAP_ABE_MIXECHO_DL 25
+#define OMAP_ABE_MIXVXREC_MM_DL 26
+#define OMAP_ABE_MIXVXREC_TONES 27
+#define OMAP_ABE_MIXVXREC_VX_UL 28
+#define OMAP_ABE_MIXVXREC_VX_DL 29
+#define OMAP_ABE_MIXAUDUL_MM_DL 30
+#define OMAP_ABE_MIXAUDUL_TONES 31
+#define OMAP_ABE_MIXAUDUL_UPLINK 32
+#define OMAP_ABE_MIXAUDUL_VX_DL 33
+#define OMAP_ABE_GAIN_BTUL_LEFT 34
+#define OMAP_ABE_GAIN_BTUL_RIGTH 35
+
+void omap_abe_reset_gain_mixer(struct omap_abe *abe, u32 id, u32 p);
+
+void abe_int_2_float16(u32 data, u32 *mantissa, u32 *exp);
+
+#endif /* _ABE_GAIN_H_ */
diff --git a/sound/soc/omap/abe/abe_ini.c b/sound/soc/omap/abe/abe_ini.c
new file mode 100644
index 0000000..288a3d3
--- /dev/null
+++ b/sound/soc/omap/abe/abe_ini.c
@@ -0,0 +1,447 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_dbg.h"
+#include "abe.h"
+#include "abe_aess.h"
+#include "abe_gain.h"
+#include "abe_mem.h"
+#include "abe_port.h"
+#include "abe_seq.h"
+
+#include "abe_taskid.h"
+
+
+#define ABE_TASK_ID(ID) (OMAP_ABE_D_TASKSLIST_ADDR + sizeof(ABE_STask)*(ID))
+void omap_abe_build_scheduler_table(struct omap_abe *abe);
+void omap_abe_reset_all_ports(struct omap_abe *abe);
+
+const u32 abe_firmware_array[ABE_FIRMWARE_MAX_SIZE] = {
+#include "abe_firmware.c"
+};
+
+
+/*
+ * initialize the default values for call-backs to subroutines
+ * - FIFO IRQ call-backs for sequenced tasks
+ * - FIFO IRQ call-backs for audio player/recorders (ping-pong protocols)
+ * - Remote debugger interface
+ * - Error monitoring
+ * - Activity Tracing
+ */
+
+/**
+ * abe_init_mem - Allocate Kernel space memory map for ABE
+ *
+ * Memory map of ABE memory space for PMEM/DMEM/SMEM/DMEM
+ */
+void abe_init_mem(void __iomem **_io_base)
+{
+ int i;
+
+ abe = kzalloc(sizeof(struct omap_abe), GFP_KERNEL);
+ if (abe == NULL)
+ printk(KERN_ERR "ABE Allocation ERROR ");
+
+ for (i = 0; i < 5; i++)
+ abe->io_base[i] = _io_base[i];
+
+ mutex_init(&abe->mutex);
+
+}
+EXPORT_SYMBOL(abe_init_mem);
+
+/**
+ * abe_load_fw_param - Load ABE Firmware memories
+ * @PMEM: Pointer of Program memory data
+ * @PMEM_SIZE: Size of PMEM data
+ * @CMEM: Pointer of Coeffients memory data
+ * @CMEM_SIZE: Size of CMEM data
+ * @SMEM: Pointer of Sample memory data
+ * @SMEM_SIZE: Size of SMEM data
+ * @DMEM: Pointer of Data memory data
+ * @DMEM_SIZE: Size of DMEM data
+ *
+ */
+int abe_load_fw_param(u32 *ABE_FW)
+{
+ u32 pmem_size, dmem_size, smem_size, cmem_size;
+ u32 *pmem_ptr, *dmem_ptr, *smem_ptr, *cmem_ptr, *fw_ptr;
+ _log(ABE_ID_LOAD_FW_param, 0, 0, 0);
+#define ABE_FW_OFFSET 5
+ fw_ptr = ABE_FW;
+ abe->firmware_version_number = *fw_ptr++;
+ pmem_size = *fw_ptr++;
+ cmem_size = *fw_ptr++;
+ dmem_size = *fw_ptr++;
+ smem_size = *fw_ptr++;
+ pmem_ptr = fw_ptr;
+ cmem_ptr = pmem_ptr + (pmem_size >> 2);
+ dmem_ptr = cmem_ptr + (cmem_size >> 2);
+ smem_ptr = dmem_ptr + (dmem_size >> 2);
+ /* do not load PMEM */
+ if (abe->warm_boot) {
+ /* Stop the event Generator */
+ omap_abe_stop_event_generator(abe);
+
+ /* Now we are sure the firmware is stalled */
+ omap_abe_mem_write(abe, OMAP_ABE_CMEM, 0, cmem_ptr,
+ cmem_size);
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, 0, smem_ptr,
+ smem_size);
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, 0, dmem_ptr,
+ dmem_size);
+ /* Restore the event Generator status */
+ omap_abe_start_event_generator(abe);
+ } else {
+ omap_abe_mem_write(abe, OMAP_ABE_PMEM, 0, pmem_ptr,
+ pmem_size);
+ omap_abe_mem_write(abe, OMAP_ABE_CMEM, 0, cmem_ptr,
+ cmem_size);
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, 0, smem_ptr,
+ smem_size);
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, 0, dmem_ptr,
+ dmem_size);
+ }
+ abe->warm_boot = 1;
+ return 0;
+}
+EXPORT_SYMBOL(abe_load_fw_param);
+
+/**
+ * omap_abe_load_fw - Load ABE Firmware and initialize memories
+ * @abe: Pointer on abe handle
+ *
+ */
+int omap_abe_load_fw(struct omap_abe *abe, u32 *firmware)
+{
+ _log(ABE_ID_LOAD_FW, 0, 0, 0);
+ abe_load_fw_param(firmware);
+ omap_abe_reset_all_ports(abe);
+ omap_abe_build_scheduler_table(abe);
+ omap_abe_reset_all_sequence(abe);
+ omap_abe_select_main_port(OMAP_ABE_PDM_DL_PORT);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_load_fw);
+
+/**
+ * abe_reload_fw - Reload ABE Firmware after OFF mode
+ */
+int omap_abe_reload_fw(struct omap_abe *abe, u32 *firmware)
+{
+ abe->warm_boot = 0;
+ abe_load_fw_param(firmware);
+ omap_abe_build_scheduler_table(abe);
+ omap_abe_dbg_reset(&abe->dbg);
+ /* IRQ circular read pointer in DMEM */
+ abe->irq_dbg_read_ptr = 0;
+ /* Restore Gains not managed by the drivers */
+ omap_abe_write_gain(abe, GAINS_SPLIT, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_SPLIT, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL1, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL1, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL2, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL2, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_reload_fw);
+
+/**
+ * omap_abe_get_default_fw
+ *
+ * Get default ABE firmware
+ */
+u32 *omap_abe_get_default_fw(struct omap_abe *abe)
+{
+ return (u32 *)abe_firmware_array;
+}
+
+/**
+ * abe_build_scheduler_table
+ *
+ */
+void omap_abe_build_scheduler_table(struct omap_abe *abe)
+{
+ u16 i, n;
+ u8 *ptr;
+ u16 aUplinkMuxing[NBROUTE_UL];
+
+ /* LOAD OF THE TASKS' MULTIFRAME */
+ /* WARNING ON THE LOCATION OF IO_MM_DL WHICH IS PATCHED
+ IN "abe_init_io_tasks" */
+ for (ptr = (u8 *) &(abe->MultiFrame[0][0]), i = 0;
+ i < sizeof(abe->MultiFrame); i++)
+ *ptr++ = 0;
+
+ abe->MultiFrame[0][2] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_VX_DL)*/;
+ abe->MultiFrame[0][3] = ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_DL_8);
+
+ abe->MultiFrame[1][3] = ABE_TASK_ID(C_ABE_FW_TASK_VX_DL_8_48_FIR);
+ abe->MultiFrame[1][6] = ABE_TASK_ID(C_ABE_FW_TASK_DL2Mixer);
+ abe->MultiFrame[1][7] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_VIB_DL)*/;
+
+ abe->MultiFrame[2][0] = ABE_TASK_ID(C_ABE_FW_TASK_DL1Mixer);
+ abe->MultiFrame[2][1] = ABE_TASK_ID(C_ABE_FW_TASK_SDTMixer);
+ abe->MultiFrame[2][5] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_DMIC)*/;
+
+ abe->MultiFrame[3][0] = ABE_TASK_ID(C_ABE_FW_TASK_DL1_GAIN);
+ abe->MultiFrame[3][6] = ABE_TASK_ID(C_ABE_FW_TASK_DL2_GAIN);
+ abe->MultiFrame[3][7] = ABE_TASK_ID(C_ABE_FW_TASK_DL2_EQ);
+
+ abe->MultiFrame[4][0] = ABE_TASK_ID(C_ABE_FW_TASK_DL1_EQ);
+ abe->MultiFrame[4][2] = ABE_TASK_ID(C_ABE_FW_TASK_VXRECMixer);
+ abe->MultiFrame[4][3] = ABE_TASK_ID(C_ABE_FW_TASK_VXREC_SPLIT);
+ abe->MultiFrame[4][6] = ABE_TASK_ID(C_ABE_FW_TASK_VIBRA1);
+ abe->MultiFrame[4][7] = ABE_TASK_ID(C_ABE_FW_TASK_VIBRA2);
+
+ abe->MultiFrame[5][0] = 0;
+ abe->MultiFrame[5][1] = ABE_TASK_ID(C_ABE_FW_TASK_EARP_48_96_LP);
+ abe->MultiFrame[5][2] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_PDM_UL)*/;
+ abe->MultiFrame[5][7] = ABE_TASK_ID(C_ABE_FW_TASK_VIBRA_SPLIT);
+
+ abe->MultiFrame[6][0] = ABE_TASK_ID(C_ABE_FW_TASK_EARP_48_96_LP);
+ abe->MultiFrame[6][4] = ABE_TASK_ID(C_ABE_FW_TASK_EchoMixer);
+ abe->MultiFrame[6][5] = ABE_TASK_ID(C_ABE_FW_TASK_BT_UL_SPLIT);
+
+ abe->MultiFrame[7][0] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_PDM_DL)*/;
+ abe->MultiFrame[7][3] = ABE_TASK_ID(C_ABE_FW_TASK_DBG_SYNC);
+ abe->MultiFrame[7][5] = ABE_TASK_ID(C_ABE_FW_TASK_ECHO_REF_SPLIT);
+
+ abe->MultiFrame[8][2] = ABE_TASK_ID(C_ABE_FW_TASK_DMIC1_96_48_LP);
+ abe->MultiFrame[8][4] = ABE_TASK_ID(C_ABE_FW_TASK_DMIC1_SPLIT);
+
+ abe->MultiFrame[9][2] = ABE_TASK_ID(C_ABE_FW_TASK_DMIC2_96_48_LP);
+ abe->MultiFrame[9][4] = ABE_TASK_ID(C_ABE_FW_TASK_DMIC2_SPLIT);
+ abe->MultiFrame[9][6] = 0;
+ abe->MultiFrame[9][7] = ABE_TASK_ID(C_ABE_FW_TASK_IHF_48_96_LP);
+
+ abe->MultiFrame[10][2] = ABE_TASK_ID(C_ABE_FW_TASK_DMIC3_96_48_LP);
+ abe->MultiFrame[10][4] = ABE_TASK_ID(C_ABE_FW_TASK_DMIC3_SPLIT);
+ abe->MultiFrame[10][7] = ABE_TASK_ID(C_ABE_FW_TASK_IHF_48_96_LP);
+
+ abe->MultiFrame[11][2] = ABE_TASK_ID(C_ABE_FW_TASK_AMIC_96_48_LP);
+ abe->MultiFrame[11][4] = ABE_TASK_ID(C_ABE_FW_TASK_AMIC_SPLIT);
+ abe->MultiFrame[11][7] = ABE_TASK_ID(C_ABE_FW_TASK_VIBRA_PACK);
+
+ abe->MultiFrame[12][3] = ABE_TASK_ID(C_ABE_FW_TASK_VX_UL_ROUTING);
+ abe->MultiFrame[12][4] = ABE_TASK_ID(C_ABE_FW_TASK_ULMixer);
+ abe->MultiFrame[12][5] = ABE_TASK_ID(C_ABE_FW_TASK_VX_UL_48_8);
+
+ abe->MultiFrame[13][2] = ABE_TASK_ID(C_ABE_FW_TASK_MM_UL2_ROUTING);
+ abe->MultiFrame[13][3] = ABE_TASK_ID(C_ABE_FW_TASK_SideTone);
+ abe->MultiFrame[13][5] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_BT_VX_DL)*/;
+
+ abe->MultiFrame[14][3] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_DMIC)*/;
+ abe->MultiFrame[14][4] = ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_8);
+
+ abe->MultiFrame[15][0] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_EXT_OUT)*/;
+ abe->MultiFrame[15][3] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_BT_VX_UL)*/;
+ abe->MultiFrame[15][6] = ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_UL_8);
+
+ abe->MultiFrame[16][2] = ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_UL_8);
+ abe->MultiFrame[16][3] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_VX_UL)*/;
+
+ abe->MultiFrame[17][2] = ABE_TASK_ID(C_ABE_FW_TASK_BT_UL_8_48);
+ abe->MultiFrame[17][3] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_UL2)*/;
+
+ abe->MultiFrame[18][0] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_DL)*/;
+ abe->MultiFrame[18][6] = ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_DL_8);
+
+ abe->MultiFrame[19][0] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_PDM_DL)*/;
+
+ /* MM_UL is moved to OPP 100% */
+ abe->MultiFrame[19][6] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_UL)*/;
+
+ abe->MultiFrame[20][0] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_TONES_DL)*/;
+ abe->MultiFrame[20][6] = ABE_TASK_ID(C_ABE_FW_TASK_ASRC_MM_EXT_IN);
+
+ abe->MultiFrame[21][1] = ABE_TASK_ID(C_ABE_FW_TASK_DEBUGTRACE_VX_ASRCs);
+ abe->MultiFrame[21][3] = 0/*ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_EXT_IN)*/;
+ /* MUST STAY ON SLOT 22 */
+ abe->MultiFrame[22][0] = ABE_TASK_ID(C_ABE_FW_TASK_DEBUG_IRQFIFO);
+ abe->MultiFrame[22][1] = ABE_TASK_ID(C_ABE_FW_TASK_INIT_FW_MEMORY);
+ abe->MultiFrame[22][2] = 0;
+ /* MM_EXT_IN_SPLIT task must be after IO_MM_EXT_IN and before
+ ASRC_MM_EXT_IN in order to manage OPP50 <-> transitions */
+ abe->MultiFrame[22][4] = ABE_TASK_ID(C_ABE_FW_TASK_MM_EXT_IN_SPLIT);
+
+ abe->MultiFrame[23][0] = ABE_TASK_ID(C_ABE_FW_TASK_GAIN_UPDATE);
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_MULTIFRAME_ADDR,
+ (u32 *) abe->MultiFrame, sizeof(abe->MultiFrame));
+ /* reset the uplink router */
+ n = (OMAP_ABE_D_AUPLINKROUTING_SIZE) >> 1;
+ for (i = 0; i < n; i++)
+ aUplinkMuxing[i] = ZERO_labelID;
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_AUPLINKROUTING_ADDR,
+ (u32 *) aUplinkMuxing, sizeof(aUplinkMuxing));
+}
+
+/**
+ * omap_abe_reset_port
+ * @id: ABE port ID
+ *
+ * stop the port activity and reload default parameters on the associated
+ * processing features.
+ * Clears the internal AE buffers.
+ */
+int omap_abe_reset_port(u32 id)
+{
+ _log(ABE_ID_RESET_PORT, id, 0, 0);
+ abe_port[id] = ((abe_port_t *) abe_port_init)[id];
+ return 0;
+}
+
+/**
+ * abe_reset_all_ports
+ *
+ * load default configuration for all features
+ */
+void omap_abe_reset_all_ports(struct omap_abe *abe)
+{
+ u16 i;
+ for (i = 0; i < LAST_PORT_ID; i++)
+ omap_abe_reset_port(i);
+ /* mixers' configuration */
+ omap_abe_write_mixer(abe, MIXDL1, MUTE_GAIN,
+ RAMP_5MS, MIX_DL1_INPUT_MM_DL);
+ omap_abe_write_mixer(abe, MIXDL1, MUTE_GAIN,
+ RAMP_5MS, MIX_DL1_INPUT_MM_UL2);
+ omap_abe_write_mixer(abe, MIXDL1, MUTE_GAIN,
+ RAMP_5MS, MIX_DL1_INPUT_VX_DL);
+ omap_abe_write_mixer(abe, MIXDL1, MUTE_GAIN,
+ RAMP_5MS, MIX_DL1_INPUT_TONES);
+ omap_abe_write_mixer(abe, MIXDL2, MUTE_GAIN,
+ RAMP_5MS, MIX_DL2_INPUT_TONES);
+ omap_abe_write_mixer(abe, MIXDL2, MUTE_GAIN,
+ RAMP_5MS, MIX_DL2_INPUT_VX_DL);
+ omap_abe_write_mixer(abe, MIXDL2, MUTE_GAIN,
+ RAMP_5MS, MIX_DL2_INPUT_MM_DL);
+ omap_abe_write_mixer(abe, MIXDL2, MUTE_GAIN,
+ RAMP_5MS, MIX_DL2_INPUT_MM_UL2);
+ omap_abe_write_mixer(abe, MIXSDT, MUTE_GAIN,
+ RAMP_5MS, MIX_SDT_INPUT_UP_MIXER);
+ omap_abe_write_mixer(abe, MIXSDT, GAIN_0dB,
+ RAMP_5MS, MIX_SDT_INPUT_DL1_MIXER);
+ omap_abe_write_mixer(abe, MIXECHO, MUTE_GAIN,
+ RAMP_5MS, MIX_ECHO_DL1);
+ omap_abe_write_mixer(abe, MIXECHO, MUTE_GAIN,
+ RAMP_5MS, MIX_ECHO_DL2);
+ omap_abe_write_mixer(abe, MIXAUDUL, MUTE_GAIN,
+ RAMP_5MS, MIX_AUDUL_INPUT_MM_DL);
+ omap_abe_write_mixer(abe, MIXAUDUL, MUTE_GAIN,
+ RAMP_5MS, MIX_AUDUL_INPUT_TONES);
+ omap_abe_write_mixer(abe, MIXAUDUL, GAIN_0dB,
+ RAMP_5MS, MIX_AUDUL_INPUT_UPLINK);
+ omap_abe_write_mixer(abe, MIXAUDUL, MUTE_GAIN,
+ RAMP_5MS, MIX_AUDUL_INPUT_VX_DL);
+ omap_abe_write_mixer(abe, MIXVXREC, MUTE_GAIN,
+ RAMP_5MS, MIX_VXREC_INPUT_TONES);
+ omap_abe_write_mixer(abe, MIXVXREC, MUTE_GAIN,
+ RAMP_5MS, MIX_VXREC_INPUT_VX_DL);
+ omap_abe_write_mixer(abe, MIXVXREC, MUTE_GAIN,
+ RAMP_5MS, MIX_VXREC_INPUT_MM_DL);
+ omap_abe_write_mixer(abe, MIXVXREC, MUTE_GAIN,
+ RAMP_5MS, MIX_VXREC_INPUT_VX_UL);
+ omap_abe_write_gain(abe, GAINS_DMIC1, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DMIC1, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DMIC2, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DMIC2, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DMIC3, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DMIC3, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_AMIC, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_AMIC, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_SPLIT, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_SPLIT, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL1, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL1, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL2, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_DL2, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_BTUL, GAIN_0dB,
+ RAMP_5MS, GAIN_LEFT_OFFSET);
+ omap_abe_write_gain(abe, GAINS_BTUL, GAIN_0dB,
+ RAMP_5MS, GAIN_RIGHT_OFFSET);
+}
diff --git a/sound/soc/omap/abe/abe_initxxx_labels.h b/sound/soc/omap/abe/abe_initxxx_labels.h
new file mode 100644
index 0000000..66f1856
--- /dev/null
+++ b/sound/soc/omap/abe/abe_initxxx_labels.h
@@ -0,0 +1,460 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#ifndef _ABE_INITXXX_LABELS_H_
+#define _ABE_INITXXX_LABELS_H_
+#define Dummy_Regs_labelID 0
+#define Dummy_AM_labelID 1
+#define Voice_8k_UL_labelID 2
+#define Voice_8k_DL_labelID 3
+#define ECHO_REF_8K_labelID 4
+#define Voice_16k_UL_labelID 5
+#define Voice_16k_DL_labelID 6
+#define ECHO_REF_16K_labelID 7
+#define MM_DL_labelID 8
+#define IO_VX_DL_ASRC_labelID 9
+#define IO_MM_EXT_IN_ASRC_labelID 10
+#define IO_VIBRA_DL_labelID 11
+#define ZERO_labelID 12
+#define GTarget_labelID 13
+#define GCurrent_labelID 14
+#define Gr_1_labelID 15
+#define Gr_2_labelID 16
+#define Gr_Regs_labelID 17
+#define DMIC0_Gain_labelID 18
+#define DMIC1_Gain_labelID 19
+#define DMIC2_Gain_labelID 20
+#define DMIC3_Gain_labelID 21
+#define AMIC_Gain_labelID 22
+#define MIXDL1_Gain_labelID 23
+#define MIXDL2_Gain_labelID 24
+#define DEFAULT_Gain_labelID 25
+#define DL1_M_G_Tones_labelID 26
+#define DL2_M_G_Tones_labelID 27
+#define Echo_M_G_labelID 28
+#define SDT_M_G_labelID 29
+#define VXREC_M_G_VX_DL_labelID 30
+#define UL_M_G_VX_DL_labelID 31
+#define BTUL_Gain_labelID 32
+#define DL1_M_labelID 33
+#define DL2_M_labelID 34
+#define MM_UL2_labelID 35
+#define VX_DL_labelID 36
+#define Tones_labelID 37
+#define DL_M_MM_UL2_VX_DL_labelID 38
+#define Echo_M_labelID 39
+#define VX_UL_labelID 40
+#define VX_UL_M_labelID 41
+#define SDT_F_labelID 42
+#define SDT_F_data_labelID 43
+#define SDT_Coef_labelID 44
+#define SDT_Regs_labelID 45
+#define SDT_M_labelID 46
+#define DL1_EQ_labelID 47
+#define DL2_EQ_labelID 48
+#define DL1_GAIN_out_labelID 49
+#define DL2_GAIN_out_labelID 50
+#define DMIC1_labelID 51
+#define DMIC1_L_labelID 52
+#define DMIC1_R_labelID 53
+#define DMIC2_labelID 54
+#define DMIC2_L_labelID 55
+#define DMIC2_R_labelID 56
+#define DMIC3_labelID 57
+#define DMIC3_L_labelID 58
+#define DMIC3_R_labelID 59
+#define SaturationMinMax_labelID 60
+#define TEMPORARY0_labelID 61
+#define TEMPORARY1_labelID 62
+#define BT_UL_L_labelID 63
+#define BT_UL_R_labelID 64
+#define AMIC_labelID 65
+#define AMIC_L_labelID 66
+#define AMIC_R_labelID 67
+#define EchoRef_L_labelID 68
+#define EchoRef_R_labelID 69
+#define MM_DL_L_labelID 70
+#define MM_DL_R_labelID 71
+#define MM_UL_labelID 72
+#define AMIC_96_labelID 73
+#define DMIC0_96_labelID 74
+#define DMIC1_96_labelID 75
+#define DMIC2_96_labelID 76
+#define UL_MIC_48K_labelID 77
+#define EQ_DL_48K_labelID 78
+#define EQ_48K_labelID 79
+#define McPDM_Out1_labelID 80
+#define McPDM_Out2_labelID 81
+#define McPDM_Out3_labelID 82
+#define VX_UL_MUX_labelID 83
+#define MM_UL2_MUX_labelID 84
+#define MM_UL_MUX_labelID 85
+#define XinASRC_DL_VX_labelID 86
+#define ASRC_DL_VX_Coefs_labelID 87
+#define ASRC_DL_VX_Alpha_labelID 88
+#define ASRC_DL_VX_VarsBeta_labelID 89
+#define ASRC_DL_VX_8k_Regs_labelID 90
+#define XinASRC_UL_VX_labelID 91
+#define ASRC_UL_VX_Coefs_labelID 92
+#define ASRC_UL_VX_Alpha_labelID 93
+#define ASRC_UL_VX_VarsBeta_labelID 94
+#define ASRC_UL_VX_8k_Regs_labelID 95
+#define UL_48_8_DEC_labelID 96
+#define ASRC_DL_VX_16k_Regs_labelID 97
+#define ASRC_UL_VX_16k_Regs_labelID 98
+#define UL_48_16_DEC_labelID 99
+#define XinASRC_MM_EXT_IN_labelID 100
+#define ASRC_MM_EXT_IN_Coefs_labelID 101
+#define ASRC_MM_EXT_IN_Alpha_labelID 102
+#define ASRC_MM_EXT_IN_VarsBeta_labelID 103
+#define ASRC_MM_EXT_IN_Regs_labelID 104
+#define VX_REC_labelID 105
+#define VXREC_UL_M_Tones_VX_UL_labelID 106
+#define VX_REC_L_labelID 107
+#define VX_REC_R_labelID 108
+#define DL2_M_L_labelID 109
+#define DL2_M_R_labelID 110
+#define DL1_M_data_labelID 111
+#define DL1_M_Coefs_labelID 112
+#define DL2_M_LR_data_labelID 113
+#define DL2_M_LR_Coefs_labelID 114
+#define SRC_6_LP_COEFS_labelID 115
+#define SRC_6_LP_GAIN_COEFS_labelID 116
+#define SRC_6_HP_COEFS_labelID 117
+#define SRC_3_LP_COEFS_labelID 118
+#define SRC_3_LP_GAIN_COEFS_labelID 119
+#define SRC_3_HP_COEFS_labelID 120
+#define VX_DL_8_48_LP_DATA_labelID 121
+#define VX_DL_8_48_HP_DATA_labelID 122
+#define VX_DL_16_48_LP_DATA_labelID 123
+#define VX_DL_16_48_HP_DATA_labelID 124
+#define VX_UL_48_8_LP_DATA_labelID 125
+#define VX_UL_48_8_HP_DATA_labelID 126
+#define VX_UL_48_16_LP_DATA_labelID 127
+#define VX_UL_48_16_HP_DATA_labelID 128
+#define BT_UL_8_48_LP_DATA_labelID 129
+#define BT_UL_8_48_HP_DATA_labelID 130
+#define BT_UL_16_48_LP_DATA_labelID 131
+#define BT_UL_16_48_HP_DATA_labelID 132
+#define BT_DL_48_8_LP_DATA_labelID 133
+#define BT_DL_48_8_HP_DATA_labelID 134
+#define BT_DL_48_16_LP_DATA_labelID 135
+#define BT_DL_48_16_HP_DATA_labelID 136
+#define ECHO_REF_48_16_LP_DATA_labelID 137
+#define ECHO_REF_48_16_HP_DATA_labelID 138
+#define ECHO_REF_48_8_LP_DATA_labelID 139
+#define ECHO_REF_48_8_HP_DATA_labelID 140
+#define ECHO_REF_DEC_labelID 141
+#define VX_UL_8_TEMP_labelID 142
+#define VX_UL_16_TEMP_labelID 143
+#define UP_DOWN_8_48_labelID 144
+#define UP_DOWN_16_48_labelID 145
+#define SRC_6_LP_48k_labelID 146
+#define SRC_6_HP_labelID 147
+#define SRC_3_LP_48k_labelID 148
+#define SRC_3_HP_labelID 149
+#define EARP_48_96_LP_DATA_labelID 150
+#define SRC_48_96_LP_labelID 151
+#define IHF_48_96_LP_DATA_labelID 152
+#define EQ_VX_UL_16K_labelID 153
+#define AB0_labelID 154
+#define AC0_labelID 155
+#define MM_DL_C_labelID 156
+#define TONES_C_labelID 157
+#define MM_DL_44P1_REGS_labelID 158
+#define TONES_44P1_REGS_labelID 159
+#define MM_DL_44P1_DRIFT_labelID 160
+#define MM_DL_44P1_XK_labelID 161
+#define TONES_44P1_DRIFT_labelID 162
+#define TONES_44P1_XK_labelID 163
+#define SRC_44P1_MULFAC1_2_labelID 164
+#define A00_labelID 165
+#define MM_DL_44P1_WPTR_labelID 166
+#define MM_DL_44P1_RPTR_labelID 167
+#define TONES_44P1_WPTR_labelID 168
+#define TONES_44P1_RPTR_labelID 169
+#define C_0DB_SAT_labelID 170
+#define AC_labelID 171
+#define AD_labelID 172
+#define AE_labelID 173
+#define AF_labelID 174
+#define AG_labelID 175
+#define AH_labelID 176
+#define AI_labelID 177
+#define AJ_labelID 178
+#define AK_labelID 179
+#define AL_labelID 180
+#define AM_labelID 181
+#define AN_labelID 182
+#define AO_labelID 183
+#define AP_labelID 184
+#define AQ_labelID 185
+#define AR_labelID 186
+#define AS_labelID 187
+#define AT_labelID 188
+#define AU_labelID 189
+#define AV_labelID 190
+#define AW_labelID 191
+#define pVIBRA1_p0_labelID 192
+#define pVIBRA1_p1_labelID 193
+#define pVIBRA1_p23_labelID 194
+#define pVIBRA1_p45_labelID 195
+#define pVibra1_pR1_labelID 196
+#define pVibra1_pR2_labelID 197
+#define pVibra1_pR3_labelID 198
+#define pVIBRA1_r_labelID 199
+#define pVIBRA2_p0_0_labelID 200
+#define pVIBRA2_p0_labelID 201
+#define pVIBRA2_p1_labelID 202
+#define pVIBRA2_p23_labelID 203
+#define pVIBRA2_p45_labelID 204
+#define pCtrl_p67_labelID 205
+#define pVIBRA2_r_labelID 206
+#define VIBRA_labelID 207
+#define UP_48_96_LP_COEFS_DC_HF_labelID 208
+#define AX_labelID 209
+#define UP_48_96_LP_COEFS_DC_HS_labelID 210
+#define AMIC_96_48_data_labelID 211
+#define DOWN_96_48_AMIC_Coefs_labelID 212
+#define DOWN_96_48_DMIC_Coefs_labelID 213
+#define DOWN_96_48_AMIC_Regs_labelID 214
+#define DOWN_96_48_DMIC_Regs_labelID 215
+#define DMIC0_96_48_data_labelID 216
+#define DMIC1_96_48_data_labelID 217
+#define DMIC2_96_48_data_labelID 218
+#define SIO_DMIC_labelID 219
+#define SIO_PDM_UL_labelID 220
+#define SIO_BT_VX_UL_labelID 221
+#define SIO_MM_UL_labelID 222
+#define SIO_MM_UL2_labelID 223
+#define SIO_VX_UL_labelID 224
+#define SIO_MM_DL_labelID 225
+#define SIO_VX_DL_labelID 226
+#define SIO_TONES_DL_labelID 227
+#define SIO_VIB_DL_labelID 228
+#define SIO_BT_VX_DL_labelID 229
+#define SIO_PDM_DL_labelID 230
+#define SIO_MM_EXT_OUT_labelID 231
+#define SIO_MM_EXT_IN_labelID 232
+#define SIO_TDM_OUT_labelID 233
+#define SIO_TDM_IN_labelID 234
+#define DMIC_ATC_PTR_labelID 235
+#define MCPDM_UL_ATC_PTR_labelID 236
+#define BT_VX_UL_ATC_PTR_labelID 237
+#define MM_UL_ATC_PTR_labelID 238
+#define MM_UL2_ATC_PTR_labelID 239
+#define VX_UL_ATC_PTR_labelID 240
+#define MM_DL_ATC_PTR_labelID 241
+#define VX_DL_ATC_PTR_labelID 242
+#define TONES_DL_ATC_PTR_labelID 243
+#define VIB_DL_ATC_PTR_labelID 244
+#define BT_VX_DL_ATC_PTR_labelID 245
+#define PDM_DL_ATC_PTR_labelID 246
+#define MM_EXT_OUT_ATC_PTR_labelID 247
+#define MM_EXT_IN_ATC_PTR_labelID 248
+#define TDM_OUT_ATC_PTR_labelID 249
+#define TDM_IN_ATC_PTR_labelID 250
+#define MCU_IRQ_FIFO_ptr_labelID 251
+#define DEBUG_IRQ_FIFO_reg_labelID 252
+#define UP_DOWN_48_96_labelID 253
+#define OSR96_2_labelID 254
+#define DEBUG_GAINS_labelID 255
+#define DBG_8K_PATTERN_labelID 256
+#define DBG_16K_PATTERN_labelID 257
+#define DBG_24K_PATTERN_labelID 258
+#define DBG_48K_PATTERN_labelID 259
+#define DBG_96K_PATTERN_labelID 260
+#define UL_VX_UL_48_8K_labelID 261
+#define UL_VX_UL_48_16K_labelID 262
+#define BT_DL_labelID 263
+#define BT_UL_labelID 264
+#define BT_DL_8k_labelID 265
+#define BT_DL_16k_labelID 266
+#define BT_UL_8k_labelID 267
+#define BT_UL_16k_labelID 268
+#define MM_EXT_IN_labelID 269
+#define MM_EXT_IN_L_labelID 270
+#define MM_EXT_IN_R_labelID 271
+#define ECHO_REF_48_16_WRAP_labelID 272
+#define ECHO_REF_48_8_WRAP_labelID 273
+#define BT_UL_16_48_WRAP_labelID 274
+#define BT_UL_8_48_WRAP_labelID 275
+#define BT_DL_48_16_WRAP_labelID 276
+#define BT_DL_48_8_WRAP_labelID 277
+#define VX_DL_16_48_WRAP_labelID 278
+#define VX_DL_8_48_WRAP_labelID 279
+#define VX_UL_48_16_WRAP_labelID 280
+#define VX_UL_48_8_WRAP_labelID 281
+#define ATC_NULL_BUFFER_labelID 282
+#define MEM_INIT_hal_mem_labelID 283
+#define MEM_INIT_write_mem_labelID 284
+#define MEM_INIT_regs_labelID 285
+#define GAIN_0DB_labelID 286
+#define XinASRC_BT_UL_labelID 287
+#define IO_BT_UL_ASRC_labelID 288
+#define ASRC_BT_UL_Coefs_labelID 289
+#define ASRC_BT_UL_Alpha_labelID 290
+#define ASRC_BT_UL_VarsBeta_labelID 291
+#define ASRC_BT_UL_8k_Regs_labelID 292
+#define ASRC_BT_UL_16k_Regs_labelID 293
+#define XinASRC_BT_DL_labelID 294
+#define DL_48_8_DEC_labelID 295
+#define DL_48_16_DEC_labelID 296
+#define BT_DL_8k_TEMP_labelID 297
+#define BT_DL_16k_TEMP_labelID 298
+#define BT_DL_8k_opp100_labelID 299
+#define BT_DL_16k_opp100_labelID 300
+#define ASRC_BT_DL_Coefs_labelID 301
+#define ASRC_BT_DL_Alpha_labelID 302
+#define ASRC_BT_DL_VarsBeta_labelID 303
+#define ASRC_BT_DL_8k_Regs_labelID 304
+#define ASRC_BT_DL_16k_Regs_labelID 305
+#define BT_DL_48_8_OPP100_WRAP_labelID 306
+#define BT_DL_48_16_OPP100_WRAP_labelID 307
+#define VX_DL_8_48_OSR_LP_labelID 308
+#define SRC_FIR6_OSR_LP_labelID 309
+#define VX_DL_8_48_FIR_WRAP_labelID 310
+#define PING_labelID 311
+#define PING_Regs_labelID 312
+#define BT_UL_8_48_FIR_WRAP_labelID 313
+#define BT_UL_8_48_OSR_LP_labelID 314
+#define Dummy_315_labelID 315
+#define Dummy_316_labelID 316
+#define Dummy_317_labelID 317
+#define Dummy_318_labelID 318
+#define Dummy_319_labelID 319
+#define Dummy_320_labelID 320
+#define Dummy_321_labelID 321
+#define Dummy_322_labelID 322
+#define Dummy_323_labelID 323
+#define Dummy_324_labelID 324
+#define Dummy_325_labelID 325
+#define Dummy_326_labelID 326
+#define Dummy_327_labelID 327
+#define Dummy_328_labelID 328
+#define Dummy_329_labelID 329
+#define Dummy_330_labelID 330
+#define Dummy_331_labelID 331
+#define Dummy_332_labelID 332
+#define Dummy_333_labelID 333
+#define Dummy_334_labelID 334
+#define Dummy_335_labelID 335
+#define Dummy_336_labelID 336
+#define Dummy_337_labelID 337
+#define Dummy_338_labelID 338
+#define Dummy_339_labelID 339
+#define Dummy_340_labelID 340
+#define Dummy_341_labelID 341
+#define Dummy_342_labelID 342
+#define Dummy_343_labelID 343
+#define Dummy_344_labelID 344
+#define Dummy_345_labelID 345
+#define Dummy_346_labelID 346
+#define Dummy_347_labelID 347
+#define Dummy_348_labelID 348
+#define Dummy_349_labelID 349
+#define Dummy_350_labelID 350
+#define Dummy_351_labelID 351
+#define Dummy_352_labelID 352
+#define Dummy_353_labelID 353
+#define Dummy_354_labelID 354
+#define Dummy_355_labelID 355
+#define Dummy_356_labelID 356
+#define Dummy_357_labelID 357
+#define Dummy_358_labelID 358
+#define Dummy_359_labelID 359
+#define Dummy_360_labelID 360
+#define Dummy_361_labelID 361
+#define Dummy_362_labelID 362
+#define Dummy_363_labelID 363
+#define Dummy_364_labelID 364
+#define Dummy_365_labelID 365
+#define Dummy_366_labelID 366
+#define Dummy_367_labelID 367
+#define Dummy_368_labelID 368
+#define Dummy_369_labelID 369
+#define Dummy_370_labelID 370
+#define Dummy_371_labelID 371
+#define Dummy_372_labelID 372
+#define Dummy_373_labelID 373
+#define Dummy_374_labelID 374
+#define Dummy_375_labelID 375
+#define Dummy_376_labelID 376
+#define Dummy_377_labelID 377
+#define Dummy_378_labelID 378
+#define Dummy_379_labelID 379
+#define Dummy_380_labelID 380
+#define Dummy_381_labelID 381
+#define Dummy_382_labelID 382
+#define Dummy_383_labelID 383
+#define Dummy_384_labelID 384
+#define Dummy_385_labelID 385
+#define Dummy_386_labelID 386
+#define Dummy_387_labelID 387
+#define Dummy_388_labelID 388
+#define Dummy_389_labelID 389
+#define Dummy_390_labelID 390
+#define Dummy_391_labelID 391
+#define Dummy_392_labelID 392
+#define Dummy_393_labelID 393
+#define Dummy_394_labelID 394
+#define Dummy_395_labelID 395
+#define Dummy_396_labelID 396
+#define Dummy_397_labelID 397
+#define Dummy_398_labelID 398
+#define Dummy_399_labelID 399
+#endif /* _ABE_INITXXXX_LABELS_H_ */
diff --git a/sound/soc/omap/abe/abe_irq.c b/sound/soc/omap/abe/abe_irq.c
new file mode 100644
index 0000000..d639894
--- /dev/null
+++ b/sound/soc/omap/abe/abe_irq.c
@@ -0,0 +1,107 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include "abe_legacy.h"
+
+extern u32 abe_irq_pingpong_player_id;
+
+/*
+ * initialize the default values for call-backs to subroutines
+ * - FIFO IRQ call-backs for sequenced tasks
+ * - FIFO IRQ call-backs for audio player/recorders (ping-pong protocols)
+ * - Remote debugger interface
+ * - Error monitoring
+ * - Activity Tracing
+ */
+/**
+ * abe_irq_ping_pong
+ *
+ * Call the respective subroutine depending on the IRQ FIFO content:
+ * APS interrupts : IRQ_FIFO[31:28] = IRQtag_APS,
+ * IRQ_FIFO[27:16] = APS_IRQs, IRQ_FIFO[15:0] = loopCounter
+ * SEQ interrupts : IRQ_FIFO[31:28] = IRQtag_COUNT,
+ * IRQ_FIFO[27:16] = Count_IRQs, IRQ_FIFO[15:0] = loopCounter
+ * Ping-Pong Interrupts : IRQ_FIFO[31:28] = IRQtag_PP,
+ * IRQ_FIFO[27:16] = PP_MCU_IRQ, IRQ_FIFO[15:0] = loopCounter
+ */
+void abe_irq_ping_pong(void)
+{
+ abe_call_subroutine(abe_irq_pingpong_player_id, NOPARAMETER,
+ NOPARAMETER, NOPARAMETER, NOPARAMETER);
+}
+/**
+ * abe_irq_check_for_sequences
+* @i: sequence ID
+ *
+ * check the active sequence list
+ *
+ */
+void abe_irq_check_for_sequences(u32 i)
+{
+}
+/**
+ * abe_irq_aps
+ *
+ * call the application subroutines that updates
+ * the acoustics protection filters
+ */
+void abe_irq_aps(u32 aps_info)
+{
+ abe_call_subroutine(abe_irq_aps_adaptation_id, NOPARAMETER, NOPARAMETER,
+ NOPARAMETER, NOPARAMETER);
+}
diff --git a/sound/soc/omap/abe/abe_legacy.h b/sound/soc/omap/abe/abe_legacy.h
new file mode 100644
index 0000000..ca73dc2
--- /dev/null
+++ b/sound/soc/omap/abe/abe_legacy.h
@@ -0,0 +1,98 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_MAIN_H_
+#define _ABE_MAIN_H_
+
+#include <linux/io.h>
+
+#include "abe_dm_addr.h"
+#include "abe_sm_addr.h"
+#include "abe_cm_addr.h"
+#include "abe_define.h"
+#include "abe_fw.h"
+#include "abe_def.h"
+#include "abe_typ.h"
+#include "abe_ext.h"
+#include "abe_dbg.h"
+#include "abe_ref.h"
+#include "abe_api.h"
+#include "abe_typedef.h"
+#include "abe_functionsid.h"
+#include "abe_taskid.h"
+#include "abe_initxxx_labels.h"
+#include "abe_fw.h"
+
+/* pipe connection to the TARGET simulator */
+#define ABE_DEBUG_CHECKERS 0
+/* simulator data extracted from a text-file */
+#define ABE_DEBUG_HWFILE 0
+/* low-level log files */
+#define ABE_DEBUG_LL_LOG 0
+
+extern struct omap_abe *abe;
+
+void omap_abe_dbg_log(struct omap_abe *abe, u32 x, u32 y, u32 z, u32 t);
+void omap_abe_dbg_error(struct omap_abe *abe, int level, int error);
+
+/*
+ * MACROS
+ */
+#define _log(x, y, z, t) { if (x & abe->dbg.mask) omap_abe_dbg_log(abe, x, y, z, t); }
+
+#endif /* _ABE_MAIN_H_ */
diff --git a/sound/soc/omap/abe/abe_main.c b/sound/soc/omap/abe/abe_main.c
new file mode 100644
index 0000000..86e969e
--- /dev/null
+++ b/sound/soc/omap/abe/abe_main.c
@@ -0,0 +1,747 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+
+#include "abe_legacy.h"
+#include "abe_dbg.h"
+#include "abe_port.h"
+
+
+struct omap_abe_equ {
+ /* type of filter */
+ u32 equ_type;
+ /* filter length */
+ u32 equ_length;
+ union {
+ /* parameters are the direct and recursive coefficients in */
+ /* Q6.26 integer fixed-point format. */
+ s32 type1[NBEQ1];
+ struct {
+ /* center frequency of the band [Hz] */
+ s32 freq[NBEQ2];
+ /* gain of each band. [dB] */
+ s32 gain[NBEQ2];
+ /* Q factor of this band [dB] */
+ s32 q[NBEQ2];
+ } type2;
+ } coef;
+ s32 equ_param3;
+};
+
+#include "abe_gain.h"
+#include "abe_aess.h"
+#include "abe_seq.h"
+
+
+int omap_abe_connect_debug_trace(struct omap_abe *abe,
+ struct omap_abe_dma *dma2);
+
+int omap_abe_reset_hal(struct omap_abe *abe);
+int omap_abe_load_fw(struct omap_abe *abe, u32 *firmware);
+int omap_abe_reload_fw(struct omap_abe *abe, u32 *firmware);
+u32* omap_abe_get_default_fw(struct omap_abe *abe);
+int omap_abe_wakeup(struct omap_abe *abe);
+int omap_abe_irq_processing(struct omap_abe *abe);
+int omap_abe_clear_irq(struct omap_abe *abe);
+int omap_abe_disable_irq(struct omap_abe *abe);
+int omap_abe_set_debug_trace(struct omap_abe_dbg *dbg, int debug);
+int omap_abe_set_ping_pong_buffer(struct omap_abe *abe,
+ u32 port, u32 n_bytes);
+int omap_abe_read_next_ping_pong_buffer(struct omap_abe *abe,
+ u32 port, u32 *p, u32 *n);
+int omap_abe_init_ping_pong_buffer(struct omap_abe *abe,
+ u32 id, u32 size_bytes, u32 n_buffers,
+ u32 *p);
+int omap_abe_read_offset_from_ping_buffer(struct omap_abe *abe,
+ u32 id, u32 *n);
+int omap_abe_set_router_configuration(struct omap_abe *abe,
+ u32 id, u32 k, u32 *param);
+int omap_abe_set_opp_processing(struct omap_abe *abe, u32 opp);
+int omap_abe_disable_data_transfer(struct omap_abe *abe, u32 id);
+int omap_abe_enable_data_transfer(struct omap_abe *abe, u32 id);
+int omap_abe_connect_cbpr_dmareq_port(struct omap_abe *abe,
+ u32 id, abe_data_format_t *f,
+ u32 d,
+ abe_dma_t *returned_dma_t);
+int omap_abe_connect_irq_ping_pong_port(struct omap_abe *abe,
+ u32 id, abe_data_format_t *f,
+ u32 subroutine_id, u32 size,
+ u32 *sink, u32 dsp_mcu_flag);
+int omap_abe_connect_serial_port(struct omap_abe *abe,
+ u32 id, abe_data_format_t *f,
+ u32 mcbsp_id);
+int omap_abe_read_port_address(struct omap_abe *abe,
+ u32 port, abe_dma_t *dma2);
+int omap_abe_check_activity(struct omap_abe *abe);
+
+int omap_abe_use_compensated_gain(struct omap_abe *abe, int on_off);
+int omap_abe_write_equalizer(struct omap_abe *abe,
+ u32 id, struct omap_abe_equ *param);
+
+int omap_abe_disable_gain(struct omap_abe *abe, u32 id, u32 p);
+int omap_abe_enable_gain(struct omap_abe *abe, u32 id, u32 p);
+int omap_abe_mute_gain(struct omap_abe *abe, u32 id, u32 p);
+int omap_abe_unmute_gain(struct omap_abe *abe, u32 id, u32 p);
+
+int omap_abe_write_gain(struct omap_abe *abe,
+ u32 id, s32 f_g, u32 ramp, u32 p);
+int omap_abe_write_mixer(struct omap_abe *abe,
+ u32 id, s32 f_g, u32 f_ramp, u32 p);
+int omap_abe_read_gain(struct omap_abe *abe,
+ u32 id, u32 *f_g, u32 p);
+int omap_abe_read_mixer(struct omap_abe *abe,
+ u32 id, u32 *f_g, u32 p);
+
+extern struct omap_abe *abe;
+
+#if 0
+/**
+ * abe_init_mem - Allocate Kernel space memory map for ABE
+ *
+ * Memory map of ABE memory space for PMEM/DMEM/SMEM/DMEM
+ */
+void abe_init_mem(void __iomem *_io_base)
+{
+ omap_abe_init_mem(abe, _io_base);
+}
+EXPORT_SYMBOL(abe_init_mem);
+
+struct omap_abe* abe_probe_aess(void)
+{
+ return omap_abe_probe_aess(abe);
+}
+EXPORT_SYMBOL(abe_probe_aess);
+
+void abe_remove_aess(void)
+{
+ omap_abe_remove_aess(abe);
+}
+EXPORT_SYMBOL(abe_remove_aess);
+
+void abe_add_subroutine(u32 *id, abe_subroutine2 f,
+ u32 nparam, u32 *params)
+{
+ omap_abe_add_subroutine(abe, id, f, nparam, params);
+}
+EXPORT_SYMBOL(abe_add_subroutine);
+
+#endif
+
+/**
+ * abe_reset_hal - reset the ABE/HAL
+ * @rdev: regulator source
+ * @constraints: constraints to apply
+ *
+ * Operations : reset the HAL by reloading the static variables and
+ * default AESS registers.
+ * Called after a PRCM cold-start reset of ABE
+ */
+u32 abe_reset_hal(void)
+{
+ omap_abe_reset_hal(abe);
+ return 0;
+}
+EXPORT_SYMBOL(abe_reset_hal);
+
+/**
+ * abe_load_fw - Load ABE Firmware and initialize memories
+ *
+ */
+u32 abe_load_fw(u32 *firmware)
+{
+ omap_abe_load_fw(abe, firmware);
+ return 0;
+}
+EXPORT_SYMBOL(abe_load_fw);
+
+/**
+ * abe_reload_fw - Reload ABE Firmware and initialize memories
+ *
+ */
+u32 abe_reload_fw(u32 *firmware)
+{
+ omap_abe_reload_fw(abe, firmware);
+ return 0;
+}
+EXPORT_SYMBOL(abe_reload_fw);
+
+u32* abe_get_default_fw(void)
+{
+ return omap_abe_get_default_fw(abe);
+}
+EXPORT_SYMBOL(abe_get_default_fw);
+
+/**
+ * abe_wakeup - Wakeup ABE
+ *
+ * Wakeup ABE in case of retention
+ */
+u32 abe_wakeup(void)
+{
+ omap_abe_wakeup(abe);
+ return 0;
+}
+EXPORT_SYMBOL(abe_wakeup);
+
+/**
+ * abe_irq_processing - Process ABE interrupt
+ *
+ * This subroutine is call upon reception of "MA_IRQ_99 ABE_MPU_IRQ" Audio
+ * back-end interrupt. This subroutine will check the ATC Hrdware, the
+ * IRQ_FIFO from the AE and act accordingly. Some IRQ source are originated
+ * for the delivery of "end of time sequenced tasks" notifications, some are
+ * originated from the Ping-Pong protocols, some are generated from
+ * the embedded debugger when the firmware stops on programmable break-points,
+ * etc ...
+ */
+u32 abe_irq_processing(void)
+{
+ omap_abe_irq_processing(abe);
+ return 0;
+}
+EXPORT_SYMBOL(abe_irq_processing);
+
+/**
+ * abe_clear_irq - clear ABE interrupt
+ *
+ * This subroutine is call to clear MCU Irq
+ */
+u32 abe_clear_irq(void)
+{
+ omap_abe_clear_irq(abe);
+ return 0;
+}
+EXPORT_SYMBOL(abe_clear_irq);
+
+/**
+ * abe_disable_irq - disable MCU/DSP ABE interrupt
+ *
+ * This subroutine is disabling ABE MCU/DSP Irq
+ */
+u32 abe_disable_irq(void)
+{
+ omap_abe_disable_irq(abe);
+
+ return 0;
+}
+EXPORT_SYMBOL(abe_disable_irq);
+
+/**
+ * abe_write_event_generator - Selects event generator source
+ * @e: Event Generation Counter, McPDM, DMIC or default.
+ *
+ * Loads the AESS event generator hardware source.
+ * Loads the firmware parameters accordingly.
+ * Indicates to the FW which data stream is the most important to preserve
+ * in case all the streams are asynchronous.
+ * If the parameter is "default", then HAL decides which Event source
+ * is the best appropriate based on the opened ports.
+ *
+ * When neither the DMIC and the McPDM are activated, the AE will have
+ * its EVENT generator programmed with the EVENT_COUNTER.
+ * The event counter will be tuned in order to deliver a pulse frequency higher
+ * than 96 kHz.
+ * The DPLL output at 100% OPP is MCLK = (32768kHz x6000) = 196.608kHz
+ * The ratio is (MCLK/96000)+(1<<1) = 2050
+ * (1<<1) in order to have the same speed at 50% and 100% OPP
+ * (only 15 MSB bits are used at OPP50%)
+ */
+u32 abe_write_event_generator(u32 e) // should integarte abe as parameter
+{
+ omap_abe_write_event_generator(abe, e);
+ return 0;
+}
+EXPORT_SYMBOL(abe_write_event_generator);
+
+/**
+ * abe_start_event_generator - Starts event generator source
+ *
+ * Start the event genrator of AESS. No more event will be send to AESS engine.
+ * Upper layer must wait 1/96kHz to be sure that engine reaches
+ * the IDLE instruction.
+ */
+u32 abe_stop_event_generator(void)
+{
+ omap_abe_stop_event_generator(abe);
+ return 0;
+}
+EXPORT_SYMBOL(abe_stop_event_generator);
+
+/**
+ * abe_connect_debug_trace
+ * @dma2:pointer to the DMEM trace buffer
+ *
+ * returns the address and size of the real-time debug trace buffer,
+ * the content of which will vary from one firmware release to another
+ */
+u32 abe_connect_debug_trace(abe_dma_t *dma2)
+{
+ omap_abe_connect_debug_trace(abe, (struct omap_abe_dma *)dma2);
+ return 0;
+}
+EXPORT_SYMBOL(abe_connect_debug_trace);
+
+/**
+ * abe_set_debug_trace
+ * @debug: debug ID from a list to be defined
+ *
+ * loads a mask which filters the debug trace to dedicated types of data
+ */
+u32 abe_set_debug_trace(abe_dbg_t debug)
+{
+ omap_abe_set_debug_trace(&abe->dbg, (int)(debug));
+ return 0;
+}
+EXPORT_SYMBOL(abe_set_debug_trace);
+
+/**
+ * abe_set_ping_pong_buffer
+ * @port: ABE port ID
+ * @n_bytes: Size of Ping/Pong buffer
+ *
+ * Updates the next ping-pong buffer with "size" bytes copied from the
+ * host processor. This API notifies the FW that the data transfer is done.
+ */
+u32 abe_set_ping_pong_buffer(u32 port, u32 n_bytes)
+{
+ omap_abe_set_ping_pong_buffer(abe, port, n_bytes);
+ return 0;
+}
+EXPORT_SYMBOL(abe_set_ping_pong_buffer);
+
+/**
+ * abe_read_next_ping_pong_buffer
+ * @port: ABE portID
+ * @p: Next buffer address (pointer)
+ * @n: Next buffer size (pointer)
+ *
+ * Tell the next base address of the next ping_pong Buffer and its size
+ */
+u32 abe_read_next_ping_pong_buffer(u32 port, u32 *p, u32 *n)
+{
+ omap_abe_read_next_ping_pong_buffer(abe, port, p, n);
+ return 0;
+}
+EXPORT_SYMBOL(abe_read_next_ping_pong_buffer);
+
+/**
+ * abe_init_ping_pong_buffer
+ * @id: ABE port ID
+ * @size_bytes:size of the ping pong
+ * @n_buffers:number of buffers (2 = ping/pong)
+ * @p:returned address of the ping-pong list of base addresses
+ * (byte offset from DMEM start)
+ *
+ * Computes the base address of the ping_pong buffers
+ */
+u32 abe_init_ping_pong_buffer(u32 id, u32 size_bytes, u32 n_buffers,
+ u32 *p)
+{
+ omap_abe_init_ping_pong_buffer(abe, id, size_bytes, n_buffers, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_init_ping_pong_buffer);
+
+/**
+ * abe_read_offset_from_ping_buffer
+ * @id: ABE port ID
+ * @n: returned address of the offset
+ * from the ping buffer start address (in samples)
+ *
+ * Computes the current firmware ping pong read pointer location,
+ * expressed in samples, as the offset from the start address of ping buffer.
+ */
+u32 abe_read_offset_from_ping_buffer(u32 id, u32 *n)
+{
+ omap_abe_read_offset_from_ping_buffer(abe, id, n);
+ return 0;
+}
+EXPORT_SYMBOL(abe_read_offset_from_ping_buffer);
+
+/**
+ * abe_write_equalizer
+ * @id: name of the equalizer
+ * @param : equalizer coefficients
+ *
+ * Load the coefficients in CMEM.
+ */
+u32 abe_write_equalizer(u32 id, abe_equ_t *param)
+{
+ omap_abe_write_equalizer(abe, id, (struct omap_abe_equ *)param);
+ return 0;
+}
+EXPORT_SYMBOL(abe_write_equalizer);
+/**
+ * abe_disable_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+u32 abe_disable_gain(u32 id, u32 p)
+{
+ omap_abe_disable_gain(abe, id, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_disable_gain);
+/**
+ * abe_enable_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+u32 abe_enable_gain(u32 id, u32 p)
+{
+ omap_abe_enable_gain(abe, id, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_enable_gain);
+
+/**
+ * abe_mute_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+u32 abe_mute_gain(u32 id, u32 p)
+{
+ omap_abe_mute_gain(abe, id, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_mute_gain);
+
+/**
+ * abe_unmute_gain
+ * Parameters:
+ * mixer id
+ * sub-port id
+ *
+ */
+u32 abe_unmute_gain(u32 id, u32 p)
+{
+ omap_abe_unmute_gain(abe, id, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_unmute_gain);
+
+/**
+ * abe_write_gain
+ * @id: gain name or mixer name
+ * @f_g: list of input gains of the mixer
+ * @ramp: gain ramp speed factor
+ * @p: list of ports corresponding to the above gains
+ *
+ * Loads the gain coefficients to FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's gain
+ * in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+u32 abe_write_gain(u32 id, s32 f_g, u32 ramp, u32 p)
+{
+ omap_abe_write_gain(abe, id, f_g, ramp, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_write_gain);
+
+/**
+ * abe_write_mixer
+ * @id: name of the mixer
+ * @param: input gains and delay ramp of the mixer
+ * @p: port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's
+ * gain in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+u32 abe_write_mixer(u32 id, s32 f_g, u32 f_ramp, u32 p)
+{
+ omap_abe_write_gain(abe, id, f_g, f_ramp, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_write_mixer);
+
+/**
+ * abe_read_gain
+ * @id: name of the mixer
+ * @param: list of input gains of the mixer
+ * @p: list of port corresponding to the above gains
+ *
+ */
+u32 abe_read_gain(u32 id, u32 *f_g, u32 p)
+{
+ omap_abe_read_gain(abe, id, f_g, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_read_gain);
+
+/**
+ * abe_read_mixer
+ * @id: name of the mixer
+ * @param: gains of the mixer
+ * @p: port corresponding to the above gains
+ *
+ * Load the gain coefficients in FW memory. This API can be called when
+ * the corresponding MIXER is not activated. After reloading the firmware
+ * the default coefficients corresponds to "all input and output mixer's
+ * gain in mute state". A mixer is disabled with a network reconfiguration
+ * corresponding to an OPP value.
+ */
+u32 abe_read_mixer(u32 id, u32 *f_g, u32 p)
+{
+ omap_abe_read_gain(abe, id, f_g, p);
+ return 0;
+}
+EXPORT_SYMBOL(abe_read_mixer);
+
+/**
+ * abe_set_router_configuration
+ * @Id: name of the router
+ * @Conf: id of the configuration
+ * @param: list of output index of the route
+ *
+ * The uplink router takes its input from DMIC (6 samples), AMIC (2 samples)
+ * and PORT1/2 (2 stereo ports). Each sample will be individually stored in
+ * an intermediate table of 10 elements.
+ *
+ * Example of router table parameter for voice uplink with phoenix microphones
+ *
+ * indexes 0 .. 9 = MM_UL description (digital MICs and MMEXTIN)
+ * DMIC1_L_labelID, DMIC1_R_labelID, DMIC2_L_labelID, DMIC2_R_labelID,
+ * MM_EXT_IN_L_labelID, MM_EXT_IN_R_labelID, ZERO_labelID, ZERO_labelID,
+ * ZERO_labelID, ZERO_labelID,
+ * indexes 10 .. 11 = MM_UL2 description (recording on DMIC3)
+ * DMIC3_L_labelID, DMIC3_R_labelID,
+ * indexes 12 .. 13 = VX_UL description (VXUL based on PDMUL data)
+ * AMIC_L_labelID, AMIC_R_labelID,
+ * indexes 14 .. 15 = RESERVED (NULL)
+ * ZERO_labelID, ZERO_labelID,
+ */
+u32 abe_set_router_configuration(u32 id, u32 k, u32 *param)
+{
+ omap_abe_set_router_configuration(abe, id, k, param);
+ return 0;
+}
+EXPORT_SYMBOL(abe_set_router_configuration);
+
+/**
+ * abe_set_opp_processing - Set OPP mode for ABE Firmware
+ * @opp: OOPP mode
+ *
+ * New processing network and OPP:
+ * 0: Ultra Lowest power consumption audio player (no post-processing, no mixer)
+ * 1: OPP 25% (simple multimedia features, including low-power player)
+ * 2: OPP 50% (multimedia and voice calls)
+ * 3: OPP100% ( multimedia complex use-cases)
+ *
+ * Rearranges the FW task network to the corresponding OPP list of features.
+ * The corresponding AE ports are supposed to be set/reset accordingly before
+ * this switch.
+ *
+ */
+u32 abe_set_opp_processing(u32 opp)
+{
+ omap_abe_set_opp_processing(abe, opp);
+ return 0;
+}
+EXPORT_SYMBOL(abe_set_opp_processing);
+
+/**
+ * abe_disable_data_transfer
+ * @id: ABE port id
+ *
+ * disables the ATC descriptor and stop IO/port activities
+ * disable the IO task (@f = 0)
+ * clear ATC DMEM buffer, ATC enabled
+ */
+u32 abe_disable_data_transfer(u32 id)
+{
+ omap_abe_disable_data_transfer(abe, id);
+ return 0;
+}
+EXPORT_SYMBOL(abe_disable_data_transfer);
+
+/**
+ * abe_enable_data_transfer
+ * @ip: ABE port id
+ *
+ * enables the ATC descriptor
+ * reset ATC pointers
+ * enable the IO task (@f <> 0)
+ */
+u32 abe_enable_data_transfer(u32 id)
+{
+ omap_abe_enable_data_transfer(abe, id);
+ return 0;
+}
+EXPORT_SYMBOL(abe_enable_data_transfer);
+
+/**
+ * abe_connect_cbpr_dmareq_port
+ * @id: port name
+ * @f: desired data format
+ * @d: desired dma_request line (0..7)
+ * @a: returned pointer to the base address of the CBPr register and number of
+ * samples to exchange during a DMA_request.
+ *
+ * enables the data echange between a DMA and the ABE through the
+ * CBPr registers of AESS.
+ */
+u32 abe_connect_cbpr_dmareq_port(u32 id, abe_data_format_t *f, u32 d,
+ abe_dma_t *returned_dma_t)
+{
+ omap_abe_connect_cbpr_dmareq_port(abe, id, f, d, returned_dma_t);
+ return 0;
+}
+EXPORT_SYMBOL(abe_connect_cbpr_dmareq_port);
+
+/**
+ * abe_connect_irq_ping_pong_port
+ * @id: port name
+ * @f: desired data format
+ * @I: index of the call-back subroutine to call
+ * @s: half-buffer (ping) size
+ * @p: returned base address of the first (ping) buffer)
+ *
+ * enables the data echanges between a direct access to the DMEM
+ * memory of ABE using cache flush. On each IRQ activation a subroutine
+ * registered with "abe_plug_subroutine" will be called. This subroutine
+ * will generate an amount of samples, send them to DMEM memory and call
+ * "abe_set_ping_pong_buffer" to notify the new amount of samples in the
+ * pong buffer.
+ */
+u32 abe_connect_irq_ping_pong_port(u32 id, abe_data_format_t *f,
+ u32 subroutine_id, u32 size,
+ u32 *sink, u32 dsp_mcu_flag)
+{
+ omap_abe_connect_irq_ping_pong_port(abe, id, f, subroutine_id, size,
+ sink, dsp_mcu_flag);
+ return 0;
+}
+EXPORT_SYMBOL(abe_connect_irq_ping_pong_port);
+
+/**
+ * abe_connect_serial_port()
+ * @id: port name
+ * @f: data format
+ * @i: peripheral ID (McBSP #1, #2, #3)
+ *
+ * Operations : enables the data echanges between a McBSP and an ATC buffer in
+ * DMEM. This API is used connect 48kHz McBSP streams to MM_DL and 8/16kHz
+ * voice streams to VX_UL, VX_DL, BT_VX_UL, BT_VX_DL. It abstracts the
+ * abe_write_port API.
+ */
+u32 abe_connect_serial_port(u32 id, abe_data_format_t *f,
+ u32 mcbsp_id)
+{
+ omap_abe_connect_serial_port(abe, id, f, mcbsp_id);
+ return 0;
+}
+EXPORT_SYMBOL(abe_connect_serial_port);
+
+/**
+ * abe_read_port_address
+ * @dma: output pointer to the DMA iteration and data destination pointer
+ *
+ * This API returns the address of the DMA register used on this audio port.
+ * Depending on the protocol being used, adds the base address offset L3
+ * (DMA) or MPU (ARM)
+ */
+u32 abe_read_port_address(u32 port, abe_dma_t *dma2)
+{
+ omap_abe_read_port_address(abe, port, dma2);
+ return 0;
+}
+EXPORT_SYMBOL(abe_read_port_address);
+
+/**
+ * abe_check_activity - Check if some ABE activity.
+ *
+ * Check if any ABE ports are running.
+ * return 1: still activity on ABE
+ * return 0: no more activity on ABE. Event generator can be stopped
+ *
+ */
+u32 abe_check_activity(void)
+{
+ return (u32)omap_abe_check_activity(abe);
+}
+EXPORT_SYMBOL(abe_check_activity);
+/**
+ * abe_use_compensated_gain
+ * @on_off:
+ *
+ * Selects the automatic Mixer's gain management
+ * on_off = 1 allows the "abe_write_gain" to adjust the overall
+ * gains of the mixer to be tuned not to create saturation
+ */
+abehal_status abe_use_compensated_gain(u32 on_off)
+{
+ omap_abe_use_compensated_gain(abe, (int)(on_off));
+ return 0;
+}
+EXPORT_SYMBOL(abe_use_compensated_gain);
diff --git a/sound/soc/omap/abe/abe_main.h b/sound/soc/omap/abe/abe_main.h
new file mode 100644
index 0000000..cf18376
--- /dev/null
+++ b/sound/soc/omap/abe/abe_main.h
@@ -0,0 +1,664 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_MAIN_H_
+#define _ABE_MAIN_H_
+
+#include <linux/io.h>
+
+#include "abe_initxxx_labels.h"
+
+#define D_DEBUG_FIFO_ADDR 8160
+#define D_DEBUG_FIFO_ADDR_END 8255
+
+#define SUB_0_PARAM 0
+#define SUB_1_PARAM 1
+
+#define ABE_DEFAULT_BASE_ADDRESS_L3 0x49000000L
+#define ABE_DMEM_BASE_ADDRESS_MPU 0x40180000L
+#define ABE_DMEM_BASE_OFFSET_MPU 0x00080000L
+#define ABE_DMEM_BASE_ADDRESS_L3 (ABE_DEFAULT_BASE_ADDRESS_L3 + \
+ ABE_DMEM_BASE_OFFSET_MPU)
+
+/*
+ * HARDWARE AND PERIPHERAL DEFINITIONS
+ */
+/* MM_DL */
+#define ABE_CBPR0_IDX 0
+/* VX_DL */
+#define ABE_CBPR1_IDX 1
+/* VX_UL */
+#define ABE_CBPR2_IDX 2
+/* MM_UL */
+#define ABE_CBPR3_IDX 3
+/* MM_UL2 */
+#define ABE_CBPR4_IDX 4
+/* TONES */
+#define ABE_CBPR5_IDX 5
+/* VIB */
+#define ABE_CBPR6_IDX 6
+/* DEBUG/CTL */
+#define ABE_CBPR7_IDX 7
+
+/*
+ * OPP TYPE
+ *
+ * 0: Ultra Lowest power consumption audio player
+ * 1: OPP 25% (simple multimedia features)
+ * 2: OPP 50% (multimedia and voice calls)
+ * 3: OPP100% (multimedia complex use-cases)
+ */
+#define ABE_OPP0 0
+#define ABE_OPP25 1
+#define ABE_OPP50 2
+#define ABE_OPP100 3
+/*
+ * SAMPLES TYPE
+ *
+ * mono 16 bit sample LSB aligned, 16 MSB bits are unused;
+ * mono right shifted to 16bits LSBs on a 32bits DMEM FIFO for McBSP
+ * TX purpose;
+ * mono sample MSB aligned (16/24/32bits);
+ * two successive mono samples in one 32bits container;
+ * Two L/R 16bits samples in a 32bits container;
+ * Two channels defined with two MSB aligned samples;
+ * Three channels defined with three MSB aligned samples (MIC);
+ * Four channels defined with four MSB aligned samples (MIC);
+ * . . .
+ * Eight channels defined with eight MSB aligned samples (MIC);
+ */
+#define MONO_MSB 1
+#define MONO_RSHIFTED_16 2
+#define STEREO_RSHIFTED_16 3
+#define STEREO_16_16 4
+#define STEREO_MSB 5
+#define THREE_MSB 6
+#define FOUR_MSB 7
+#define FIVE_MSB 8
+#define SIX_MSB 9
+#define SEVEN_MSB 10
+#define EIGHT_MSB 11
+#define NINE_MSB 12
+#define TEN_MSB 13
+/*
+ * PORT PROTOCOL TYPE - abe_port_protocol_switch_id
+ */
+#define SLIMBUS_PORT_PROT 1
+#define SERIAL_PORT_PROT 2
+#define TDM_SERIAL_PORT_PROT 3
+#define DMIC_PORT_PROT 4
+#define MCPDMDL_PORT_PROT 5
+#define MCPDMUL_PORT_PROT 6
+#define PINGPONG_PORT_PROT 7
+#define DMAREQ_PORT_PROT 8
+/*
+ * PORT IDs, this list is aligned with the FW data mapping
+ */
+#define DMIC_PORT 0
+#define PDM_UL_PORT 1
+#define BT_VX_UL_PORT 2
+#define MM_UL_PORT 3
+#define MM_UL2_PORT 4
+#define VX_UL_PORT 5
+#define MM_DL_PORT 6
+#define VX_DL_PORT 7
+#define TONES_DL_PORT 8
+#define VIB_DL_PORT 9
+#define BT_VX_DL_PORT 10
+#define PDM_DL_PORT 11
+#define MM_EXT_OUT_PORT 12
+#define MM_EXT_IN_PORT 13
+#define TDM_DL_PORT 14
+#define TDM_UL_PORT 15
+#define DEBUG_PORT 16
+#define LAST_PORT_ID 17
+/* definitions for the compatibility with HAL05xx */
+#define PDM_DL1_PORT 18
+#define PDM_DL2_PORT 19
+#define PDM_VIB_PORT 20
+/* There is only one DMIC port, always used with 6 samples
+ per 96kHz periods */
+#define DMIC_PORT1 DMIC_PORT
+#define DMIC_PORT2 DMIC_PORT
+#define DMIC_PORT3 DMIC_PORT
+/*
+ * Signal processing module names - EQ APS MIX ROUT
+ */
+/* equalizer downlink path headset + earphone */
+#define FEAT_EQ1 1
+/* equalizer downlink path integrated handsfree LEFT */
+#define FEAT_EQ2L (FEAT_EQ1+1)
+/* equalizer downlink path integrated handsfree RIGHT */
+#define FEAT_EQ2R (FEAT_EQ2L+1)
+/* equalizer downlink path side-tone */
+#define FEAT_EQSDT (FEAT_EQ2R+1)
+/* equalizer uplink path AMIC */
+#define FEAT_EQAMIC (FEAT_EQSDT+1)
+/* equalizer uplink path DMIC */
+#define FEAT_EQDMIC (FEAT_EQAMIC+1)
+/* Acoustic protection for headset */
+#define FEAT_APS1 (FEAT_EQDMIC+1)
+/* acoustic protection high-pass filter for handsfree "Left" */
+#define FEAT_APS2 (FEAT_APS1+1)
+/* acoustic protection high-pass filter for handsfree "Right" */
+#define FEAT_APS3 (FEAT_APS2+1)
+/* asynchronous sample-rate-converter for the downlink voice path */
+#define FEAT_ASRC1 (FEAT_APS3+1)
+/* asynchronous sample-rate-converter for the uplink voice path */
+#define FEAT_ASRC2 (FEAT_ASRC1+1)
+/* asynchronous sample-rate-converter for the multimedia player */
+#define FEAT_ASRC3 (FEAT_ASRC2+1)
+/* asynchronous sample-rate-converter for the echo reference */
+#define FEAT_ASRC4 (FEAT_ASRC3+1)
+/* mixer of the headset and earphone path */
+#define FEAT_MIXDL1 (FEAT_ASRC4+1)
+/* mixer of the hands-free path */
+#define FEAT_MIXDL2 (FEAT_MIXDL1+1)
+/* mixer for audio being sent on the voice_ul path */
+#define FEAT_MIXAUDUL (FEAT_MIXDL2+1)
+/* mixer for voice communication recording */
+#define FEAT_MIXVXREC (FEAT_MIXAUDUL+1)
+/* mixer for side-tone */
+#define FEAT_MIXSDT (FEAT_MIXVXREC+1)
+/* mixer for echo reference */
+#define FEAT_MIXECHO (FEAT_MIXSDT+1)
+/* router of the uplink path */
+#define FEAT_UPROUTE (FEAT_MIXECHO+1)
+/* all gains */
+#define FEAT_GAINS (FEAT_UPROUTE+1)
+#define FEAT_GAINS_DMIC1 (FEAT_GAINS+1)
+#define FEAT_GAINS_DMIC2 (FEAT_GAINS_DMIC1+1)
+#define FEAT_GAINS_DMIC3 (FEAT_GAINS_DMIC2+1)
+#define FEAT_GAINS_AMIC (FEAT_GAINS_DMIC3+1)
+#define FEAT_GAINS_SPLIT (FEAT_GAINS_AMIC+1)
+#define FEAT_GAINS_DL1 (FEAT_GAINS_SPLIT+1)
+#define FEAT_GAINS_DL2 (FEAT_GAINS_DL1+1)
+#define FEAT_GAIN_BTUL (FEAT_GAINS_DL2+1)
+/* sequencing queue of micro tasks */
+#define FEAT_SEQ (FEAT_GAIN_BTUL+1)
+/* Phoenix control queue through McPDM */
+#define FEAT_CTL (FEAT_SEQ+1)
+/* list of features of the firmware -------------------------------*/
+#define MAXNBFEATURE FEAT_CTL
+/* abe_equ_id */
+/* equalizer downlink path headset + earphone */
+#define EQ1 FEAT_EQ1
+/* equalizer downlink path integrated handsfree LEFT */
+#define EQ2L FEAT_EQ2L
+#define EQ2R FEAT_EQ2R
+/* equalizer downlink path side-tone */
+#define EQSDT FEAT_EQSDT
+#define EQAMIC FEAT_EQAMIC
+#define EQDMIC FEAT_EQDMIC
+/* abe_aps_id */
+/* Acoustic protection for headset */
+#define APS1 FEAT_APS1
+#define APS2L FEAT_APS2
+#define APS2R FEAT_APS3
+/* abe_asrc_id */
+/* asynchronous sample-rate-converter for the downlink voice path */
+#define ASRC1 FEAT_ASRC1
+/* asynchronous sample-rate-converter for the uplink voice path */
+#define ASRC2 FEAT_ASRC2
+/* asynchronous sample-rate-converter for the multimedia player */
+#define ASRC3 FEAT_ASRC3
+/* asynchronous sample-rate-converter for the voice uplink echo_reference */
+#define ASRC4 FEAT_ASRC4
+/* abe_mixer_id */
+#define MIXDL1 FEAT_MIXDL1
+#define MIXDL2 FEAT_MIXDL2
+#define MIXSDT FEAT_MIXSDT
+#define MIXECHO FEAT_MIXECHO
+#define MIXAUDUL FEAT_MIXAUDUL
+#define MIXVXREC FEAT_MIXVXREC
+/* abe_router_id */
+/* there is only one router up to now */
+#define UPROUTE FEAT_UPROUTE
+/*
+ * gain controls
+ */
+#define GAIN_LEFT_OFFSET 0
+#define GAIN_RIGHT_OFFSET 1
+/*
+ * GAIN IDs
+ */
+#define GAINS_DMIC1 FEAT_GAINS_DMIC1
+#define GAINS_DMIC2 FEAT_GAINS_DMIC2
+#define GAINS_DMIC3 FEAT_GAINS_DMIC3
+#define GAINS_AMIC FEAT_GAINS_AMIC
+#define GAINS_SPLIT FEAT_GAINS_SPLIT
+#define GAINS_DL1 FEAT_GAINS_DL1
+#define GAINS_DL2 FEAT_GAINS_DL2
+#define GAINS_BTUL FEAT_GAIN_BTUL
+/*
+ * ABE CONST AREA FOR PARAMETERS TRANSLATION
+ */
+#define sizeof_alpha_iir_table 61
+#define sizeof_beta_iir_table 61
+#define GAIN_MAXIMUM 3000L
+#define GAIN_24dB 2400L
+#define GAIN_18dB 1800L
+#define GAIN_12dB 1200L
+#define GAIN_6dB 600L
+/* default gain = 1 */
+#define GAIN_0dB 0L
+#define GAIN_M6dB -600L
+#define GAIN_M12dB -1200L
+#define GAIN_M18dB -1800L
+#define GAIN_M24dB -2400L
+#define GAIN_M30dB -3000L
+#define GAIN_M40dB -4000L
+#define GAIN_M50dB -5000L
+/* muted gain = -120 decibels */
+#define MUTE_GAIN -12000L
+#define GAIN_TOOLOW -13000L
+#define GAIN_MUTE MUTE_GAIN
+#define RAMP_MINLENGTH 3L
+/* ramp_t is in milli- seconds */
+#define RAMP_0MS 0L
+#define RAMP_1MS 1L
+#define RAMP_2MS 2L
+#define RAMP_5MS 5L
+#define RAMP_10MS 10L
+#define RAMP_20MS 20L
+#define RAMP_50MS 50L
+#define RAMP_100MS 100L
+#define RAMP_200MS 200L
+#define RAMP_500MS 500L
+#define RAMP_1000MS 1000L
+#define RAMP_MAXLENGTH 10000L
+/* for abe_translate_gain_format */
+#define LINABE_TO_DECIBELS 1
+#define DECIBELS_TO_LINABE 2
+/* for abe_translate_ramp_format */
+#define IIRABE_TO_MICROS 1
+#define MICROS_TO_IIABE 2
+/*
+ * EVENT GENERATORS - abe_event_id
+ */
+#define EVENT_TIMER 0
+#define EVENT_44100 1
+/*
+ * DMA requests
+ */
+/*Internal connection doesn't connect at ABE boundary */
+#define External_DMA_0 0
+/*Transmit request digital microphone */
+#define DMIC_DMA_REQ 1
+/*Multichannel PDM downlink */
+#define McPDM_DMA_DL 2
+/*Multichannel PDM uplink */
+#define McPDM_DMA_UP 3
+/*MCBSP module 1 - transmit request */
+#define MCBSP1_DMA_TX 4
+/*MCBSP module 1 - receive request */
+#define MCBSP1_DMA_RX 5
+/*MCBSP module 2 - transmit request */
+#define MCBSP2_DMA_TX 6
+/*MCBSP module 2 - receive request */
+#define MCBSP2_DMA_RX 7
+/*MCBSP module 3 - transmit request */
+#define MCBSP3_DMA_TX 8
+/*MCBSP module 3 - receive request */
+#define MCBSP3_DMA_RX 9
+/*
+ * SERIAL PORTS IDs - abe_mcbsp_id
+ */
+#define MCBSP1_TX MCBSP1_DMA_TX
+#define MCBSP1_RX MCBSP1_DMA_RX
+#define MCBSP2_TX MCBSP2_DMA_TX
+#define MCBSP2_RX MCBSP2_DMA_RX
+#define MCBSP3_TX MCBSP3_DMA_TX
+#define MCBSP3_RX MCBSP3_DMA_RX
+
+#define PING_PONG_WITH_MCU_IRQ 1
+#define PING_PONG_WITH_DSP_IRQ 2
+
+/*
+ Mixer ID Input port ID Comments
+ DL1_MIXER 0 MMDL path
+ 1 MMUL2 path
+ 2 VXDL path
+ 3 TONES path
+ SDT_MIXER 0 Uplink path
+ 1 Downlink path
+ ECHO_MIXER 0 DL1_MIXER path
+ 1 DL2_MIXER path
+ AUDUL_MIXER 0 TONES_DL path
+ 1 Uplink path
+ 2 MM_DL path
+ VXREC_MIXER 0 TONES_DL path
+ 1 VX_DL path
+ 2 MM_DL path
+ 3 VX_UL path
+*/
+#define MIX_VXUL_INPUT_MM_DL 0
+#define MIX_VXUL_INPUT_TONES 1
+#define MIX_VXUL_INPUT_VX_UL 2
+#define MIX_VXUL_INPUT_VX_DL 3
+#define MIX_DL1_INPUT_MM_DL 0
+#define MIX_DL1_INPUT_MM_UL2 1
+#define MIX_DL1_INPUT_VX_DL 2
+#define MIX_DL1_INPUT_TONES 3
+#define MIX_DL2_INPUT_MM_DL 0
+#define MIX_DL2_INPUT_MM_UL2 1
+#define MIX_DL2_INPUT_VX_DL 2
+#define MIX_DL2_INPUT_TONES 3
+#define MIX_SDT_INPUT_UP_MIXER 0
+#define MIX_SDT_INPUT_DL1_MIXER 1
+#define MIX_AUDUL_INPUT_MM_DL 0
+#define MIX_AUDUL_INPUT_TONES 1
+#define MIX_AUDUL_INPUT_UPLINK 2
+#define MIX_AUDUL_INPUT_VX_DL 3
+#define MIX_VXREC_INPUT_MM_DL 0
+#define MIX_VXREC_INPUT_TONES 1
+#define MIX_VXREC_INPUT_VX_UL 2
+#define MIX_VXREC_INPUT_VX_DL 3
+#define MIX_ECHO_DL1 0
+#define MIX_ECHO_DL2 1
+/* nb of samples to route */
+#define NBROUTE_UL 16
+/* 10 routing tables max */
+#define NBROUTE_CONFIG_MAX 10
+/* 5 pre-computed routing tables */
+#define NBROUTE_CONFIG 6
+/* AMIC on VX_UL */
+#define UPROUTE_CONFIG_AMIC 0
+/* DMIC first pair on VX_UL */
+#define UPROUTE_CONFIG_DMIC1 1
+/* DMIC second pair on VX_UL */
+#define UPROUTE_CONFIG_DMIC2 2
+/* DMIC last pair on VX_UL */
+#define UPROUTE_CONFIG_DMIC3 3
+/* BT_UL on VX_UL */
+#define UPROUTE_CONFIG_BT 4
+/* ECHO_REF on MM_UL2 */
+#define UPROUTE_ECHO_MMUL2 5
+
+/*
+ * DMA_T
+ *
+ * dma structure for easing programming
+ */
+typedef struct {
+ /* OCP L3 pointer to the first address of the */
+ void *data;
+ /* destination buffer (either DMA or Ping-Pong read/write pointers). */
+ /* address L3 when addressing the DMEM buffer instead of CBPr */
+ void *l3_dmem;
+ /* address L3 translated to L4 the ARM memory space */
+ void *l4_dmem;
+ /* number of iterations for the DMA data moves. */
+ u32 iter;
+} abe_dma_t;
+typedef u32 abe_dbg_t;
+/*
+ * ROUTER_T
+ *
+ * table of indexes in unsigned bytes
+ */
+typedef u16 abe_router_t;
+/*
+ * DATA_FORMAT_T
+ *
+ * used in port declaration
+ */
+typedef struct {
+ /* Sampling frequency of the stream */
+ u32 f;
+ /* Sample format type */
+ u32 samp_format;
+} abe_data_format_t;
+/*
+ * PORT_PROTOCOL_T
+ *
+ * port declaration
+ */
+typedef struct {
+ /* Direction=0 means input from AESS point of view */
+ u32 direction;
+ /* Protocol type (switch) during the data transfers */
+ u32 protocol_switch;
+ union {
+ /* Slimbus peripheral connected to ATC */
+ struct {
+ /* Address of ATC Slimbus descriptor's index */
+ u32 desc_addr1;
+ /* DMEM address 1 in bytes */
+ u32 buf_addr1;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ /* Second ATC index for SlimBus reception (or NULL) */
+ u32 desc_addr2;
+ /* DMEM address 2 in bytes */
+ u32 buf_addr2;
+ } prot_slimbus;
+ /* McBSP/McASP peripheral connected to ATC */
+ struct {
+ u32 desc_addr;
+ /* Address of ATC McBSP/McASP descriptor's in bytes */
+ u32 buf_addr;
+ /* DMEM address in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ } prot_serial;
+ /* DMIC peripheral connected to ATC */
+ struct {
+ /* DMEM address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size in bytes */
+ u32 buf_size;
+ /* Number of activated DMIC */
+ u32 nbchan;
+ } prot_dmic;
+ /* McPDMDL peripheral connected to ATC */
+ struct {
+ /* DMEM address in bytes */
+ u32 buf_addr;
+ /* DMEM size in bytes */
+ u32 buf_size;
+ /* Control allowed on McPDM DL */
+ u32 control;
+ } prot_mcpdmdl;
+ /* McPDMUL peripheral connected to ATC */
+ struct {
+ /* DMEM address size in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ } prot_mcpdmul;
+ /* Ping-Pong interface to the Host using cache-flush */
+ struct {
+ /* Address of ATC descriptor's */
+ u32 desc_addr;
+ /* DMEM buffer base address in bytes */
+ u32 buf_addr;
+ /* DMEM size in bytes for each ping and pong buffers */
+ u32 buf_size;
+ /* IRQ address (either DMA (0) MCU (1) or DSP(2)) */
+ u32 irq_addr;
+ /* IRQ data content loaded in the AESS IRQ register */
+ u32 irq_data;
+ /* Call-back function upon IRQ reception */
+ u32 callback;
+ } prot_pingpong;
+ /* DMAreq line to CBPr */
+ struct {
+ /* Address of ATC descriptor's */
+ u32 desc_addr;
+ /* DMEM buffer address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ /* DMAreq address */
+ u32 dma_addr;
+ /* DMA/AESS = 1 << #DMA */
+ u32 dma_data;
+ } prot_dmareq;
+ /* Circular buffer - direct addressing to DMEM */
+ struct {
+ /* DMEM buffer base address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size in bytes */
+ u32 buf_size;
+ /* DMAreq address */
+ u32 dma_addr;
+ /* DMA/AESS = 1 << #DMA */
+ u32 dma_data;
+ } prot_circular_buffer;
+ } p;
+} abe_port_protocol_t;
+
+/*
+ * EQU_T
+ *
+ * coefficients of the equalizer
+ */
+/* 24 Q6.26 coefficients */
+#define NBEQ1 25
+/* 2x12 Q6.26 coefficients */
+#define NBEQ2 13
+
+typedef struct {
+ /* type of filter */
+ u32 equ_type;
+ /* filter length */
+ u32 equ_length;
+ union {
+ /* parameters are the direct and recursive coefficients in */
+ /* Q6.26 integer fixed-point format. */
+ s32 type1[NBEQ1];
+ struct {
+ /* center frequency of the band [Hz] */
+ s32 freq[NBEQ2];
+ /* gain of each band. [dB] */
+ s32 gain[NBEQ2];
+ /* Q factor of this band [dB] */
+ s32 q[NBEQ2];
+ } type2;
+ } coef;
+ s32 equ_param3;
+} abe_equ_t;
+
+
+/* subroutine with no parameter */
+typedef void (*abe_subroutine0) (void);
+/* subroutine with one parameter */
+typedef void (*abe_subroutine1) (u32);
+typedef void (*abe_subroutine2) (u32, u32);
+typedef void (*abe_subroutine3) (u32, u32, u32);
+typedef void (*abe_subroutine4) (u32, u32, u32, u32);
+
+
+extern u32 abe_irq_pingpong_player_id;
+
+
+void abe_init_mem(void __iomem **_io_base);
+u32 abe_reset_hal(void);
+int abe_load_fw(u32 *firmware);
+int abe_reload_fw(u32 *firmware);
+u32 *abe_get_default_fw(void);
+u32 abe_wakeup(void);
+u32 abe_irq_processing(void);
+u32 abe_clear_irq(void);
+u32 abe_disable_irq(void);
+u32 abe_write_event_generator(u32 e);
+u32 abe_stop_event_generator(void);
+u32 abe_connect_debug_trace(abe_dma_t *dma2);
+u32 abe_set_debug_trace(abe_dbg_t debug);
+u32 abe_set_ping_pong_buffer(u32 port, u32 n_bytes);
+u32 abe_read_next_ping_pong_buffer(u32 port, u32 *p, u32 *n);
+u32 abe_init_ping_pong_buffer(u32 id, u32 size_bytes, u32 n_buffers,
+ u32 *p);
+u32 abe_read_offset_from_ping_buffer(u32 id, u32 *n);
+u32 abe_write_equalizer(u32 id, abe_equ_t *param);
+u32 abe_disable_gain(u32 id, u32 p);
+u32 abe_enable_gain(u32 id, u32 p);
+u32 abe_mute_gain(u32 id, u32 p);
+u32 abe_unmute_gain(u32 id, u32 p);
+u32 abe_write_gain(u32 id, s32 f_g, u32 ramp, u32 p);
+u32 abe_write_mixer(u32 id, s32 f_g, u32 f_ramp, u32 p);
+u32 abe_read_gain(u32 id, u32 *f_g, u32 p);
+u32 abe_read_mixer(u32 id, u32 *f_g, u32 p);
+u32 abe_set_router_configuration(u32 id, u32 k, u32 *param);
+u32 abe_set_opp_processing(u32 opp);
+u32 abe_disable_data_transfer(u32 id);
+u32 abe_enable_data_transfer(u32 id);
+u32 abe_connect_cbpr_dmareq_port(u32 id, abe_data_format_t *f, u32 d,
+ abe_dma_t *returned_dma_t);
+u32 abe_connect_irq_ping_pong_port(u32 id, abe_data_format_t *f,
+ u32 subroutine_id, u32 size,
+ u32 *sink, u32 dsp_mcu_flag);
+u32 abe_connect_serial_port(u32 id, abe_data_format_t *f,
+ u32 mcbsp_id);
+u32 abe_read_port_address(u32 port, abe_dma_t *dma2);
+void abe_add_subroutine(u32 *id, abe_subroutine2 f, u32 nparam, u32 *params);
+u32 abe_read_next_ping_pong_buffer(u32 port, u32 *p, u32 *n);
+u32 abe_check_activity(void);
+void abe_add_subroutine(u32 *id, abe_subroutine2 f,
+ u32 nparam, u32 *params);
+
+u32 abe_plug_subroutine(u32 *id, abe_subroutine2 f, u32 n,
+ u32 *params);
+
+#endif /* _ABE_MAIN_H_ */
diff --git a/sound/soc/omap/abe/abe_mem.h b/sound/soc/omap/abe/abe_mem.h
new file mode 100644
index 0000000..683968e
--- /dev/null
+++ b/sound/soc/omap/abe/abe_mem.h
@@ -0,0 +1,99 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_MEM_H_
+#define _ABE_MEM_H_
+
+#define OMAP_ABE_DMEM 0
+#define OMAP_ABE_CMEM 1
+#define OMAP_ABE_SMEM 2
+#define OMAP_ABE_PMEM 3
+#define OMAP_ABE_AESS 4
+
+/* Distinction between Read and Write from/to ABE memory
+ * is useful for simulation tool */
+static inline void omap_abe_mem_write(struct omap_abe *abe, int bank,
+ u32 offset, u32 *src, size_t bytes)
+{
+ memcpy((abe->io_base[bank] + offset), src, bytes);
+}
+
+static inline void omap_abe_mem_read(struct omap_abe *abe, int bank,
+ u32 offset, u32 *dest, size_t bytes)
+{
+ memcpy(dest, (abe->io_base[bank] + offset), bytes);
+}
+
+static inline u32 omap_abe_reg_readl(struct omap_abe *abe, u32 offset)
+{
+ return __raw_readl(abe->io_base[OMAP_ABE_AESS] + offset);
+}
+
+static inline void omap_abe_reg_writel(struct omap_abe *abe,
+ u32 offset, u32 val)
+{
+ __raw_writel(val, (abe->io_base[OMAP_ABE_AESS] + offset));
+}
+
+static inline void *omap_abe_reset_mem(struct omap_abe *abe, int bank,
+ u32 offset, size_t bytes)
+{
+ return memset(abe->io_base[bank] + offset, 0, bytes);
+}
+
+#endif /*_ABE_MEM_H_*/
diff --git a/sound/soc/omap/abe/abe_port.c b/sound/soc/omap/abe/abe_port.c
new file mode 100644
index 0000000..1923d35
--- /dev/null
+++ b/sound/soc/omap/abe/abe_port.c
@@ -0,0 +1,1747 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#include "abe_legacy.h"
+#include "abe_port.h"
+#include "abe_dbg.h"
+#include "abe_mem.h"
+#include "abe_gain.h"
+
+/**
+ * abe_clean_temporay buffers
+ *
+ * clear temporary buffers
+ */
+void omap_abe_clean_temporary_buffers(struct omap_abe *abe, u32 id)
+{
+ switch (id) {
+ case OMAP_ABE_DMIC_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_DMIC_UL_FIFO_ADDR,
+ OMAP_ABE_D_DMIC_UL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_DMIC0_96_48_DATA_ADDR,
+ OMAP_ABE_S_DMIC0_96_48_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_DMIC1_96_48_DATA_ADDR,
+ OMAP_ABE_S_DMIC1_96_48_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_DMIC2_96_48_DATA_ADDR,
+ OMAP_ABE_S_DMIC2_96_48_DATA_SIZE);
+ /* reset working values of the gain, target gain is preserved */
+ omap_abe_reset_gain_mixer(abe, GAINS_DMIC1, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DMIC1, GAIN_RIGHT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DMIC2, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DMIC2, GAIN_RIGHT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DMIC3, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DMIC3, GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_PDM_UL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MCPDM_UL_FIFO_ADDR,
+ OMAP_ABE_D_MCPDM_UL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_AMIC_96_48_DATA_ADDR,
+ OMAP_ABE_S_AMIC_96_48_DATA_SIZE);
+ /* reset working values of the gain, target gain is preserved */
+ omap_abe_reset_gain_mixer(abe, GAINS_AMIC, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_AMIC, GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_BT_VX_UL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_BT_UL_FIFO_ADDR,
+ OMAP_ABE_D_BT_UL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_UL_ADDR,
+ OMAP_ABE_S_BT_UL_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_UL_8_48_HP_DATA_ADDR,
+ OMAP_ABE_S_BT_UL_8_48_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_UL_8_48_LP_DATA_ADDR,
+ OMAP_ABE_S_BT_UL_8_48_LP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_UL_16_48_HP_DATA_ADDR,
+ OMAP_ABE_S_BT_UL_16_48_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_UL_16_48_LP_DATA_ADDR,
+ OMAP_ABE_S_BT_UL_16_48_LP_DATA_SIZE);
+ /* reset working values of the gain, target gain is preserved */
+ omap_abe_reset_gain_mixer(abe, GAINS_BTUL, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_BTUL, GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_MM_UL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MM_UL_FIFO_ADDR,
+ OMAP_ABE_D_MM_UL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_MM_UL_ADDR,
+ OMAP_ABE_S_MM_UL_SIZE);
+ break;
+ case OMAP_ABE_MM_UL2_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MM_UL2_FIFO_ADDR,
+ OMAP_ABE_D_MM_UL2_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_MM_UL2_ADDR,
+ OMAP_ABE_S_MM_UL2_SIZE);
+ break;
+ case OMAP_ABE_VX_UL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_VX_UL_FIFO_ADDR,
+ OMAP_ABE_D_VX_UL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_UL_ADDR,
+ OMAP_ABE_S_VX_UL_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_UL_48_8_HP_DATA_ADDR,
+ OMAP_ABE_S_VX_UL_48_8_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_UL_48_8_LP_DATA_ADDR,
+ OMAP_ABE_S_VX_UL_48_8_LP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_UL_48_16_HP_DATA_ADDR,
+ OMAP_ABE_S_VX_UL_48_16_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_UL_48_16_LP_DATA_ADDR,
+ OMAP_ABE_S_VX_UL_48_16_LP_DATA_SIZE);
+ omap_abe_reset_gain_mixer(abe, MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+ break;
+ case OMAP_ABE_MM_DL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MM_DL_FIFO_ADDR,
+ OMAP_ABE_D_MM_DL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_MM_DL_ADDR,
+ OMAP_ABE_S_MM_DL_SIZE);
+ omap_abe_reset_gain_mixer(abe, MIXDL1, MIX_DL1_INPUT_MM_DL);
+ omap_abe_reset_gain_mixer(abe, MIXDL2, MIX_DL2_INPUT_MM_DL);
+ break;
+ case OMAP_ABE_VX_DL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_VX_DL_FIFO_ADDR,
+ OMAP_ABE_D_VX_DL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_DL_ADDR,
+ OMAP_ABE_S_VX_DL_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_DL_8_48_HP_DATA_ADDR,
+ OMAP_ABE_S_VX_DL_8_48_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_DL_8_48_LP_DATA_ADDR,
+ OMAP_ABE_S_VX_DL_8_48_LP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_DL_8_48_OSR_LP_DATA_ADDR,
+ OMAP_ABE_S_VX_DL_8_48_OSR_LP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_DL_16_48_HP_DATA_ADDR,
+ OMAP_ABE_S_VX_DL_16_48_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VX_DL_16_48_LP_DATA_ADDR,
+ OMAP_ABE_S_VX_DL_16_48_LP_DATA_SIZE);
+ omap_abe_reset_gain_mixer(abe, MIXDL1, MIX_DL1_INPUT_VX_DL);
+ omap_abe_reset_gain_mixer(abe, MIXDL2, MIX_DL2_INPUT_VX_DL);
+ break;
+ case OMAP_ABE_TONES_DL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_TONES_DL_FIFO_ADDR,
+ OMAP_ABE_D_TONES_DL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_TONES_ADDR,
+ OMAP_ABE_S_TONES_SIZE);
+ omap_abe_reset_gain_mixer(abe, MIXDL1, MIX_DL1_INPUT_TONES);
+ omap_abe_reset_gain_mixer(abe, MIXDL2, MIX_DL2_INPUT_TONES);
+ break;
+ case OMAP_ABE_VIB_DL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_VIB_DL_FIFO_ADDR,
+ OMAP_ABE_D_VIB_DL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_VIBRA_ADDR,
+ OMAP_ABE_S_VIBRA_SIZE);
+ break;
+ case OMAP_ABE_BT_VX_DL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_BT_DL_FIFO_ADDR,
+ OMAP_ABE_D_BT_DL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_DL_ADDR,
+ OMAP_ABE_S_BT_DL_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_DL_48_8_HP_DATA_ADDR,
+ OMAP_ABE_S_BT_DL_48_8_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_DL_48_8_LP_DATA_ADDR,
+ OMAP_ABE_S_BT_DL_48_8_LP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_DL_48_16_HP_DATA_ADDR,
+ OMAP_ABE_S_BT_DL_48_16_HP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_BT_DL_48_16_LP_DATA_ADDR,
+ OMAP_ABE_S_BT_DL_48_16_LP_DATA_SIZE);
+ break;
+ case OMAP_ABE_PDM_DL_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MCPDM_DL_FIFO_ADDR,
+ OMAP_ABE_D_MCPDM_DL_FIFO_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_DL2_M_LR_EQ_DATA_ADDR,
+ OMAP_ABE_S_DL2_M_LR_EQ_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_DL1_M_EQ_DATA_ADDR,
+ OMAP_ABE_S_DL1_M_EQ_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_EARP_48_96_LP_DATA_ADDR,
+ OMAP_ABE_S_EARP_48_96_LP_DATA_SIZE);
+ omap_abe_reset_mem(abe, OMAP_ABE_SMEM,
+ OMAP_ABE_S_IHF_48_96_LP_DATA_ADDR,
+ OMAP_ABE_S_IHF_48_96_LP_DATA_SIZE);
+ omap_abe_reset_gain_mixer(abe, GAINS_DL1, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DL1, GAIN_RIGHT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DL2, GAIN_LEFT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, GAINS_DL2, GAIN_RIGHT_OFFSET);
+ omap_abe_reset_gain_mixer(abe, MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+ omap_abe_reset_gain_mixer(abe, MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+ break;
+ case OMAP_ABE_MM_EXT_OUT_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MM_EXT_OUT_FIFO_ADDR,
+ OMAP_ABE_D_MM_EXT_OUT_FIFO_SIZE);
+ break;
+ case OMAP_ABE_MM_EXT_IN_PORT:
+ omap_abe_reset_mem(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MM_EXT_IN_FIFO_ADDR,
+ OMAP_ABE_D_MM_EXT_IN_FIFO_SIZE);
+ break;
+ }
+}
+
+/**
+ * omap_abe_disable_enable_dma_request
+ * Parameter:
+ * Operations:
+ * Return value:
+ */
+void omap_abe_disable_enable_dma_request(struct omap_abe *abe, u32 id,
+ u32 on_off)
+{
+ u8 desc_third_word[4], irq_dmareq_field;
+ u32 sio_desc_address;
+ u32 struct_offset;
+ struct ABE_SIODescriptor sio_desc;
+ struct ABE_SPingPongDescriptor desc_pp;
+
+ if (abe_port[id].protocol.protocol_switch == PINGPONG_PORT_PROT) {
+ irq_dmareq_field =
+ (u8) (on_off *
+ abe_port[id].protocol.p.prot_pingpong.irq_data);
+ sio_desc_address = OMAP_ABE_D_PINGPONGDESC_ADDR;
+ struct_offset = (u32) &(desc_pp.data_size) - (u32) &(desc_pp);
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ sio_desc_address + struct_offset,
+ (u32 *) desc_third_word, 4);
+ desc_third_word[2] = irq_dmareq_field;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ sio_desc_address + struct_offset,
+ (u32 *) desc_third_word, 4);
+ } else {
+ /* serial interface: sync ATC with Firmware activity */
+ sio_desc_address =
+ OMAP_ABE_D_IODESCR_ADDR +
+ (id * sizeof(struct ABE_SIODescriptor));
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ sio_desc_address, (u32 *) &sio_desc,
+ sizeof(sio_desc));
+ if (on_off) {
+ if (abe_port[id].protocol.protocol_switch != SERIAL_PORT_PROT)
+ sio_desc.atc_irq_data =
+ (u8) abe_port[id].protocol.p.prot_dmareq.
+ dma_data;
+ sio_desc.on_off = 0x80;
+ } else {
+ sio_desc.atc_irq_data = 0;
+ sio_desc.on_off = 0;
+ }
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ sio_desc_address, (u32 *) &sio_desc,
+ sizeof(sio_desc));
+ }
+
+}
+
+/**
+ * omap_abe_enable_dma_request
+ *
+ * Parameter:
+ * Operations:
+ * Return value:
+ *
+ */
+void omap_abe_enable_dma_request(struct omap_abe *abe, u32 id)
+{
+ omap_abe_disable_enable_dma_request(abe, id, 1);
+}
+
+/**
+ * omap_abe_disable_dma_request
+ *
+ * Parameter:
+ * Operations:
+ * Return value:
+ *
+ */
+void omap_abe_disable_dma_request(struct omap_abe *abe, u32 id)
+{
+ omap_abe_disable_enable_dma_request(abe, id, 0);
+}
+
+/**
+ * abe_init_atc
+ * @id: ABE port ID
+ *
+ * load the DMEM ATC/AESS descriptors
+ */
+void omap_abe_init_atc(struct omap_abe *abe, u32 id)
+{
+ u8 iter;
+ s32 datasize;
+ struct omap_abe_atc_desc atc_desc;
+
+#define JITTER_MARGIN 4
+ /* load default values of the descriptor */
+ atc_desc.rdpt = 0;
+ atc_desc.wrpt = 0;
+ atc_desc.irqdest = 0;
+ atc_desc.cberr = 0;
+ atc_desc.desen = 0;
+ atc_desc.nw = 0;
+ atc_desc.reserved0 = 0;
+ atc_desc.reserved1 = 0;
+ atc_desc.reserved2 = 0;
+ atc_desc.srcid = 0;
+ atc_desc.destid = 0;
+ atc_desc.badd = 0;
+ atc_desc.iter = 0;
+ atc_desc.cbsize = 0;
+ datasize = abe_dma_port_iter_factor(&((abe_port[id]).format));
+ iter = (u8) abe_dma_port_iteration(&((abe_port[id]).format));
+ /* if the ATC FIFO is too small there will be two ABE firmware
+ utasks to do the copy this happems on DMIC and MCPDMDL */
+ /* VXDL_8kMono = 4 = 2 + 2x1 */
+ /* VXDL_16kstereo = 12 = 8 + 2x2 */
+ /* MM_DL_1616 = 14 = 12 + 2x1 */
+ /* DMIC = 84 = 72 + 2x6 */
+ /* VXUL_8kMono = 2 */
+ /* VXUL_16kstereo = 4 */
+ /* MM_UL2_Stereo = 4 */
+ /* PDMDL = 12 */
+ /* IN from AESS point of view */
+ if (abe_port[id].protocol.direction == ABE_ATC_DIRECTION_IN)
+ if (iter + 2 * datasize > 126)
+ atc_desc.wrpt = (iter >> 1) +
+ ((JITTER_MARGIN-1) * datasize);
+ else
+ atc_desc.wrpt = iter + ((JITTER_MARGIN-1) * datasize);
+ else
+ atc_desc.wrpt = 0 + ((JITTER_MARGIN+1) * datasize);
+ switch ((abe_port[id]).protocol.protocol_switch) {
+ case SLIMBUS_PORT_PROT:
+ atc_desc.cbdir = (abe_port[id]).protocol.direction;
+ atc_desc.cbsize =
+ (abe_port[id]).protocol.p.prot_slimbus.buf_size;
+ atc_desc.badd =
+ ((abe_port[id]).protocol.p.prot_slimbus.buf_addr1) >> 4;
+ atc_desc.iter = (abe_port[id]).protocol.p.prot_slimbus.iter;
+ atc_desc.srcid =
+ abe_atc_srcid[(abe_port[id]).protocol.p.prot_slimbus.
+ desc_addr1 >> 3];
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_slimbus.
+ desc_addr1, (u32 *) &atc_desc, sizeof(atc_desc));
+ atc_desc.badd =
+ (abe_port[id]).protocol.p.prot_slimbus.buf_addr2;
+ atc_desc.srcid =
+ abe_atc_srcid[(abe_port[id]).protocol.p.prot_slimbus.
+ desc_addr2 >> 3];
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_slimbus.
+ desc_addr2, (u32 *) &atc_desc, sizeof(atc_desc));
+ break;
+ case SERIAL_PORT_PROT:
+ atc_desc.cbdir = (abe_port[id]).protocol.direction;
+ atc_desc.cbsize =
+ (abe_port[id]).protocol.p.prot_serial.buf_size;
+ atc_desc.badd =
+ ((abe_port[id]).protocol.p.prot_serial.buf_addr) >> 4;
+ atc_desc.iter = (abe_port[id]).protocol.p.prot_serial.iter;
+ atc_desc.srcid =
+ abe_atc_srcid[(abe_port[id]).protocol.p.prot_serial.
+ desc_addr >> 3];
+ atc_desc.destid =
+ abe_atc_dstid[(abe_port[id]).protocol.p.prot_serial.
+ desc_addr >> 3];
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_serial.desc_addr,
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ break;
+ case DMIC_PORT_PROT:
+ atc_desc.cbdir = ABE_ATC_DIRECTION_IN;
+ atc_desc.cbsize = (abe_port[id]).protocol.p.prot_dmic.buf_size;
+ atc_desc.badd =
+ ((abe_port[id]).protocol.p.prot_dmic.buf_addr) >> 4;
+ atc_desc.iter = DMIC_ITER;
+ atc_desc.srcid = abe_atc_srcid[ABE_ATC_DMIC_DMA_REQ];
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (ABE_ATC_DMIC_DMA_REQ*ATC_SIZE),
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ break;
+ case MCPDMDL_PORT_PROT:
+ atc_desc.cbdir = ABE_ATC_DIRECTION_OUT;
+ atc_desc.cbsize =
+ (abe_port[id]).protocol.p.prot_mcpdmdl.buf_size;
+ atc_desc.badd =
+ ((abe_port[id]).protocol.p.prot_mcpdmdl.buf_addr) >> 4;
+ atc_desc.iter = MCPDM_DL_ITER;
+ atc_desc.destid = abe_atc_dstid[ABE_ATC_MCPDMDL_DMA_REQ];
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (ABE_ATC_MCPDMDL_DMA_REQ*ATC_SIZE),
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ break;
+ case MCPDMUL_PORT_PROT:
+ atc_desc.cbdir = ABE_ATC_DIRECTION_IN;
+ atc_desc.cbsize =
+ (abe_port[id]).protocol.p.prot_mcpdmul.buf_size;
+ atc_desc.badd =
+ ((abe_port[id]).protocol.p.prot_mcpdmul.buf_addr) >> 4;
+ atc_desc.iter = MCPDM_UL_ITER;
+ atc_desc.srcid = abe_atc_srcid[ABE_ATC_MCPDMUL_DMA_REQ];
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (ABE_ATC_MCPDMUL_DMA_REQ*ATC_SIZE),
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ break;
+ case PINGPONG_PORT_PROT:
+ /* software protocol, nothing to do on ATC */
+ break;
+ case DMAREQ_PORT_PROT:
+ atc_desc.cbdir = (abe_port[id]).protocol.direction;
+ atc_desc.cbsize =
+ (abe_port[id]).protocol.p.prot_dmareq.buf_size;
+ atc_desc.badd =
+ ((abe_port[id]).protocol.p.prot_dmareq.buf_addr) >> 4;
+ /* CBPr needs ITER=1.
+ It is the job of eDMA to do the iterations */
+ atc_desc.iter = 1;
+ /* input from ABE point of view */
+ if (abe_port[id].protocol.direction == ABE_ATC_DIRECTION_IN) {
+ /* atc_atc_desc.rdpt = 127; */
+ /* atc_atc_desc.wrpt = 0; */
+ atc_desc.srcid = abe_atc_srcid
+ [(abe_port[id]).protocol.p.prot_dmareq.
+ desc_addr >> 3];
+ } else {
+ /* atc_atc_desc.rdpt = 0; */
+ /* atc_atc_desc.wrpt = 127; */
+ atc_desc.destid = abe_atc_dstid
+ [(abe_port[id]).protocol.p.prot_dmareq.
+ desc_addr >> 3];
+ }
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_dmareq.desc_addr,
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ break;
+ }
+}
+
+/**
+ * omap_abe_enable_pp_io_task
+ * @id: port_id
+ *
+ *
+ */
+void omap_abe_enable_pp_io_task(struct omap_abe *abe, u32 id)
+{
+ if (OMAP_ABE_MM_DL_PORT == id) {
+ /* MM_DL managed in ping-pong */
+ abe->MultiFrame[TASK_IO_MM_DL_SLT][TASK_IO_MM_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_IO_PING_PONG);
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MULTIFRAME_ADDR, (u32 *) abe->MultiFrame,
+ sizeof(abe->MultiFrame));
+ } else {
+ /* ping_pong is only supported on MM_DL */
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+}
+/**
+ * omap_abe_disable_pp_io_task
+ * @id: port_id
+ *
+ *
+ */
+void omap_abe_disable_pp_io_task(struct omap_abe *abe, u32 id)
+{
+ if (OMAP_ABE_MM_DL_PORT == id) {
+ /* MM_DL managed in ping-pong */
+ abe->MultiFrame[TASK_IO_MM_DL_SLT][TASK_IO_MM_DL_IDX] = 0;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MULTIFRAME_ADDR, (u32 *) abe->MultiFrame,
+ sizeof(abe->MultiFrame));
+ } else {
+ /* ping_pong is only supported on MM_DL */
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+}
+
+/**
+ * omap_abe_disable_data_transfer
+ * @id: ABE port id
+ *
+ * disables the ATC descriptor and stop IO/port activities
+ * disable the IO task (@f = 0)
+ * clear ATC DMEM buffer, ATC enabled
+ */
+int omap_abe_disable_data_transfer(struct omap_abe *abe, u32 id)
+{
+
+ _log(ABE_ID_DISABLE_DATA_TRANSFER, id, 0, 0);
+
+ /* MM_DL managed in ping-pong */
+ if (id == OMAP_ABE_MM_DL_PORT) {
+ if (abe_port[OMAP_ABE_MM_DL_PORT].protocol.protocol_switch == PINGPONG_PORT_PROT)
+ omap_abe_disable_pp_io_task(abe, OMAP_ABE_MM_DL_PORT);
+ }
+ /* local host variable status= "port is running" */
+ abe_port[id].status = OMAP_ABE_PORT_ACTIVITY_IDLE;
+ /* disable DMA requests */
+ omap_abe_disable_dma_request(abe, id);
+ /* disable ATC transfers */
+ omap_abe_init_atc(abe, id);
+ omap_abe_clean_temporary_buffers(abe, id);
+ /* select the main port based on the desactivation of this port */
+ abe_decide_main_port();
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_disable_data_transfer);
+
+/**
+ * omap_abe_enable_data_transfer
+ * @ip: ABE port id
+ *
+ * enables the ATC descriptor
+ * reset ATC pointers
+ * enable the IO task (@f <> 0)
+ */
+int omap_abe_enable_data_transfer(struct omap_abe *abe, u32 id)
+{
+ abe_port_protocol_t *protocol;
+ abe_data_format_t format;
+
+ _log(ABE_ID_ENABLE_DATA_TRANSFER, id, 0, 0);
+ omap_abe_clean_temporary_buffers(abe, id);
+ if (id == OMAP_ABE_PDM_UL_PORT) {
+ /* initializes the ABE ATC descriptors in DMEM - MCPDM_UL */
+ protocol = &(abe_port[OMAP_ABE_PDM_UL_PORT].protocol);
+ format = abe_port[OMAP_ABE_PDM_UL_PORT].format;
+ omap_abe_init_atc(abe, OMAP_ABE_PDM_UL_PORT);
+ abe_init_io_tasks(OMAP_ABE_PDM_UL_PORT, &format, protocol);
+ }
+ if (id == OMAP_ABE_PDM_DL_PORT) {
+ /* initializes the ABE ATC descriptors in DMEM - MCPDM_DL */
+ protocol = &(abe_port[OMAP_ABE_PDM_DL_PORT].protocol);
+ format = abe_port[OMAP_ABE_PDM_DL_PORT].format;
+ omap_abe_init_atc(abe, OMAP_ABE_PDM_DL_PORT);
+ abe_init_io_tasks(OMAP_ABE_PDM_DL_PORT, &format, protocol);
+ }
+ /* MM_DL managed in ping-pong */
+ if (id == OMAP_ABE_MM_DL_PORT) {
+ protocol = &(abe_port[OMAP_ABE_MM_DL_PORT].protocol);
+ if (protocol->protocol_switch == PINGPONG_PORT_PROT)
+ omap_abe_enable_pp_io_task(abe, OMAP_ABE_MM_DL_PORT);
+ }
+ if (id == OMAP_ABE_DMIC_PORT) {
+ /* one DMIC port enabled = all DMICs enabled,
+ * since there is a single DMIC path for all DMICs */
+ protocol = &(abe_port[OMAP_ABE_DMIC_PORT].protocol);
+ format = abe_port[OMAP_ABE_DMIC_PORT].format;
+ omap_abe_init_atc(abe, OMAP_ABE_DMIC_PORT);
+ abe_init_io_tasks(OMAP_ABE_DMIC_PORT, &format, protocol);
+ }
+ if (id == OMAP_ABE_VX_UL_PORT) {
+ if (abe_port[OMAP_ABE_VX_DL_PORT].status == OMAP_ABE_PORT_ACTIVITY_RUNNING) {
+ /* VX_DL port already started, hence no need to
+ initialize ASRC */
+ } else {
+ /* Init VX_UL ASRC & VX_DL ASRC and enable its adaptation */
+ abe_init_asrc_vx_ul(-250);
+ abe_init_asrc_vx_dl(250);
+ }
+ }
+ if (id == OMAP_ABE_VX_DL_PORT) {
+ if (abe_port[OMAP_ABE_VX_UL_PORT].status == OMAP_ABE_PORT_ACTIVITY_RUNNING) {
+ /* VX_UL port already started, hence no need to
+ initialize ASRC */
+ } else {
+ /* Init VX_UL ASRC & VX_DL ASRC and enable its adaptation */
+ abe_init_asrc_vx_ul(-250);
+ abe_init_asrc_vx_dl(250);
+ }
+ }
+
+ /* local host variable status= "port is running" */
+ abe_port[id].status = OMAP_ABE_PORT_ACTIVITY_RUNNING;
+ /* enable DMA requests */
+ omap_abe_enable_dma_request(abe, id);
+ /* select the main port based on the activation of this new port */
+ abe_decide_main_port();
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_enable_data_transfer);
+
+/**
+ * omap_abe_connect_cbpr_dmareq_port
+ * @id: port name
+ * @f: desired data format
+ * @d: desired dma_request line (0..7)
+ * @a: returned pointer to the base address of the CBPr register and number of
+ * samples to exchange during a DMA_request.
+ *
+ * enables the data echange between a DMA and the ABE through the
+ * CBPr registers of AESS.
+ */
+int omap_abe_connect_cbpr_dmareq_port(struct omap_abe *abe,
+ u32 id, abe_data_format_t *f,
+ u32 d,
+ abe_dma_t *returned_dma_t)
+{
+ _log(ABE_ID_CONNECT_CBPR_DMAREQ_PORT, id, f->f, f->samp_format);
+
+ abe_port[id] = ((abe_port_t *) abe_port_init)[id];
+ (abe_port[id]).format = (*f);
+ abe_port[id].protocol.protocol_switch = DMAREQ_PORT_PROT;
+ abe_port[id].protocol.p.prot_dmareq.iter = abe_dma_port_iteration(f);
+ abe_port[id].protocol.p.prot_dmareq.dma_addr = ABE_DMASTATUS_RAW;
+ abe_port[id].protocol.p.prot_dmareq.dma_data = (1 << d);
+ /* load the dma_t with physical information from AE memory mapping */
+ abe_init_dma_t(id, &((abe_port[id]).protocol));
+
+ /* load the micro-task parameters */
+ abe_init_io_tasks(id, &((abe_port[id]).format),
+ &((abe_port[id]).protocol));
+ abe_port[id].status = OMAP_ABE_PORT_INITIALIZED;
+
+ /* load the ATC descriptors - disabled */
+ omap_abe_init_atc(abe, id);
+ /* return the dma pointer address */
+ abe_read_port_address(id, returned_dma_t);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_connect_cbpr_dmareq_port);
+
+/**
+ * omap_abe_connect_irq_ping_pong_port
+ * @id: port name
+ * @f: desired data format
+ * @I: index of the call-back subroutine to call
+ * @s: half-buffer (ping) size
+ * @p: returned base address of the first (ping) buffer)
+ *
+ * enables the data echanges between a direct access to the DMEM
+ * memory of ABE using cache flush. On each IRQ activation a subroutine
+ * registered with "abe_plug_subroutine" will be called. This subroutine
+ * will generate an amount of samples, send them to DMEM memory and call
+ * "abe_set_ping_pong_buffer" to notify the new amount of samples in the
+ * pong buffer.
+ */
+int omap_abe_connect_irq_ping_pong_port(struct omap_abe *abe,
+ u32 id, abe_data_format_t *f,
+ u32 subroutine_id, u32 size,
+ u32 *sink, u32 dsp_mcu_flag)
+{
+ _log(ABE_ID_CONNECT_IRQ_PING_PONG_PORT, id, f->f, f->samp_format);
+
+ /* ping_pong is only supported on MM_DL */
+ if (id != OMAP_ABE_MM_DL_PORT) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+ abe_port[id] = ((abe_port_t *) abe_port_init)[id];
+ (abe_port[id]).format = (*f);
+ (abe_port[id]).protocol.protocol_switch = PINGPONG_PORT_PROT;
+ (abe_port[id]).protocol.p.prot_pingpong.buf_addr =
+ OMAP_ABE_D_PING_ADDR;
+ (abe_port[id]).protocol.p.prot_pingpong.buf_size = size;
+ (abe_port[id]).protocol.p.prot_pingpong.irq_data = (1);
+ abe_init_ping_pong_buffer(OMAP_ABE_MM_DL_PORT, size, 2, sink);
+ if (dsp_mcu_flag == PING_PONG_WITH_MCU_IRQ)
+ (abe_port[id]).protocol.p.prot_pingpong.irq_addr =
+ ABE_MCU_IRQSTATUS_RAW;
+ if (dsp_mcu_flag == PING_PONG_WITH_DSP_IRQ)
+ (abe_port[id]).protocol.p.prot_pingpong.irq_addr =
+ ABE_DSP_IRQSTATUS_RAW;
+ abe_port[id].status = OMAP_ABE_PORT_INITIALIZED;
+ /* load the micro-task parameters */
+ abe_init_io_tasks(id, &((abe_port[id]).format),
+ &((abe_port[id]).protocol));
+ /* load the ATC descriptors - disabled */
+ omap_abe_init_atc(abe, id);
+ *sink = (abe_port[id]).protocol.p.prot_pingpong.buf_addr;
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_connect_irq_ping_pong_port);
+
+/**
+ * omap_abe_connect_serial_port()
+ * @id: port name
+ * @f: data format
+ * @i: peripheral ID (McBSP #1, #2, #3)
+ *
+ * Operations : enables the data echanges between a McBSP and an ATC buffer in
+ * DMEM. This API is used connect 48kHz McBSP streams to MM_DL and 8/16kHz
+ * voice streams to VX_UL, VX_DL, BT_VX_UL, BT_VX_DL. It abstracts the
+ * abe_write_port API.
+ */
+int omap_abe_connect_serial_port(struct omap_abe *abe,
+ u32 id, abe_data_format_t *f,
+ u32 mcbsp_id)
+{
+ _log(ABE_ID_CONNECT_SERIAL_PORT, id, f->samp_format, mcbsp_id);
+
+ abe_port[id] = ((abe_port_t *) abe_port_init)[id];
+ (abe_port[id]).format = (*f);
+ (abe_port[id]).protocol.protocol_switch = SERIAL_PORT_PROT;
+ /* McBSP peripheral connected to ATC */
+ (abe_port[id]).protocol.p.prot_serial.desc_addr = mcbsp_id*ATC_SIZE;
+ /* check the iteration of ATC */
+ (abe_port[id]).protocol.p.prot_serial.iter =
+ abe_dma_port_iter_factor(f);
+
+ /* load the micro-task parameters */
+ abe_init_io_tasks(id, &((abe_port[id]).format),
+ &((abe_port[id]).protocol));
+ abe_port[id].status = OMAP_ABE_PORT_INITIALIZED;
+
+ /* load the ATC descriptors - disabled */
+ omap_abe_init_atc(abe, id);
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_connect_serial_port);
+
+/**
+ * omap_abe_read_port_address
+ * @dma: output pointer to the DMA iteration and data destination pointer
+ *
+ * This API returns the address of the DMA register used on this audio port.
+ * Depending on the protocol being used, adds the base address offset L3
+ * (DMA) or MPU (ARM)
+ */
+int omap_abe_read_port_address(struct omap_abe *abe,
+ u32 port, abe_dma_t *dma2)
+{
+ abe_dma_t_offset dma1;
+ u32 protocol_switch;
+
+ _log(ABE_ID_READ_PORT_ADDRESS, port, 0, 0);
+
+ dma1 = (abe_port[port]).dma;
+ protocol_switch = abe_port[port].protocol.protocol_switch;
+ switch (protocol_switch) {
+ case PINGPONG_PORT_PROT:
+ /* return the base address of the buffer in L3 and L4 spaces */
+ (*dma2).data = (void *)(dma1.data +
+ ABE_DEFAULT_BASE_ADDRESS_L3 + ABE_DMEM_BASE_OFFSET_MPU);
+ (*dma2).l3_dmem = (void *)(dma1.data +
+ ABE_DEFAULT_BASE_ADDRESS_L3 + ABE_DMEM_BASE_OFFSET_MPU);
+ (*dma2).l4_dmem = (void *)(dma1.data +
+ ABE_DEFAULT_BASE_ADDRESS_L4 + ABE_DMEM_BASE_OFFSET_MPU);
+ break;
+ case DMAREQ_PORT_PROT:
+ /* return the CBPr(L3), DMEM(L3), DMEM(L4) address */
+ (*dma2).data = (void *)(dma1.data +
+ ABE_DEFAULT_BASE_ADDRESS_L3 + ABE_ATC_BASE_OFFSET_MPU);
+ (*dma2).l3_dmem =
+ (void *)((abe_port[port]).protocol.p.prot_dmareq.buf_addr +
+ ABE_DEFAULT_BASE_ADDRESS_L3 + ABE_DMEM_BASE_OFFSET_MPU);
+ (*dma2).l4_dmem =
+ (void *)((abe_port[port]).protocol.p.prot_dmareq.buf_addr +
+ ABE_DEFAULT_BASE_ADDRESS_L4 + ABE_DMEM_BASE_OFFSET_MPU);
+ break;
+ default:
+ break;
+ }
+ (*dma2).iter = (dma1.iter);
+
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_read_port_address);
+
+/**
+ * abe_init_dma_t
+ * @ id: ABE port ID
+ * @ prot: protocol being used
+ *
+ * load the dma_t with physical information from AE memory mapping
+ */
+void abe_init_dma_t(u32 id, abe_port_protocol_t *prot)
+{
+ abe_dma_t_offset dma;
+ u32 idx;
+ /* default dma_t points to address 0000... */
+ dma.data = 0;
+ dma.iter = 0;
+ switch (prot->protocol_switch) {
+ case PINGPONG_PORT_PROT:
+ for (idx = 0; idx < 32; idx++) {
+ if (((prot->p).prot_pingpong.irq_data) ==
+ (u32) (1 << idx))
+ break;
+ }
+ (prot->p).prot_dmareq.desc_addr =
+ ((CBPr_DMA_RTX0 + idx)*ATC_SIZE);
+ /* translate byte address/size in DMEM words */
+ dma.data = (prot->p).prot_pingpong.buf_addr >> 2;
+ dma.iter = (prot->p).prot_pingpong.buf_size >> 2;
+ break;
+ case DMAREQ_PORT_PROT:
+ for (idx = 0; idx < 32; idx++) {
+ if (((prot->p).prot_dmareq.dma_data) ==
+ (u32) (1 << idx))
+ break;
+ }
+ dma.data = (CIRCULAR_BUFFER_PERIPHERAL_R__0 + (idx << 2));
+ dma.iter = (prot->p).prot_dmareq.iter;
+ (prot->p).prot_dmareq.desc_addr =
+ ((CBPr_DMA_RTX0 + idx)*ATC_SIZE);
+ break;
+ case SLIMBUS_PORT_PROT:
+ case SERIAL_PORT_PROT:
+ case DMIC_PORT_PROT:
+ case MCPDMDL_PORT_PROT:
+ case MCPDMUL_PORT_PROT:
+ default:
+ break;
+ }
+ /* upload the dma type */
+ abe_port[id].dma = dma;
+}
+
+/**
+ * abe_enable_atc
+ * Parameter:
+ * Operations:
+ * Return value:
+ */
+void abe_enable_atc(u32 id)
+{
+ struct omap_abe_atc_desc atc_desc;
+
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_dmareq.desc_addr,
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ atc_desc.desen = 1;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_dmareq.desc_addr,
+ (u32 *) &atc_desc, sizeof(atc_desc));
+
+}
+/**
+ * abe_disable_atc
+ * Parameter:
+ * Operations:
+ * Return value:
+ */
+void abe_disable_atc(u32 id)
+{
+ struct omap_abe_atc_desc atc_desc;
+
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_dmareq.desc_addr,
+ (u32 *) &atc_desc, sizeof(atc_desc));
+ atc_desc.desen = 0;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ (abe_port[id]).protocol.p.prot_dmareq.desc_addr,
+ (u32 *) &atc_desc, sizeof(atc_desc));
+
+}
+/**
+ * abe_init_io_tasks
+ * @prot : protocol being used
+ *
+ * load the micro-task parameters doing to DMEM <==> SMEM data moves
+ *
+ * I/O descriptors input parameters :
+ * For Read from DMEM usually THR1/THR2 = X+1/X-1
+ * For Write to DMEM usually THR1/THR2 = 2/0
+ * UP_1/2 =X+1/X-1
+ */
+void abe_init_io_tasks(u32 id, abe_data_format_t *format,
+ abe_port_protocol_t *prot)
+{
+ u32 x_io, direction, iter_samples, smem1, smem2, smem3, io_sub_id,
+ io_flag;
+ u32 copy_func_index, before_func_index, after_func_index;
+ u32 dmareq_addr, dmareq_field;
+ u32 sio_desc_address, datasize, iter, nsamp, datasize2, dOppMode32;
+ u32 atc_ptr_saved, atc_ptr_saved2, copy_func_index1;
+ u32 copy_func_index2, atc_desc_address1, atc_desc_address2;
+ struct ABE_SPingPongDescriptor desc_pp;
+ struct ABE_SIODescriptor sio_desc;
+
+ if (prot->protocol_switch == PINGPONG_PORT_PROT) {
+ /* ping_pong is only supported on MM_DL */
+ if (OMAP_ABE_MM_DL_PORT != id) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_PARAMETER_ERROR);
+ }
+ smem1 = smem_mm_dl;
+ copy_func_index = (u8) abe_dma_port_copy_subroutine_id(id);
+ dmareq_addr = abe_port[id].protocol.p.prot_pingpong.irq_addr;
+ dmareq_field = abe_port[id].protocol.p.prot_pingpong.irq_data;
+ datasize = abe_dma_port_iter_factor(format);
+ /* number of "samples" either mono or stereo */
+ iter = abe_dma_port_iteration(format);
+ iter_samples = (iter / datasize);
+ /* load the IO descriptor */
+ /* no drift */
+ desc_pp.drift_ASRC = 0;
+ /* no drift */
+ desc_pp.drift_io = 0;
+ desc_pp.hw_ctrl_addr = (u16) dmareq_addr;
+ desc_pp.copy_func_index = (u8) copy_func_index;
+ desc_pp.smem_addr = (u8) smem1;
+ /* DMA req 0 is used for CBPr0 */
+ desc_pp.atc_irq_data = (u8) dmareq_field;
+ /* size of block transfer */
+ desc_pp.x_io = (u8) iter_samples;
+ desc_pp.data_size = (u8) datasize;
+ /* address comunicated in Bytes */
+ desc_pp.workbuff_BaseAddr =
+ (u16) (abe_base_address_pingpong[1]);
+ /* size comunicated in XIO sample */
+ desc_pp.workbuff_Samples = 0;
+ desc_pp.nextbuff0_BaseAddr =
+ (u16) (abe_base_address_pingpong[0]);
+ desc_pp.nextbuff1_BaseAddr =
+ (u16) (abe_base_address_pingpong[1]);
+ if (dmareq_addr == ABE_DMASTATUS_RAW) {
+ desc_pp.nextbuff0_Samples =
+ (u16) ((abe_size_pingpong >> 2) / datasize);
+ desc_pp.nextbuff1_Samples =
+ (u16) ((abe_size_pingpong >> 2) / datasize);
+ } else {
+ desc_pp.nextbuff0_Samples = 0;
+ desc_pp.nextbuff1_Samples = 0;
+ }
+ /* next buffer to send is B1, first IRQ fills B0 */
+ desc_pp.counter = 0;
+ /* send a DMA req to fill B0 with N samples
+ abe_block_copy (COPY_FROM_HOST_TO_ABE,
+ ABE_ATC,
+ ABE_DMASTATUS_RAW,
+ &(abe_port[id].protocol.p.prot_pingpong.irq_data),
+ 4); */
+ sio_desc_address = OMAP_ABE_D_PINGPONGDESC_ADDR;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ sio_desc_address, (u32 *) &desc_pp,
+ sizeof(desc_pp));
+ } else {
+ io_sub_id = dmareq_addr = ABE_DMASTATUS_RAW;
+ dmareq_field = 0;
+ atc_desc_address1 = atc_desc_address2 = 0;
+ /* default: repeat of the last downlink samples in case of
+ DMA errors, (disable=0x00) */
+ io_flag = 0xFF;
+ datasize2 = datasize = abe_dma_port_iter_factor(format);
+ x_io = (u8) abe_dma_port_iteration(format);
+ nsamp = (x_io / datasize);
+ atc_ptr_saved2 = atc_ptr_saved = DMIC_ATC_PTR_labelID + id;
+ smem1 = abe_port[id].smem_buffer1;
+ smem3 = smem2 = abe_port[id].smem_buffer2;
+ copy_func_index1 = (u8) abe_dma_port_copy_subroutine_id(id);
+ before_func_index = after_func_index =
+ copy_func_index2 = NULL_COPY_CFPID;
+ switch (prot->protocol_switch) {
+ case DMIC_PORT_PROT:
+ /* DMIC port is read in two steps */
+ x_io = x_io >> 1;
+ nsamp = nsamp >> 1;
+ atc_desc_address1 = (ABE_ATC_DMIC_DMA_REQ*ATC_SIZE);
+ io_sub_id = IO_IP_CFPID;
+ break;
+ case MCPDMDL_PORT_PROT:
+ /* PDMDL port is written to in two steps */
+ x_io = x_io >> 1;
+ atc_desc_address1 =
+ (ABE_ATC_MCPDMDL_DMA_REQ*ATC_SIZE);
+ io_sub_id = IO_IP_CFPID;
+ break;
+ case MCPDMUL_PORT_PROT:
+ atc_desc_address1 =
+ (ABE_ATC_MCPDMUL_DMA_REQ*ATC_SIZE);
+ io_sub_id = IO_IP_CFPID;
+ break;
+ case SLIMBUS_PORT_PROT:
+ atc_desc_address1 =
+ abe_port[id].protocol.p.prot_slimbus.desc_addr1;
+ atc_desc_address2 =
+ abe_port[id].protocol.p.prot_slimbus.desc_addr2;
+ copy_func_index2 = NULL_COPY_CFPID;
+ /* @@@@@@
+ #define SPLIT_SMEM_CFPID 9
+ #define MERGE_SMEM_CFPID 10
+ #define SPLIT_TDM_12_CFPID 11
+ #define MERGE_TDM_12_CFPID 12
+ */
+ io_sub_id = IO_IP_CFPID;
+ break;
+ case SERIAL_PORT_PROT: /* McBSP/McASP */
+ atc_desc_address1 =
+ (s16) abe_port[id].protocol.p.prot_serial.
+ desc_addr;
+ io_sub_id = IO_IP_CFPID;
+ break;
+ case DMAREQ_PORT_PROT: /* DMA w/wo CBPr */
+ dmareq_addr =
+ abe_port[id].protocol.p.prot_dmareq.dma_addr;
+ dmareq_field = 0;
+ atc_desc_address1 =
+ abe_port[id].protocol.p.prot_dmareq.desc_addr;
+ io_sub_id = IO_IP_CFPID;
+ break;
+ }
+ /* special situation of the PING_PONG protocol which
+ has its own SIO descriptor format */
+ /*
+ Sequence of operations on ping-pong buffers B0/B1
+ -------------- time ---------------------------->>>>
+ Host Application is ready to send data from DDR to B0
+ SDMA is initialized from "abe_connect_irq_ping_pong_port" to B0
+ FIRMWARE starts with #12 B1 data,
+ sends IRQ/DMAreq, sends #pong B1 data,
+ sends IRQ/DMAreq, sends #ping B0,
+ sends B1 samples
+ ARM / SDMA | fills B0 | fills B1 ... | fills B0 ...
+ Counter 0 1 2 3
+ */
+ switch (id) {
+ case OMAP_ABE_PDM_DL_PORT:
+ abe->MultiFrame[7][0] = ABE_TASK_ID(C_ABE_FW_TASK_IO_PDM_DL);
+ abe->MultiFrame[19][0] = ABE_TASK_ID(C_ABE_FW_TASK_IO_PDM_DL);
+ break;
+ case OMAP_ABE_TONES_DL_PORT:
+ abe->MultiFrame[20][0] = ABE_TASK_ID(C_ABE_FW_TASK_IO_TONES_DL);
+ break;
+ case OMAP_ABE_PDM_UL_PORT:
+ abe->MultiFrame[5][2] = ABE_TASK_ID(C_ABE_FW_TASK_IO_PDM_UL);
+ break;
+ case OMAP_ABE_DMIC_PORT:
+ abe->MultiFrame[2][5] = ABE_TASK_ID(C_ABE_FW_TASK_IO_DMIC);
+ abe->MultiFrame[14][3] = ABE_TASK_ID(C_ABE_FW_TASK_IO_DMIC);
+ break;
+ case OMAP_ABE_MM_UL_PORT:
+ copy_func_index1 = COPY_MM_UL_CFPID;
+ before_func_index = ROUTE_MM_UL_CFPID;
+ abe->MultiFrame[19][6] = ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_UL);
+ break;
+ case OMAP_ABE_MM_UL2_PORT:
+ abe->MultiFrame[17][3] = ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_UL2);
+ break;
+ case OMAP_ABE_VX_DL_PORT:
+ /* check for 8kHz/16kHz */
+ if (abe_port[id].format.f == 8000) {
+ abe->MultiFrame[TASK_VX_DL_SLT][TASK_VX_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_VX_DL_8_48_FIR);
+ /*Voice_8k_DL_labelID */
+ smem1 = IO_VX_DL_ASRC_labelID;
+
+ if ((abe_port[OMAP_ABE_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is VX_DL_PORT
+ * both VX_UL ASRC and VX_DL ASRC will add/remove sample
+ * referring to VX_DL flow_counter */
+ abe->MultiFrame[TASK_ASRC_VX_DL_SLT][TASK_ASRC_VX_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_DL_8);
+ abe->MultiFrame[TASK_ASRC_VX_UL_SLT][TASK_ASRC_VX_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_UL_8_SIB);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ } else {
+ abe->MultiFrame[TASK_VX_DL_SLT][TASK_VX_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_VX_DL_16_48);
+ /* Voice_16k_DL_labelID */
+ smem1 = IO_VX_DL_ASRC_labelID;
+
+ if ((abe_port[OMAP_ABE_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is VX_DL_PORT
+ * both VX_UL ASRC and VX_DL ASRC will add/remove sample
+ * referring to VX_DL flow_counter */
+ abe->MultiFrame[TASK_ASRC_VX_DL_SLT][TASK_ASRC_VX_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_DL_16);
+ abe->MultiFrame[TASK_ASRC_VX_UL_SLT][TASK_ASRC_VX_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_UL_16_SIB);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ }
+ abe->MultiFrame[0][2] = ABE_TASK_ID(C_ABE_FW_TASK_IO_VX_DL);
+ break;
+ case OMAP_ABE_VX_UL_PORT:
+ /* check for 8kHz/16kHz */
+ if (abe_port[id].format.f == 8000) {
+ abe->MultiFrame[TASK_VX_UL_SLT][TASK_VX_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_VX_UL_48_8);
+ /* MultiFrame[TASK_ECHO_SLT][TASK_ECHO_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ECHO_REF_48_8); */
+ smem1 = Voice_8k_UL_labelID;
+
+ if ((abe_port[OMAP_ABE_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is VX_UL_PORT
+ * both VX_UL ASRC and VX_DL ASRC will add/remove sample
+ * referring to VX_UL flow_counter */
+ abe->MultiFrame[TASK_ASRC_VX_DL_SLT][TASK_ASRC_VX_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_DL_8_SIB);
+ abe->MultiFrame[TASK_ASRC_VX_UL_SLT][TASK_ASRC_VX_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_UL_8);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ } else {
+ abe->MultiFrame[TASK_VX_UL_SLT][TASK_VX_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_VX_UL_48_16);
+ /* MultiFrame[TASK_ECHO_SLT][TASK_ECHO_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ECHO_REF_48_16); */
+ smem1 = Voice_16k_UL_labelID;
+
+ if ((abe_port[OMAP_ABE_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is VX_UL_PORT
+ * both VX_UL ASRC and VX_DL ASRC will add/remove sample
+ * referring to VX_UL flow_counter */
+ abe->MultiFrame[TASK_ASRC_VX_DL_SLT][TASK_ASRC_VX_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_DL_16_SIB);
+ abe->MultiFrame[TASK_ASRC_VX_UL_SLT][TASK_ASRC_VX_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_VX_UL_16);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ }
+ abe->MultiFrame[16][3] = ABE_TASK_ID(C_ABE_FW_TASK_IO_VX_UL);
+ break;
+ case OMAP_ABE_BT_VX_DL_PORT:
+ /* check for 8kHz/16kHz */
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MAXTASKBYTESINSLOT_ADDR, &dOppMode32,
+ sizeof(u32));
+
+ if (abe_port[id].format.f == 8000) {
+ if (dOppMode32 == DOPPMODE32_OPP100) {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_8_OPP100);
+ smem1 = BT_DL_8k_opp100_labelID;
+ } else {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_8);
+ smem1 = BT_DL_8k_labelID;
+ }
+ if ((abe_port[OMAP_ABE_BT_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_BT_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is BT_VX_DL_PORT
+ * both BT_VX_DL ASRC and BT_VX_UL ASRC will add/remove sample
+ * referring to BT_VX_DL flow_counter */
+ abe->MultiFrame[TASK_ASRC_BT_DL_SLT][TASK_ASRC_BT_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_DL_8);
+ abe->MultiFrame[TASK_ASRC_BT_UL_SLT][TASK_ASRC_BT_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_UL_8_SIB);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ } else {
+ if (dOppMode32 == DOPPMODE32_OPP100) {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_16_OPP100);
+ smem1 = BT_DL_16k_opp100_labelID;
+ } else {
+ abe->MultiFrame[TASK_BT_DL_48_8_SLT][TASK_BT_DL_48_8_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_DL_48_16);
+ smem1 = BT_DL_16k_labelID;
+ }
+ if ((abe_port[OMAP_ABE_BT_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_BT_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is BT_VX_DL_PORT
+ * both BT_VX_DL ASRC and BT_VX_UL ASRC will add/remove sample
+ * referring to BT_VX_DL flow_counter */
+ abe->MultiFrame[TASK_ASRC_BT_DL_SLT][TASK_ASRC_BT_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_DL_16);
+ abe->MultiFrame[TASK_ASRC_BT_UL_SLT][TASK_ASRC_BT_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_UL_16_SIB);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ }
+ abe->MultiFrame[13][5] = ABE_TASK_ID(C_ABE_FW_TASK_IO_BT_VX_DL);
+ break;
+ case OMAP_ABE_BT_VX_UL_PORT:
+ /* check for 8kHz/16kHz */
+ /* set the SMEM buffer -- programming sequence */
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MAXTASKBYTESINSLOT_ADDR, &dOppMode32,
+ sizeof(u32));
+
+ if (abe_port[id].format.f == 8000) {
+ abe->MultiFrame[TASK_BT_UL_8_48_SLT][TASK_BT_UL_8_48_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_UL_8_48);
+ if (dOppMode32 == DOPPMODE32_OPP100)
+ /* ASRC input buffer, size 40 */
+ smem1 = smem_bt_vx_ul_opp100;
+ else
+ /* at OPP 50 without ASRC */
+ smem1 = BT_UL_8k_labelID;
+ if ((abe_port[OMAP_ABE_BT_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_BT_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is BT_VX_UL_PORT */
+ /* both BT_VX_UL ASRC and BT_VX_DL ASRC will add/remove sample
+ referring to BT_VX_UL flow_counter */
+ abe->MultiFrame[TASK_ASRC_BT_UL_SLT][TASK_ASRC_BT_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_UL_8);
+ abe->MultiFrame[TASK_ASRC_BT_DL_SLT][TASK_ASRC_BT_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_DL_8_SIB);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ } else {
+ abe->MultiFrame[TASK_BT_UL_8_48_SLT][TASK_BT_UL_8_48_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_BT_UL_16_48);
+ if (dOppMode32 == DOPPMODE32_OPP100)
+ /* ASRC input buffer, size 40 */
+ smem1 = smem_bt_vx_ul_opp100;
+ else
+ /* at OPP 50 without ASRC */
+ smem1 = BT_UL_16k_labelID;
+ if ((abe_port[OMAP_ABE_BT_VX_UL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE) &&
+ (abe_port[OMAP_ABE_BT_VX_DL_PORT].status ==
+ OMAP_ABE_PORT_ACTIVITY_IDLE)) {
+ /* the 1st opened port is BT_VX_UL_PORT */
+ /* both BT_VX_UL ASRC and BT_VX_DL ASRC will add/remove sample
+ referring to BT_VX_UL flow_counter */
+ abe->MultiFrame[TASK_ASRC_BT_UL_SLT][TASK_ASRC_BT_UL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_UL_16);
+ abe->MultiFrame[TASK_ASRC_BT_DL_SLT][TASK_ASRC_BT_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ASRC_BT_DL_16_SIB);
+ } else {
+ /* Do nothing, Scheduling Table has already been patched */
+ }
+ }
+ abe->MultiFrame[15][3] = ABE_TASK_ID(C_ABE_FW_TASK_IO_BT_VX_UL);
+ break;
+ case OMAP_ABE_MM_DL_PORT:
+ /* check for CBPr / serial_port / Ping-pong access */
+ abe->MultiFrame[TASK_IO_MM_DL_SLT][TASK_IO_MM_DL_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_DL);
+ smem1 = smem_mm_dl;
+ break;
+ case OMAP_ABE_MM_EXT_IN_PORT:
+ /* set the SMEM buffer -- programming sequence */
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MAXTASKBYTESINSLOT_ADDR, &dOppMode32,
+ sizeof(u32));
+ if (dOppMode32 == DOPPMODE32_OPP100)
+ /* ASRC input buffer, size 40 */
+ smem1 = smem_mm_ext_in_opp100;
+ else
+ /* at OPP 50 without ASRC */
+ smem1 = smem_mm_ext_in_opp50;
+
+ abe->MultiFrame[21][3] = ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_EXT_IN);
+ break;
+ case OMAP_ABE_MM_EXT_OUT_PORT:
+ abe->MultiFrame[15][0] = ABE_TASK_ID(C_ABE_FW_TASK_IO_MM_EXT_OUT);
+ break;
+ default:
+ break;
+ }
+
+ if (abe_port[id].protocol.direction == ABE_ATC_DIRECTION_IN)
+ direction = 0;
+ else
+ /* offset of the write pointer in the ATC descriptor */
+ direction = 3;
+
+ sio_desc.drift_ASRC = 0;
+ sio_desc.drift_io = 0;
+ sio_desc.io_type_idx = (u8) io_sub_id;
+ sio_desc.samp_size = (u8) datasize;
+ sio_desc.hw_ctrl_addr = (u16) (dmareq_addr << 2);
+ sio_desc.atc_irq_data = (u8) dmareq_field;
+ sio_desc.flow_counter = (u16) 0;
+ sio_desc.direction_rw = (u8) direction;
+ sio_desc.repeat_last_samp = (u8) io_flag;
+ sio_desc.nsamp = (u8) nsamp;
+ sio_desc.x_io = (u8) x_io;
+ /* set ATC ON */
+ sio_desc.on_off = 0x80;
+ sio_desc.split_addr1 = (u16) smem1;
+ sio_desc.split_addr2 = (u16) smem2;
+ sio_desc.split_addr3 = (u16) smem3;
+ sio_desc.before_f_index = (u8) before_func_index;
+ sio_desc.after_f_index = (u8) after_func_index;
+ sio_desc.smem_addr1 = (u16) smem1;
+ sio_desc.atc_address1 = (u16) atc_desc_address1;
+ sio_desc.atc_pointer_saved1 = (u16) atc_ptr_saved;
+ sio_desc.data_size1 = (u8) datasize;
+ sio_desc.copy_f_index1 = (u8) copy_func_index1;
+ sio_desc.smem_addr2 = (u16) smem2;
+ sio_desc.atc_address2 = (u16) atc_desc_address2;
+ sio_desc.atc_pointer_saved2 = (u16) atc_ptr_saved2;
+ sio_desc.data_size2 = (u8) datasize2;
+ sio_desc.copy_f_index2 = (u8) copy_func_index2;
+ sio_desc_address = OMAP_ABE_D_IODESCR_ADDR + (id *
+ sizeof(struct ABE_SIODescriptor));
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ sio_desc_address, (u32 *) &sio_desc,
+ sizeof(sio_desc));
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM,
+ OMAP_ABE_D_MULTIFRAME_ADDR, (u32 *) abe->MultiFrame,
+ sizeof(abe->MultiFrame));
+ }
+
+}
+
+/**
+ * omap_abe_select_main_port - Select stynchronization port for Event generator.
+ * @id: audio port name
+ *
+ * tells the FW which is the reference stream for adjusting
+ * the processing on 23/24/25 slots
+ */
+int omap_abe_select_main_port(u32 id)
+{
+ u32 selection;
+
+ _log(ABE_ID_SELECT_MAIN_PORT, id, 0, 0);
+
+ /* flow control */
+ selection = OMAP_ABE_D_IODESCR_ADDR + id * sizeof(struct ABE_SIODescriptor) +
+ flow_counter_;
+ /* when the main port is a sink port from AESS point of view
+ the sign the firmware task analysis must be changed */
+ selection &= 0xFFFFL;
+ if (abe_port[id].protocol.direction == ABE_ATC_DIRECTION_IN)
+ selection |= 0x80000;
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_SLOT23_CTRL_ADDR,
+ &selection, 4);
+ return 0;
+}
+/**
+ * abe_decide_main_port - Select stynchronization port for Event generator.
+ * @id: audio port name
+ *
+ * tells the FW which is the reference stream for adjusting
+ * the processing on 23/24/25 slots
+ *
+ * takes the first port in a list which is slave on the data interface
+ */
+u32 abe_valid_port_for_synchro(u32 id)
+{
+ if ((abe_port[id].protocol.protocol_switch ==
+ DMAREQ_PORT_PROT) ||
+ (abe_port[id].protocol.protocol_switch ==
+ PINGPONG_PORT_PROT) ||
+ (abe_port[id].status != OMAP_ABE_PORT_ACTIVITY_RUNNING))
+ return 0;
+ else
+ return 1;
+}
+void abe_decide_main_port(void)
+{
+ u32 id, id_not_found;
+ id_not_found = 1;
+ for (id = 0; id < LAST_PORT_ID - 1; id++) {
+ if (abe_valid_port_for_synchro(abe_port_priority[id])) {
+ id_not_found = 0;
+ break;
+ }
+ }
+ /* if no port is currently activated, the default one is PDM_DL */
+ if (id_not_found)
+ omap_abe_select_main_port(OMAP_ABE_PDM_DL_PORT);
+ else
+ omap_abe_select_main_port(abe_port_priority[id]);
+}
+/**
+ * abe_format_switch
+ * @f: port format
+ * @iter: port iteration
+ * @mulfac: multiplication factor
+ *
+ * translates the sampling and data length to ITER number for the DMA
+ * and the multiplier factor to apply during data move with DMEM
+ *
+ */
+void abe_format_switch(abe_data_format_t *f, u32 *iter, u32 *mulfac)
+{
+ u32 n_freq;
+#if FW_SCHED_LOOP_FREQ == 4000
+ switch (f->f) {
+ /* nb of samples processed by scheduling loop */
+ case 8000:
+ n_freq = 2;
+ break;
+ case 16000:
+ n_freq = 4;
+ break;
+ case 24000:
+ n_freq = 6;
+ break;
+ case 44100:
+ n_freq = 12;
+ break;
+ case 96000:
+ n_freq = 24;
+ break;
+ default/*case 48000 */ :
+ n_freq = 12;
+ break;
+ }
+#else
+ /* erroneous cases */
+ n_freq = 0;
+#endif
+ switch (f->samp_format) {
+ case MONO_MSB:
+ case MONO_RSHIFTED_16:
+ case STEREO_16_16:
+ *mulfac = 1;
+ break;
+ case STEREO_MSB:
+ case STEREO_RSHIFTED_16:
+ *mulfac = 2;
+ break;
+ case THREE_MSB:
+ *mulfac = 3;
+ break;
+ case FOUR_MSB:
+ *mulfac = 4;
+ break;
+ case FIVE_MSB:
+ *mulfac = 5;
+ break;
+ case SIX_MSB:
+ *mulfac = 6;
+ break;
+ case SEVEN_MSB:
+ *mulfac = 7;
+ break;
+ case EIGHT_MSB:
+ *mulfac = 8;
+ break;
+ case NINE_MSB:
+ *mulfac = 9;
+ break;
+ default:
+ *mulfac = 1;
+ break;
+ }
+ *iter = (n_freq * (*mulfac));
+}
+/**
+ * abe_dma_port_iteration
+ * @f: port format
+ *
+ * translates the sampling and data length to ITER number for the DMA
+ */
+u32 abe_dma_port_iteration(abe_data_format_t *f)
+{
+ u32 iter, mulfac;
+ abe_format_switch(f, &iter, &mulfac);
+ return iter;
+}
+/**
+ * abe_dma_port_iter_factor
+ * @f: port format
+ *
+ * returns the multiplier factor to apply during data move with DMEM
+ */
+u32 abe_dma_port_iter_factor(abe_data_format_t *f)
+{
+ u32 iter, mulfac;
+ abe_format_switch(f, &iter, &mulfac);
+ return mulfac;
+}
+/**
+ * omap_abe_dma_port_iter_factor
+ * @f: port format
+ *
+ * returns the multiplier factor to apply during data move with DMEM
+ */
+u32 omap_abe_dma_port_iter_factor(struct omap_abe_data_format *f)
+{
+ u32 iter, mulfac;
+ abe_format_switch((abe_data_format_t *)f, &iter, &mulfac);
+ return mulfac;
+}
+/**
+ * abe_dma_port_copy_subroutine_id
+ *
+ * @port_id: ABE port ID
+ *
+ * returns the index of the function doing the copy in I/O tasks
+ */
+u32 abe_dma_port_copy_subroutine_id(u32 port_id)
+{
+ u32 sub_id;
+ if (abe_port[port_id].protocol.direction == ABE_ATC_DIRECTION_IN) {
+ switch (abe_port[port_id].format.samp_format) {
+ case MONO_MSB:
+ sub_id = D2S_MONO_MSB_CFPID;
+ break;
+ case MONO_RSHIFTED_16:
+ sub_id = D2S_MONO_RSHIFTED_16_CFPID;
+ break;
+ case STEREO_RSHIFTED_16:
+ sub_id = D2S_STEREO_RSHIFTED_16_CFPID;
+ break;
+ case STEREO_16_16:
+ sub_id = D2S_STEREO_16_16_CFPID;
+ break;
+ case STEREO_MSB:
+ sub_id = D2S_STEREO_MSB_CFPID;
+ break;
+ case SIX_MSB:
+ if (port_id == OMAP_ABE_DMIC_PORT) {
+ sub_id = COPY_DMIC_CFPID;
+ break;
+ }
+ default:
+ sub_id = NULL_COPY_CFPID;
+ break;
+ }
+ } else {
+ switch (abe_port[port_id].format.samp_format) {
+ case MONO_MSB:
+ sub_id = S2D_MONO_MSB_CFPID;
+ break;
+ case MONO_RSHIFTED_16:
+ sub_id = S2D_MONO_RSHIFTED_16_CFPID;
+ break;
+ case STEREO_RSHIFTED_16:
+ sub_id = S2D_STEREO_RSHIFTED_16_CFPID;
+ break;
+ case STEREO_16_16:
+ sub_id = S2D_STEREO_16_16_CFPID;
+ break;
+ case STEREO_MSB:
+ sub_id = S2D_STEREO_MSB_CFPID;
+ break;
+ case SIX_MSB:
+ if (port_id == OMAP_ABE_PDM_DL_PORT) {
+ sub_id = COPY_MCPDM_DL_CFPID;
+ break;
+ }
+ if (port_id == OMAP_ABE_MM_UL_PORT) {
+ sub_id = COPY_MM_UL_CFPID;
+ break;
+ }
+ case THREE_MSB:
+ case FOUR_MSB:
+ case FIVE_MSB:
+ case SEVEN_MSB:
+ case EIGHT_MSB:
+ case NINE_MSB:
+ sub_id = COPY_MM_UL_CFPID;
+ break;
+ default:
+ sub_id = NULL_COPY_CFPID;
+ break;
+ }
+ }
+ return sub_id;
+}
+
+/**
+ * abe_read_remaining_data
+ * @id: ABE port_ID
+ * @n: size pointer to the remaining number of 32bits words
+ *
+ * computes the remaining amount of data in the buffer.
+ */
+abehal_status abe_read_remaining_data(u32 port, u32 *n)
+{
+ u32 sio_pp_desc_address;
+ struct ABE_SPingPongDescriptor desc_pp;
+
+ _log(ABE_ID_READ_REMAINING_DATA, port, 0, 0);
+
+ /*
+ * read the port SIO descriptor and extract the
+ * current pointer address after reading the counter
+ */
+ sio_pp_desc_address = OMAP_ABE_D_PINGPONGDESC_ADDR;
+ omap_abe_mem_read(abe, OMAP_ABE_DMEM, sio_pp_desc_address,
+ (u32 *) &desc_pp, sizeof(struct ABE_SPingPongDescriptor));
+ *n = desc_pp.workbuff_Samples;
+
+ return 0;
+}
+EXPORT_SYMBOL(abe_read_remaining_data);
+
+/**
+ * abe_mono_mixer
+ * @id: name of the mixer (MIXDL1 or MIXDL2)
+ * on_off: enable\disable flag
+ *
+ * This API Programs DL1Mixer or DL2Mixer to output mono data
+ * on both left and right data paths.
+ */
+abehal_status abe_mono_mixer(u32 id, u32 on_off)
+{
+
+ switch (id) {
+ case MIXDL1:
+ if (on_off)
+ abe->MultiFrame[TASK_DL1Mixer_SLT][TASK_DL1Mixer_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_DL1Mixer_dual_mono);
+ else
+ abe->MultiFrame[TASK_DL1Mixer_SLT][TASK_DL1Mixer_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_DL1Mixer);
+ break;
+ case MIXDL2:
+ if (on_off)
+ abe->MultiFrame[TASK_DL2Mixer_SLT][TASK_DL2Mixer_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_DL2Mixer_dual_mono);
+ else
+ abe->MultiFrame[TASK_DL2Mixer_SLT][TASK_DL2Mixer_IDX] =
+ ABE_TASK_ID(C_ABE_FW_TASK_DL2Mixer);
+ break;
+ case MIXAUDUL:
+ if (on_off)
+ abe->MultiFrame[12][4] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ULMixer_dual_mono);
+ else
+ abe->MultiFrame[12][4] =
+ ABE_TASK_ID(C_ABE_FW_TASK_ULMixer);
+ break;
+ default:
+ break;
+ }
+
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_MULTIFRAME_ADDR,
+ (u32 *) abe->MultiFrame, sizeof(abe->MultiFrame));
+
+ return 0;
+}
+EXPORT_SYMBOL(abe_mono_mixer);
+/**
+ * abe_write_pdmdl_offset - write the desired offset on the DL1/DL2 paths
+ *
+ * Parameters:
+ * path: 1 for the DL1 ABE path, 2 for the DL2 ABE path
+ * offset_left: integer value that will be added on all PDM left samples
+ * offset_right: integer value that will be added on all PDM right samples
+ *
+ */
+void abe_write_pdmdl_offset(u32 path, u32 offset_left, u32 offset_right)
+{
+ switch (path) {
+ case 1:
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, OMAP_ABE_S_DC_HS_ADDR + 4,
+ &offset_left, sizeof(u32));
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, OMAP_ABE_S_DC_HS_ADDR,
+ &offset_right, sizeof(u32));
+ break;
+ case 2:
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, OMAP_ABE_S_DC_HF_ADDR + 4,
+ &offset_left, sizeof(u32));
+ omap_abe_mem_write(abe, OMAP_ABE_SMEM, OMAP_ABE_S_DC_HF_ADDR,
+ &offset_right, sizeof(u32));
+ break;
+ default:
+ break;
+ }
+}
+EXPORT_SYMBOL(abe_write_pdmdl_offset);
+
diff --git a/sound/soc/omap/abe/abe_port.h b/sound/soc/omap/abe/abe_port.h
new file mode 100644
index 0000000..290f8b5
--- /dev/null
+++ b/sound/soc/omap/abe/abe_port.h
@@ -0,0 +1,161 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_PORT_H_
+#define _ABE_PORT_H_
+
+struct omap_abe_data_format {
+ /* Sampling frequency of the stream */
+ u32 f;
+ /* Sample format type */
+ u32 samp_format;
+};
+
+struct omap_abe_port_protocol {
+ /* Direction=0 means input from AESS point of view */
+ u32 direction;
+ /* Protocol type (switch) during the data transfers */
+ u32 protocol_switch;
+ union {
+ /* McBSP/McASP peripheral connected to ATC */
+ struct {
+ u32 desc_addr;
+ /* Address of ATC McBSP/McASP descriptor's in bytes */
+ u32 buf_addr;
+ /* DMEM address in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ } serial;
+ /* DMIC peripheral connected to ATC */
+ struct {
+ /* DMEM address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size in bytes */
+ u32 buf_size;
+ /* Number of activated DMIC */
+ u32 nbchan;
+ } dmic;
+ /* McPDMDL peripheral connected to ATC */
+ struct {
+ /* DMEM address in bytes */
+ u32 buf_addr;
+ /* DMEM size in bytes */
+ u32 buf_size;
+ /* Control allowed on McPDM DL */
+ u32 control;
+ } mcpdmdl;
+ /* McPDMUL peripheral connected to ATC */
+ struct {
+ /* DMEM address size in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ } mcpdmul;
+ /* Ping-Pong interface to the Host using cache-flush */
+ struct {
+ /* Address of ATC descriptor's */
+ u32 desc_addr;
+ /* DMEM buffer base address in bytes */
+ u32 buf_addr;
+ /* DMEM size in bytes for each ping and pong buffers */
+ u32 buf_size;
+ /* IRQ address (either DMA (0) MCU (1) or DSP(2)) */
+ u32 irq_addr;
+ /* IRQ data content loaded in the AESS IRQ register */
+ u32 irq_data;
+ /* Call-back function upon IRQ reception */
+ u32 callback;
+ } pingpong;
+ /* DMAreq line to CBPr */
+ struct {
+ /* Address of ATC descriptor's */
+ u32 desc_addr;
+ /* DMEM buffer address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ /* DMAreq address */
+ u32 dma_addr;
+ /* DMA/AESS = 1 << #DMA */
+ u32 dma_data;
+ } dmareq;
+ /* Circular buffer - direct addressing to DMEM */
+ struct {
+ /* DMEM buffer base address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size in bytes */
+ u32 buf_size;
+ /* DMAreq address */
+ u32 dma_addr;
+ /* DMA/AESS = 1 << #DMA */
+ u32 dma_data;
+ } circular_buffer;
+ } port;
+};
+
+extern const abe_port_t abe_port_init[];
+extern abe_port_t abe_port[];
+extern const u32 abe_port_priority[];
+
+int omap_abe_select_main_port(u32 id);
+u32 omap_abe_dma_port_iter_factor(struct omap_abe_data_format *f);
+
+#endif/* _ABE_PORT_H_ */
diff --git a/sound/soc/omap/abe/abe_ref.h b/sound/soc/omap/abe/abe_ref.h
new file mode 100644
index 0000000..4c8a9bb
--- /dev/null
+++ b/sound/soc/omap/abe/abe_ref.h
@@ -0,0 +1,152 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_REF_H_
+#define _ABE_REF_H_
+
+#include "abe_api.h"
+
+/*
+ * 'ABE_PRO.H' all non-API prototypes for INI, IRQ, SEQ ...
+ */
+/*
+ * HAL EXTERNAL AP
+ */
+/*
+ * HAL INTERNAL AP
+ */
+void abe_decide_main_port(void);
+void abe_reset_all_ports(void);
+void abe_reset_all_fifo(void);
+void abe_reset_all_sequence(void);
+u32 abe_dma_port_iteration(abe_data_format_t *format);
+void abe_read_sys_clock(u32 *time);
+void abe_enable_atc(u32 id);
+void abe_disable_atc(u32 id);
+void abe_init_io_tasks(u32 id, abe_data_format_t *format,
+ abe_port_protocol_t *prot);
+void abe_init_dma_t(u32 id, abe_port_protocol_t *prot);
+u32 abe_dma_port_iter_factor(abe_data_format_t *f);
+u32 abe_dma_port_copy_subroutine_id(u32 i);
+void abe_call_subroutine(u32 idx, u32 p1, u32 p2, u32 p3, u32 p4);
+void abe_monitoring(void);
+void abe_add_subroutine(u32 *id, abe_subroutine2 f, u32 nparam, u32 *params);
+abehal_status abe_read_next_ping_pong_buffer(u32 port, u32 *p, u32 *n);
+void abe_irq_ping_pong(void);
+void abe_irq_check_for_sequences(u32 seq_info);
+void abe_default_irq_pingpong_player(void);
+void abe_default_irq_pingpong_player_32bits(void);
+void abe_rshifted16_irq_pingpong_player_32bits(void);
+void abe_1616_irq_pingpong_player_1616bits(void);
+void abe_default_irq_aps_adaptation(void);
+void abe_irq_aps(u32 aps_info);
+void abe_dbg_error_log(u32 x);
+void abe_init_asrc_vx_dl(s32 dppm);
+void abe_init_asrc_vx_ul(s32 dppm);
+void abe_init_asrc_mm_ext_in(s32 dppm);
+void abe_init_asrc_bt_ul(s32 dppm);
+void abe_init_asrc_bt_dl(s32 dppm);
+
+void omap_abe_hw_configuration(struct omap_abe *abe);
+void omap_abe_gain_offset(struct omap_abe *abe, u32 id, u32 *mixer_offset);
+int omap_abe_use_compensated_gain(struct omap_abe *abe, int on_off);
+
+/*
+ * HAL INTERNAL DATA
+ */
+extern const u32 abe_port_priority[LAST_PORT_ID - 1];
+extern const u32 abe_firmware_array[ABE_FIRMWARE_MAX_SIZE];
+extern const u32 abe_atc_srcid[];
+extern const u32 abe_atc_dstid[];
+extern const abe_port_t abe_port_init[];
+extern const abe_seq_t all_sequence_init[];
+extern const abe_router_t abe_router_ul_table_preset
+ [NBROUTE_CONFIG][NBROUTE_UL];
+extern const abe_sequence_t seq_null;
+
+extern abe_port_t abe_port[];
+extern abe_seq_t all_sequence[];
+extern abe_router_t abe_router_ul_table[NBROUTE_CONFIG_MAX][NBROUTE_UL];
+/* table of new subroutines called in the sequence */
+extern abe_subroutine2 abe_all_subsubroutine[MAXNBSUBROUTINE];
+/* number of parameters per calls */
+extern u32 abe_all_subsubroutine_nparam[MAXNBSUBROUTINE];
+extern u32 abe_subroutine_id[MAXNBSUBROUTINE];
+extern u32 *abe_all_subroutine_params[MAXNBSUBROUTINE];
+extern u32 abe_subroutine_write_pointer;
+extern abe_sequence_t abe_all_sequence[MAXNBSEQUENCE];
+extern u32 abe_sequence_write_pointer;
+/* current number of pending sequences (avoids to look in the table) */
+extern u32 abe_nb_pending_sequences;
+/* pending sequences due to ressource collision */
+extern u32 abe_pending_sequences[MAXNBSEQUENCE];
+/* mask of unsharable ressources among other sequences */
+extern u32 abe_global_sequence_mask;
+/* table of active sequences */
+extern abe_seq_t abe_active_sequence[MAXACTIVESEQUENCE][MAXSEQUENCESTEPS];
+/* index of the plugged subroutine doing ping-pong cache-flush
+ DMEM accesses */
+extern u32 abe_irq_aps_adaptation_id;
+/* base addresses of the ping pong buffers */
+extern u32 abe_base_address_pingpong[MAX_PINGPONG_BUFFERS];
+/* size of each ping/pong buffers */
+extern u32 abe_size_pingpong;
+/* number of ping/pong buffer being used */
+extern u32 abe_nb_pingpong;
+
+#endif/* _ABE_REF_H_ */
diff --git a/sound/soc/omap/abe/abe_seq.c b/sound/soc/omap/abe/abe_seq.c
new file mode 100644
index 0000000..6ae2aa5
--- /dev/null
+++ b/sound/soc/omap/abe/abe_seq.c
@@ -0,0 +1,308 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include "abe_legacy.h"
+
+#include "abe_mem.h"
+
+extern struct omap_abe *abe;
+extern u32 abe_irq_pingpong_player_id;
+
+/**
+ * abe_null_subroutine
+ *
+ */
+void abe_null_subroutine_0(void)
+{
+}
+void abe_null_subroutine_2(u32 a, u32 b)
+{
+}
+void abe_null_subroutine_4(u32 a, u32 b, u32 c, u32 d)
+{
+}
+/**
+ * abe_init_subroutine_table - initializes the default table of pointers
+ * to subroutines
+ *
+ * initializes the default table of pointers to subroutines
+ *
+ */
+void abe_init_subroutine_table(void)
+{
+ u32 id;
+ /* reset the table's pointers */
+ abe_subroutine_write_pointer = 0;
+ /* the first index is the NULL task */
+ abe_add_subroutine(&id, (abe_subroutine2) abe_null_subroutine_2,
+ SUB_0_PARAM, (u32 *) 0);
+ /* write mixer has 4 parameters */
+ abe_add_subroutine(&(abe_subroutine_id[SUB_WRITE_MIXER]),
+ (abe_subroutine2) abe_write_mixer, SUB_4_PARAM,
+ (u32 *) 0);
+ /* ping-pong player IRQ */
+ abe_add_subroutine(&abe_irq_pingpong_player_id,
+ (abe_subroutine2) abe_null_subroutine_0, SUB_0_PARAM,
+ (u32 *) 0);
+}
+/**
+ * abe_add_subroutine
+ * @id: ABE port id
+ * @f: pointer to the subroutines
+ * @nparam: number of parameters
+ * @params: pointer to the psrameters
+ *
+ * add one function pointer more and returns the index to it
+ */
+void abe_add_subroutine(u32 *id, abe_subroutine2 f, u32 nparam, u32 *params)
+{
+ u32 i, i_found;
+ if ((abe_subroutine_write_pointer >= MAXNBSUBROUTINE) ||
+ ((u32) f == 0)) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_SEQ,
+ ABE_PARAMETER_OVERFLOW);
+ } else {
+ /* search if this subroutine address was not already
+ * declared, then return the previous index
+ */
+ for (i_found = abe_subroutine_write_pointer, i = 0;
+ i < abe_subroutine_write_pointer; i++) {
+ if (f == abe_all_subsubroutine[i])
+ i_found = i;
+ }
+ if (i_found == abe_subroutine_write_pointer) {
+ *id = abe_subroutine_write_pointer;
+ abe_all_subsubroutine
+ [abe_subroutine_write_pointer] = (f);
+ abe_all_subroutine_params
+ [abe_subroutine_write_pointer] = params;
+ abe_all_subsubroutine_nparam
+ [abe_subroutine_write_pointer] = nparam;
+ abe_subroutine_write_pointer++;
+ } else {
+ abe_all_subroutine_params[i_found] = params;
+ *id = i_found;
+ }
+ }
+}
+/**
+ * abe_add_sequence
+ * @id: returned sequence index after pluging a new sequence
+ * (index in the tables)
+ * @s: sequence to be inserted
+ *
+ * Load a time-sequenced operations.
+ */
+void abe_add_sequence(u32 *id, abe_sequence_t *s)
+{
+ abe_seq_t *seq_src, *seq_dst;
+ u32 i, no_end_of_sequence_found;
+ seq_src = &(s->seq1);
+ seq_dst = &((abe_all_sequence[abe_sequence_write_pointer]).seq1);
+ if ((abe_sequence_write_pointer >= MAXNBSEQUENCE) || ((u32) s == 0)) {
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_SEQ,
+ ABE_PARAMETER_OVERFLOW);
+ } else {
+ *id = abe_subroutine_write_pointer;
+ /* copy the mask */
+ (abe_all_sequence[abe_sequence_write_pointer]).mask = s->mask;
+ for (no_end_of_sequence_found = 1, i = 0; i < MAXSEQUENCESTEPS;
+ i++, seq_src++, seq_dst++) {
+ /* sequence copied line by line */
+ (*seq_dst) = (*seq_src);
+ /* stop when the line start with time=(-1) */
+ if ((*(s32 *) seq_src) == (-1)) {
+ /* stop when the line start with time=(-1) */
+ no_end_of_sequence_found = 0;
+ break;
+ }
+ }
+ abe_subroutine_write_pointer++;
+ if (no_end_of_sequence_found)
+ omap_abe_dbg_error(abe, OMAP_ABE_ERR_API,
+ ABE_SEQTOOLONG);
+ }
+}
+/**
+ * abe_reset_one_sequence
+ * @id: sequence ID
+ *
+ * load default configuration for that sequence
+ * kill running activities
+ */
+void abe_reset_one_sequence(u32 id)
+{
+}
+/**
+ * abe_reset_all_sequence
+ *
+ * load default configuration for all sequences
+ * kill any running activities
+ */
+void omap_abe_reset_all_sequence(struct omap_abe *abe)
+{
+ u32 i;
+ abe_init_subroutine_table();
+ /* arrange to have the first sequence index=0 to the NULL operation
+ sequence */
+ abe_add_sequence(&i, (abe_sequence_t *) &seq_null);
+ /* reset the the collision protection mask */
+ abe_global_sequence_mask = 0;
+ /* reset the pending sequences list */
+ for (abe_nb_pending_sequences = i = 0; i < MAXNBSEQUENCE; i++)
+ abe_pending_sequences[i] = 0;
+}
+/**
+ * abe_call_subroutine
+ * @idx: index to the table of all registered Call-backs and subroutines
+ *
+ * run and log a subroutine
+ */
+void abe_call_subroutine(u32 idx, u32 p1, u32 p2, u32 p3, u32 p4)
+{
+ abe_subroutine0 f0;
+ abe_subroutine1 f1;
+ abe_subroutine2 f2;
+ abe_subroutine3 f3;
+ abe_subroutine4 f4;
+ u32 *params;
+ if (idx > MAXNBSUBROUTINE)
+ return;
+ switch (idx) {
+ /* call the subroutines defined at compilation time
+ (const .. sequences) */
+#if 0
+ case SUB_WRITE_MIXER_DL1:
+ abe_write_mixer_dl1(p1, p2, p3)
+ abe_fprintf("write_mixer");
+ break;
+#endif
+ /* call the subroutines defined at execution time
+ (dynamic sequences) */
+ default:
+ switch (abe_all_subsubroutine_nparam[idx]) {
+ case SUB_0_PARAM:
+ f0 = (abe_subroutine0) abe_all_subsubroutine[idx];
+ (*f0) ();
+ break;
+ case SUB_1_PARAM:
+ f1 = (abe_subroutine1) abe_all_subsubroutine[idx];
+ params = abe_all_subroutine_params
+ [abe_irq_pingpong_player_id];
+ if (params != (u32 *) 0)
+ p1 = params[0];
+ (*f1) (p1);
+ break;
+ case SUB_2_PARAM:
+ f2 = abe_all_subsubroutine[idx];
+ params = abe_all_subroutine_params
+ [abe_irq_pingpong_player_id];
+ if (params != (u32 *) 0) {
+ p1 = params[0];
+ p2 = params[1];
+ }
+ (*f2) (p1, p2);
+ break;
+ case SUB_3_PARAM:
+ f3 = (abe_subroutine3) abe_all_subsubroutine[idx];
+ params = abe_all_subroutine_params
+ [abe_irq_pingpong_player_id];
+ if (params != (u32 *) 0) {
+ p1 = params[0];
+ p2 = params[1];
+ p3 = params[2];
+ }
+ (*f3) (p1, p2, p3);
+ break;
+ case SUB_4_PARAM:
+ f4 = (abe_subroutine4) abe_all_subsubroutine[idx];
+ params = abe_all_subroutine_params
+ [abe_irq_pingpong_player_id];
+ if (params != (u32 *) 0) {
+ p1 = params[0];
+ p2 = params[1];
+ p3 = params[2];
+ p4 = params[3];
+ }
+ (*f4) (p1, p2, p3, p4);
+ break;
+ default:
+ break;
+ }
+ }
+}
+
+/**
+ * abe_set_sequence_time_accuracy
+ * @fast: fast counter
+ * @slow: slow counter
+ *
+ */
+abehal_status abe_set_sequence_time_accuracy(u32 fast, u32 slow)
+{
+ u32 data;
+ _log(ABE_ID_SET_SEQUENCE_TIME_ACCURACY, fast, slow, 0);
+ data = minimum(MAX_UINT16, fast / FW_SCHED_LOOP_FREQ_DIV1000);
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_FASTCOUNTER_ADDR,
+ &data, sizeof(data));
+ data = minimum(MAX_UINT16, slow / FW_SCHED_LOOP_FREQ_DIV1000);
+ omap_abe_mem_write(abe, OMAP_ABE_DMEM, OMAP_ABE_D_SLOWCOUNTER_ADDR,
+ &data, sizeof(data));
+ return 0;
+}
+EXPORT_SYMBOL(abe_set_sequence_time_accuracy);
diff --git a/sound/soc/omap/abe/abe_seq.h b/sound/soc/omap/abe/abe_seq.h
new file mode 100644
index 0000000..e5047ad
--- /dev/null
+++ b/sound/soc/omap/abe/abe_seq.h
@@ -0,0 +1,64 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_SEQ_H_
+#define _ABE_SEQ_H_
+
+void omap_abe_reset_all_sequence(struct omap_abe *abe);
+
+#endif /* _ABE_SEQ_H_ */
diff --git a/sound/soc/omap/abe/abe_sm_addr.h b/sound/soc/omap/abe/abe_sm_addr.h
new file mode 100644
index 0000000..a9e28ac
--- /dev/null
+++ b/sound/soc/omap/abe/abe_sm_addr.h
@@ -0,0 +1,353 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#define OMAP_ABE_INIT_SM_ADDR 0x0
+#define OMAP_ABE_INIT_SM_SIZE 0xC80
+#define OMAP_ABE_S_DATA0_ADDR 0xC80
+#define OMAP_ABE_S_DATA0_SIZE 0x8
+#define OMAP_ABE_S_TEMP_ADDR 0xC88
+#define OMAP_ABE_S_TEMP_SIZE 0x8
+#define OMAP_ABE_S_PHOENIXOFFSET_ADDR 0xC90
+#define OMAP_ABE_S_PHOENIXOFFSET_SIZE 0x8
+#define OMAP_ABE_S_GTARGET1_ADDR 0xC98
+#define OMAP_ABE_S_GTARGET1_SIZE 0x38
+#define OMAP_ABE_S_GTARGET_DL1_ADDR 0xCD0
+#define OMAP_ABE_S_GTARGET_DL1_SIZE 0x10
+#define OMAP_ABE_S_GTARGET_DL2_ADDR 0xCE0
+#define OMAP_ABE_S_GTARGET_DL2_SIZE 0x10
+#define OMAP_ABE_S_GTARGET_ECHO_ADDR 0xCF0
+#define OMAP_ABE_S_GTARGET_ECHO_SIZE 0x8
+#define OMAP_ABE_S_GTARGET_SDT_ADDR 0xCF8
+#define OMAP_ABE_S_GTARGET_SDT_SIZE 0x8
+#define OMAP_ABE_S_GTARGET_VXREC_ADDR 0xD00
+#define OMAP_ABE_S_GTARGET_VXREC_SIZE 0x10
+#define OMAP_ABE_S_GTARGET_UL_ADDR 0xD10
+#define OMAP_ABE_S_GTARGET_UL_SIZE 0x10
+#define OMAP_ABE_S_GTARGET_BTUL_ADDR 0xD20
+#define OMAP_ABE_S_GTARGET_BTUL_SIZE 0x8
+#define OMAP_ABE_S_GCURRENT_ADDR 0xD28
+#define OMAP_ABE_S_GCURRENT_SIZE 0x90
+#define OMAP_ABE_S_GAIN_ONE_ADDR 0xDB8
+#define OMAP_ABE_S_GAIN_ONE_SIZE 0x8
+#define OMAP_ABE_S_TONES_ADDR 0xDC0
+#define OMAP_ABE_S_TONES_SIZE 0x60
+#define OMAP_ABE_S_VX_DL_ADDR 0xE20
+#define OMAP_ABE_S_VX_DL_SIZE 0x60
+#define OMAP_ABE_S_MM_UL2_ADDR 0xE80
+#define OMAP_ABE_S_MM_UL2_SIZE 0x60
+#define OMAP_ABE_S_MM_DL_ADDR 0xEE0
+#define OMAP_ABE_S_MM_DL_SIZE 0x60
+#define OMAP_ABE_S_DL1_M_OUT_ADDR 0xF40
+#define OMAP_ABE_S_DL1_M_OUT_SIZE 0x60
+#define OMAP_ABE_S_DL2_M_OUT_ADDR 0xFA0
+#define OMAP_ABE_S_DL2_M_OUT_SIZE 0x60
+#define OMAP_ABE_S_ECHO_M_OUT_ADDR 0x1000
+#define OMAP_ABE_S_ECHO_M_OUT_SIZE 0x60
+#define OMAP_ABE_S_SDT_M_OUT_ADDR 0x1060
+#define OMAP_ABE_S_SDT_M_OUT_SIZE 0x60
+#define OMAP_ABE_S_VX_UL_ADDR 0x10C0
+#define OMAP_ABE_S_VX_UL_SIZE 0x60
+#define OMAP_ABE_S_VX_UL_M_ADDR 0x1120
+#define OMAP_ABE_S_VX_UL_M_SIZE 0x60
+#define OMAP_ABE_S_BT_DL_ADDR 0x1180
+#define OMAP_ABE_S_BT_DL_SIZE 0x60
+#define OMAP_ABE_S_BT_UL_ADDR 0x11E0
+#define OMAP_ABE_S_BT_UL_SIZE 0x60
+#define OMAP_ABE_S_BT_DL_8K_ADDR 0x1240
+#define OMAP_ABE_S_BT_DL_8K_SIZE 0x18
+#define OMAP_ABE_S_BT_DL_16K_ADDR 0x1258
+#define OMAP_ABE_S_BT_DL_16K_SIZE 0x28
+#define OMAP_ABE_S_BT_UL_8K_ADDR 0x1280
+#define OMAP_ABE_S_BT_UL_8K_SIZE 0x10
+#define OMAP_ABE_S_BT_UL_16K_ADDR 0x1290
+#define OMAP_ABE_S_BT_UL_16K_SIZE 0x20
+#define OMAP_ABE_S_SDT_F_ADDR 0x12B0
+#define OMAP_ABE_S_SDT_F_SIZE 0x60
+#define OMAP_ABE_S_SDT_F_DATA_ADDR 0x1310
+#define OMAP_ABE_S_SDT_F_DATA_SIZE 0x48
+#define OMAP_ABE_S_MM_DL_OSR_ADDR 0x1358
+#define OMAP_ABE_S_MM_DL_OSR_SIZE 0xC0
+#define OMAP_ABE_S_24_ZEROS_ADDR 0x1418
+#define OMAP_ABE_S_24_ZEROS_SIZE 0xC0
+#define OMAP_ABE_S_DMIC1_ADDR 0x14D8
+#define OMAP_ABE_S_DMIC1_SIZE 0x60
+#define OMAP_ABE_S_DMIC2_ADDR 0x1538
+#define OMAP_ABE_S_DMIC2_SIZE 0x60
+#define OMAP_ABE_S_DMIC3_ADDR 0x1598
+#define OMAP_ABE_S_DMIC3_SIZE 0x60
+#define OMAP_ABE_S_AMIC_ADDR 0x15F8
+#define OMAP_ABE_S_AMIC_SIZE 0x60
+#define OMAP_ABE_S_DMIC1_L_ADDR 0x1658
+#define OMAP_ABE_S_DMIC1_L_SIZE 0x60
+#define OMAP_ABE_S_DMIC1_R_ADDR 0x16B8
+#define OMAP_ABE_S_DMIC1_R_SIZE 0x60
+#define OMAP_ABE_S_DMIC2_L_ADDR 0x1718
+#define OMAP_ABE_S_DMIC2_L_SIZE 0x60
+#define OMAP_ABE_S_DMIC2_R_ADDR 0x1778
+#define OMAP_ABE_S_DMIC2_R_SIZE 0x60
+#define OMAP_ABE_S_DMIC3_L_ADDR 0x17D8
+#define OMAP_ABE_S_DMIC3_L_SIZE 0x60
+#define OMAP_ABE_S_DMIC3_R_ADDR 0x1838
+#define OMAP_ABE_S_DMIC3_R_SIZE 0x60
+#define OMAP_ABE_S_BT_UL_L_ADDR 0x1898
+#define OMAP_ABE_S_BT_UL_L_SIZE 0x60
+#define OMAP_ABE_S_BT_UL_R_ADDR 0x18F8
+#define OMAP_ABE_S_BT_UL_R_SIZE 0x60
+#define OMAP_ABE_S_AMIC_L_ADDR 0x1958
+#define OMAP_ABE_S_AMIC_L_SIZE 0x60
+#define OMAP_ABE_S_AMIC_R_ADDR 0x19B8
+#define OMAP_ABE_S_AMIC_R_SIZE 0x60
+#define OMAP_ABE_S_ECHOREF_L_ADDR 0x1A18
+#define OMAP_ABE_S_ECHOREF_L_SIZE 0x60
+#define OMAP_ABE_S_ECHOREF_R_ADDR 0x1A78
+#define OMAP_ABE_S_ECHOREF_R_SIZE 0x60
+#define OMAP_ABE_S_MM_DL_L_ADDR 0x1AD8
+#define OMAP_ABE_S_MM_DL_L_SIZE 0x60
+#define OMAP_ABE_S_MM_DL_R_ADDR 0x1B38
+#define OMAP_ABE_S_MM_DL_R_SIZE 0x60
+#define OMAP_ABE_S_MM_UL_ADDR 0x1B98
+#define OMAP_ABE_S_MM_UL_SIZE 0x3C0
+#define OMAP_ABE_S_AMIC_96K_ADDR 0x1F58
+#define OMAP_ABE_S_AMIC_96K_SIZE 0xC0
+#define OMAP_ABE_S_DMIC0_96K_ADDR 0x2018
+#define OMAP_ABE_S_DMIC0_96K_SIZE 0xC0
+#define OMAP_ABE_S_DMIC1_96K_ADDR 0x20D8
+#define OMAP_ABE_S_DMIC1_96K_SIZE 0xC0
+#define OMAP_ABE_S_DMIC2_96K_ADDR 0x2198
+#define OMAP_ABE_S_DMIC2_96K_SIZE 0xC0
+#define OMAP_ABE_S_UL_VX_UL_48_8K_ADDR 0x2258
+#define OMAP_ABE_S_UL_VX_UL_48_8K_SIZE 0x60
+#define OMAP_ABE_S_UL_VX_UL_48_16K_ADDR 0x22B8
+#define OMAP_ABE_S_UL_VX_UL_48_16K_SIZE 0x60
+#define OMAP_ABE_S_UL_MIC_48K_ADDR 0x2318
+#define OMAP_ABE_S_UL_MIC_48K_SIZE 0x60
+#define OMAP_ABE_S_VOICE_8K_UL_ADDR 0x2378
+#define OMAP_ABE_S_VOICE_8K_UL_SIZE 0x18
+#define OMAP_ABE_S_VOICE_8K_DL_ADDR 0x2390
+#define OMAP_ABE_S_VOICE_8K_DL_SIZE 0x10
+#define OMAP_ABE_S_MCPDM_OUT1_ADDR 0x23A0
+#define OMAP_ABE_S_MCPDM_OUT1_SIZE 0xC0
+#define OMAP_ABE_S_MCPDM_OUT2_ADDR 0x2460
+#define OMAP_ABE_S_MCPDM_OUT2_SIZE 0xC0
+#define OMAP_ABE_S_MCPDM_OUT3_ADDR 0x2520
+#define OMAP_ABE_S_MCPDM_OUT3_SIZE 0xC0
+#define OMAP_ABE_S_VOICE_16K_UL_ADDR 0x25E0
+#define OMAP_ABE_S_VOICE_16K_UL_SIZE 0x28
+#define OMAP_ABE_S_VOICE_16K_DL_ADDR 0x2608
+#define OMAP_ABE_S_VOICE_16K_DL_SIZE 0x20
+#define OMAP_ABE_S_XINASRC_DL_VX_ADDR 0x2628
+#define OMAP_ABE_S_XINASRC_DL_VX_SIZE 0x140
+#define OMAP_ABE_S_XINASRC_UL_VX_ADDR 0x2768
+#define OMAP_ABE_S_XINASRC_UL_VX_SIZE 0x140
+#define OMAP_ABE_S_XINASRC_MM_EXT_IN_ADDR 0x28A8
+#define OMAP_ABE_S_XINASRC_MM_EXT_IN_SIZE 0x140
+#define OMAP_ABE_S_VX_REC_ADDR 0x29E8
+#define OMAP_ABE_S_VX_REC_SIZE 0x60
+#define OMAP_ABE_S_VX_REC_L_ADDR 0x2A48
+#define OMAP_ABE_S_VX_REC_L_SIZE 0x60
+#define OMAP_ABE_S_VX_REC_R_ADDR 0x2AA8
+#define OMAP_ABE_S_VX_REC_R_SIZE 0x60
+#define OMAP_ABE_S_DL2_M_L_ADDR 0x2B08
+#define OMAP_ABE_S_DL2_M_L_SIZE 0x60
+#define OMAP_ABE_S_DL2_M_R_ADDR 0x2B68
+#define OMAP_ABE_S_DL2_M_R_SIZE 0x60
+#define OMAP_ABE_S_DL2_M_LR_EQ_DATA_ADDR 0x2BC8
+#define OMAP_ABE_S_DL2_M_LR_EQ_DATA_SIZE 0xC8
+#define OMAP_ABE_S_DL1_M_EQ_DATA_ADDR 0x2C90
+#define OMAP_ABE_S_DL1_M_EQ_DATA_SIZE 0xC8
+#define OMAP_ABE_S_EARP_48_96_LP_DATA_ADDR 0x2D58
+#define OMAP_ABE_S_EARP_48_96_LP_DATA_SIZE 0x78
+#define OMAP_ABE_S_IHF_48_96_LP_DATA_ADDR 0x2DD0
+#define OMAP_ABE_S_IHF_48_96_LP_DATA_SIZE 0x78
+#define OMAP_ABE_S_VX_UL_8_TEMP_ADDR 0x2E48
+#define OMAP_ABE_S_VX_UL_8_TEMP_SIZE 0x10
+#define OMAP_ABE_S_VX_UL_16_TEMP_ADDR 0x2E58
+#define OMAP_ABE_S_VX_UL_16_TEMP_SIZE 0x20
+#define OMAP_ABE_S_VX_DL_8_48_LP_DATA_ADDR 0x2E78
+#define OMAP_ABE_S_VX_DL_8_48_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_VX_DL_8_48_HP_DATA_ADDR 0x2EE0
+#define OMAP_ABE_S_VX_DL_8_48_HP_DATA_SIZE 0x38
+#define OMAP_ABE_S_VX_DL_16_48_LP_DATA_ADDR 0x2F18
+#define OMAP_ABE_S_VX_DL_16_48_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_VX_DL_16_48_HP_DATA_ADDR 0x2F80
+#define OMAP_ABE_S_VX_DL_16_48_HP_DATA_SIZE 0x28
+#define OMAP_ABE_S_VX_UL_48_8_LP_DATA_ADDR 0x2FA8
+#define OMAP_ABE_S_VX_UL_48_8_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_VX_UL_48_8_HP_DATA_ADDR 0x3010
+#define OMAP_ABE_S_VX_UL_48_8_HP_DATA_SIZE 0x38
+#define OMAP_ABE_S_VX_UL_48_16_LP_DATA_ADDR 0x3048
+#define OMAP_ABE_S_VX_UL_48_16_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_VX_UL_48_16_HP_DATA_ADDR 0x30B0
+#define OMAP_ABE_S_VX_UL_48_16_HP_DATA_SIZE 0x28
+#define OMAP_ABE_S_BT_UL_8_48_LP_DATA_ADDR 0x30D8
+#define OMAP_ABE_S_BT_UL_8_48_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_BT_UL_8_48_HP_DATA_ADDR 0x3140
+#define OMAP_ABE_S_BT_UL_8_48_HP_DATA_SIZE 0x38
+#define OMAP_ABE_S_BT_UL_16_48_LP_DATA_ADDR 0x3178
+#define OMAP_ABE_S_BT_UL_16_48_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_BT_UL_16_48_HP_DATA_ADDR 0x31E0
+#define OMAP_ABE_S_BT_UL_16_48_HP_DATA_SIZE 0x28
+#define OMAP_ABE_S_BT_DL_48_8_LP_DATA_ADDR 0x3208
+#define OMAP_ABE_S_BT_DL_48_8_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_BT_DL_48_8_HP_DATA_ADDR 0x3270
+#define OMAP_ABE_S_BT_DL_48_8_HP_DATA_SIZE 0x38
+#define OMAP_ABE_S_BT_DL_48_16_LP_DATA_ADDR 0x32A8
+#define OMAP_ABE_S_BT_DL_48_16_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_BT_DL_48_16_HP_DATA_ADDR 0x3310
+#define OMAP_ABE_S_BT_DL_48_16_HP_DATA_SIZE 0x28
+#define OMAP_ABE_S_ECHO_REF_48_8_LP_DATA_ADDR 0x3338
+#define OMAP_ABE_S_ECHO_REF_48_8_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_ECHO_REF_48_8_HP_DATA_ADDR 0x33A0
+#define OMAP_ABE_S_ECHO_REF_48_8_HP_DATA_SIZE 0x38
+#define OMAP_ABE_S_ECHO_REF_48_16_LP_DATA_ADDR 0x33D8
+#define OMAP_ABE_S_ECHO_REF_48_16_LP_DATA_SIZE 0x68
+#define OMAP_ABE_S_ECHO_REF_48_16_HP_DATA_ADDR 0x3440
+#define OMAP_ABE_S_ECHO_REF_48_16_HP_DATA_SIZE 0x28
+#define OMAP_ABE_S_XINASRC_ECHO_REF_ADDR 0x3468
+#define OMAP_ABE_S_XINASRC_ECHO_REF_SIZE 0x140
+#define OMAP_ABE_S_ECHO_REF_16K_ADDR 0x35A8
+#define OMAP_ABE_S_ECHO_REF_16K_SIZE 0x28
+#define OMAP_ABE_S_ECHO_REF_8K_ADDR 0x35D0
+#define OMAP_ABE_S_ECHO_REF_8K_SIZE 0x18
+#define OMAP_ABE_S_DL1_EQ_ADDR 0x35E8
+#define OMAP_ABE_S_DL1_EQ_SIZE 0x60
+#define OMAP_ABE_S_DL2_EQ_ADDR 0x3648
+#define OMAP_ABE_S_DL2_EQ_SIZE 0x60
+#define OMAP_ABE_S_DL1_GAIN_OUT_ADDR 0x36A8
+#define OMAP_ABE_S_DL1_GAIN_OUT_SIZE 0x60
+#define OMAP_ABE_S_DL2_GAIN_OUT_ADDR 0x3708
+#define OMAP_ABE_S_DL2_GAIN_OUT_SIZE 0x60
+#define OMAP_ABE_S_DC_HS_ADDR 0x3768
+#define OMAP_ABE_S_DC_HS_SIZE 0x8
+#define OMAP_ABE_S_DC_HF_ADDR 0x3770
+#define OMAP_ABE_S_DC_HF_SIZE 0x8
+#define OMAP_ABE_S_VIBRA_ADDR 0x3778
+#define OMAP_ABE_S_VIBRA_SIZE 0x30
+#define OMAP_ABE_S_VIBRA2_IN_ADDR 0x37A8
+#define OMAP_ABE_S_VIBRA2_IN_SIZE 0x30
+#define OMAP_ABE_S_VIBRA2_ADDR_ADDR 0x37D8
+#define OMAP_ABE_S_VIBRA2_ADDR_SIZE 0x8
+#define OMAP_ABE_S_VIBRACTRL_FORRIGHTSM_ADDR 0x37E0
+#define OMAP_ABE_S_VIBRACTRL_FORRIGHTSM_SIZE 0xC0
+#define OMAP_ABE_S_RNOISE_MEM_ADDR 0x38A0
+#define OMAP_ABE_S_RNOISE_MEM_SIZE 0x8
+#define OMAP_ABE_S_CTRL_ADDR 0x38A8
+#define OMAP_ABE_S_CTRL_SIZE 0x90
+#define OMAP_ABE_S_VIBRA1_IN_ADDR 0x3938
+#define OMAP_ABE_S_VIBRA1_IN_SIZE 0x30
+#define OMAP_ABE_S_VIBRA1_TEMP_ADDR 0x3968
+#define OMAP_ABE_S_VIBRA1_TEMP_SIZE 0xC0
+#define OMAP_ABE_S_VIBRACTRL_FORLEFTSM_ADDR 0x3A28
+#define OMAP_ABE_S_VIBRACTRL_FORLEFTSM_SIZE 0xC0
+#define OMAP_ABE_S_VIBRA1_MEM_ADDR 0x3AE8
+#define OMAP_ABE_S_VIBRA1_MEM_SIZE 0x58
+#define OMAP_ABE_S_VIBRACTRL_STEREO_ADDR 0x3B40
+#define OMAP_ABE_S_VIBRACTRL_STEREO_SIZE 0xC0
+#define OMAP_ABE_S_AMIC_96_48_DATA_ADDR 0x3C00
+#define OMAP_ABE_S_AMIC_96_48_DATA_SIZE 0x98
+#define OMAP_ABE_S_DMIC0_96_48_DATA_ADDR 0x3C98
+#define OMAP_ABE_S_DMIC0_96_48_DATA_SIZE 0x98
+#define OMAP_ABE_S_DMIC1_96_48_DATA_ADDR 0x3D30
+#define OMAP_ABE_S_DMIC1_96_48_DATA_SIZE 0x98
+#define OMAP_ABE_S_DMIC2_96_48_DATA_ADDR 0x3DC8
+#define OMAP_ABE_S_DMIC2_96_48_DATA_SIZE 0x98
+#define OMAP_ABE_S_DBG_8K_PATTERN_ADDR 0x3E60
+#define OMAP_ABE_S_DBG_8K_PATTERN_SIZE 0x10
+#define OMAP_ABE_S_DBG_16K_PATTERN_ADDR 0x3E70
+#define OMAP_ABE_S_DBG_16K_PATTERN_SIZE 0x20
+#define OMAP_ABE_S_DBG_24K_PATTERN_ADDR 0x3E90
+#define OMAP_ABE_S_DBG_24K_PATTERN_SIZE 0x30
+#define OMAP_ABE_S_DBG_48K_PATTERN_ADDR 0x3EC0
+#define OMAP_ABE_S_DBG_48K_PATTERN_SIZE 0x60
+#define OMAP_ABE_S_DBG_96K_PATTERN_ADDR 0x3F20
+#define OMAP_ABE_S_DBG_96K_PATTERN_SIZE 0xC0
+#define OMAP_ABE_S_MM_EXT_IN_ADDR 0x3FE0
+#define OMAP_ABE_S_MM_EXT_IN_SIZE 0x60
+#define OMAP_ABE_S_MM_EXT_IN_L_ADDR 0x4040
+#define OMAP_ABE_S_MM_EXT_IN_L_SIZE 0x60
+#define OMAP_ABE_S_MM_EXT_IN_R_ADDR 0x40A0
+#define OMAP_ABE_S_MM_EXT_IN_R_SIZE 0x60
+#define OMAP_ABE_S_MIC4_ADDR 0x4100
+#define OMAP_ABE_S_MIC4_SIZE 0x60
+#define OMAP_ABE_S_MIC4_L_ADDR 0x4160
+#define OMAP_ABE_S_MIC4_L_SIZE 0x60
+#define OMAP_ABE_S_SATURATION_7FFF_ADDR 0x41C0
+#define OMAP_ABE_S_SATURATION_7FFF_SIZE 0x8
+#define OMAP_ABE_S_SATURATION_ADDR 0x41C8
+#define OMAP_ABE_S_SATURATION_SIZE 0x8
+#define OMAP_ABE_S_XINASRC_BT_UL_ADDR 0x41D0
+#define OMAP_ABE_S_XINASRC_BT_UL_SIZE 0x140
+#define OMAP_ABE_S_XINASRC_BT_DL_ADDR 0x4310
+#define OMAP_ABE_S_XINASRC_BT_DL_SIZE 0x140
+#define OMAP_ABE_S_BT_DL_8K_TEMP_ADDR 0x4450
+#define OMAP_ABE_S_BT_DL_8K_TEMP_SIZE 0x10
+#define OMAP_ABE_S_BT_DL_16K_TEMP_ADDR 0x4460
+#define OMAP_ABE_S_BT_DL_16K_TEMP_SIZE 0x20
+#define OMAP_ABE_S_VX_DL_8_48_OSR_LP_DATA_ADDR 0x4480
+#define OMAP_ABE_S_VX_DL_8_48_OSR_LP_DATA_SIZE 0xE0
+#define OMAP_ABE_S_BT_UL_8_48_OSR_LP_DATA_ADDR 0x4560
+#define OMAP_ABE_S_BT_UL_8_48_OSR_LP_DATA_SIZE 0xE0
+#define OMAP_ABE_S_MM_DL_44P1_ADDR 0x4640
+#define OMAP_ABE_S_MM_DL_44P1_SIZE 0x300
+#define OMAP_ABE_S_TONES_44P1_ADDR 0x4940
+#define OMAP_ABE_S_TONES_44P1_SIZE 0x300
+#define OMAP_ABE_S_MM_DL_44P1_XK_ADDR 0x4C40
+#define OMAP_ABE_S_MM_DL_44P1_XK_SIZE 0x10
+#define OMAP_ABE_S_TONES_44P1_XK_ADDR 0x4C50
+#define OMAP_ABE_S_TONES_44P1_XK_SIZE 0x10
+#define OMAP_ABE_S_SRC_44P1_MULFAC1_ADDR 0x4C60
+#define OMAP_ABE_S_SRC_44P1_MULFAC1_SIZE 0x8
diff --git a/sound/soc/omap/abe/abe_taskid.h b/sound/soc/omap/abe/abe_taskid.h
new file mode 100644
index 0000000..b72c3a5
--- /dev/null
+++ b/sound/soc/omap/abe/abe_taskid.h
@@ -0,0 +1,187 @@
+/*
+ *
+ * This file is provided under a dual BSD/GPLv2 license. When using or
+ * redistributing this file, you may do so under either license.
+ *
+ * GPL LICENSE SUMMARY
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of version 2 of the GNU General Public License as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ * The full GNU General Public License is included in this distribution
+ * in the file called LICENSE.GPL.
+ *
+ * BSD LICENSE
+ *
+ * Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * * Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * * Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in
+ * the documentation and/or other materials provided with the
+ * distribution.
+ * * Neither the name of Texas Instruments Incorporated nor the names of
+ * its contributors may be used to endorse or promote products derived
+ * from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#ifndef _ABE_TASKID_H_
+#define _ABE_TASKID_H_
+#define C_ABE_FW_TASK_ASRC_VX_DL_8 0
+#define C_ABE_FW_TASK_ASRC_VX_DL_16 1
+#define C_ABE_FW_TASK_ASRC_VX_DL_8_SIB 2
+#define C_ABE_FW_TASK_ASRC_VX_DL_16_SIB 3
+#define C_ABE_FW_TASK_ASRC_MM_EXT_IN 4
+#define C_ABE_FW_TASK_ASRC_VX_UL_8 5
+#define C_ABE_FW_TASK_ASRC_VX_UL_16 6
+#define C_ABE_FW_TASK_ASRC_VX_UL_8_SIB 7
+#define C_ABE_FW_TASK_ASRC_VX_UL_16_SIB 8
+#define C_ABE_FW_TASK_VX_UL_48_8_DEC 9
+#define C_ABE_FW_TASK_VX_UL_48_16_DEC 10
+#define C_ABE_FW_TASK_BT_DL_48_8_DEC 11
+#define C_ABE_FW_TASK_BT_DL_48_16_DEC 12
+#define C_ABE_FW_TASK_ECHO_REF_48_8_DEC 13
+#define C_ABE_FW_TASK_ECHO_REF_48_16_DEC 14
+#define C_ABE_FW_TASK_DL2_EQ 15
+#define C_ABE_FW_TASK_ECHO_REF_48_16 16
+#define C_ABE_FW_TASK_ECHO_REF_48_8 17
+#define C_ABE_FW_TASK_GAIN_UPDATE 18
+#define C_ABE_FW_TASK_SideTone 19
+#define C_ABE_FW_TASK_VX_DL_8_48_LP 20
+#define C_ABE_FW_TASK_VX_DL_8_48_HP 21
+#define C_ABE_FW_TASK_VX_DL_16_48_LP 22
+#define C_ABE_FW_TASK_VX_DL_16_48_HP 23
+#define C_ABE_FW_TASK_VX_UL_48_8_LP 24
+#define C_ABE_FW_TASK_VX_UL_48_8_HP 25
+#define C_ABE_FW_TASK_VX_UL_48_16_LP 26
+#define C_ABE_FW_TASK_VX_UL_48_16_HP 27
+#define C_ABE_FW_TASK_BT_UL_8_48_LP 28
+#define C_ABE_FW_TASK_BT_UL_8_48_HP 29
+#define C_ABE_FW_TASK_BT_UL_16_48_LP 30
+#define C_ABE_FW_TASK_BT_UL_16_48_HP 31
+#define C_ABE_FW_TASK_BT_DL_48_8_LP 32
+#define C_ABE_FW_TASK_BT_DL_48_8_HP 33
+#define C_ABE_FW_TASK_BT_DL_48_16_LP 34
+#define C_ABE_FW_TASK_BT_DL_48_16_HP 35
+#define C_ABE_FW_TASK_ECHO_REF_48_8_LP 36
+#define C_ABE_FW_TASK_ECHO_REF_48_8_HP 37
+#define C_ABE_FW_TASK_ECHO_REF_48_16_LP 38
+#define C_ABE_FW_TASK_ECHO_REF_48_16_HP 39
+#define C_ABE_FW_TASK_DL1_EQ 40
+#define C_ABE_FW_TASK_IHF_48_96_LP 41
+#define C_ABE_FW_TASK_EARP_48_96_LP 42
+#define C_ABE_FW_TASK_DL1_GAIN 43
+#define C_ABE_FW_TASK_DL2_GAIN 44
+#define C_ABE_FW_TASK_IO_PING_PONG 45
+#define C_ABE_FW_TASK_IO_DMIC 46
+#define C_ABE_FW_TASK_IO_PDM_UL 47
+#define C_ABE_FW_TASK_IO_BT_VX_UL 48
+#define C_ABE_FW_TASK_IO_MM_UL 49
+#define C_ABE_FW_TASK_IO_MM_UL2 50
+#define C_ABE_FW_TASK_IO_VX_UL 51
+#define C_ABE_FW_TASK_IO_MM_DL 52
+#define C_ABE_FW_TASK_IO_VX_DL 53
+#define C_ABE_FW_TASK_IO_TONES_DL 54
+#define C_ABE_FW_TASK_IO_VIB_DL 55
+#define C_ABE_FW_TASK_IO_BT_VX_DL 56
+#define C_ABE_FW_TASK_IO_PDM_DL 57
+#define C_ABE_FW_TASK_IO_MM_EXT_OUT 58
+#define C_ABE_FW_TASK_IO_MM_EXT_IN 59
+#define C_ABE_FW_TASK_DEBUG_IRQFIFO 60
+#define C_ABE_FW_TASK_EchoMixer 61
+#define C_ABE_FW_TASK_SDTMixer 62
+#define C_ABE_FW_TASK_DL1Mixer 63
+#define C_ABE_FW_TASK_DL2Mixer 64
+#define C_ABE_FW_TASK_DL1Mixer_dual_mono 65
+#define C_ABE_FW_TASK_DL2Mixer_dual_mono 66
+#define C_ABE_FW_TASK_VXRECMixer 67
+#define C_ABE_FW_TASK_ULMixer 68
+#define C_ABE_FW_TASK_ULMixer_dual_mono 69
+#define C_ABE_FW_TASK_VIBRA_PACK 70
+#define C_ABE_FW_TASK_VX_DL_8_48_0SR 71
+#define C_ABE_FW_TASK_VX_DL_16_48_0SR 72
+#define C_ABE_FW_TASK_BT_UL_8_48_0SR 73
+#define C_ABE_FW_TASK_BT_UL_16_48_0SR 74
+#define C_ABE_FW_TASK_IHF_48_96_0SR 75
+#define C_ABE_FW_TASK_EARP_48_96_0SR 76
+#define C_ABE_FW_TASK_AMIC_SPLIT 77
+#define C_ABE_FW_TASK_DMIC1_SPLIT 78
+#define C_ABE_FW_TASK_DMIC2_SPLIT 79
+#define C_ABE_FW_TASK_DMIC3_SPLIT 80
+#define C_ABE_FW_TASK_VXREC_SPLIT 81
+#define C_ABE_FW_TASK_BT_UL_SPLIT 82
+#define C_ABE_FW_TASK_MM_SPLIT 83
+#define C_ABE_FW_TASK_VIBRA_SPLIT 84
+#define C_ABE_FW_TASK_MM_EXT_IN_SPLIT 85
+#define C_ABE_FW_TASK_ECHO_REF_SPLIT 86
+#define C_ABE_FW_TASK_UNUSED_1 87
+#define C_ABE_FW_TASK_VX_UL_ROUTING 88
+#define C_ABE_FW_TASK_MM_UL2_ROUTING 89
+#define C_ABE_FW_TASK_VIBRA1 90
+#define C_ABE_FW_TASK_VIBRA2 91
+#define C_ABE_FW_TASK_BT_UL_16_48 92
+#define C_ABE_FW_TASK_BT_UL_8_48 93
+#define C_ABE_FW_TASK_BT_DL_48_16 94
+#define C_ABE_FW_TASK_BT_DL_48_8 95
+#define C_ABE_FW_TASK_VX_DL_16_48 96
+#define C_ABE_FW_TASK_VX_DL_8_48 97
+#define C_ABE_FW_TASK_VX_UL_48_16 98
+#define C_ABE_FW_TASK_VX_UL_48_8 99
+#define C_ABE_FW_TASK_DBG_SYNC 100
+#define C_ABE_FW_TASK_AMIC_96_48_LP 101
+#define C_ABE_FW_TASK_DMIC1_96_48_LP 102
+#define C_ABE_FW_TASK_DMIC2_96_48_LP 103
+#define C_ABE_FW_TASK_DMIC3_96_48_LP 104
+#define C_ABE_FW_TASK_INIT_FW_MEMORY 105
+#define C_ABE_FW_TASK_DEBUGTRACE_VX_ASRCs 106
+#define C_ABE_FW_TASK_ASRC_BT_UL_8 107
+#define C_ABE_FW_TASK_ASRC_BT_UL_16 108
+#define C_ABE_FW_TASK_ASRC_BT_UL_8_SIB 109
+#define C_ABE_FW_TASK_ASRC_BT_UL_16_SIB 110
+#define C_ABE_FW_TASK_ASRC_BT_DL_8 111
+#define C_ABE_FW_TASK_ASRC_BT_DL_16 112
+#define C_ABE_FW_TASK_ASRC_BT_DL_8_SIB 113
+#define C_ABE_FW_TASK_ASRC_BT_DL_16_SIB 114
+#define C_ABE_FW_TASK_BT_DL_48_8_HP_OPP100 115
+#define C_ABE_FW_TASK_BT_DL_48_16_HP_OPP100 116
+#define C_ABE_FW_TASK_BT_DL_48_8_OPP100 117
+#define C_ABE_FW_TASK_BT_DL_48_16_OPP100 118
+#define C_ABE_FW_TASK_VX_DL_8_48_OSR_LP 119
+#define C_ABE_FW_TASK_VX_DL_8_48_FIR 120
+#define C_ABE_FW_TASK_BT_UL_8_48_OSR_LP 121
+#define C_ABE_FW_TASK_BT_UL_8_48_FIR 122
+#define C_ABE_FW_TASK_SRC44P1_MMDL 123
+#define C_ABE_FW_TASK_SRC44P1_TONES 124
+#define C_ABE_FW_TASK_SRC44P1_MMDL_1211 125
+#define C_ABE_FW_TASK_SRC44P1_TONES_1211 126
+#endif /* _ABE_TASKID_H_ */
diff --git a/sound/soc/omap/abe/abe_typ.h b/sound/soc/omap/abe/abe_typ.h
new file mode 100644
index 0000000..650d043
--- /dev/null
+++ b/sound/soc/omap/abe/abe_typ.h
@@ -0,0 +1,654 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#include "abe_def.h"
+#include "abe_initxxx_labels.h"
+
+#ifndef _ABE_TYP_H_
+#define _ABE_TYP_H_
+/*
+ * BASIC TYPES
+ */
+#define MAX_UINT8 ((((1L << 7) - 1) << 1) + 1)
+#define MAX_UINT16 ((((1L << 15) - 1) << 1) + 1)
+#define MAX_UINT32 ((((1L << 31) - 1) << 1) + 1)
+#define s8 char
+#define u8 unsigned char
+#define s16 short
+#define u16 unsigned short
+#define s32 int
+#define u32 unsigned int
+/* returned status from HAL APIs */
+#define abehal_status u32
+/* 4 bytes Bit field indicating the type of informations to be traced */
+typedef u32 abe_dbg_mask_t;
+/* scheduling task loops (250us / 272us with respectively 48kHz /
+ 44.1kHz on Phoenix). */
+typedef u32 abe_dbg_t;
+/* Index to the table of sequences */
+typedef u32 abe_seq_code_t;
+/* Index to the table of subroutines called in the sequence */
+typedef u32 abe_sub_code_t;
+/* subroutine with no parameter */
+typedef void (*abe_subroutine0) (void);
+/* subroutine with one parameter */
+typedef void (*abe_subroutine1) (u32);
+typedef void (*abe_subroutine2) (u32, u32);
+typedef void (*abe_subroutine3) (u32, u32, u32);
+typedef void (*abe_subroutine4) (u32, u32, u32, u32);
+/*
+ * CODE PORTABILITY - FUTURE PATCHES
+ *
+ * 32bits field for having the code compatible with future revisions of
+ * the hardware (audio integration) or evolution of the software
+ * partitionning. Used for the highest level APIs (launch_sequences)
+ */
+typedef u32 abe_patch_rev;
+/*
+ * ENUMS
+ */
+/*
+ * MEMORY CONFIG TYPE
+ *
+ * 0: Ultra Lowest power consumption audio player
+ * 1: OPP 25% (simple multimedia features)
+ * 2: OPP 50% (multimedia and voice calls)
+ * 3: OPP100% (multimedia complex use-cases)
+ */
+#define ABE_AUDIO_PLAYER_ON_HEADSET_OR_EARPHONE 1
+#define ABE_DRIFT_MANAGEMENT_FOR_AUDIO_PLAYER 2
+#define ABE_DRIFT_MANAGEMENT_FOR_VOICE_CALL 3
+#define ABE_VOICE_CALL_ON_HEADSET_OR_EARPHONE_OR_BT 4
+#define ABE_MULTIMEDIA_AUDIO_RECORDER 5
+#define ABE_VIBRATOR_OR_HAPTICS 6
+#define ABE_VOICE_CALL_ON_HANDS_FREE_SPEAKER 7
+#define ABE_RINGER_TONES 8
+#define ABE_VOICE_CALL_WITH_EARPHONE_ACTIVE_NOISE_CANCELLER 9
+#define ABE_LAST_USE_CASE 10
+/*
+ * OPP TYPE
+ *
+ * 0: Ultra Lowest power consumption audio player
+ * 1: OPP 25% (simple multimedia features)
+ * 2: OPP 50% (multimedia and voice calls)
+ * 3: OPP100% (multimedia complex use-cases)
+ */
+#define ABE_OPP0 0
+#define ABE_OPP25 1
+#define ABE_OPP50 2
+#define ABE_OPP100 3
+/*
+ * DMIC DECIMATION RATIO
+ *
+ */
+#define ABE_DEC16 16
+#define ABE_DEC25 25
+#define ABE_DEC32 32
+#define ABE_DEC40 40
+/*
+ * SAMPLES TYPE
+ *
+ * mono 16 bit sample LSB aligned, 16 MSB bits are unused;
+ * mono right shifted to 16bits LSBs on a 32bits DMEM FIFO for McBSP
+ * TX purpose;
+ * mono sample MSB aligned (16/24/32bits);
+ * two successive mono samples in one 32bits container;
+ * Two L/R 16bits samples in a 32bits container;
+ * Two channels defined with two MSB aligned samples;
+ * Three channels defined with three MSB aligned samples (MIC);
+ * Four channels defined with four MSB aligned samples (MIC);
+ * . . .
+ * Eight channels defined with eight MSB aligned samples (MIC);
+ */
+#define MONO_MSB 1
+#define MONO_RSHIFTED_16 2
+#define STEREO_RSHIFTED_16 3
+#define STEREO_16_16 4
+#define STEREO_MSB 5
+#define THREE_MSB 6
+#define FOUR_MSB 7
+#define FIVE_MSB 8
+#define SIX_MSB 9
+#define SEVEN_MSB 10
+#define EIGHT_MSB 11
+#define NINE_MSB 12
+#define TEN_MSB 13
+/*
+ * PORT PROTOCOL TYPE - abe_port_protocol_switch_id
+ */
+#define SLIMBUS_PORT_PROT 1
+#define SERIAL_PORT_PROT 2
+#define TDM_SERIAL_PORT_PROT 3
+#define DMIC_PORT_PROT 4
+#define MCPDMDL_PORT_PROT 5
+#define MCPDMUL_PORT_PROT 6
+#define PINGPONG_PORT_PROT 7
+#define DMAREQ_PORT_PROT 8
+/*
+ * PORT IDs, this list is aligned with the FW data mapping
+ */
+#define OMAP_ABE_DMIC_PORT 0
+#define OMAP_ABE_PDM_UL_PORT 1
+#define OMAP_ABE_BT_VX_UL_PORT 2
+#define OMAP_ABE_MM_UL_PORT 3
+#define OMAP_ABE_MM_UL2_PORT 4
+#define OMAP_ABE_VX_UL_PORT 5
+#define OMAP_ABE_MM_DL_PORT 6
+#define OMAP_ABE_VX_DL_PORT 7
+#define OMAP_ABE_TONES_DL_PORT 8
+#define OMAP_ABE_VIB_DL_PORT 9
+#define OMAP_ABE_BT_VX_DL_PORT 10
+#define OMAP_ABE_PDM_DL_PORT 11
+#define OMAP_ABE_MM_EXT_OUT_PORT 12
+#define OMAP_ABE_MM_EXT_IN_PORT 13
+#define TDM_DL_PORT 14
+#define TDM_UL_PORT 15
+#define DEBUG_PORT 16
+#define LAST_PORT_ID 17
+/* definitions for the compatibility with HAL05xx */
+#define PDM_DL1_PORT 18
+#define PDM_DL2_PORT 19
+#define PDM_VIB_PORT 20
+/* There is only one DMIC port, always used with 6 samples
+ per 96kHz periods */
+#define DMIC_PORT1 DMIC_PORT
+#define DMIC_PORT2 DMIC_PORT
+#define DMIC_PORT3 DMIC_PORT
+/*
+ * ABE_DL_SRC_ID source of samples
+ */
+#define SRC_DL1_MIXER_OUTPUT DL1_M_labelID
+#define SRC_SDT_MIXER_OUTPUT SDT_M_labelID
+#define SRC_DL1_GAIN_OUTPUT DL1_GAIN_out_labelID
+#define SRC_DL1_EQ_OUTPUT DL1_EQ_labelID
+#define SRC_DL2_GAIN_OUTPUT DL2_GAIN_out_labelID
+#define SRC_DL2_EQ_OUTPUT DL2_EQ_labelID
+#define SRC_MM_DL MM_DL_labelID
+#define SRC_TONES_DL Tones_labelID
+#define SRC_VX_DL VX_DL_labelID
+#define SRC_VX_UL VX_UL_labelID
+#define SRC_MM_UL2 MM_UL2_labelID
+#define SRC_MM_UL MM_UL_labelID
+/*
+ * abe_patched_pattern_id
+ * selection of the audio engine signal to
+ * replace by a precomputed pattern
+ */
+#define DBG_PATCH_AMIC 1
+#define DBG_PATCH_DMIC1 2
+#define DBG_PATCH_DMIC2 3
+#define DBG_PATCH_DMIC3 4
+#define DBG_PATCH_VX_REC 5
+#define DBG_PATCH_BT_UL 6
+#define DBG_PATCH_MM_DL 7
+#define DBG_PATCH_DL2_EQ 8
+#define DBG_PATCH_VIBRA 9
+#define DBG_PATCH_MM_EXT_IN 10
+#define DBG_PATCH_EANC_FBK_Out 11
+#define DBG_PATCH_MIC4 12
+#define DBG_PATCH_MM_DL_MIXDL1 13
+#define DBG_PATCH_MM_DL_MIXDL2 14
+/*
+ * Signal processing module names - EQ APS MIX ROUT
+ */
+/* equalizer downlink path headset + earphone */
+#define FEAT_EQ1 1
+/* equalizer downlink path integrated handsfree LEFT */
+#define FEAT_EQ2L (FEAT_EQ1+1)
+/* equalizer downlink path integrated handsfree RIGHT */
+#define FEAT_EQ2R (FEAT_EQ2L+1)
+/* equalizer downlink path side-tone */
+#define FEAT_EQSDT (FEAT_EQ2R+1)
+/* equalizer uplink path AMIC */
+#define FEAT_EQAMIC (FEAT_EQSDT+1)
+/* equalizer uplink path DMIC */
+#define FEAT_EQDMIC (FEAT_EQAMIC+1)
+/* Acoustic protection for headset */
+#define FEAT_APS1 (FEAT_EQDMIC+1)
+/* acoustic protection high-pass filter for handsfree "Left" */
+#define FEAT_APS2 (FEAT_APS1+1)
+/* acoustic protection high-pass filter for handsfree "Right" */
+#define FEAT_APS3 (FEAT_APS2+1)
+/* asynchronous sample-rate-converter for the downlink voice path */
+#define FEAT_ASRC1 (FEAT_APS3+1)
+/* asynchronous sample-rate-converter for the uplink voice path */
+#define FEAT_ASRC2 (FEAT_ASRC1+1)
+/* asynchronous sample-rate-converter for the multimedia player */
+#define FEAT_ASRC3 (FEAT_ASRC2+1)
+/* asynchronous sample-rate-converter for the echo reference */
+#define FEAT_ASRC4 (FEAT_ASRC3+1)
+/* mixer of the headset and earphone path */
+#define FEAT_MIXDL1 (FEAT_ASRC4+1)
+/* mixer of the hands-free path */
+#define FEAT_MIXDL2 (FEAT_MIXDL1+1)
+/* mixer for audio being sent on the voice_ul path */
+#define FEAT_MIXAUDUL (FEAT_MIXDL2+1)
+/* mixer for voice communication recording */
+#define FEAT_MIXVXREC (FEAT_MIXAUDUL+1)
+/* mixer for side-tone */
+#define FEAT_MIXSDT (FEAT_MIXVXREC+1)
+/* mixer for echo reference */
+#define FEAT_MIXECHO (FEAT_MIXSDT+1)
+/* router of the uplink path */
+#define FEAT_UPROUTE (FEAT_MIXECHO+1)
+/* all gains */
+#define FEAT_GAINS (FEAT_UPROUTE+1)
+#define FEAT_GAINS_DMIC1 (FEAT_GAINS+1)
+#define FEAT_GAINS_DMIC2 (FEAT_GAINS_DMIC1+1)
+#define FEAT_GAINS_DMIC3 (FEAT_GAINS_DMIC2+1)
+#define FEAT_GAINS_AMIC (FEAT_GAINS_DMIC3+1)
+#define FEAT_GAINS_SPLIT (FEAT_GAINS_AMIC+1)
+#define FEAT_GAINS_DL1 (FEAT_GAINS_SPLIT+1)
+#define FEAT_GAINS_DL2 (FEAT_GAINS_DL1+1)
+#define FEAT_GAIN_BTUL (FEAT_GAINS_DL2+1)
+/* sequencing queue of micro tasks */
+#define FEAT_SEQ (FEAT_GAIN_BTUL+1)
+/* Phoenix control queue through McPDM */
+#define FEAT_CTL (FEAT_SEQ+1)
+/* list of features of the firmware -------------------------------*/
+#define MAXNBFEATURE FEAT_CTL
+/* abe_equ_id */
+/* equalizer downlink path headset + earphone */
+#define EQ1 FEAT_EQ1
+/* equalizer downlink path integrated handsfree LEFT */
+#define EQ2L FEAT_EQ2L
+#define EQ2R FEAT_EQ2R
+/* equalizer downlink path side-tone */
+#define EQSDT FEAT_EQSDT
+#define EQAMIC FEAT_EQAMIC
+#define EQDMIC FEAT_EQDMIC
+/* abe_aps_id */
+/* Acoustic protection for headset */
+#define APS1 FEAT_APS1
+#define APS2L FEAT_APS2
+#define APS2R FEAT_APS3
+/* abe_asrc_id */
+/* asynchronous sample-rate-converter for the downlink voice path */
+#define ASRC1 FEAT_ASRC1
+/* asynchronous sample-rate-converter for the uplink voice path */
+#define ASRC2 FEAT_ASRC2
+/* asynchronous sample-rate-converter for the multimedia player */
+#define ASRC3 FEAT_ASRC3
+/* asynchronous sample-rate-converter for the voice uplink echo_reference */
+#define ASRC4 FEAT_ASRC4
+/* abe_mixer_id */
+#define MIXDL1 FEAT_MIXDL1
+#define MIXDL2 FEAT_MIXDL2
+#define MIXSDT FEAT_MIXSDT
+#define MIXECHO FEAT_MIXECHO
+#define MIXAUDUL FEAT_MIXAUDUL
+#define MIXVXREC FEAT_MIXVXREC
+/* abe_router_id */
+/* there is only one router up to now */
+#define UPROUTE FEAT_UPROUTE
+/*
+ * GAIN IDs
+ */
+#define GAINS_DMIC1 FEAT_GAINS_DMIC1
+#define GAINS_DMIC2 FEAT_GAINS_DMIC2
+#define GAINS_DMIC3 FEAT_GAINS_DMIC3
+#define GAINS_AMIC FEAT_GAINS_AMIC
+#define GAINS_SPLIT FEAT_GAINS_SPLIT
+#define GAINS_DL1 FEAT_GAINS_DL1
+#define GAINS_DL2 FEAT_GAINS_DL2
+#define GAINS_BTUL FEAT_GAIN_BTUL
+/*
+ * EVENT GENERATORS - abe_event_id
+ */
+#define EVENT_TIMER 0
+#define EVENT_44100 1
+/*
+ * SERIAL PORTS IDs - abe_mcbsp_id
+ */
+#define MCBSP1_TX MCBSP1_DMA_TX
+#define MCBSP1_RX MCBSP1_DMA_RX
+#define MCBSP2_TX MCBSP2_DMA_TX
+#define MCBSP2_RX MCBSP2_DMA_RX
+#define MCBSP3_TX MCBSP3_DMA_TX
+#define MCBSP3_RX MCBSP3_DMA_RX
+/*
+ * SERIAL PORTS IDs - abe_slimbus_id;
+ */
+#define SLIMBUS1_TX0 SLIMBUS1_DMA_TX0
+#define SLIMBUS1_TX1 SLIMBUS1_DMA_TX1
+#define SLIMBUS1_TX2 SLIMBUS1_DMA_TX2
+#define SLIMBUS1_TX3 SLIMBUS1_DMA_TX3
+#define SLIMBUS1_TX4 SLIMBUS1_DMA_TX4
+#define SLIMBUS1_TX5 SLIMBUS1_DMA_TX5
+#define SLIMBUS1_TX6 SLIMBUS1_DMA_TX6
+#define SLIMBUS1_TX7 SLIMBUS1_DMA_TX7
+#define SLIMBUS1_RX0 SLIMBUS1_DMA_RX0
+#define SLIMBUS1_RX1 SLIMBUS1_DMA_RX1
+#define SLIMBUS1_RX2 SLIMBUS1_DMA_RX2
+#define SLIMBUS1_RX3 SLIMBUS1_DMA_RX3
+#define SLIMBUS1_RX4 SLIMBUS1_DMA_RX4
+#define SLIMBUS1_RX5 SLIMBUS1_DMA_RX5
+#define SLIMBUS1_RX6 SLIMBUS1_DMA_RX6
+#define SLIMBUS1_RX7 SLIMBUS1_DMA_RX7
+#define SLIMBUS_UNUSED _DUMMY_FIFO_
+/*
+ * ----------------- TYPES USED FOR APIS ---------------
+ */
+
+/*
+ * EQU_T
+ *
+ * coefficients of the equalizer
+ */
+typedef struct {
+ /* type of filter */
+ u32 equ_type;
+ /* filter length */
+ u32 equ_length;
+ union {
+ /* parameters are the direct and recursive coefficients in */
+ /* Q6.26 integer fixed-point format. */
+ s32 type1[NBEQ1];
+ struct {
+ /* center frequency of the band [Hz] */
+ s32 freq[NBEQ2];
+ /* gain of each band. [dB] */
+ s32 gain[NBEQ2];
+ /* Q factor of this band [dB] */
+ s32 q[NBEQ2];
+ } type2;
+ } coef;
+ s32 equ_param3;
+} abe_equ_t;
+
+/*
+ * APS_T
+ *
+ * coefficients of the Acoustics Protection and Safety
+ */
+struct abe_aps_t {
+ s32 coef1[NBAPS1];
+ s32 coef2[NBAPS2];
+};
+
+struct abe_aps_energy_t {
+ /* structure of two energy_t estimation for coil and membrane */
+ u32 e1;
+ u32 e2;
+};
+/*
+ * ROUTER_T
+ *
+ * table of indexes in unsigned bytes
+ */
+typedef u16 abe_router_t;
+/*
+ * DATA_FORMAT_T
+ *
+ * used in port declaration
+ */
+typedef struct {
+ /* Sampling frequency of the stream */
+ u32 f;
+ /* Sample format type */
+ u32 samp_format;
+} abe_data_format_t;
+/*
+ * PORT_PROTOCOL_T
+ *
+ * port declaration
+ */
+typedef struct {
+ /* Direction=0 means input from AESS point of view */
+ u32 direction;
+ /* Protocol type (switch) during the data transfers */
+ u32 protocol_switch;
+ union {
+ /* Slimbus peripheral connected to ATC */
+ struct {
+ /* Address of ATC Slimbus descriptor's index */
+ u32 desc_addr1;
+ /* DMEM address 1 in bytes */
+ u32 buf_addr1;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ /* Second ATC index for SlimBus reception (or NULL) */
+ u32 desc_addr2;
+ /* DMEM address 2 in bytes */
+ u32 buf_addr2;
+ } prot_slimbus;
+ /* McBSP/McASP peripheral connected to ATC */
+ struct {
+ u32 desc_addr;
+ /* Address of ATC McBSP/McASP descriptor's in bytes */
+ u32 buf_addr;
+ /* DMEM address in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ } prot_serial;
+ /* DMIC peripheral connected to ATC */
+ struct {
+ /* DMEM address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size in bytes */
+ u32 buf_size;
+ /* Number of activated DMIC */
+ u32 nbchan;
+ } prot_dmic;
+ /* McPDMDL peripheral connected to ATC */
+ struct {
+ /* DMEM address in bytes */
+ u32 buf_addr;
+ /* DMEM size in bytes */
+ u32 buf_size;
+ /* Control allowed on McPDM DL */
+ u32 control;
+ } prot_mcpdmdl;
+ /* McPDMUL peripheral connected to ATC */
+ struct {
+ /* DMEM address size in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ } prot_mcpdmul;
+ /* Ping-Pong interface to the Host using cache-flush */
+ struct {
+ /* Address of ATC descriptor's */
+ u32 desc_addr;
+ /* DMEM buffer base address in bytes */
+ u32 buf_addr;
+ /* DMEM size in bytes for each ping and pong buffers */
+ u32 buf_size;
+ /* IRQ address (either DMA (0) MCU (1) or DSP(2)) */
+ u32 irq_addr;
+ /* IRQ data content loaded in the AESS IRQ register */
+ u32 irq_data;
+ /* Call-back function upon IRQ reception */
+ u32 callback;
+ } prot_pingpong;
+ /* DMAreq line to CBPr */
+ struct {
+ /* Address of ATC descriptor's */
+ u32 desc_addr;
+ /* DMEM buffer address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size size in bytes */
+ u32 buf_size;
+ /* ITERation on each DMAreq signals */
+ u32 iter;
+ /* DMAreq address */
+ u32 dma_addr;
+ /* DMA/AESS = 1 << #DMA */
+ u32 dma_data;
+ } prot_dmareq;
+ /* Circular buffer - direct addressing to DMEM */
+ struct {
+ /* DMEM buffer base address in bytes */
+ u32 buf_addr;
+ /* DMEM buffer size in bytes */
+ u32 buf_size;
+ /* DMAreq address */
+ u32 dma_addr;
+ /* DMA/AESS = 1 << #DMA */
+ u32 dma_data;
+ } prot_circular_buffer;
+ } p;
+} abe_port_protocol_t;
+/*
+ * DMA_T
+ *
+ * dma structure for easing programming
+ */
+typedef struct {
+ /* OCP L3 pointer to the first address of the */
+ void *data;
+ /* destination buffer (either DMA or Ping-Pong read/write pointers). */
+ /* address L3 when addressing the DMEM buffer instead of CBPr */
+ void *l3_dmem;
+ /* address L3 translated to L4 the ARM memory space */
+ void *l4_dmem;
+ /* number of iterations for the DMA data moves. */
+ u32 iter;
+} abe_dma_t;
+
+typedef struct {
+ /* Offset to the first address of the */
+ u32 data;
+ /* number of iterations for the DMA data moves. */
+ u32 iter;
+} abe_dma_t_offset;
+/*
+ * SEQ_T
+ *
+ * struct {
+ * micros_t time; Waiting time before executing next line
+ * seq_code_t code Subroutine index interpreted in the HAL
+ * and translated to FW subroutine codes
+ * in case of ABE tasks
+ * int32 param[2] Two parameters
+ * } seq_t
+ *
+ */
+typedef struct {
+ u32 delta_time;
+ u32 code;
+ u32 param[4];
+ u8 tag;
+} abe_seq_t;
+
+typedef struct {
+ u32 mask;
+ abe_seq_t seq1;
+ abe_seq_t seq2;
+} abe_sequence_t;
+/*
+ * DRIFT_T abe_drift_t = s32
+ *
+ * ASRC drift parameter in [ppm] value
+ */
+/*
+ * -------------------- INTERNAL DATA TYPES ---------------------
+ */
+/*
+ * ABE_IRQ_DATA_T
+ *
+ * IRQ FIFO content declaration
+ * APS interrupts : IRQ_FIFO[31:28] = IRQtag_APS,
+ * IRQ_FIFO[27:16] = APS_IRQs, IRQ_FIFO[15:0] = loopCounter
+ * SEQ interrupts : IRQ_FIFO[31:28] IRQtag_COUNT,
+ * IRQ_FIFO[27:16] = Count_IRQs, IRQ_FIFO[15:0] = loopCounter
+ * Ping-Pong Interrupts : IRQ_FIFO[31:28] = IRQtag_PP,
+ * IRQ_FIFO[27:16] = PP_MCU_IRQ, IRQ_FIFO[15:0] = loopCounter
+ */
+typedef struct {
+ unsigned int counter:16;
+ unsigned int data:12;
+ unsigned int tag:4;
+} abe_irq_data_t;
+/*
+ * ABE_PORT_T status / format / sampling / protocol(call_back) /
+ * features / gain / name ..
+ *
+ */
+typedef struct {
+ /* running / idled */
+ u16 status;
+ /* Sample format type */
+ abe_data_format_t format;
+ /* API : for ASRC */
+ s32 drift;
+ /* optionnal call-back index for errors and ack */
+ u16 callback;
+ /* IO tasks buffers */
+ u16 smem_buffer1;
+ u16 smem_buffer2;
+ abe_port_protocol_t protocol;
+ /* pointer and iteration counter of the xDMA */
+ abe_dma_t_offset dma;
+ /* list of features associated to a port (EQ, APS, ... , ends with 0) */
+ u16 feature_index[MAXFEATUREPORT];
+ char name[NBCHARPORTNAME];
+} abe_port_t;
+/*
+ * ABE_SUBROUTINE_T
+ *
+ */
+typedef struct {
+ u32 sub_id;
+ s32 param[4];
+} abe_subroutine_t;
+
+#endif/* ifndef _ABE_TYP_H_ */
diff --git a/sound/soc/omap/abe/abe_typedef.h b/sound/soc/omap/abe/abe_typedef.h
new file mode 100644
index 0000000..59d7221
--- /dev/null
+++ b/sound/soc/omap/abe/abe_typedef.h
@@ -0,0 +1,232 @@
+/*
+
+ This file is provided under a dual BSD/GPLv2 license. When using or
+ redistributing this file, you may do so under either license.
+
+ GPL LICENSE SUMMARY
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of version 2 of the GNU General Public License as
+ published by the Free Software Foundation.
+
+ This program is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA.
+ The full GNU General Public License is included in this distribution
+ in the file called LICENSE.GPL.
+
+ BSD LICENSE
+
+ Copyright(c) 2010-2011 Texas Instruments Incorporated,
+ All rights reserved.
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions
+ are met:
+
+ * Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+ * Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in
+ the documentation and/or other materials provided with the
+ distribution.
+ * Neither the name of Texas Instruments Incorporated nor the names of
+ its contributors may be used to endorse or promote products derived
+ from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+*/
+
+#ifndef _ABE_TYPEDEF_H_
+#define _ABE_TYPEDEF_H_
+
+#include "abe_define.h"
+#include "abe_typ.h"
+
+/*
+ * Basic types definition
+ */
+/*
+ * Commonly used structures
+ */
+typedef struct abetaskTag {
+ /* 0 ... Index of called function */
+ u16 iF;
+ /* 2 ... for INITPTR of A0 */
+ u16 A0;
+ /* 4 ... for INITPTR of A1 */
+ u16 A1;
+ /* 6 ... for INITPTR of A2 & A3 */
+ u16 A2_3;
+ /* 8 ... for INITPTR of A4 & A5 */
+ u16 A4_5;
+ /* 10 ... for INITREG of R0, R1, R2, R3 */
+ u16 R;
+ /* 12 */
+ u16 misc0;
+ /* 14 */
+ u16 misc1;
+} ABE_STask;
+typedef ABE_STask *pABE_STask;
+typedef ABE_STask **ppABE_STask;
+
+struct ABE_SIODescriptor {
+ /* 0 */
+ u16 drift_ASRC;
+ /* 2 */
+ u16 drift_io;
+ /* 4 "Function index" of XLS sheet "Functions" */
+ u8 io_type_idx;
+ /* 5 1 = MONO or Stereo1616, 2= STEREO, ... */
+ u8 samp_size;
+ /* 6 drift "issues" for ASRC */
+ s16 flow_counter;
+ /* 8 address for IRQ or DMArequests */
+ u16 hw_ctrl_addr;
+ /* 10 DMA request bit-field or IRQ (DSP/MCU) */
+ u8 atc_irq_data;
+ /* 11 0 = Read, 3 = Write */
+ u8 direction_rw;
+ /* 12 */
+ u8 repeat_last_samp;
+ /* 13 12 at 48kHz, ... */
+ u8 nsamp;
+ /* 14 nsamp x samp_size */
+ u8 x_io;
+ /* 15 ON = 0x80, OFF = 0x00 */
+ u8 on_off;
+ /* 16 For Slimbus and TDM purpose */
+ u16 split_addr1;
+ /* 18 */
+ u16 split_addr2;
+ /* 20 */
+ u16 split_addr3;
+ /* 22 */
+ u8 before_f_index;
+ /* 23 */
+ u8 after_f_index;
+ /* 24 SM/CM INITPTR field */
+ u16 smem_addr1;
+ /* 26 in bytes */
+ u16 atc_address1;
+ /* 28 DMIC_ATC_PTR, MCPDM_UL_ATC_PTR, ... */
+ u16 atc_pointer_saved1;
+ /* 30 samp_size (except in TDM or Slimbus) */
+ u8 data_size1;
+ /* 31 "Function index" of XLS sheet "Functions" */
+ u8 copy_f_index1;
+ /* 32 For Slimbus and TDM purpose */
+ u16 smem_addr2;
+ /* 34 */
+ u16 atc_address2;
+ /* 36 */
+ u16 atc_pointer_saved2;
+ /* 38 */
+ u8 data_size2;
+ /* 39 */
+ u8 copy_f_index2;
+};
+
+/* [w] asrc output used for the next asrc call (+/- 1 / 0) */
+#define drift_asrc_ 0
+/* [w] asrc output used for controlling the number of samples to be
+ exchanged (+/- 1 / 0) */
+#define drift_io_ 2
+/* address of the IO subroutine */
+#define io_type_idx_ 4
+#define samp_size_ 5
+/* flow error counter */
+#define flow_counter_ 6
+/* dmareq address or host irq buffer address (atc address) */
+#define hw_ctrl_addr_ 8
+/* data content to be loaded to "hw_ctrl_addr" */
+#define atc_irq_data_ 10
+/* read dmem =0, write dmem =3 (atc offset of the access pointer) */
+#define direction_rw_ 11
+/* flag set to allow repeating the last sample on downlink paths */
+#define repeat_last_samp_ 12
+/* number of samples (either mono stereo...) */
+#define nsamp_ 13
+/* x number of raw DMEM data moved */
+#define x_io_ 14
+#define on_off_ 15
+/* internal smem buffer initptr pointer index */
+#define split_addr1_ 16
+/* internal smem buffer initptr pointer index */
+#define split_addr2_ 18
+/* internal smem buffer initptr pointer index */
+#define split_addr3_ 20
+/* index of the copy subroutine */
+#define before_f_index_ 22
+/* index of the copy subroutine */
+#define after_f_index_ 23
+#define minidesc1_ 24
+/* internal smem buffer initptr pointer index */
+#define rel_smem_ 0
+/* atc descriptor address (byte address x4) */
+#define rel_atc_ 2
+/* location of the saved ATC pointer (+debug info) */
+#define rel_atc_saved 4
+/* size of each sample (1:mono/1616 2:stereo ... ) */
+#define rel_size_ 6
+/* index of the copy subroutine */
+#define rel_f_ 7
+#define s_mem_mm_ul 24
+#define s_mm_ul_size 30
+#define minidesc2_ 32
+#define Struct_Size 40
+
+struct ABE_SPingPongDescriptor {
+ /* 0: [W] asrc output used for the next ASRC call (+/- 1 / 0) */
+ u16 drift_ASRC;
+ /* 2: [W] asrc output used for controlling the number of
+ samples to be exchanged (+/- 1 / 0) */
+ u16 drift_io;
+ /* 4: DMAReq address or HOST IRQ buffer address (ATC ADDRESS) */
+ u16 hw_ctrl_addr;
+ /* 6: index of the copy subroutine */
+ u8 copy_func_index;
+ /* 7: X number of SMEM samples to move */
+ u8 x_io;
+ /* 8: 0 for mono data, 1 for stereo data */
+ u8 data_size;
+ /* 9: internal SMEM buffer INITPTR pointer index */
+ u8 smem_addr;
+ /* 10: data content to be loaded to "hw_ctrl_addr" */
+ u8 atc_irq_data;
+ /* 11: ping/pong buffer flag */
+ u8 counter;
+ /* 12: current Base address of the working buffer */
+ u16 workbuff_BaseAddr;
+ /* 14: samples left in the working buffer */
+ u16 workbuff_Samples;
+ /* 16: Base address of the ping/pong buffer 0 */
+ u16 nextbuff0_BaseAddr;
+ /* 18: samples available in the ping/pong buffer 0 */
+ u16 nextbuff0_Samples;
+ /* 20: Base address of the ping/pong buffer 1 */
+ u16 nextbuff1_BaseAddr;
+ /* 22: samples available in the ping/pong buffer 1 */
+ u16 nextbuff1_Samples;
+};
+
+#endif/* _ABE_TYPEDEF_H_ */
diff --git a/sound/soc/omap/abe/port_mgr.c b/sound/soc/omap/abe/port_mgr.c
new file mode 100644
index 0000000..aef5cee
--- /dev/null
+++ b/sound/soc/omap/abe/port_mgr.c
@@ -0,0 +1,341 @@
+/*
+ * ALSA SoC OMAP ABE port manager
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+//#define DEBUG
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/spinlock.h>
+#include <linux/list.h>
+#include <linux/debugfs.h>
+#include <linux/device.h>
+#include "port_mgr.h"
+#include "abe_main.h"
+
+/* this must match logical ID numbers in port_mgr.h */
+static const char *lport_name[] = {
+ "dmic0", "dmic1", "dmic2", "pdmdl1", "pdmdl2", "pdmvib",
+ "pdmul1", "bt_vx_dl", "bt_vx_ul", "mm_ext_ul", "mm_ext_dl",
+ "mm_dl1", "mm_ul1", "mm_ul2", "vx_dl", "vx_ul", "vib", "tones"
+};
+
+static DEFINE_MUTEX(port_mgr_mutex);
+static struct abe *the_abe = NULL;
+static int users = 0;
+
+/*
+ * Get the Physical port ID based on the logical port ID
+ *
+ * FE and BE ports have unique ID's within the driver but share
+ * ID's within the ABE. This maps a driver port ID to an ABE port ID.
+ */
+static int get_physical_id(int logical_id)
+{
+ switch (logical_id) {
+ /* backend ports */
+ case OMAP_ABE_BE_PORT_DMIC0:
+ case OMAP_ABE_BE_PORT_DMIC1:
+ case OMAP_ABE_BE_PORT_DMIC2:
+ return DMIC_PORT;
+ case OMAP_ABE_BE_PORT_PDM_DL1:
+ case OMAP_ABE_BE_PORT_PDM_DL2:
+ return PDM_DL_PORT;
+ case OMAP_ABE_BE_PORT_PDM_VIB:
+ return VIB_DL_PORT;
+ case OMAP_ABE_BE_PORT_PDM_UL1:
+ return PDM_UL_PORT;
+ case OMAP_ABE_BE_PORT_BT_VX_DL:
+ return BT_VX_DL_PORT;
+ case OMAP_ABE_BE_PORT_BT_VX_UL:
+ return BT_VX_UL_PORT;
+ case OMAP_ABE_BE_PORT_MM_EXT_UL:
+ return MM_EXT_OUT_PORT;
+ case OMAP_ABE_BE_PORT_MM_EXT_DL:
+ return MM_EXT_IN_PORT;
+ /* front end ports */
+ case OMAP_ABE_FE_PORT_MM_DL1:
+ return MM_DL_PORT;
+ case OMAP_ABE_FE_PORT_MM_UL1:
+ return MM_UL_PORT;
+ case OMAP_ABE_FE_PORT_MM_UL2:
+ return MM_UL2_PORT;
+ case OMAP_ABE_FE_PORT_VX_DL:
+ return VX_DL_PORT;
+ case OMAP_ABE_FE_PORT_VX_UL:
+ return VX_UL_PORT;
+ case OMAP_ABE_FE_PORT_VIB:
+ return VIB_DL_PORT;
+ case OMAP_ABE_FE_PORT_TONES:
+ return TONES_DL_PORT;
+ }
+ return -EINVAL;
+}
+
+/*
+ * Get the number of enabled users of the physical port shared by this client.
+ * Locks held by callers.
+ */
+static int port_get_num_users(struct abe *abe, struct omap_abe_port *port)
+{
+ struct omap_abe_port *p;
+ int users = 0;
+
+ list_for_each_entry(p, &abe->ports, list) {
+ if (p->physical_id == port->physical_id && p->state == PORT_ENABLED)
+ users++;
+ }
+ return users;
+}
+
+static int port_is_open(struct abe *abe, int phy_port)
+{
+ struct omap_abe_port *p;
+
+ list_for_each_entry(p, &abe->ports, list) {
+ if (p->physical_id == phy_port && p->state == PORT_ENABLED)
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * Check whether the physical port is enabled for this PHY port ID.
+ * Locks held by callers.
+ */
+int omap_abe_port_is_enabled(struct abe *abe, struct omap_abe_port *port)
+{
+ struct omap_abe_port *p;
+ unsigned long flags;
+
+ spin_lock_irqsave(&abe->lock, flags);
+
+ list_for_each_entry(p, &abe->ports, list) {
+ if (p->physical_id == port->physical_id && p->state == PORT_ENABLED) {
+ spin_unlock_irqrestore(&abe->lock, flags);
+ return 1;
+ }
+ }
+
+ spin_unlock_irqrestore(&abe->lock, flags);
+ return 0;
+}
+EXPORT_SYMBOL(omap_abe_port_is_enabled);
+
+/*
+ * omap_abe_port_enable - enable ABE logical port
+ *
+ * @abe - ABE.
+ * @port - logical ABE port ID to be enabled.
+ */
+int omap_abe_port_enable(struct abe *abe, struct omap_abe_port *port)
+{
+ int ret = 0;
+ unsigned long flags;
+
+ /* only enable the physical port iff it is disabled */
+ pr_debug("port %s increment count %d\n",
+ lport_name[port->logical_id], port->users);
+
+ spin_lock_irqsave(&abe->lock, flags);
+ if (port->users == 0 && port_get_num_users(abe, port) == 0) {
+
+ /* enable the physical port */
+ pr_debug("port %s phy port %d enabled\n",
+ lport_name[port->logical_id], port->physical_id);
+ abe_enable_data_transfer(port->physical_id);
+ }
+
+ port->state = PORT_ENABLED;
+ port->users++;
+ spin_unlock_irqrestore(&abe->lock, flags);
+ return ret;
+}
+EXPORT_SYMBOL(omap_abe_port_enable);
+
+/*
+ * omap_abe_port_disable - disable ABE logical port
+ *
+ * @abe - ABE.
+ * @port - logical ABE port ID to be disabled.
+ */
+int omap_abe_port_disable(struct abe *abe, struct omap_abe_port *port)
+{
+ int ret = 0;
+ unsigned long flags;
+
+ /* only disable the port iff no other users are using it */
+ pr_debug("port %s decrement count %d\n",
+ lport_name[port->logical_id], port->users);
+
+ spin_lock_irqsave(&abe->lock, flags);
+ if (port->users == 1 && port_get_num_users(abe, port) == 1) {
+ /* disable the physical port */
+ pr_debug("port %s phy port %d disabled\n",
+ lport_name[port->logical_id], port->physical_id);
+
+ abe_disable_data_transfer(port->physical_id);
+ }
+
+ port->state = PORT_DISABLED;
+ port->users--;
+ spin_unlock_irqrestore(&abe->lock, flags);
+ return ret;
+}
+EXPORT_SYMBOL(omap_abe_port_disable);
+
+/*
+ * omap_abe_port_open - open ABE logical port
+ *
+ * @abe - ABE.
+ * @logical_id - logical ABE port ID to be opened.
+ */
+struct omap_abe_port *omap_abe_port_open(struct abe *abe, int logical_id)
+{
+ struct omap_abe_port *port;
+ unsigned long flags;
+
+#ifdef CONFIG_DEBUG_FS
+ char debug_fs_name[32];
+#endif
+
+ if (logical_id < 0 || logical_id > OMAP_ABE_MAX_PORT_ID)
+ return NULL;
+
+ if (port_is_open(abe, logical_id))
+ return NULL;
+
+ port = kzalloc(sizeof(struct omap_abe_port), GFP_KERNEL);
+ if (port == NULL)
+ return NULL;
+
+ port->logical_id = logical_id;
+ port->physical_id = get_physical_id(logical_id);
+ port->state = PORT_DISABLED;
+ port->abe = abe;
+
+ spin_lock_irqsave(&abe->lock, flags);
+ list_add(&port->list, &abe->ports);
+ spin_unlock_irqrestore(&abe->lock, flags);
+ port->physical_users = port_get_num_users(abe, port);
+
+#ifdef CONFIG_DEBUG_FS
+ sprintf(debug_fs_name, "%s_state", lport_name[logical_id]);
+ port->debugfs_lstate = debugfs_create_u32(debug_fs_name, 0644,
+ abe->debugfs_root, &port->state);
+ sprintf(debug_fs_name, "%s_phy", lport_name[logical_id]);
+ port->debugfs_lphy = debugfs_create_u32(debug_fs_name, 0644,
+ abe->debugfs_root, &port->physical_id);
+ sprintf(debug_fs_name, "%s_users", lport_name[logical_id]);
+ port->debugfs_lusers = debugfs_create_u32(debug_fs_name, 0644,
+ abe->debugfs_root, &port->users);
+#endif
+
+ pr_debug("opened port %s\n", lport_name[logical_id]);
+ return port;
+}
+EXPORT_SYMBOL(omap_abe_port_open);
+
+/*
+ * omap_abe_port_close - close ABE logical port
+ *
+ * @port - logical ABE port to be closed (and disabled).
+ */
+void omap_abe_port_close(struct abe *abe, struct omap_abe_port *port)
+{
+ unsigned long flags;
+
+ /* disable the port */
+ omap_abe_port_disable(abe, port);
+
+ spin_lock_irqsave(&abe->lock, flags);
+ list_del(&port->list);
+ spin_unlock_irqrestore(&abe->lock, flags);
+
+ pr_debug("closed port %s\n", lport_name[port->logical_id]);
+ kfree(port);
+}
+EXPORT_SYMBOL(omap_abe_port_close);
+
+static struct abe *omap_abe_port_mgr_init(void)
+{
+ struct abe *abe;
+
+ abe = kzalloc(sizeof(struct abe), GFP_KERNEL);
+ if (abe == NULL)
+ return NULL;
+
+ spin_lock_init(&abe->lock);
+
+ INIT_LIST_HEAD(&abe->ports);
+ the_abe = abe;
+
+#ifdef CONFIG_DEBUG_FS
+ abe->debugfs_root = debugfs_create_dir("abe_port", NULL);
+ if (!abe->debugfs_root) {
+ pr_debug( "Failed to create port manager debugfs directory\n");
+ }
+#endif
+ return abe;
+}
+
+static void omap_abe_port_mgr_free(struct abe *abe)
+{
+#ifdef CONFIG_DEBUG_FS
+ debugfs_remove_recursive(abe->debugfs_root);
+#endif
+ kfree(abe);
+ the_abe = NULL;
+}
+
+struct abe *omap_abe_port_mgr_get(void)
+{
+ struct abe * abe;
+
+ mutex_lock(&port_mgr_mutex);
+
+ if (the_abe)
+ abe = the_abe;
+ else
+ abe = omap_abe_port_mgr_init();
+
+ users++;
+ mutex_unlock(&port_mgr_mutex);
+ return abe;
+}
+EXPORT_SYMBOL(omap_abe_port_mgr_get);
+
+void omap_abe_port_mgr_put(struct abe *abe)
+{
+ mutex_lock(&port_mgr_mutex);
+
+ if (users == 0)
+ goto out;
+
+ if (--users == 0)
+ omap_abe_port_mgr_free(abe);
+
+out:
+ mutex_unlock(&port_mgr_mutex);
+}
+EXPORT_SYMBOL(omap_abe_port_mgr_put);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/abe/port_mgr.h b/sound/soc/omap/abe/port_mgr.h
new file mode 100644
index 0000000..a65b0d3
--- /dev/null
+++ b/sound/soc/omap/abe/port_mgr.h
@@ -0,0 +1,98 @@
+/*
+ * ABE Port manager
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_OMAP_PORT_MGR_H
+#define __LINUX_SND_SOC_OMAP_PORT_MGR_H
+
+#include <linux/debugfs.h>
+
+/*
+ * TODO: These structures, enums and port ID macros should be moved to the
+ * new public ABE API header.
+ */
+
+/* Logical PORT IDs - Backend */
+#define OMAP_ABE_BE_PORT_DMIC0 0
+#define OMAP_ABE_BE_PORT_DMIC1 1
+#define OMAP_ABE_BE_PORT_DMIC2 2
+#define OMAP_ABE_BE_PORT_PDM_DL1 3
+#define OMAP_ABE_BE_PORT_PDM_DL2 4
+#define OMAP_ABE_BE_PORT_PDM_VIB 5
+#define OMAP_ABE_BE_PORT_PDM_UL1 6
+#define OMAP_ABE_BE_PORT_BT_VX_DL 7
+#define OMAP_ABE_BE_PORT_BT_VX_UL 8
+#define OMAP_ABE_BE_PORT_MM_EXT_UL 9
+#define OMAP_ABE_BE_PORT_MM_EXT_DL 10
+
+/* Logical PORT IDs - Frontend */
+#define OMAP_ABE_FE_PORT_MM_DL1 11
+#define OMAP_ABE_FE_PORT_MM_UL1 12
+#define OMAP_ABE_FE_PORT_MM_UL2 13
+#define OMAP_ABE_FE_PORT_VX_DL 14
+#define OMAP_ABE_FE_PORT_VX_UL 15
+#define OMAP_ABE_FE_PORT_VIB 16
+#define OMAP_ABE_FE_PORT_TONES 17
+
+#define OMAP_ABE_MAX_PORT_ID OMAP_ABE_FE_PORT_TONES
+
+/* ports can either be enabled or disabled */
+enum port_state {
+ PORT_DISABLED = 0,
+ PORT_ENABLED,
+};
+
+/* structure used for client port info */
+struct omap_abe_port {
+
+ /* logical and physical port IDs that correspond this port */
+ int logical_id;
+ int physical_id;
+ int physical_users;
+
+ /* enabled or disabled */
+ enum port_state state;
+
+ /* logical port ref count */
+ int users;
+
+ struct list_head list;
+ struct abe *abe;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_lstate;
+ struct dentry *debugfs_lphy;
+ struct dentry *debugfs_lusers;
+#endif
+};
+
+/* main ABE structure */
+struct abe {
+
+ /* List of open ABE logical ports */
+ struct list_head ports;
+
+ /* spinlock */
+ spinlock_t lock;
+
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_root;
+#endif
+};
+
+struct omap_abe_port *omap_abe_port_open(struct abe *abe, int logical_id);
+void omap_abe_port_close(struct abe *abe, struct omap_abe_port *port);
+int omap_abe_port_enable(struct abe *abe, struct omap_abe_port *port);
+int omap_abe_port_disable(struct abe *abe, struct omap_abe_port *port);
+int omap_abe_port_is_enabled(struct abe *abe, struct omap_abe_port *port);
+struct abe *omap_abe_port_mgr_get(void);
+void omap_abe_port_mgr_put(struct abe *abe);
+
+#endif /* __LINUX_SND_SOC_OMAP_PORT_MGR_H */
diff --git a/sound/soc/omap/omap-abe-dsp.c b/sound/soc/omap/omap-abe-dsp.c
new file mode 100644
index 0000000..13ae8e9
--- /dev/null
+++ b/sound/soc/omap/omap-abe-dsp.c
@@ -0,0 +1,2443 @@
+/*
+ * omap-abe-dsp.c
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ * Copyright (C) 2010 Texas Instruments Inc.
+ *
+ * Authors: Liam Girdwood <lrg@ti.com>
+ * Misael Lopez Cruz <misael.lopez@ti.com>
+ * Sebastien Guiriec <s-guiriec@ti.com>
+ *
+ */
+
+#define DEBUG
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/platform_device.h>
+#include <linux/workqueue.h>
+#include <linux/i2c/twl.h>
+#include <linux/clk.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+#include <linux/pm_runtime.h>
+#include <linux/dma-mapping.h>
+#include <linux/wait.h>
+#include <linux/firmware.h>
+#include <linux/debugfs.h>
+
+#include <plat/omap_hwmod.h>
+#include <plat/omap_device.h>
+#include <plat/dma.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/omap-abe-dsp.h>
+
+#include "omap-abe-dsp.h"
+#include "omap-abe.h"
+#include "abe/abe_main.h"
+#include "abe/port_mgr.h"
+
+#warning need omap_device_set_rate
+#define omap_device_set_rate(x, y, z)
+
+static const char *abe_memory_bank[5] = {
+ "dmem",
+ "cmem",
+ "smem",
+ "pmem",
+ "mpu"
+};
+
+
+/*
+ * ABE loadable coefficients.
+ * The coefficient and their mixer configurations are loaded with the firmware
+ * blob duing probe().
+ */
+
+struct coeff_config {
+ char name[ABE_COEFF_NAME_SIZE];
+ u32 count;
+ u32 coeff;
+ char texts[ABE_COEFF_NUM_TEXTS][ABE_COEFF_TEXT_SIZE];
+};
+
+/*
+ * ABE Firmware Header.
+ * The ABE firmware blob has a header that describes each data section. This
+ * way we can store coefficients etc in the firmware.
+ */
+struct fw_header {
+ u32 magic; /* magic number */
+ u32 crc; /* optional crc */
+ u32 firmware_size; /* payload size */
+ u32 coeff_size; /* payload size */
+ u32 coeff_version; /* coefficent version */
+ u32 firmware_version; /* min version of ABE firmware required */
+ u32 num_equ; /* number of equalizers */
+};
+
+/*
+ * ABE private data.
+ */
+struct abe_data {
+ struct omap4_abe_dsp_pdata *abe_pdata;
+ struct device *dev;
+ struct snd_soc_platform *platform;
+ struct delayed_work delayed_work;
+ struct mutex mutex;
+ struct mutex opp_mutex;
+ struct clk *clk;
+ void __iomem *io_base[5];
+ int irq;
+ int opp;
+ int active;
+
+ /* coefficients */
+ struct fw_header hdr;
+ u32 *firmware;
+ s32 *equ[ABE_MAX_EQU];
+ int equ_profile[ABE_MAX_EQU];
+ struct soc_enum equalizer_enum[ABE_MAX_EQU];
+ struct snd_kcontrol_new equalizer_control[ABE_MAX_EQU];
+ struct coeff_config *equ_texts;
+
+ /* DAPM mixer config - TODO: some of this can be replaced with HAL update */
+ u32 widget_opp[ABE_NUM_DAPM_REG + 1];
+
+ u16 router[16];
+ int loss_count;
+
+ struct snd_pcm_substream *ping_pong_substream;
+ int first_irq;
+
+ struct snd_pcm_substream *psubs;
+
+#ifdef CONFIG_DEBUG_FS
+ /* ABE runtime debug config */
+
+ /* its intended we can switch on/off individual debug items */
+ u32 dbg_format1; /* TODO: match flag names here to debug format flags */
+ u32 dbg_format2;
+ u32 dbg_format3;
+
+ u32 dbg_buffer_bytes;
+ u32 dbg_circular;
+ u32 dbg_buffer_msecs; /* size of buffer in secs */
+ u32 dbg_elem_bytes;
+ dma_addr_t dbg_buffer_addr;
+ wait_queue_head_t wait;
+ int dbg_reader_offset;
+ int dbg_dma_offset;
+ int dbg_complete;
+ struct dentry *debugfs_root;
+ struct dentry *debugfs_fmt1;
+ struct dentry *debugfs_fmt2;
+ struct dentry *debugfs_fmt3;
+ struct dentry *debugfs_size;
+ struct dentry *debugfs_data;
+ struct dentry *debugfs_circ;
+ struct dentry *debugfs_elem_bytes;
+ struct dentry *debugfs_opp_level;
+ char *dbg_buffer;
+ struct omap_pcm_dma_data *dma_data;
+ int dma_ch;
+ int dma_req;
+#endif
+};
+
+static struct abe_data *the_abe;
+
+// TODO: map to the new version of HAL
+static unsigned int abe_dsp_read(struct snd_soc_platform *platform,
+ unsigned int reg)
+{
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+
+ BUG_ON(reg > ABE_NUM_DAPM_REG);
+ return abe->widget_opp[reg];
+}
+
+static int abe_dsp_write(struct snd_soc_platform *platform, unsigned int reg,
+ unsigned int val)
+{
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+
+ BUG_ON(reg > ABE_NUM_DAPM_REG);
+ abe->widget_opp[reg] = val;
+ return 0;
+}
+
+static void abe_irq_pingpong_subroutine(u32 *data)
+{
+ u32 dst, n_bytes;
+
+ abe_read_next_ping_pong_buffer(MM_DL_PORT, &dst, &n_bytes);
+ abe_set_ping_pong_buffer(MM_DL_PORT, n_bytes);
+
+ /* Do not call ALSA function for first IRQ */
+ if (the_abe->first_irq) {
+ the_abe->first_irq = 0;
+ } else {
+ if (the_abe->ping_pong_substream)
+ snd_pcm_period_elapsed(the_abe->ping_pong_substream);
+ }
+}
+
+static irqreturn_t abe_irq_handler(int irq, void *dev_id)
+{
+ struct abe_data *abe = dev_id;
+
+ /* TODO: handle underruns/overruns/errors */
+ pm_runtime_get_sync(abe->dev);
+ abe_clear_irq(); // TODO: why is IRQ not cleared after processing ?
+ abe_irq_processing();
+ pm_runtime_put_sync(abe->dev);
+ return IRQ_HANDLED;
+}
+
+// TODO: these should really be called internally since we will know the McPDM state
+void abe_dsp_pm_get(void)
+{
+ pm_runtime_get_sync(the_abe->dev);
+}
+EXPORT_SYMBOL_GPL(abe_dsp_pm_get);
+
+void abe_dsp_pm_put(void)
+{
+ pm_runtime_put_sync(the_abe->dev);
+}
+EXPORT_SYMBOL_GPL(abe_dsp_pm_put);
+
+void abe_dsp_shutdown(void)
+{
+ if (!the_abe->active && !abe_check_activity()) {
+ abe_set_opp_processing(ABE_OPP25);
+ the_abe->opp = 25;
+ abe_stop_event_generator();
+ udelay(250);
+ omap_device_set_rate(the_abe->dev, the_abe->dev, 0);
+ }
+}
+EXPORT_SYMBOL_GPL(abe_dsp_shutdown);
+
+/*
+ * These TLV settings will need fine tuned for each individual control
+ */
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(mm_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(tones_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(voice_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(capture_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(mm_dl2_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(tones_dl2_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(voice_dl2_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(capture_dl2_tlv, -12000, 100, 3000);
+
+/* SDT volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(sdt_ul_tlv, -12000, 100, 3000);
+
+/* SDT volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(sdt_dl_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_mm_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_tones_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_vx_ul_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_vx_dl_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_mm_dl_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_tones_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_vx_dl_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_vx_ul_tlv, -12000, 100, 3000);
+
+/* DMIC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(dmic_tlv, -12000, 100, 3000);
+
+/* BT UL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(btul_tlv, -12000, 100, 3000);
+
+/* AMIC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(amic_tlv, -12000, 100, 3000);
+
+//TODO: we have to use the shift value atm to represent register id due to current HAL
+static int dl1_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ // TODO: optimise all of these to call HAL abe_enable_gain(mixer, enable)
+ if (ucontrol->value.integer.value[0]) {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ abe_enable_gain(MIXDL1, mc->reg);
+ } else {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ abe_disable_gain(MIXDL1, mc->reg);
+ }
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int dl2_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ if (ucontrol->value.integer.value[0]) {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ abe_enable_gain(MIXDL2, mc->reg);
+ } else {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ abe_disable_gain(MIXDL2, mc->reg);
+ }
+
+ pm_runtime_put_sync(the_abe->dev);
+ return 1;
+}
+
+static int audio_ul_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ if (ucontrol->value.integer.value[0]) {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ abe_enable_gain(MIXAUDUL, mc->reg);
+ } else {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ abe_disable_gain(MIXAUDUL, mc->reg);
+ }
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int vxrec_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ if (ucontrol->value.integer.value[0]) {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ abe_enable_gain(MIXVXREC, mc->reg);
+ } else {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ abe_disable_gain(MIXVXREC, mc->reg);
+ }
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int sdt_put_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ if (ucontrol->value.integer.value[0]) {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ abe_enable_gain(MIXSDT, mc->reg);
+ } else {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ abe_disable_gain(MIXSDT, mc->reg);
+ }
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int abe_get_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = the_abe->widget_opp[mc->shift];
+ return 0;
+}
+
+/* router IDs that match our mixer strings */
+static const abe_router_t router[] = {
+ ZERO_labelID, /* strangely this is not 0 */
+ DMIC1_L_labelID, DMIC1_R_labelID,
+ DMIC2_L_labelID, DMIC2_R_labelID,
+ DMIC3_L_labelID, DMIC3_R_labelID,
+ BT_UL_L_labelID, BT_UL_R_labelID,
+ MM_EXT_IN_L_labelID, MM_EXT_IN_R_labelID,
+ AMIC_L_labelID, AMIC_R_labelID,
+ VX_REC_L_labelID, VX_REC_R_labelID,
+};
+
+static int ul_mux_put_route(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ int mux = ucontrol->value.enumerated.item[0];
+ int reg = e->reg - ABE_MUX(0);
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ if (mux > ABE_ROUTES_UL)
+ return 0;
+
+ // TODO: get all this via firmware
+ if (reg < 8) {
+ /* 0 .. 9 = MM_UL */
+ the_abe->router[reg] = router[mux];
+ } else if (reg < 12) {
+ /* 10 .. 11 = MM_UL2 */
+ /* 12 .. 13 = VX_UL */
+ the_abe->router[reg + 2] = router[mux];
+ }
+
+ /* 2nd arg here is unused */
+ abe_set_router_configuration(UPROUTE, 0, (u32 *)the_abe->router);
+
+ if (router[mux] != ZERO_labelID)
+ the_abe->widget_opp[e->reg] = e->shift_l;
+ else
+ the_abe->widget_opp[e->reg] = 0;
+
+ snd_soc_dapm_mux_update_power(widget, kcontrol, 1, mux, e);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int ul_mux_get_route(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_enum *e =
+ (struct soc_enum *)kcontrol->private_value;
+ int reg = e->reg - ABE_MUX(0), i, rval = 0;
+
+ // TODO: get all this via firmware
+ if (reg < 8) {
+ /* 0 .. 9 = MM_UL */
+ rval = the_abe->router[reg];
+ } else if (reg < 12) {
+ /* 10 .. 11 = MM_UL2 */
+ /* 12 .. 13 = VX_UL */
+ rval = the_abe->router[reg + 2];
+ }
+
+ for (i = 0; i < ARRAY_SIZE(router); i++) {
+ if (router[i] == rval) {
+ ucontrol->value.integer.value[0] = i;
+ return 0;
+ }
+ }
+
+ return 1;
+}
+
+
+static int abe_put_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ if (ucontrol->value.integer.value[0]) {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+ } else {
+ the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+ snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+ }
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+
+static int volume_put_sdt_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+
+ abe_write_mixer(MIXSDT, abe_val_to_gain(ucontrol->value.integer.value[0]),
+ RAMP_0MS, mc->reg);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int volume_put_audul_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_write_mixer(MIXAUDUL, abe_val_to_gain(ucontrol->value.integer.value[0]),
+ RAMP_0MS, mc->reg);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int volume_put_vxrec_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_write_mixer(MIXVXREC, abe_val_to_gain(ucontrol->value.integer.value[0]),
+ RAMP_0MS, mc->reg);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int volume_put_dl1_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_write_mixer(MIXDL1, abe_val_to_gain(ucontrol->value.integer.value[0]),
+ RAMP_0MS, mc->reg);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int volume_put_dl2_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_write_mixer(MIXDL2, abe_val_to_gain(ucontrol->value.integer.value[0]),
+ RAMP_0MS, mc->reg);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int volume_put_gain(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_write_gain(mc->reg,
+ abe_val_to_gain(ucontrol->value.integer.value[0]),
+ RAMP_20MS, mc->shift);
+ abe_write_gain(mc->reg,
+ -12000 + (ucontrol->value.integer.value[1] * 100),
+ RAMP_20MS, mc->rshift);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+static int volume_get_dl1_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_read_mixer(MIXDL1, &val, mc->reg);
+ ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 0;
+}
+
+static int volume_get_dl2_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_read_mixer(MIXDL2, &val, mc->reg);
+ ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 0;
+}
+
+static int volume_get_audul_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_read_mixer(MIXAUDUL, &val, mc->reg);
+ ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 0;
+}
+
+static int volume_get_vxrec_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_read_mixer(MIXVXREC, &val, mc->reg);
+ ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 0;
+}
+
+static int volume_get_sdt_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_read_mixer(MIXSDT, &val, mc->reg);
+ ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 0;
+}
+
+static int volume_get_gain(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ u32 val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_read_gain(mc->reg, &val, mc->shift);
+ ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+ abe_read_gain(mc->reg, &val, mc->rshift);
+ ucontrol->value.integer.value[1] = abe_gain_to_val(val);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 0;
+}
+
+static int abe_get_equalizer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_enum *eqc = (struct soc_enum *)kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = the_abe->equ_profile[eqc->reg];
+ return 0;
+}
+
+static int abe_put_equalizer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct soc_enum *eqc = (struct soc_enum *)kcontrol->private_value;
+ u16 val = ucontrol->value.enumerated.item[0];
+ abe_equ_t equ_params;
+ int len;
+
+ if (eqc->reg >= the_abe->hdr.num_equ)
+ return -EINVAL;
+
+ if (val >= the_abe->equ_texts[eqc->reg].count)
+ return -EINVAL;
+
+ len = the_abe->equ_texts[eqc->reg].coeff;
+ equ_params.equ_length = len;
+ memcpy(equ_params.coef.type1, the_abe->equ[eqc->reg] + val * len,
+ len * sizeof(u32));
+ the_abe->equ_profile[eqc->reg] = val;
+
+ pm_runtime_get_sync(the_abe->dev);
+ abe_write_equalizer(eqc->reg + 1, &equ_params);
+ pm_runtime_put_sync(the_abe->dev);
+
+ return 1;
+}
+
+int snd_soc_info_enum_ext1(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = e->max;
+
+ if (uinfo->value.enumerated.item > e->max - 1)
+ uinfo->value.enumerated.item = e->max - 1;
+ strcpy(uinfo->value.enumerated.name,
+ snd_soc_get_enum_text(e, uinfo->value.enumerated.item));
+
+ return 0;
+}
+
+static const char *route_ul_texts[] = {
+ "None", "DMic0L", "DMic0R", "DMic1L", "DMic1R", "DMic2L", "DMic2R",
+ "BT Left", "BT Right", "MMExt Left", "MMExt Right", "AMic0", "AMic1",
+ "VX Left", "VX Right"
+};
+
+/* ROUTE_UL Mux table */
+static const struct soc_enum abe_enum[] = {
+ SOC_ENUM_SINGLE(MUX_MM_UL10, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL11, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL12, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL13, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL14, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL15, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL16, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL17, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL20, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_MM_UL21, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_VX_UL0, 0, 15, route_ul_texts),
+ SOC_ENUM_SINGLE(MUX_VX_UL1, 0, 15, route_ul_texts),
+};
+
+static const struct snd_kcontrol_new mm_ul00_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[0],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul01_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[1],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul02_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[2],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul03_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[3],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul04_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[4],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul05_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[5],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul06_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[6],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul07_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[7],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul10_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[8],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul11_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[9],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_vx0_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[10],
+ ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_vx1_control =
+ SOC_DAPM_ENUM_EXT("Route", abe_enum[11],
+ ul_mux_get_route, ul_mux_put_route);
+
+/* DL1 mixer paths */
+static const struct snd_kcontrol_new dl1_mixer_controls[] = {
+ SOC_SINGLE_EXT("Tones", MIX_DL1_INPUT_TONES, MIX_DL1_TONES, 1, 0,
+ abe_get_mixer, dl1_put_mixer),
+ SOC_SINGLE_EXT("Voice", MIX_DL1_INPUT_VX_DL, MIX_DL1_VOICE, 1, 0,
+ abe_get_mixer, dl1_put_mixer),
+ SOC_SINGLE_EXT("Capture", MIX_DL1_INPUT_MM_UL2, MIX_DL1_CAPTURE, 1, 0,
+ abe_get_mixer, dl1_put_mixer),
+ SOC_SINGLE_EXT("Multimedia", MIX_DL1_INPUT_MM_DL, MIX_DL1_MEDIA, 1, 0,
+ abe_get_mixer, dl1_put_mixer),
+};
+
+/* DL2 mixer paths */
+static const struct snd_kcontrol_new dl2_mixer_controls[] = {
+ SOC_SINGLE_EXT("Tones", MIX_DL2_INPUT_TONES, MIX_DL2_TONES, 1, 0,
+ abe_get_mixer, dl2_put_mixer),
+ SOC_SINGLE_EXT("Voice", MIX_DL2_INPUT_VX_DL, MIX_DL2_VOICE, 1, 0,
+ abe_get_mixer, dl2_put_mixer),
+ SOC_SINGLE_EXT("Capture", MIX_DL2_INPUT_MM_UL2, MIX_DL2_CAPTURE, 1, 0,
+ abe_get_mixer, dl2_put_mixer),
+ SOC_SINGLE_EXT("Multimedia", MIX_DL2_INPUT_MM_DL, MIX_DL2_MEDIA, 1, 0,
+ abe_get_mixer, dl2_put_mixer),
+};
+
+/* AUDUL ("Voice Capture Mixer") mixer paths */
+static const struct snd_kcontrol_new audio_ul_mixer_controls[] = {
+ SOC_SINGLE_EXT("Tones Playback", MIX_AUDUL_INPUT_TONES, MIX_AUDUL_TONES, 1, 0,
+ abe_get_mixer, audio_ul_put_mixer),
+ SOC_SINGLE_EXT("Media Playback", MIX_AUDUL_INPUT_MM_DL, MIX_AUDUL_MEDIA, 1, 0,
+ abe_get_mixer, audio_ul_put_mixer),
+ SOC_SINGLE_EXT("Capture", MIX_AUDUL_INPUT_UPLINK, MIX_AUDUL_CAPTURE, 1, 0,
+ abe_get_mixer, audio_ul_put_mixer),
+};
+
+/* VXREC ("Capture Mixer") mixer paths */
+static const struct snd_kcontrol_new vx_rec_mixer_controls[] = {
+ SOC_SINGLE_EXT("Tones", MIX_VXREC_INPUT_TONES, MIX_VXREC_TONES, 1, 0,
+ abe_get_mixer, vxrec_put_mixer),
+ SOC_SINGLE_EXT("Voice Playback", MIX_VXREC_INPUT_VX_DL,
+ MIX_VXREC_VOICE_PLAYBACK, 1, 0, abe_get_mixer, vxrec_put_mixer),
+ SOC_SINGLE_EXT("Voice Capture", MIX_VXREC_INPUT_VX_UL,
+ MIX_VXREC_VOICE_CAPTURE, 1, 0, abe_get_mixer, vxrec_put_mixer),
+ SOC_SINGLE_EXT("Media Playback", MIX_VXREC_INPUT_MM_DL,
+ MIX_VXREC_MEDIA, 1, 0, abe_get_mixer, vxrec_put_mixer),
+};
+
+/* SDT ("Sidetone Mixer") mixer paths */
+static const struct snd_kcontrol_new sdt_mixer_controls[] = {
+ SOC_SINGLE_EXT("Capture", MIX_SDT_INPUT_UP_MIXER, MIX_SDT_CAPTURE, 1, 0,
+ abe_get_mixer, sdt_put_mixer),
+ SOC_SINGLE_EXT("Playback", MIX_SDT_INPUT_DL1_MIXER, MIX_SDT_PLAYBACK, 1, 0,
+ abe_get_mixer, sdt_put_mixer),
+};
+
+/* Virtual PDM_DL Switch */
+static const struct snd_kcontrol_new pdm_dl1_switch_controls =
+ SOC_SINGLE_EXT("Switch", ABE_VIRTUAL_SWITCH, MIX_SWITCH_PDM_DL, 1, 0,
+ abe_get_mixer, abe_put_switch);
+
+/* Virtual BT_VX_DL Switch */
+static const struct snd_kcontrol_new bt_vx_dl_switch_controls =
+ SOC_SINGLE_EXT("Switch", ABE_VIRTUAL_SWITCH, MIX_SWITCH_BT_VX_DL, 1, 0,
+ abe_get_mixer, abe_put_switch);
+
+/* Virtual MM_EXT_DL Switch */
+static const struct snd_kcontrol_new mm_ext_dl_switch_controls =
+ SOC_SINGLE_EXT("Switch", ABE_VIRTUAL_SWITCH, MIX_SWITCH_MM_EXT_DL, 1, 0,
+ abe_get_mixer, abe_put_switch);
+
+static const struct snd_kcontrol_new abe_controls[] = {
+ /* DL1 mixer gains */
+ SOC_SINGLE_EXT_TLV("DL1 Media Playback Volume",
+ MIX_DL1_INPUT_MM_DL, 0, 149, 0,
+ volume_get_dl1_mixer, volume_put_dl1_mixer, mm_dl1_tlv),
+ SOC_SINGLE_EXT_TLV("DL1 Tones Playback Volume",
+ MIX_DL1_INPUT_TONES, 0, 149, 0,
+ volume_get_dl1_mixer, volume_put_dl1_mixer, tones_dl1_tlv),
+ SOC_SINGLE_EXT_TLV("DL1 Voice Playback Volume",
+ MIX_DL1_INPUT_VX_DL, 0, 149, 0,
+ volume_get_dl1_mixer, volume_put_dl1_mixer, voice_dl1_tlv),
+ SOC_SINGLE_EXT_TLV("DL1 Capture Playback Volume",
+ MIX_DL1_INPUT_MM_UL2, 0, 149, 0,
+ volume_get_dl1_mixer, volume_put_dl1_mixer, capture_dl1_tlv),
+
+ /* DL2 mixer gains */
+ SOC_SINGLE_EXT_TLV("DL2 Media Playback Volume",
+ MIX_DL2_INPUT_MM_DL, 0, 149, 0,
+ volume_get_dl2_mixer, volume_put_dl2_mixer, mm_dl2_tlv),
+ SOC_SINGLE_EXT_TLV("DL2 Tones Playback Volume",
+ MIX_DL2_INPUT_TONES, 0, 149, 0,
+ volume_get_dl2_mixer, volume_put_dl2_mixer, tones_dl2_tlv),
+ SOC_SINGLE_EXT_TLV("DL2 Voice Playback Volume",
+ MIX_DL2_INPUT_VX_DL, 0, 149, 0,
+ volume_get_dl2_mixer, volume_put_dl2_mixer, voice_dl2_tlv),
+ SOC_SINGLE_EXT_TLV("DL2 Capture Playback Volume",
+ MIX_DL2_INPUT_MM_UL2, 0, 149, 0,
+ volume_get_dl2_mixer, volume_put_dl2_mixer, capture_dl2_tlv),
+
+ /* VXREC mixer gains */
+ SOC_SINGLE_EXT_TLV("VXREC Media Volume",
+ MIX_VXREC_INPUT_MM_DL, 0, 149, 0,
+ volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_mm_dl_tlv),
+ SOC_SINGLE_EXT_TLV("VXREC Tones Volume",
+ MIX_VXREC_INPUT_TONES, 0, 149, 0,
+ volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_tones_tlv),
+ SOC_SINGLE_EXT_TLV("VXREC Voice DL Volume",
+ MIX_VXREC_INPUT_VX_UL, 0, 149, 0,
+ volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_vx_dl_tlv),
+ SOC_SINGLE_EXT_TLV("VXREC Voice UL Volume",
+ MIX_VXREC_INPUT_VX_DL, 0, 149, 0,
+ volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_vx_ul_tlv),
+
+ /* AUDUL mixer gains */
+ SOC_SINGLE_EXT_TLV("AUDUL Media Volume",
+ MIX_AUDUL_INPUT_MM_DL, 0, 149, 0,
+ volume_get_audul_mixer, volume_put_audul_mixer, audul_mm_tlv),
+ SOC_SINGLE_EXT_TLV("AUDUL Tones Volume",
+ MIX_AUDUL_INPUT_TONES, 0, 149, 0,
+ volume_get_audul_mixer, volume_put_audul_mixer, audul_tones_tlv),
+ SOC_SINGLE_EXT_TLV("AUDUL Voice UL Volume",
+ MIX_AUDUL_INPUT_UPLINK, 0, 149, 0,
+ volume_get_audul_mixer, volume_put_audul_mixer, audul_vx_ul_tlv),
+ SOC_SINGLE_EXT_TLV("AUDUL Voice DL Volume",
+ MIX_AUDUL_INPUT_VX_DL, 0, 149, 0,
+ volume_get_audul_mixer, volume_put_audul_mixer, audul_vx_dl_tlv),
+
+ /* SDT mixer gains */
+ SOC_SINGLE_EXT_TLV("SDT UL Volume",
+ MIX_SDT_INPUT_UP_MIXER, 0, 149, 0,
+ volume_get_sdt_mixer, volume_put_sdt_mixer, sdt_ul_tlv),
+ SOC_SINGLE_EXT_TLV("SDT DL Volume",
+ MIX_SDT_INPUT_DL1_MIXER, 0, 149, 0,
+ volume_get_sdt_mixer, volume_put_sdt_mixer, sdt_dl_tlv),
+
+ /* DMIC gains */
+ SOC_DOUBLE_EXT_TLV("DMIC1 UL Volume",
+ GAINS_DMIC1, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+ volume_get_gain, volume_put_gain, dmic_tlv),
+
+ SOC_DOUBLE_EXT_TLV("DMIC2 UL Volume",
+ GAINS_DMIC2, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+ volume_get_gain, volume_put_gain, dmic_tlv),
+
+ SOC_DOUBLE_EXT_TLV("DMIC3 UL Volume",
+ GAINS_DMIC3, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+ volume_get_gain, volume_put_gain, dmic_tlv),
+
+ SOC_DOUBLE_EXT_TLV("AMIC UL Volume",
+ GAINS_AMIC, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+ volume_get_gain, volume_put_gain, amic_tlv),
+
+ SOC_DOUBLE_EXT_TLV("BT UL Volume",
+ GAINS_BTUL, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+ volume_get_gain, volume_put_gain, btul_tlv),
+};
+
+static const struct snd_soc_dapm_widget abe_dapm_widgets[] = {
+
+ /* Frontend AIFs */
+ SND_SOC_DAPM_AIF_IN("TONES_DL", "Tones Playback", 0,
+ W_AIF_TONES_DL, ABE_OPP_25, 0),
+ SND_SOC_DAPM_AIF_IN("VX_DL", "Voice Playback", 0,
+ W_AIF_VX_DL, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_OUT("VX_UL", "Voice Capture", 0,
+ W_AIF_VX_UL, ABE_OPP_50, 0),
+ /* the MM_UL mapping is intentional */
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0,
+ W_AIF_MM_UL1, ABE_OPP_100, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0,
+ W_AIF_MM_UL2, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL", " MultiMedia1 Playback", 0,
+ W_AIF_MM_DL, ABE_OPP_25, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL_LP", " MultiMedia1 LP Playback", 0,
+ W_AIF_MM_DL_LP, ABE_OPP_25, 0),
+ SND_SOC_DAPM_AIF_IN("VIB_DL", "Vibra Playback", 0,
+ W_AIF_VIB_DL, ABE_OPP_100, 0),
+ SND_SOC_DAPM_AIF_IN("MODEM_DL", "MODEM Playback", 0,
+ W_AIF_MODEM_DL, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_OUT("MODEM_UL", "MODEM Capture", 0,
+ W_AIF_MODEM_UL, ABE_OPP_50, 0),
+
+ /* Backend DAIs */
+ SND_SOC_DAPM_AIF_IN("PDM_UL1", "Analog Capture", 0,
+ W_AIF_PDM_UL1, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_OUT("PDM_DL1", "HS Playback", 0,
+ W_AIF_PDM_DL1, ABE_OPP_25, 0),
+ SND_SOC_DAPM_AIF_OUT("PDM_DL2", "HF Playback", 0,
+ W_AIF_PDM_DL2, ABE_OPP_100, 0),
+ SND_SOC_DAPM_AIF_OUT("PDM_VIB", "Vibra Playback", 0,
+ W_AIF_PDM_VIB, ABE_OPP_100, 0),
+ SND_SOC_DAPM_AIF_IN("BT_VX_UL", "BT Capture", 0,
+ W_AIF_BT_VX_UL, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_OUT("BT_VX_DL", "BT Playback", 0,
+ W_AIF_BT_VX_DL, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_IN("MM_EXT_UL", "FM Capture", 0,
+ W_AIF_MM_EXT_UL, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_EXT_DL", "FM Playback", 0,
+ W_AIF_MM_EXT_DL, ABE_OPP_25, 0),
+ SND_SOC_DAPM_AIF_IN("DMIC0", "DMIC0 Capture", 0,
+ W_AIF_DMIC0, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_IN("DMIC1", "DMIC1 Capture", 0,
+ W_AIF_DMIC1, ABE_OPP_50, 0),
+ SND_SOC_DAPM_AIF_IN("DMIC2", "DMIC2 Capture", 0,
+ W_AIF_DMIC2, ABE_OPP_50, 0),
+
+ /* ROUTE_UL Capture Muxes */
+ SND_SOC_DAPM_MUX("MUX_UL00",
+ W_MUX_UL00, ABE_OPP_50, 0, &mm_ul00_control),
+ SND_SOC_DAPM_MUX("MUX_UL01",
+ W_MUX_UL01, ABE_OPP_50, 0, &mm_ul01_control),
+ SND_SOC_DAPM_MUX("MUX_UL02",
+ W_MUX_UL02, ABE_OPP_50, 0, &mm_ul02_control),
+ SND_SOC_DAPM_MUX("MUX_UL03",
+ W_MUX_UL03, ABE_OPP_50, 0, &mm_ul03_control),
+ SND_SOC_DAPM_MUX("MUX_UL04",
+ W_MUX_UL04, ABE_OPP_50, 0, &mm_ul04_control),
+ SND_SOC_DAPM_MUX("MUX_UL05",
+ W_MUX_UL05, ABE_OPP_50, 0, &mm_ul05_control),
+ SND_SOC_DAPM_MUX("MUX_UL06",
+ W_MUX_UL06, ABE_OPP_50, 0, &mm_ul06_control),
+ SND_SOC_DAPM_MUX("MUX_UL07",
+ W_MUX_UL07, ABE_OPP_50, 0, &mm_ul07_control),
+ SND_SOC_DAPM_MUX("MUX_UL10",
+ W_MUX_UL10, ABE_OPP_50, 0, &mm_ul10_control),
+ SND_SOC_DAPM_MUX("MUX_UL11",
+ W_MUX_UL11, ABE_OPP_50, 0, &mm_ul11_control),
+ SND_SOC_DAPM_MUX("MUX_VX0",
+ W_MUX_VX00, ABE_OPP_50, 0, &mm_vx0_control),
+ SND_SOC_DAPM_MUX("MUX_VX1",
+ W_MUX_VX01, ABE_OPP_50, 0, &mm_vx1_control),
+
+ /* DL1 & DL2 Playback Mixers */
+ SND_SOC_DAPM_MIXER("DL1 Mixer",
+ W_MIXER_DL1, ABE_OPP_25, 0, dl1_mixer_controls,
+ ARRAY_SIZE(dl1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("DL2 Mixer",
+ W_MIXER_DL2, ABE_OPP_100, 0, dl2_mixer_controls,
+ ARRAY_SIZE(dl2_mixer_controls)),
+
+ /* DL1 Mixer Input volumes ?????*/
+ SND_SOC_DAPM_PGA("DL1 Media Volume",
+ W_VOLUME_DL1, 0, 0, NULL, 0),
+
+ /* AUDIO_UL_MIXER */
+ SND_SOC_DAPM_MIXER("Voice Capture Mixer",
+ W_MIXER_AUDIO_UL, ABE_OPP_50, 0, audio_ul_mixer_controls,
+ ARRAY_SIZE(audio_ul_mixer_controls)),
+
+ /* VX_REC_MIXER */
+ SND_SOC_DAPM_MIXER("Capture Mixer",
+ W_MIXER_VX_REC, ABE_OPP_50, 0, vx_rec_mixer_controls,
+ ARRAY_SIZE(vx_rec_mixer_controls)),
+
+ /* SDT_MIXER - TODO: shoult this not be OPP25 ??? */
+ SND_SOC_DAPM_MIXER("Sidetone Mixer",
+ W_MIXER_SDT, ABE_OPP_25, 0, sdt_mixer_controls,
+ ARRAY_SIZE(sdt_mixer_controls)),
+
+ /*
+ * The Following three are virtual switches to select the output port
+ * after DL1 Gain.
+ */
+
+ /* Virtual PDM_DL1 Switch */
+ SND_SOC_DAPM_MIXER("DL1 PDM",
+ W_VSWITCH_DL1_PDM, ABE_OPP_25, 0, &pdm_dl1_switch_controls, 1),
+
+ /* Virtual BT_VX_DL Switch */
+ SND_SOC_DAPM_MIXER("DL1 BT_VX",
+ W_VSWITCH_DL1_BT_VX, ABE_OPP_50, 0, &bt_vx_dl_switch_controls, 1),
+
+ /* Virtual MM_EXT_DL Switch TODO: confrm OPP level here */
+ SND_SOC_DAPM_MIXER("DL1 MM_EXT",
+ W_VSWITCH_DL1_MM_EXT, ABE_OPP_50, 0, &mm_ext_dl_switch_controls, 1),
+
+ /* Virtuals to join our capture sources */
+ SND_SOC_DAPM_MIXER("Sidetone Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Voice Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("DL1 Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("DL2 Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Join our MM_DL and MM_DL_LP playback */
+ SND_SOC_DAPM_MIXER("MM_DL VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Virtual MODEM and VX_UL mixer */
+ SND_SOC_DAPM_MIXER("VX UL VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("VX DL VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+ /* Virtual Pins to force backends ON atm */
+ SND_SOC_DAPM_OUTPUT("BE_OUT"),
+ SND_SOC_DAPM_INPUT("BE_IN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+ /* MUX_UL00 - ROUTE_UL - Chan 0 */
+ {"MUX_UL00", "DMic0L", "DMIC0"},
+ {"MUX_UL00", "DMic0R", "DMIC0"},
+ {"MUX_UL00", "DMic1L", "DMIC1"},
+ {"MUX_UL00", "DMic1R", "DMIC1"},
+ {"MUX_UL00", "DMic2L", "DMIC2"},
+ {"MUX_UL00", "DMic2R", "DMIC2"},
+ {"MUX_UL00", "BT Left", "BT_VX_UL"},
+ {"MUX_UL00", "BT Right", "BT_VX_UL"},
+ {"MUX_UL00", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL00", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL00", "AMic0", "PDM_UL1"},
+ {"MUX_UL00", "AMic1", "PDM_UL1"},
+ {"MUX_UL00", "VX Left", "Capture Mixer"},
+ {"MUX_UL00", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL00"},
+
+ /* MUX_UL01 - ROUTE_UL - Chan 1 */
+ {"MUX_UL01", "DMic0L", "DMIC0"},
+ {"MUX_UL01", "DMic0R", "DMIC0"},
+ {"MUX_UL01", "DMic1L", "DMIC1"},
+ {"MUX_UL01", "DMic1R", "DMIC1"},
+ {"MUX_UL01", "DMic2L", "DMIC2"},
+ {"MUX_UL01", "DMic2R", "DMIC2"},
+ {"MUX_UL01", "BT Left", "BT_VX_UL"},
+ {"MUX_UL01", "BT Right", "BT_VX_UL"},
+ {"MUX_UL01", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL01", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL01", "AMic0", "PDM_UL1"},
+ {"MUX_UL01", "AMic1", "PDM_UL1"},
+ {"MUX_UL01", "VX Left", "Capture Mixer"},
+ {"MUX_UL01", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL01"},
+
+ /* MUX_UL02 - ROUTE_UL - Chan 2 */
+ {"MUX_UL02", "DMic0L", "DMIC0"},
+ {"MUX_UL02", "DMic0R", "DMIC0"},
+ {"MUX_UL02", "DMic1L", "DMIC1"},
+ {"MUX_UL02", "DMic1R", "DMIC1"},
+ {"MUX_UL02", "DMic2L", "DMIC2"},
+ {"MUX_UL02", "DMic2R", "DMIC2"},
+ {"MUX_UL02", "BT Left", "BT_VX_UL"},
+ {"MUX_UL02", "BT Right", "BT_VX_UL"},
+ {"MUX_UL02", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL02", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL02", "AMic0", "PDM_UL1"},
+ {"MUX_UL02", "AMic1", "PDM_UL1"},
+ {"MUX_UL02", "VX Left", "Capture Mixer"},
+ {"MUX_UL02", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL02"},
+
+ /* MUX_UL03 - ROUTE_UL - Chan 3 */
+ {"MUX_UL03", "DMic0L", "DMIC0"},
+ {"MUX_UL03", "DMic0R", "DMIC0"},
+ {"MUX_UL03", "DMic1L", "DMIC1"},
+ {"MUX_UL03", "DMic1R", "DMIC1"},
+ {"MUX_UL03", "DMic2L", "DMIC2"},
+ {"MUX_UL03", "DMic2R", "DMIC2"},
+ {"MUX_UL03", "BT Left", "BT_VX_UL"},
+ {"MUX_UL03", "BT Right", "BT_VX_UL"},
+ {"MUX_UL03", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL03", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL03", "AMic0", "PDM_UL1"},
+ {"MUX_UL03", "AMic1", "PDM_UL1"},
+ {"MUX_UL03", "VX Left", "Capture Mixer"},
+ {"MUX_UL03", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL03"},
+
+ /* MUX_UL04 - ROUTE_UL - Chan 4 */
+ {"MUX_UL04", "DMic0L", "DMIC0"},
+ {"MUX_UL04", "DMic0R", "DMIC0"},
+ {"MUX_UL04", "DMic1L", "DMIC1"},
+ {"MUX_UL04", "DMic1R", "DMIC1"},
+ {"MUX_UL04", "DMic2L", "DMIC2"},
+ {"MUX_UL04", "DMic2R", "DMIC2"},
+ {"MUX_UL04", "BT Left", "BT_VX_UL"},
+ {"MUX_UL04", "BT Right", "BT_VX_UL"},
+ {"MUX_UL04", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL04", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL04", "AMic0", "PDM_UL1"},
+ {"MUX_UL04", "AMic1", "PDM_UL1"},
+ {"MUX_UL04", "VX Left", "Capture Mixer"},
+ {"MUX_UL04", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL04"},
+
+ /* MUX_UL05 - ROUTE_UL - Chan 5 */
+ {"MUX_UL05", "DMic0L", "DMIC0"},
+ {"MUX_UL05", "DMic0R", "DMIC0"},
+ {"MUX_UL05", "DMic1L", "DMIC1"},
+ {"MUX_UL05", "DMic1R", "DMIC1"},
+ {"MUX_UL05", "DMic2L", "DMIC2"},
+ {"MUX_UL05", "DMic2R", "DMIC2"},
+ {"MUX_UL05", "BT Left", "BT_VX_UL"},
+ {"MUX_UL05", "BT Right", "BT_VX_UL"},
+ {"MUX_UL05", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL05", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL05", "AMic0", "PDM_UL1"},
+ {"MUX_UL05", "AMic1", "PDM_UL1"},
+ {"MUX_UL05", "VX Left", "Capture Mixer"},
+ {"MUX_UL05", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL05"},
+
+ /* MUX_UL06 - ROUTE_UL - Chan 6 */
+ {"MUX_UL06", "DMic0L", "DMIC0"},
+ {"MUX_UL06", "DMic0R", "DMIC0"},
+ {"MUX_UL06", "DMic1L", "DMIC1"},
+ {"MUX_UL06", "DMic1R", "DMIC1"},
+ {"MUX_UL06", "DMic2L", "DMIC2"},
+ {"MUX_UL06", "DMic2R", "DMIC2"},
+ {"MUX_UL06", "BT Left", "BT_VX_UL"},
+ {"MUX_UL06", "BT Right", "BT_VX_UL"},
+ {"MUX_UL06", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL06", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL06", "AMic0", "PDM_UL1"},
+ {"MUX_UL06", "AMic1", "PDM_UL1"},
+ {"MUX_UL06", "VX Left", "Capture Mixer"},
+ {"MUX_UL06", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL06"},
+
+ /* MUX_UL07 - ROUTE_UL - Chan 7 */
+ {"MUX_UL07", "DMic0L", "DMIC0"},
+ {"MUX_UL07", "DMic0R", "DMIC0"},
+ {"MUX_UL07", "DMic1L", "DMIC1"},
+ {"MUX_UL07", "DMic1R", "DMIC1"},
+ {"MUX_UL07", "DMic2L", "DMIC2"},
+ {"MUX_UL07", "DMic2R", "DMIC2"},
+ {"MUX_UL07", "BT Left", "BT_VX_UL"},
+ {"MUX_UL07", "BT Right", "BT_VX_UL"},
+ {"MUX_UL07", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL07", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL07", "AMic0", "PDM_UL1"},
+ {"MUX_UL07", "AMic1", "PDM_UL1"},
+ {"MUX_UL07", "VX Left", "Capture Mixer"},
+ {"MUX_UL07", "VX Right", "Capture Mixer"},
+ {"MM_UL1", NULL, "MUX_UL07"},
+
+ /* MUX_UL10 - ROUTE_UL - Chan 10 */
+ {"MUX_UL10", "DMic0L", "DMIC0"},
+ {"MUX_UL10", "DMic0R", "DMIC0"},
+ {"MUX_UL10", "DMic1L", "DMIC1"},
+ {"MUX_UL10", "DMic1R", "DMIC1"},
+ {"MUX_UL10", "DMic2L", "DMIC2"},
+ {"MUX_UL10", "DMic2R", "DMIC2"},
+ {"MUX_UL10", "BT Left", "BT_VX_UL"},
+ {"MUX_UL10", "BT Right", "BT_VX_UL"},
+ {"MUX_UL10", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL10", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL10", "AMic0", "PDM_UL1"},
+ {"MUX_UL10", "AMic1", "PDM_UL1"},
+ {"MUX_UL10", "VX Left", "Capture Mixer"},
+ {"MUX_UL10", "VX Right", "Capture Mixer"},
+ {"MM_UL2", NULL, "MUX_UL10"},
+
+ /* MUX_UL11 - ROUTE_UL - Chan 11 */
+ {"MUX_UL11", "DMic0L", "DMIC0"},
+ {"MUX_UL11", "DMic0R", "DMIC0"},
+ {"MUX_UL11", "DMic1L", "DMIC1"},
+ {"MUX_UL11", "DMic1R", "DMIC1"},
+ {"MUX_UL11", "DMic2L", "DMIC2"},
+ {"MUX_UL11", "DMic2R", "DMIC2"},
+ {"MUX_UL11", "BT Left", "BT_VX_UL"},
+ {"MUX_UL11", "BT Right", "BT_VX_UL"},
+ {"MUX_UL11", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_UL11", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_UL11", "AMic0", "PDM_UL1"},
+ {"MUX_UL11", "AMic1", "PDM_UL1"},
+ {"MUX_UL11", "VX Left", "Capture Mixer"},
+ {"MUX_UL11", "VX Right", "Capture Mixer"},
+ {"MM_UL2", NULL, "MUX_UL11"},
+
+ /* MUX_VX0 - ROUTE_UL - Chan 20 */
+ {"MUX_VX0", "DMic0L", "DMIC0"},
+ {"MUX_VX0", "DMic0R", "DMIC0"},
+ {"MUX_VX0", "DMic1L", "DMIC1"},
+ {"MUX_VX0", "DMic1R", "DMIC1"},
+ {"MUX_VX0", "DMic2L", "DMIC2"},
+ {"MUX_VX0", "DMic2R", "DMIC2"},
+ {"MUX_VX0", "BT Left", "BT_VX_UL"},
+ {"MUX_VX0", "BT Right", "BT_VX_UL"},
+ {"MUX_VX0", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_VX0", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_VX0", "AMic0", "PDM_UL1"},
+ {"MUX_VX0", "AMic1", "PDM_UL1"},
+ {"MUX_VX0", "VX Left", "Capture Mixer"},
+ {"MUX_VX0", "VX Right", "Capture Mixer"},
+
+ /* MUX_VX1 - ROUTE_UL - Chan 20 */
+ {"MUX_VX1", "DMic0L", "DMIC0"},
+ {"MUX_VX1", "DMic0R", "DMIC0"},
+ {"MUX_VX1", "DMic1L", "DMIC1"},
+ {"MUX_VX1", "DMic1R", "DMIC1"},
+ {"MUX_VX1", "DMic2L", "DMIC2"},
+ {"MUX_VX1", "DMic2R", "DMIC2"},
+ {"MUX_VX1", "BT Left", "BT_VX_UL"},
+ {"MUX_VX1", "BT Right", "BT_VX_UL"},
+ {"MUX_VX1", "MMExt Left", "MM_EXT_UL"},
+ {"MUX_VX1", "MMExt Right", "MM_EXT_UL"},
+ {"MUX_VX1", "AMic0", "PDM_UL1"},
+ {"MUX_VX1", "AMic1", "PDM_UL1"},
+ {"MUX_VX1", "VX Left", "Capture Mixer"},
+ {"MUX_VX1", "VX Right", "Capture Mixer"},
+
+ /* Headset (DL1) playback path */
+ {"DL1 Mixer", "Tones", "TONES_DL"},
+ {"DL1 Mixer", "Voice", "VX DL VMixer"},
+ {"DL1 Mixer", "Capture", "DL1 Capture VMixer"},
+ {"DL1 Capture VMixer", NULL, "MUX_UL10"},
+ {"DL1 Capture VMixer", NULL, "MUX_UL11"},
+ {"DL1 Mixer", "Multimedia", "MM_DL VMixer"},
+ {"MM_DL VMixer", NULL, "MM_DL"},
+ {"MM_DL VMixer", NULL, "MM_DL_LP"},
+
+ /* Sidetone Mixer */
+ {"Sidetone Mixer", "Playback", "DL1 Mixer"},
+ {"Sidetone Mixer", "Capture", "Sidetone Capture VMixer"},
+ {"Sidetone Capture VMixer", NULL, "MUX_VX0"},
+ {"Sidetone Capture VMixer", NULL, "MUX_VX1"},
+
+ /* Playback Output selection after DL1 Gain */
+ {"DL1 BT_VX", "Switch", "Sidetone Mixer"},
+ {"DL1 MM_EXT", "Switch", "Sidetone Mixer"},
+ {"DL1 PDM", "Switch", "Sidetone Mixer"},
+ {"PDM_DL1", NULL, "DL1 PDM"},
+ {"BT_VX_DL", NULL, "DL1 BT_VX"},
+ {"MM_EXT_DL", NULL, "DL1 MM_EXT"},
+
+ /* Handsfree (DL2) playback path */
+ {"DL2 Mixer", "Tones", "TONES_DL"},
+ {"DL2 Mixer", "Voice", "VX DL VMixer"},
+ {"DL2 Mixer", "Capture", "DL2 Capture VMixer"},
+ {"DL2 Capture VMixer", NULL, "MUX_UL10"},
+ {"DL2 Capture VMixer", NULL, "MUX_UL11"},
+ {"DL2 Mixer", "Multimedia", "MM_DL VMixer"},
+ {"MM_DL VMixer", NULL, "MM_DL"},
+ {"MM_DL VMixer", NULL, "MM_DL_LP"},
+ {"PDM_DL2", NULL, "DL2 Mixer"},
+
+ /* VxREC Mixer */
+ {"Capture Mixer", "Tones", "TONES_DL"},
+ {"Capture Mixer", "Voice Playback", "VX DL VMixer"},
+ {"Capture Mixer", "Voice Capture", "VX UL VMixer"},
+ {"Capture Mixer", "Media Playback", "MM_DL VMixer"},
+ {"MM_DL VMixer", NULL, "MM_DL"},
+ {"MM_DL VMixer", NULL, "MM_DL_LP"},
+
+ /* Audio UL mixer */
+ {"Voice Capture Mixer", "Tones Playback", "TONES_DL"},
+ {"Voice Capture Mixer", "Media Playback", "MM_DL VMixer"},
+ {"MM_DL VMixer", NULL, "MM_DL"},
+ {"MM_DL VMixer", NULL, "MM_DL_LP"},
+ {"Voice Capture Mixer", "Capture", "Voice Capture VMixer"},
+ {"Voice Capture VMixer", NULL, "MUX_VX0"},
+ {"Voice Capture VMixer", NULL, "MUX_VX1"},
+
+ /* BT */
+ {"VX UL VMixer", NULL, "Voice Capture Mixer"},
+
+ /* Vibra */
+ {"PDM_VIB", NULL, "VIB_DL"},
+
+ /* VX and MODEM */
+ {"VX_UL", NULL, "VX UL VMixer"},
+ {"MODEM_UL", NULL, "VX UL VMixer"},
+ {"VX DL VMixer", NULL, "VX_DL"},
+ {"VX DL VMixer", NULL, "MODEM_DL"},
+
+ /* Backend Enablement - TODO: maybe re-work*/
+ {"BE_OUT", NULL, "PDM_DL1"},
+ {"BE_OUT", NULL, "PDM_DL2"},
+ {"BE_OUT", NULL, "PDM_VIB"},
+ {"BE_OUT", NULL, "MM_EXT_DL"},
+ {"BE_OUT", NULL, "BT_VX_DL"},
+ {"PDM_UL1", NULL, "BE_IN"},
+ {"BT_VX_UL", NULL, "BE_IN"},
+ {"MM_EXT_UL", NULL, "BE_IN"},
+ {"DMIC0", NULL, "BE_IN"},
+ {"DMIC1", NULL, "BE_IN"},
+ {"DMIC2", NULL, "BE_IN"},
+};
+
+#ifdef CONFIG_DEBUG_FS
+
+static int abe_dbg_get_dma_pos(struct abe_data *abe)
+{
+ return omap_get_dma_dst_pos(abe->dma_ch) - abe->dbg_buffer_addr;
+}
+
+static void abe_dbg_dma_irq(int ch, u16 stat, void *data)
+{
+}
+
+static int abe_dbg_start_dma(struct abe_data *abe, int circular)
+{
+ struct omap_dma_channel_params dma_params;
+ int err;
+
+ /* TODO: start the DMA in either :-
+ *
+ * 1) circular buffer mode where the DMA will restart when it get to
+ * the end of the buffer.
+ * 2) default mode, where DMA stops at the end of the buffer.
+ */
+
+ abe->dma_req = OMAP44XX_DMA_ABE_REQ_7;
+ err = omap_request_dma(abe->dma_req, "ABE debug",
+ abe_dbg_dma_irq, abe, &abe->dma_ch);
+ if (abe->dbg_circular) {
+ /*
+ * Link channel with itself so DMA doesn't need any
+ * reprogramming while looping the buffer
+ */
+ omap_dma_link_lch(abe->dma_ch, abe->dma_ch);
+ }
+
+ memset(&dma_params, 0, sizeof(dma_params));
+ dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
+ dma_params.trigger = abe->dma_req;
+ dma_params.sync_mode = OMAP_DMA_SYNC_FRAME;
+ dma_params.src_amode = OMAP_DMA_AMODE_DOUBLE_IDX;
+ dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
+ dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
+ dma_params.src_start = D_DEBUG_FIFO_ADDR + ABE_DMEM_BASE_ADDRESS_L3;
+ dma_params.dst_start = abe->dbg_buffer_addr;
+ dma_params.src_port = OMAP_DMA_PORT_MPUI;
+ dma_params.src_ei = 1;
+ dma_params.src_fi = 1 - abe->dbg_elem_bytes;
+
+ dma_params.elem_count = abe->dbg_elem_bytes >> 2; /* 128 bytes shifted into words */
+ dma_params.frame_count = abe->dbg_buffer_bytes / abe->dbg_elem_bytes;
+ omap_set_dma_params(abe->dma_ch, &dma_params);
+
+ omap_enable_dma_irq(abe->dma_ch, OMAP_DMA_FRAME_IRQ);
+ omap_set_dma_src_burst_mode(abe->dma_ch, OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(abe->dma_ch, OMAP_DMA_DATA_BURST_16);
+
+ abe->dbg_reader_offset = 0;
+
+ pm_runtime_get_sync(abe->dev);
+ omap_start_dma(abe->dma_ch);
+ return 0;
+}
+
+static void abe_dbg_stop_dma(struct abe_data *abe)
+{
+ while (omap_get_dma_active_status(abe->dma_ch))
+ omap_stop_dma(abe->dma_ch);
+
+ if (abe->dbg_circular)
+ omap_dma_unlink_lch(abe->dma_ch, abe->dma_ch);
+ omap_free_dma(abe->dma_ch);
+ pm_runtime_put_sync(abe->dev);
+}
+
+static int abe_open_data(struct inode *inode, struct file *file)
+{
+ struct abe_data *abe = inode->i_private;
+
+ abe->dbg_elem_bytes = 128; /* size of debug data per tick */
+
+ if (abe->dbg_format1)
+ abe->dbg_elem_bytes += ABE_DBG_FLAG1_SIZE;
+ if (abe->dbg_format2)
+ abe->dbg_elem_bytes += ABE_DBG_FLAG2_SIZE;
+ if (abe->dbg_format3)
+ abe->dbg_elem_bytes += ABE_DBG_FLAG3_SIZE;
+
+ abe->dbg_buffer_bytes = abe->dbg_elem_bytes * 4 *
+ abe->dbg_buffer_msecs;
+
+ abe->dbg_buffer = dma_alloc_writecombine(abe->dev,
+ abe->dbg_buffer_bytes, &abe->dbg_buffer_addr, GFP_KERNEL);
+ if (abe->dbg_buffer == NULL)
+ return -ENOMEM;
+
+ file->private_data = inode->i_private;
+ abe->dbg_complete = 0;
+ abe_dbg_start_dma(abe, abe->dbg_circular);
+
+ return 0;
+}
+
+static int abe_release_data(struct inode *inode, struct file *file)
+{
+ struct abe_data *abe = inode->i_private;
+
+ abe_dbg_stop_dma(abe);
+
+ dma_free_writecombine(abe->dev, abe->dbg_buffer_bytes,
+ abe->dbg_buffer, abe->dbg_buffer_addr);
+ return 0;
+}
+
+static ssize_t abe_copy_to_user(struct abe_data *abe, char __user *user_buf,
+ size_t count)
+{
+ /* check for reader buffer wrap */
+ if (abe->dbg_reader_offset + count > abe->dbg_buffer_bytes) {
+ int size = abe->dbg_buffer_bytes - abe->dbg_reader_offset;
+
+ /* wrap */
+ if (copy_to_user(user_buf,
+ abe->dbg_buffer + abe->dbg_reader_offset, size))
+ return -EFAULT;
+
+ /* need to just return if non circular */
+ if (!abe->dbg_circular) {
+ abe->dbg_complete = 1;
+ return count;
+ }
+
+ if (copy_to_user(user_buf,
+ abe->dbg_buffer, count - size))
+ return -EFAULT;
+ abe->dbg_reader_offset = count - size;
+ return count;
+ } else {
+ /* no wrap */
+ if (copy_to_user(user_buf,
+ abe->dbg_buffer + abe->dbg_reader_offset, count))
+ return -EFAULT;
+ abe->dbg_reader_offset += count;
+
+ if (!abe->dbg_circular &&
+ abe->dbg_reader_offset == abe->dbg_buffer_bytes)
+ abe->dbg_complete = 1;
+
+ return count;
+ }
+}
+
+static ssize_t abe_read_data(struct file *file, char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ ssize_t ret = 0;
+ struct abe_data *abe = file->private_data;
+ DECLARE_WAITQUEUE(wait, current);
+ int dma_offset, bytes;
+
+ add_wait_queue(&abe->wait, &wait);
+ do {
+ set_current_state(TASK_INTERRUPTIBLE);
+ /* TODO: Check if really needed. Or adjust sleep delay
+ * If not delay trace is not working */
+ msleep_interruptible(1);
+ dma_offset = abe_dbg_get_dma_pos(abe);
+
+ /* is DMA finished ? */
+ if (abe->dbg_complete)
+ break;
+
+ /* get maximum amount of debug bytes we can read */
+ if (dma_offset >= abe->dbg_reader_offset) {
+ /* dma ptr is ahead of reader */
+ bytes = dma_offset - abe->dbg_reader_offset;
+ } else {
+ /* dma ptr is behind reader */
+ bytes = dma_offset + abe->dbg_buffer_bytes -
+ abe->dbg_reader_offset;
+ }
+
+ if (count > bytes)
+ count = bytes;
+
+ if (count > 0) {
+ ret = abe_copy_to_user(abe, user_buf, count);
+ break;
+ }
+
+ if (file->f_flags & O_NONBLOCK) {
+ ret = -EAGAIN;
+ break;
+ }
+
+ if (signal_pending(current)) {
+ ret = -ERESTARTSYS;
+ break;
+ }
+
+ schedule();
+
+ } while (1);
+
+ __set_current_state(TASK_RUNNING);
+ remove_wait_queue(&abe->wait, &wait);
+
+ return ret;
+}
+
+static const struct file_operations abe_data_fops = {
+ .open = abe_open_data,
+ .read = abe_read_data,
+ .release = abe_release_data,
+};
+
+static void abe_init_debugfs(struct abe_data *abe)
+{
+ abe->debugfs_root = debugfs_create_dir("omap4-abe", NULL);
+ if (!abe->debugfs_root) {
+ printk(KERN_WARNING "ABE: Failed to create debugfs directory\n");
+ return;
+ }
+
+ abe->debugfs_fmt1 = debugfs_create_bool("format1", 0644,
+ abe->debugfs_root,
+ &abe->dbg_format1);
+ if (!abe->debugfs_fmt1)
+ printk(KERN_WARNING "ABE: Failed to create format1 debugfs file\n");
+
+ abe->debugfs_fmt2 = debugfs_create_bool("format2", 0644,
+ abe->debugfs_root,
+ &abe->dbg_format2);
+ if (!abe->debugfs_fmt2)
+ printk(KERN_WARNING "ABE: Failed to create format2 debugfs file\n");
+
+ abe->debugfs_fmt3 = debugfs_create_bool("format3", 0644,
+ abe->debugfs_root,
+ &abe->dbg_format3);
+ if (!abe->debugfs_fmt3)
+ printk(KERN_WARNING "ABE: Failed to create format3 debugfs file\n");
+
+ abe->debugfs_elem_bytes = debugfs_create_u32("element_bytes", 0604,
+ abe->debugfs_root,
+ &abe->dbg_elem_bytes);
+ if (!abe->debugfs_elem_bytes)
+ printk(KERN_WARNING "ABE: Failed to create element size debugfs file\n");
+
+ abe->debugfs_size = debugfs_create_u32("msecs", 0644,
+ abe->debugfs_root,
+ &abe->dbg_buffer_msecs);
+ if (!abe->debugfs_size)
+ printk(KERN_WARNING "ABE: Failed to create buffer size debugfs file\n");
+
+ abe->debugfs_circ = debugfs_create_bool("circular", 0644,
+ abe->debugfs_root,
+ &abe->dbg_circular);
+ if (!abe->debugfs_size)
+ printk(KERN_WARNING "ABE: Failed to create circular mode debugfs file\n");
+
+ abe->debugfs_data = debugfs_create_file("debug", 0644,
+ abe->debugfs_root,
+ abe, &abe_data_fops);
+ if (!abe->debugfs_data)
+ printk(KERN_WARNING "ABE: Failed to create data debugfs file\n");
+
+ abe->debugfs_opp_level = debugfs_create_u32("opp_level", 0604,
+ abe->debugfs_root,
+ &abe->opp);
+ if (!abe->debugfs_opp_level)
+ printk(KERN_WARNING "ABE: Failed to create OPP level debugfs file\n");
+
+ abe->dbg_buffer_msecs = 500;
+ init_waitqueue_head(&abe->wait);
+}
+
+static void abe_cleanup_debugfs(struct abe_data *abe)
+{
+ debugfs_remove_recursive(abe->debugfs_root);
+}
+
+#else
+
+static inline void abe_init_debugfs(struct abe_data *abe)
+{
+}
+
+static inline void abe_cleanup_debugfs(struct abe_data *abe)
+{
+}
+#endif
+
+static const struct snd_pcm_hardware omap_abe_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = 4 * 1024,
+ .period_bytes_max = 24 * 1024,
+ .periods_min = 2,
+ .periods_max = 2,
+ .buffer_bytes_max = 24 * 1024 * 2,
+};
+
+
+static int abe_set_opp_mode(struct abe_data *abe)
+{
+ int i, opp = 0;
+
+ /* now calculate OPP level based upon DAPM widget status */
+ for (i = 0; i < ABE_NUM_WIDGETS; i++) {
+ if (abe->widget_opp[ABE_WIDGET(i)]) {
+ dev_dbg(abe->dev, "OPP: id %d = %d%%\n", i,
+ abe->widget_opp[ABE_WIDGET(i)] * 25);
+ opp |= abe->widget_opp[ABE_WIDGET(i)];
+ }
+ }
+ opp = (1 << (fls(opp) - 1)) * 25;
+
+ if (abe->opp > opp) {
+ /* Decrease OPP mode - no need of OPP100% */
+ switch (opp) {
+ case 25:
+ abe_set_opp_processing(ABE_OPP25);
+ udelay(250);
+ omap_device_set_rate(abe->dev, abe->dev, 49150000);
+ break;
+ case 50:
+ default:
+ abe_set_opp_processing(ABE_OPP50);
+ udelay(250);
+ omap_device_set_rate(abe->dev, abe->dev, 98300000);
+ break;
+ }
+ } else if (abe->opp < opp) {
+ /* Increase OPP mode */
+ switch (opp) {
+ case 25:
+ omap_device_set_rate(abe->dev, abe->dev, 49000000);
+ abe_set_opp_processing(ABE_OPP25);
+ break;
+ case 50:
+ omap_device_set_rate(abe->dev, abe->dev, 98300000);
+ abe_set_opp_processing(ABE_OPP50);
+ break;
+ case 100:
+ default:
+ omap_device_set_rate(abe->dev, abe->dev, 196600000);
+ abe_set_opp_processing(ABE_OPP100);
+ break;
+ }
+ }
+ abe->opp = opp;
+ dev_dbg(abe->dev, "new OPP level is %d\n", opp);
+
+ return 0;
+}
+
+static int aess_set_runtime_opp_level(struct abe_data *abe)
+{
+ mutex_lock(&abe->opp_mutex);
+
+ pm_runtime_get_sync(abe->dev);
+ abe_set_opp_mode(abe);
+ pm_runtime_put_sync(abe->dev);
+
+ mutex_unlock(&abe->opp_mutex);
+
+ return 0;
+}
+
+static int aess_save_context(struct abe_data *abe)
+{
+ struct omap4_abe_dsp_pdata *pdata = abe->abe_pdata;
+
+ /* TODO: Find a better way to save/retore gains after OFF mode */
+
+ abe_mute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+ abe_mute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+ abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
+ abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
+ abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+ abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
+ abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
+ abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
+ abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
+ abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
+ abe_mute_gain(MIXECHO, MIX_ECHO_DL1);
+ abe_mute_gain(MIXECHO, MIX_ECHO_DL2);
+ abe_mute_gain(GAINS_DMIC1, GAIN_LEFT_OFFSET);
+ abe_mute_gain(GAINS_DMIC1, GAIN_RIGHT_OFFSET);
+ abe_mute_gain(GAINS_DMIC2, GAIN_LEFT_OFFSET);
+ abe_mute_gain(GAINS_DMIC2, GAIN_RIGHT_OFFSET);
+ abe_mute_gain(GAINS_DMIC3, GAIN_LEFT_OFFSET);
+ abe_mute_gain(GAINS_DMIC3, GAIN_RIGHT_OFFSET);
+ abe_mute_gain(GAINS_AMIC, GAIN_LEFT_OFFSET);
+ abe_mute_gain(GAINS_AMIC, GAIN_RIGHT_OFFSET);
+
+ if (pdata->get_context_loss_count)
+ abe->loss_count = pdata->get_context_loss_count(abe->dev);
+
+ return 0;
+}
+
+static int aess_restore_context(struct abe_data *abe)
+{
+ struct omap4_abe_dsp_pdata *pdata = abe->abe_pdata;
+ int loss_count = 0;
+
+ omap_device_set_rate(&abe->dev, &abe->dev, 98000000);
+
+ if (pdata->get_context_loss_count)
+ loss_count = pdata->get_context_loss_count(abe->dev);
+
+ if (loss_count != the_abe->loss_count)
+ abe_reload_fw(abe->firmware);
+
+ /* TODO: Find a better way to save/retore gains after dor OFF mode */
+ abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+ abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+ abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
+ abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
+ abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+ abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
+ abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
+ abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
+ abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
+ abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_UL2);
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_UL2);
+ abe_unmute_gain(MIXECHO, MIX_ECHO_DL1);
+ abe_unmute_gain(MIXECHO, MIX_ECHO_DL2);
+ abe_unmute_gain(GAINS_DMIC1, GAIN_LEFT_OFFSET);
+ abe_unmute_gain(GAINS_DMIC1, GAIN_RIGHT_OFFSET);
+ abe_unmute_gain(GAINS_DMIC2, GAIN_LEFT_OFFSET);
+ abe_unmute_gain(GAINS_DMIC2, GAIN_RIGHT_OFFSET);
+ abe_unmute_gain(GAINS_DMIC3, GAIN_LEFT_OFFSET);
+ abe_unmute_gain(GAINS_DMIC3, GAIN_RIGHT_OFFSET);
+ abe_unmute_gain(GAINS_AMIC, GAIN_LEFT_OFFSET);
+ abe_unmute_gain(GAINS_AMIC, GAIN_RIGHT_OFFSET);
+/*
+ abe_dsp_set_equalizer(EQ1, abe->dl1_equ_profile);
+ abe_dsp_set_equalizer(EQ2L, abe->dl20_equ_profile);
+ abe_dsp_set_equalizer(EQ2R, abe->dl21_equ_profile);
+ abe_dsp_set_equalizer(EQAMIC, abe->amic_equ_profile);
+ abe_dsp_set_equalizer(EQDMIC, abe->dmic_equ_profile);
+ abe_dsp_set_equalizer(EQSDT, abe->sdt_equ_profile);
+*/
+ abe_set_router_configuration(UPROUTE, 0, (u32 *)abe->router);
+
+ return 0;
+}
+
+static int aess_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ int ret = 0;
+
+ mutex_lock(&abe->mutex);
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ pm_runtime_get_sync(abe->dev);
+
+ if (!abe->active++) {
+ abe->opp = 0;
+ aess_restore_context(abe);
+ abe_set_opp_mode(abe);
+ abe_wakeup();
+ }
+
+ switch (dai->id) {
+ case ABE_FRONTEND_DAI_MODEM:
+ break;
+ case ABE_FRONTEND_DAI_LP_MEDIA:
+ snd_soc_set_runtime_hwparams(substream, &omap_abe_hardware);
+ ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 1024);
+ break;
+ default:
+ break;
+ }
+
+ mutex_unlock(&abe->mutex);
+ return ret;
+}
+
+static int aess_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ abe_data_format_t format;
+ size_t period_size;
+ u32 dst;
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ if (dai->id != ABE_FRONTEND_DAI_LP_MEDIA)
+ return 0;
+
+ /*Storing substream pointer for irq*/
+ abe->ping_pong_substream = substream;
+
+ format.f = params_rate(params);
+ if (params_format(params) == SNDRV_PCM_FORMAT_S32_LE)
+ format.samp_format = STEREO_MSB;
+ else
+ format.samp_format = STEREO_16_16;
+
+ if (format.f == 44100)
+ abe_write_event_generator(EVENT_44100);
+
+ period_size = params_period_bytes(params);
+
+ /*Adding ping pong buffer subroutine*/
+ abe_plug_subroutine(&abe_irq_pingpong_player_id,
+ (abe_subroutine2) abe_irq_pingpong_subroutine,
+ SUB_1_PARAM, (u32 *)abe);
+
+ /* Connect a Ping-Pong cache-flush protocol to MM_DL port */
+ abe_connect_irq_ping_pong_port(MM_DL_PORT, &format,
+ abe_irq_pingpong_player_id,
+ period_size, &dst,
+ PING_PONG_WITH_MCU_IRQ);
+
+ /* Memory mapping for hw params */
+ runtime->dma_area = abe->io_base[0] + dst;
+ runtime->dma_addr = 0;
+ runtime->dma_bytes = period_size * 2;
+
+ /* Need to set the first buffer in order to get interrupt */
+ abe_set_ping_pong_buffer(MM_DL_PORT, period_size);
+ abe->first_irq = 1;
+
+ return 0;
+}
+
+static int aess_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+
+ mutex_lock(&abe->mutex);
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+ aess_set_runtime_opp_level(abe);
+ mutex_unlock(&abe->mutex);
+ return 0;
+}
+
+static int aess_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+
+ mutex_lock(&abe->mutex);
+ aess_set_runtime_opp_level(abe);
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ if (!--abe->active) {
+ abe_disable_irq();
+ aess_save_context(abe);
+ abe_dsp_shutdown();
+ }
+
+ pm_runtime_put_sync(abe->dev);
+
+ mutex_unlock(&abe->mutex);
+ return 0;
+}
+
+static int aess_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *dai = rtd->cpu_dai;
+ int offset, size, err;
+
+ if (dai->id != ABE_FRONTEND_DAI_LP_MEDIA)
+ return -EINVAL;
+
+ vma->vm_flags |= VM_IO | VM_RESERVED;
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ size = vma->vm_end - vma->vm_start;
+ offset = vma->vm_pgoff << PAGE_SHIFT;
+
+ err = io_remap_pfn_range(vma, vma->vm_start,
+ (ABE_DMEM_BASE_ADDRESS_MPU +
+ ABE_DMEM_BASE_OFFSET_PING_PONG + offset) >> PAGE_SHIFT,
+ size, vma->vm_page_prot);
+
+ if (err)
+ return -EAGAIN;
+
+ return 0;
+}
+
+static snd_pcm_uframes_t aess_pointer(struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t offset;
+ u32 pingpong;
+
+ abe_read_offset_from_ping_buffer(MM_DL_PORT, &pingpong);
+ offset = (snd_pcm_uframes_t)pingpong;
+
+ return offset;
+}
+
+static struct snd_pcm_ops omap_aess_pcm_ops = {
+ .open = aess_open,
+ .hw_params = aess_hw_params,
+ .prepare = aess_prepare,
+ .close = aess_close,
+ .pointer = aess_pointer,
+ .mmap = aess_mmap,
+};
+
+#if CONFIG_PM
+static int aess_suspend(struct device *dev)
+{
+ struct abe_data *abe = dev_get_drvdata(dev);
+
+ pm_runtime_get_sync(abe->dev);
+
+ aess_save_context(abe);
+
+ pm_runtime_put_sync(abe->dev);
+
+ return 0;
+}
+
+static int aess_resume(struct device *dev)
+{
+ struct abe_data *abe = dev_get_drvdata(dev);
+
+ pm_runtime_get_sync(abe->dev);
+
+ aess_restore_context(abe);
+
+ pm_runtime_put_sync(abe->dev);
+
+ return 0;
+}
+
+#else
+#define aess_suspend NULL
+#define aess_resume NULL
+#endif
+
+static const struct dev_pm_ops aess_pm_ops = {
+ .suspend = aess_suspend,
+ .resume = aess_resume,
+};
+
+static int aess_stream_event(struct snd_soc_dapm_context *dapm)
+{
+ struct snd_soc_platform *platform = dapm->platform;
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+
+ pm_runtime_get_sync(abe->dev);
+
+ if (abe->active)
+ aess_set_runtime_opp_level(abe);
+
+ pm_runtime_put_sync(abe->dev);
+
+ return 0;
+}
+
+static int abe_add_widgets(struct snd_soc_platform *platform)
+{
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ struct fw_header *hdr = &abe->hdr;
+ int i, j;
+
+ /* create equalizer controls */
+ for (i = 0; i < hdr->num_equ; i++) {
+ struct soc_enum *equalizer_enum = &abe->equalizer_enum[i];
+ struct snd_kcontrol_new *equalizer_control =
+ &abe->equalizer_control[i];
+
+ equalizer_enum->reg = i;
+ equalizer_enum->max = abe->equ_texts[i].count;
+ for (j = 0; j < abe->equ_texts[i].count; j++)
+ equalizer_enum->dtexts[j] = abe->equ_texts[i].texts[j];
+
+ equalizer_control->name = abe->equ_texts[i].name;
+ equalizer_control->private_value = (unsigned long)equalizer_enum;
+ equalizer_control->get = abe_get_equalizer;
+ equalizer_control->put = abe_put_equalizer;
+ equalizer_control->info = snd_soc_info_enum_ext1;
+ equalizer_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+
+ dev_dbg(platform->dev, "added EQU mixer: %s profiles %d\n",
+ abe->equ_texts[i].name, abe->equ_texts[i].count);
+
+ for (j = 0; j < abe->equ_texts[i].count; j++)
+ dev_dbg(platform->dev, " %s\n", equalizer_enum->dtexts[j]);
+ }
+
+ snd_soc_add_platform_controls(platform, abe->equalizer_control,
+ hdr->num_equ);
+
+ snd_soc_add_platform_controls(platform, abe_controls,
+ ARRAY_SIZE(abe_controls));
+
+ snd_soc_dapm_new_controls(&platform->dapm, abe_dapm_widgets,
+ ARRAY_SIZE(abe_dapm_widgets));
+
+ snd_soc_dapm_add_routes(&platform->dapm, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_dapm_new_widgets(&platform->dapm);
+
+ return 0;
+}
+
+static int abe_probe(struct snd_soc_platform *platform)
+{
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ u8 *fw_data;
+ int i, offset = 0;
+ int ret = 0;
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+ const struct firmware *fw;
+#endif
+
+ abe->platform = platform;
+
+ pm_runtime_enable(abe->dev);
+
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+ /* request firmware & coefficients */
+ ret = request_firmware(&fw, "omap4_abe", platform->dev);
+ if (ret != 0) {
+ dev_err(abe->dev, "Failed to load firmware: %d\n", ret);
+ return ret;
+ }
+ fw_data = fw->data;
+#else
+ fw_data = (u8 *)abe_get_default_fw();
+#endif
+
+ /* get firmware and coefficients header info */
+ memcpy(&abe->hdr, fw_data, sizeof(struct fw_header));
+ if (abe->hdr.firmware_size > ABE_MAX_FW_SIZE) {
+ dev_err(abe->dev, "Firmware too large at %d bytes: %d\n",
+ abe->hdr.firmware_size, ret);
+ ret = -EINVAL;
+ goto err_fw;
+ }
+ dev_dbg(abe->dev, "ABE firmware size %d bytes\n", abe->hdr.firmware_size);
+
+ if (abe->hdr.coeff_size > ABE_MAX_COEFF_SIZE) {
+ dev_err(abe->dev, "Coefficients too large at %d bytes: %d\n",
+ abe->hdr.coeff_size, ret);
+ ret = -EINVAL;
+ goto err_fw;
+ }
+ dev_dbg(abe->dev, "ABE coefficients size %d bytes\n", abe->hdr.coeff_size);
+
+ /* get coefficient EQU mixer strings */
+ if (abe->hdr.num_equ >= ABE_MAX_EQU) {
+ dev_err(abe->dev, "Too many equalizers got %d\n", abe->hdr.num_equ);
+ ret = -EINVAL;
+ goto err_fw;
+ }
+ abe->equ_texts = kzalloc(abe->hdr.num_equ * sizeof(struct coeff_config),
+ GFP_KERNEL);
+ if (abe->equ_texts == NULL) {
+ ret = -ENOMEM;
+ goto err_fw;
+ }
+ offset = sizeof(struct fw_header);
+ memcpy(abe->equ_texts, fw_data + offset,
+ abe->hdr.num_equ * sizeof(struct coeff_config));
+
+ /* get coefficients from firmware */
+ abe->equ[0] = kmalloc(abe->hdr.coeff_size, GFP_KERNEL);
+ if (abe->equ[0] == NULL) {
+ ret = -ENOMEM;
+ goto err_equ;
+ }
+ offset += abe->hdr.num_equ * sizeof(struct coeff_config);
+ memcpy(abe->equ[0], fw_data + offset, abe->hdr.coeff_size);
+
+ /* allocate coefficient mixer texts */
+ dev_dbg(abe->dev, "loaded %d equalizers\n", abe->hdr.num_equ);
+ for (i = 0; i < abe->hdr.num_equ; i++) {
+ dev_dbg(abe->dev, "equ %d: %s profiles %d\n", i,
+ abe->equ_texts[i].name, abe->equ_texts[i].count);
+ if (abe->equ_texts[i].count >= ABE_MAX_PROFILES) {
+ dev_err(abe->dev, "Too many profiles got %d for equ %d\n",
+ abe->equ_texts[i].count, i);
+ ret = -EINVAL;
+ goto err_texts;
+ }
+ abe->equalizer_enum[i].dtexts =
+ kzalloc(abe->equ_texts[i].count * sizeof(char *), GFP_KERNEL);
+ if (abe->equalizer_enum[i].dtexts == NULL) {
+ ret = -ENOMEM;
+ goto err_texts;
+ }
+ }
+
+ /* initialise coefficient equalizers */
+ for (i = 1; i < abe->hdr.num_equ; i++) {
+ abe->equ[i] = abe->equ[i - 1] +
+ abe->equ_texts[i - 1].count * abe->equ_texts[i - 1].coeff;
+ }
+
+ /* store ABE firmware for later context restore */
+ abe->firmware = kzalloc(abe->hdr.firmware_size, GFP_KERNEL);
+ memcpy(abe->firmware,
+ fw_data + sizeof(struct fw_header) + abe->hdr.coeff_size,
+ abe->hdr.firmware_size);
+
+ ret = request_irq(abe->irq, abe_irq_handler, 0, "ABE", (void *)abe);
+ if (ret) {
+ dev_err(platform->dev, "request for ABE IRQ %d failed %d\n",
+ abe->irq, ret);
+ goto err_texts;
+ }
+
+ /* aess_clk has to be enabled to access hal register.
+ * Disable the clk after it has been used.
+ */
+ pm_runtime_get_sync(abe->dev);
+
+ abe_init_mem(abe->io_base);
+
+ abe_reset_hal();
+
+ abe_load_fw(abe->firmware);
+
+ /* Config OPP 100 for now */
+ abe_set_opp_processing(ABE_OPP100);
+
+ /* "tick" of the audio engine */
+ abe_write_event_generator(EVENT_TIMER);
+ /* Stop the engine */
+ abe_stop_event_generator();
+ abe_disable_irq();
+
+ pm_runtime_put_sync(abe->dev);
+ abe_add_widgets(platform);
+
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+ release_firmware(fw);
+#endif
+ return ret;
+
+err_texts:
+ kfree(abe->firmware);
+ for (i = 0; i < abe->hdr.num_equ; i++)
+ kfree(abe->equalizer_enum[i].texts);
+ kfree(abe->equ[0]);
+err_equ:
+ kfree(abe->equ_texts);
+err_fw:
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+ release_firmware(fw);
+#endif
+ return ret;
+}
+
+static int abe_remove(struct snd_soc_platform *platform)
+{
+ struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+ int i;
+
+ free_irq(abe->irq, (void *)abe);
+
+ for (i = 0; i < abe->hdr.num_equ; i++)
+ kfree(abe->equalizer_enum[i].texts);
+
+ kfree(abe->equ[0]);
+ kfree(abe->equ_texts);
+ kfree(abe->firmware);
+
+ pm_runtime_disable(abe->dev);
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver omap_aess_platform = {
+ .ops = &omap_aess_pcm_ops,
+ .probe = abe_probe,
+ .remove = abe_remove,
+ .read = abe_dsp_read,
+ .write = abe_dsp_write,
+ .stream_event = aess_stream_event,
+};
+
+static int __devinit abe_engine_probe(struct platform_device *pdev)
+{
+ struct resource *res;
+ struct omap4_abe_dsp_pdata *pdata = pdev->dev.platform_data;
+ struct abe_data *abe;
+ int ret = -EINVAL, i;
+
+ abe = kzalloc(sizeof(struct abe_data), GFP_KERNEL);
+ if (abe == NULL)
+ return -ENOMEM;
+ dev_set_drvdata(&pdev->dev, abe);
+ the_abe = abe;
+
+ /* ZERO_labelID should really be 0 */
+ for (i = 0; i < ABE_ROUTES_UL + 2; i++)
+ abe->router[i] = ZERO_labelID;
+
+ for (i = 0; i < 5; i++) {
+ res = platform_get_resource_byname(pdev, IORESOURCE_MEM,
+ abe_memory_bank[i]);
+ if (res == NULL) {
+ dev_err(&pdev->dev, "no resource %s\n",
+ abe_memory_bank[i]);
+ goto err;
+ }
+ abe->io_base[i] = ioremap(res->start, resource_size(res));
+ if (!abe->io_base[i]) {
+ ret = -ENOMEM;
+ goto err;
+ }
+ }
+
+ abe->irq = platform_get_irq(pdev, 0);
+ if (abe->irq < 0) {
+ ret = abe->irq;
+ goto err;
+ }
+
+ abe->abe_pdata = pdata;
+ abe->dev = &pdev->dev;
+ mutex_init(&abe->mutex);
+ mutex_init(&abe->opp_mutex);
+
+ ret = snd_soc_register_platform(abe->dev,
+ &omap_aess_platform);
+ if (ret < 0)
+ return ret;
+
+ abe_init_debugfs(abe);
+ return ret;
+
+err:
+ for (--i; i >= 0; i--)
+ iounmap(abe->io_base[i]);
+ kfree(abe);
+ return ret;
+}
+
+static int __devexit abe_engine_remove(struct platform_device *pdev)
+{
+ struct abe_data *abe = dev_get_drvdata(&pdev->dev);
+ int i;
+
+ abe_cleanup_debugfs(abe);
+ snd_soc_unregister_platform(&pdev->dev);
+ for (i = 0; i < 5; i++)
+ iounmap(abe->io_base[i]);
+ kfree(abe);
+ return 0;
+}
+
+static struct platform_driver omap_aess_driver = {
+ .driver = {
+ .name = "aess",
+ .owner = THIS_MODULE,
+ .pm = &aess_pm_ops,
+ },
+ .probe = abe_engine_probe,
+ .remove = __devexit_p(abe_engine_remove),
+};
+
+static int __init abe_engine_init(void)
+{
+ return platform_driver_register(&omap_aess_driver);
+}
+module_init(abe_engine_init);
+
+static void __exit abe_engine_exit(void)
+{
+ platform_driver_unregister(&omap_aess_driver);
+}
+module_exit(abe_engine_exit);
+
+MODULE_DESCRIPTION("ASoC OMAP4 ABE");
+MODULE_AUTHOR("Liam Girdwood <lrg@ti.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-abe-dsp.h b/sound/soc/omap/omap-abe-dsp.h
new file mode 100644
index 0000000..5d7016e
--- /dev/null
+++ b/sound/soc/omap/omap-abe-dsp.h
@@ -0,0 +1,163 @@
+/*
+ * omap-abe-dsp.h
+ *
+ * Copyright (C) 2010 Texas Instruments
+ *
+ * Contact: Liam Girdwood <lrg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_ABE_DSP_H__
+#define __OMAP_ABE_DSP_H__
+
+#define ABE_MIXER(x) (x)
+
+#define MIX_DL1_TONES ABE_MIXER(0)
+#define MIX_DL1_VOICE ABE_MIXER(1)
+#define MIX_DL1_CAPTURE ABE_MIXER(2)
+#define MIX_DL1_MEDIA ABE_MIXER(3)
+#define MIX_DL2_TONES ABE_MIXER(4)
+#define MIX_DL2_VOICE ABE_MIXER(5)
+#define MIX_DL2_CAPTURE ABE_MIXER(6)
+#define MIX_DL2_MEDIA ABE_MIXER(7)
+#define MIX_AUDUL_TONES ABE_MIXER(8)
+#define MIX_AUDUL_MEDIA ABE_MIXER(9)
+#define MIX_AUDUL_CAPTURE ABE_MIXER(10)
+#define MIX_VXREC_TONES ABE_MIXER(11)
+#define MIX_VXREC_VOICE_PLAYBACK ABE_MIXER(12)
+#define MIX_VXREC_VOICE_CAPTURE ABE_MIXER(13)
+#define MIX_VXREC_MEDIA ABE_MIXER(14)
+#define MIX_SDT_CAPTURE ABE_MIXER(15)
+#define MIX_SDT_PLAYBACK ABE_MIXER(16)
+#define MIX_SWITCH_PDM_DL ABE_MIXER(17)
+#define MIX_SWITCH_BT_VX_DL ABE_MIXER(18)
+#define MIX_SWITCH_MM_EXT_DL ABE_MIXER(19)
+
+#define ABE_NUM_MIXERS (MIX_SWITCH_MM_EXT_DL + 1)
+
+#define ABE_MUX(x) (x + ABE_NUM_MIXERS)
+
+#define MUX_MM_UL10 ABE_MUX(0)
+#define MUX_MM_UL11 ABE_MUX(1)
+#define MUX_MM_UL12 ABE_MUX(2)
+#define MUX_MM_UL13 ABE_MUX(3)
+#define MUX_MM_UL14 ABE_MUX(4)
+#define MUX_MM_UL15 ABE_MUX(5)
+#define MUX_MM_UL16 ABE_MUX(6)
+#define MUX_MM_UL17 ABE_MUX(7)
+#define MUX_MM_UL20 ABE_MUX(8)
+#define MUX_MM_UL21 ABE_MUX(9)
+#define MUX_VX_UL0 ABE_MUX(10)
+#define MUX_VX_UL1 ABE_MUX(11)
+
+#define ABE_NUM_MUXES (MUX_VX_UL1 - MUX_MM_UL10)
+
+#define ABE_WIDGET(x) (x + ABE_NUM_MIXERS + ABE_NUM_MUXES)
+
+/* ABE AIF Frontend Widgets */
+#define W_AIF_TONES_DL ABE_WIDGET(0)
+#define W_AIF_VX_DL ABE_WIDGET(1)
+#define W_AIF_VX_UL ABE_WIDGET(2)
+#define W_AIF_MM_UL1 ABE_WIDGET(3)
+#define W_AIF_MM_UL2 ABE_WIDGET(4)
+#define W_AIF_MM_DL ABE_WIDGET(5)
+#define W_AIF_MM_DL_LP W_AIF_MM_DL
+#define W_AIF_VIB_DL ABE_WIDGET(6)
+#define W_AIF_MODEM_DL ABE_WIDGET(7)
+#define W_AIF_MODEM_UL ABE_WIDGET(8)
+
+/* ABE AIF Backend Widgets */
+#define W_AIF_PDM_UL1 ABE_WIDGET(9)
+#define W_AIF_PDM_DL1 ABE_WIDGET(10)
+#define W_AIF_PDM_DL2 ABE_WIDGET(11)
+#define W_AIF_PDM_VIB ABE_WIDGET(12)
+#define W_AIF_BT_VX_UL ABE_WIDGET(13)
+#define W_AIF_BT_VX_DL ABE_WIDGET(14)
+#define W_AIF_MM_EXT_UL ABE_WIDGET(15)
+#define W_AIF_MM_EXT_DL ABE_WIDGET(16)
+#define W_AIF_DMIC0 ABE_WIDGET(17)
+#define W_AIF_DMIC1 ABE_WIDGET(18)
+#define W_AIF_DMIC2 ABE_WIDGET(19)
+
+/* ABE ROUTE_UL MUX Widgets */
+#define W_MUX_UL00 ABE_WIDGET(20)
+#define W_MUX_UL01 ABE_WIDGET(21)
+#define W_MUX_UL02 ABE_WIDGET(22)
+#define W_MUX_UL03 ABE_WIDGET(23)
+#define W_MUX_UL04 ABE_WIDGET(24)
+#define W_MUX_UL05 ABE_WIDGET(25)
+#define W_MUX_UL06 ABE_WIDGET(26)
+#define W_MUX_UL07 ABE_WIDGET(27)
+#define W_MUX_UL10 ABE_WIDGET(28)
+#define W_MUX_UL11 ABE_WIDGET(29)
+#define W_MUX_VX00 ABE_WIDGET(30)
+#define W_MUX_VX01 ABE_WIDGET(31)
+
+/* ABE Volume and Mixer Widgets */
+#define W_MIXER_DL1 ABE_WIDGET(32)
+#define W_MIXER_DL2 ABE_WIDGET(33)
+#define W_VOLUME_DL1 ABE_WIDGET(34)
+#define W_MIXER_AUDIO_UL ABE_WIDGET(35)
+#define W_MIXER_VX_REC ABE_WIDGET(36)
+#define W_MIXER_SDT ABE_WIDGET(37)
+#define W_VSWITCH_DL1_PDM ABE_WIDGET(38)
+#define W_VSWITCH_DL1_BT_VX ABE_WIDGET(39)
+#define W_VSWITCH_DL1_MM_EXT ABE_WIDGET(40)
+
+#define ABE_NUM_WIDGETS (W_VSWITCH_DL1_MM_EXT - W_AIF_TONES_DL)
+#define ABE_WIDGET_LAST W_VSWITCH_DL1_MM_EXT
+
+#define ABE_NUM_DAPM_REG \
+ (ABE_NUM_MIXERS + ABE_NUM_MUXES + ABE_NUM_WIDGETS)
+
+#define ABE_VIRTUAL_SWITCH 0
+#define ABE_ROUTES_UL 14
+
+// TODO: OPP bitmask - Use HAL version after update
+#define ABE_OPP_25 0
+#define ABE_OPP_50 1
+#define ABE_OPP_100 2
+
+/* TODO: size in bytes of debug options */
+#define ABE_DBG_FLAG1_SIZE 0
+#define ABE_DBG_FLAG2_SIZE 0
+#define ABE_DBG_FLAG3_SIZE 0
+
+/* TODO: Pong start offset of DMEM */
+/* Ping pong buffer DMEM offset */
+#define ABE_DMEM_BASE_OFFSET_PING_PONG 0x4000
+
+/* Gain value conversion */
+#define ABE_MAX_GAIN 12000
+#define ABE_GAIN_SCALE 100
+#define abe_gain_to_val(gain) ((val + ABE_MAX_GAIN) / ABE_GAIN_SCALE)
+#define abe_val_to_gain(val) (-ABE_MAX_GAIN + (val * ABE_GAIN_SCALE))
+
+/* Firmware coefficients and equalizers */
+#define ABE_MAX_FW_SIZE (1024 * 128)
+#define ABE_MAX_COEFF_SIZE (1024 * 4)
+#define ABE_COEFF_NAME_SIZE 20
+#define ABE_COEFF_TEXT_SIZE 20
+#define ABE_COEFF_NUM_TEXTS 10
+#define ABE_MAX_EQU 10
+#define ABE_MAX_PROFILES 30
+
+void abe_dsp_shutdown(void);
+void abe_dsp_pm_get(void);
+void abe_dsp_pm_put(void);
+
+#endif /* End of __OMAP_ABE_DSP_H__ */
diff --git a/sound/soc/omap/omap-abe.c b/sound/soc/omap/omap-abe.c
new file mode 100644
index 0000000..049f8b6
--- /dev/null
+++ b/sound/soc/omap/omap-abe.c
@@ -0,0 +1,1255 @@
+/*
+ * omap-abe.c -- OMAP ALSA SoC DAI driver using Audio Backend
+ *
+ * Copyright (C) 2010 Texas Instruments
+ *
+ * Contact: Liam Girdwood <lrg@ti.com>
+ * Misael Lopez Cruz <misael.lopez@ti.com>
+ * Sebastien Guiriec <s-guiriec@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#define DEBUG
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc-dsp.h>
+
+#include <plat/dma-44xx.h>
+#include <plat/dma.h>
+#include "omap-pcm.h"
+#include "omap-abe.h"
+#include "omap-abe-dsp.h"
+#include "abe/abe_main.h"
+#include "abe/port_mgr.h"
+
+#define OMAP_ABE_FORMATS SNDRV_PCM_FMTBIT_S32_LE
+
+struct omap_abe_data {
+ /* MODEM FE*/
+ struct snd_pcm_substream *modem_substream[2];
+ struct snd_soc_dai *modem_dai;
+
+ struct abe *abe;
+
+ /* BE & FE Ports */
+ struct omap_abe_port *port[OMAP_ABE_MAX_PORT_ID + 1];
+};
+
+/*
+ * Stream DMA parameters
+ */
+static struct omap_pcm_dma_data omap_abe_dai_dma_params[7][2] = {
+{
+ {
+ .name = "Media Playback",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_0,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+ {
+ .name = "Media Capture1",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_3,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+},
+{
+ {},
+ {
+ .name = "Media Capture2",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_4,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+},
+{
+ {
+ .name = "Voice Playback",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_1,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+ {
+ .name = "Voice Capture",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_2,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+},
+{
+ {
+ .name = "Tones Playback",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_5,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },{},
+},
+{
+ {
+ .name = "Vibra Playback",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_6,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },{},
+},
+{
+ {
+ .name = "MODEM Playback",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_1,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+ {
+ .name = "MODEM Capture",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_2,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },
+},
+{
+ {
+ .name = "Low Power Playback",
+ .dma_req = OMAP44XX_DMA_ABE_REQ_0,
+ .data_type = OMAP_DMA_DATA_TYPE_S32,
+ .sync_mode = OMAP_DMA_SYNC_PACKET,
+ },{},
+},};
+
+static int modem_get_dai(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ struct snd_soc_pcm_runtime *modem_rtd;
+
+ abe_priv->modem_substream[substream->stream] =
+ snd_soc_get_dai_substream(rtd->card,
+ OMAP_ABE_BE_MM_EXT1, substream->stream);
+
+ if (abe_priv->modem_substream[substream->stream] == NULL)
+ return -ENODEV;
+
+ modem_rtd = abe_priv->modem_substream[substream->stream]->private_data;
+ abe_priv->modem_substream[substream->stream]->runtime = substream->runtime;
+ abe_priv->modem_dai = modem_rtd->cpu_dai;
+
+ return 0;
+}
+
+static void mute_be(struct snd_soc_pcm_runtime *be,
+ struct snd_soc_dai *dai, int stream)
+{
+ dev_dbg(&be->dev, "%s: %s %d\n", __func__, be->cpu_dai->name, stream);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ switch (be->dai_link->be_id) {
+ case OMAP_ABE_DAI_PDM_DL1:
+ abe_write_gain(GAINS_DL1, MUTE_GAIN, RAMP_5MS,
+ GAIN_LEFT_OFFSET);
+ abe_write_gain(GAINS_DL1, MUTE_GAIN, RAMP_5MS,
+ GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_DAI_PDM_DL2:
+ abe_write_gain(GAINS_DL2, MUTE_GAIN, RAMP_5MS,
+ GAIN_LEFT_OFFSET);
+ abe_write_gain(GAINS_DL2, MUTE_GAIN, RAMP_5MS,
+ GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_DAI_PDM_VIB:
+ case OMAP_ABE_DAI_BT_VX:
+ case OMAP_ABE_DAI_MM_FM:
+ case OMAP_ABE_DAI_MODEM:
+ break;
+ }
+ } else {
+ switch (be->dai_link->be_id) {
+ case OMAP_ABE_DAI_PDM_UL:
+ break;
+ case OMAP_ABE_DAI_BT_VX:
+ case OMAP_ABE_DAI_MM_FM:
+ case OMAP_ABE_DAI_MODEM:
+ case OMAP_ABE_DAI_DMIC0:
+ case OMAP_ABE_DAI_DMIC1:
+ case OMAP_ABE_DAI_DMIC2:
+ break;
+ }
+ }
+}
+
+static void unmute_be(struct snd_soc_pcm_runtime *be,
+ struct snd_soc_dai *dai, int stream)
+{
+ dev_dbg(&be->dev, "%s: %s %d\n", __func__, be->cpu_dai->name, stream);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ switch (be->dai_link->be_id) {
+ case OMAP_ABE_DAI_PDM_DL1:
+ abe_write_gain(GAINS_DL1, GAIN_0dB, RAMP_5MS,
+ GAIN_LEFT_OFFSET);
+ abe_write_gain(GAINS_DL1, GAIN_0dB, RAMP_5MS,
+ GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_DAI_PDM_DL2:
+ abe_write_gain(GAINS_DL2, GAIN_0dB, RAMP_5MS,
+ GAIN_LEFT_OFFSET);
+ abe_write_gain(GAINS_DL2, GAIN_0dB, RAMP_5MS,
+ GAIN_RIGHT_OFFSET);
+ break;
+ case OMAP_ABE_DAI_PDM_VIB:
+ case OMAP_ABE_DAI_BT_VX:
+ case OMAP_ABE_DAI_MM_FM:
+ case OMAP_ABE_DAI_MODEM:
+ break;
+ }
+ } else {
+
+ switch (be->dai_link->be_id) {
+ case OMAP_ABE_DAI_PDM_UL:
+ break;
+ case OMAP_ABE_DAI_BT_VX:
+ case OMAP_ABE_DAI_MM_FM:
+ case OMAP_ABE_DAI_MODEM:
+ case OMAP_ABE_DAI_DMIC0:
+ case OMAP_ABE_DAI_DMIC1:
+ case OMAP_ABE_DAI_DMIC2:
+ break;
+ }
+ }
+}
+
+static void enable_be_port(struct snd_soc_pcm_runtime *be,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ abe_data_format_t format;
+
+ dev_dbg(&be->dev, "%s: %s %d\n", __func__, be->cpu_dai->name, stream);
+
+ switch (be->dai_link->be_id) {
+ /* McPDM Downlink is special case and handled by McPDM driver */
+ case OMAP_ABE_DAI_PDM_DL1:
+ case OMAP_ABE_DAI_PDM_DL2:
+ case OMAP_ABE_DAI_PDM_VIB:
+ case OMAP_ABE_DAI_PDM_UL:
+ break;
+ case OMAP_ABE_DAI_BT_VX:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ /* port can only be configured if it's not running */
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_DL]))
+ return;
+
+ /* BT_DL connection to McBSP 1 ports */
+ format.f = 8000;
+ format.samp_format = MONO_RSHIFTED_16;
+ abe_connect_serial_port(BT_VX_DL_PORT, &format, MCBSP1_TX);
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_DL]);
+ } else {
+
+ /* port can only be configured if it's not running */
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_UL]))
+ return;
+
+ /* BT_UL connection to McBSP 1 ports */
+ format.f = 8000;
+ format.samp_format = MONO_RSHIFTED_16;
+ abe_connect_serial_port(BT_VX_UL_PORT, &format, MCBSP1_RX);
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_UL]);
+ }
+ break;
+ case OMAP_ABE_DAI_MM_FM:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ /* port can only be configured if it's not running */
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_DL]))
+ return;
+
+ /* MM_EXT connection to McBSP 2 ports */
+ format.f = 48000;
+ format.samp_format = STEREO_RSHIFTED_16;
+ abe_connect_serial_port(MM_EXT_OUT_PORT, &format, MCBSP2_TX);
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_DL]);
+ } else {
+
+ /* port can only be configured if it's not running */
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_UL]))
+ return;
+
+ /* MM_EXT connection to McBSP 2 ports */
+ format.f = 48000;
+ format.samp_format = STEREO_RSHIFTED_16;
+ abe_connect_serial_port(MM_EXT_IN_PORT, &format, MCBSP2_RX);
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_UL]);
+ }
+ break;
+ case OMAP_ABE_DAI_DMIC0:
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_DMIC0]);
+ break;
+ case OMAP_ABE_DAI_DMIC1:
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_DMIC1]);
+ break;
+ case OMAP_ABE_DAI_DMIC2:
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_DMIC2]);
+ break;
+ }
+}
+
+static void enable_fe_port(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+
+ switch(dai->id) {
+ case ABE_FRONTEND_DAI_MEDIA:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_MM_DL1]);
+ else
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_MM_UL1]);
+ break;
+ case ABE_FRONTEND_DAI_LP_MEDIA:
+ abe_enable_data_transfer(MM_DL_PORT);
+ break;
+ case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_MM_UL2]);
+ break;
+ case ABE_FRONTEND_DAI_MODEM:
+ case ABE_FRONTEND_DAI_VOICE:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_VX_DL]);
+ else
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_VX_UL]);
+ break;
+ case ABE_FRONTEND_DAI_TONES:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_TONES]);
+ break;
+ case ABE_FRONTEND_DAI_VIBRA:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_enable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_VIB]);
+ break;
+ }
+}
+
+static void disable_be_port(struct snd_soc_pcm_runtime *be,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(&be->dev, "%s: %s %d\n", __func__, be->cpu_dai->name, stream);
+
+ switch (be->dai_link->be_id) {
+ /* McPDM Downlink is special case and handled by McPDM driver */
+ case OMAP_ABE_DAI_PDM_DL1:
+ case OMAP_ABE_DAI_PDM_DL2:
+ case OMAP_ABE_DAI_PDM_VIB:
+ case OMAP_ABE_DAI_PDM_UL:
+ break;
+ case OMAP_ABE_DAI_BT_VX:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_DL]);
+ else
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_BT_VX_UL]);
+ break;
+ case OMAP_ABE_DAI_MM_FM:
+ case OMAP_ABE_DAI_MODEM:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_DL]);
+ else
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_MM_EXT_UL]);
+ break;
+ case OMAP_ABE_DAI_DMIC0:
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_DMIC0]);
+ break;
+ case OMAP_ABE_DAI_DMIC1:
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_DMIC1]);
+ break;
+ case OMAP_ABE_DAI_DMIC2:
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_DMIC2]);
+ break;
+ }
+}
+
+static void disable_fe_port(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+
+ switch(dai->id) {
+ case ABE_FRONTEND_DAI_MEDIA:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_MM_DL1]);
+ else
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_MM_UL1]);
+ break;
+ case ABE_FRONTEND_DAI_LP_MEDIA:
+ abe_disable_data_transfer(MM_DL_PORT);
+ break;
+ case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+ if (stream == SNDRV_PCM_STREAM_CAPTURE)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_MM_UL2]);
+ break;
+ case ABE_FRONTEND_DAI_MODEM:
+ case ABE_FRONTEND_DAI_VOICE:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_VX_DL]);
+ else
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_VX_UL]);
+ break;
+ case ABE_FRONTEND_DAI_TONES:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_TONES]);
+ break;
+ case ABE_FRONTEND_DAI_VIBRA:
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ omap_abe_port_disable(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_FE_PORT_VIB]);
+ break;
+ }
+}
+
+static void mute_fe_port(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+
+ switch(dai->id) {
+ case ABE_FRONTEND_DAI_MEDIA:
+ case ABE_FRONTEND_DAI_LP_MEDIA:
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+ break;
+ case ABE_FRONTEND_DAI_VOICE:
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+ break;
+ case ABE_FRONTEND_DAI_TONES:
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
+ abe_mute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
+ abe_mute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+ break;
+ case ABE_FRONTEND_DAI_VIBRA:
+ case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+ break;
+ }
+}
+
+static void unmute_fe_port(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(&fe->dev, "%s: %s %d\n", __func__, dai->name, stream);
+
+ switch(dai->id) {
+ case ABE_FRONTEND_DAI_MEDIA:
+ case ABE_FRONTEND_DAI_LP_MEDIA:
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_MM_DL);
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_MM_DL);
+ break;
+ case ABE_FRONTEND_DAI_VOICE:
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_VX_DL);
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_VX_DL);
+ break;
+ case ABE_FRONTEND_DAI_TONES:
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL2]))
+ abe_unmute_gain(MIXDL2, MIX_DL2_INPUT_TONES);
+ if (omap_abe_port_is_enabled(abe_priv->abe,
+ abe_priv->port[OMAP_ABE_BE_PORT_PDM_DL1]))
+ abe_unmute_gain(MIXDL1, MIX_DL1_INPUT_TONES);
+ break;
+ case ABE_FRONTEND_DAI_VIBRA:
+ case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+ break;
+ }
+}
+
+static void capture_trigger(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai, int cmd)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct snd_soc_dsp_params *dsp_params, *tmp;
+ struct snd_pcm_substream *be_substream;
+ int stream = substream->stream;
+
+ dev_dbg(&fe->dev, "%s: %s %d\n", __func__, fe->cpu_dai->name, stream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+
+ /* mute and enable BE ports */
+ list_for_each_entry_safe(dsp_params, tmp, &fe->dsp[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+ /* does this trigger() apply to this BE and stream ? */
+ if (!snd_soc_dsp_is_trigger_for_be(fe, be, stream))
+ continue;
+
+ /* is the BE already in the trigger START state ? */
+ if (dsp_params->state == SND_SOC_DSP_LINK_STATE_START)
+ continue;
+
+ be_substream = snd_soc_dsp_get_substream(dsp_params->be, stream);
+
+ /* mute the BE port */
+ mute_be(be, dai, stream);
+
+ /* enable the BE port */
+ enable_be_port(be, dai, stream);
+
+ /* DAI work must be started/stopped at least 250us after ABE */
+ udelay(250);
+
+ /* trigger the BE port */
+ snd_soc_dai_trigger(be_substream, cmd, be->cpu_dai);
+ }
+
+ /* does this trigger() apply to the FE ? */
+ if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
+ /* Enable Frontend sDMA */
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+ enable_fe_port(substream, dai, stream);
+ }
+
+ /* Restore ABE GAINS AMIC */
+ list_for_each_entry(dsp_params, &fe->dsp[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+ /* does this trigger() apply to this BE and stream ? */
+ if (!snd_soc_dsp_is_trigger_for_be(fe, be, stream))
+ continue;
+
+ /* unmute this BE port */
+ unmute_be(be, dai, stream);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /* Enable sDMA */
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+ enable_fe_port(substream, dai, stream);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ /* Disable sDMA */
+ disable_fe_port(substream, dai, stream);
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+
+ /* does this trigger() apply to the FE ? */
+ if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
+ /* Disable sDMA */
+ disable_fe_port(substream, dai, stream);
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+ }
+
+ /* disable BE ports */
+ list_for_each_entry_safe(dsp_params, tmp, &fe->dsp[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+ /* does this trigger() apply to this BE and stream ? */
+ if (!snd_soc_dsp_is_trigger_for_be(fe, be, stream))
+ continue;
+
+ /* only STOP BE in FREE state */
+ /* REVISIT: Investigate the appropriate state to check against */
+ //if (dsp_params->state != SND_SOC_DSP_LINK_STATE_FREE)
+ // continue;
+
+ be_substream = snd_soc_dsp_get_substream(dsp_params->be, stream);
+
+ /* disable the BE port */
+ disable_be_port(be, dai, stream);
+
+ /* DAI work must be started/stopped at least 250us after ABE */
+ udelay(250);
+
+ /* trigger BE port */
+ snd_soc_dai_trigger(be_substream, cmd, be->cpu_dai);
+ }
+ break;
+ default:
+ break;
+ }
+}
+
+static void playback_trigger(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai, int cmd)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct snd_soc_dsp_params *dsp_params, *tmp;
+ struct snd_pcm_substream *be_substream;
+ int stream = substream->stream;
+
+ dev_dbg(&fe->dev, "%s: %s %d\n", __func__, fe->cpu_dai->name, stream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+
+ /* mute and enable ports */
+ list_for_each_entry_safe(dsp_params, tmp, &fe->dsp[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+ /* does this trigger() apply to the FE ? */
+ if (!snd_soc_dsp_is_trigger_for_be(fe, be, stream))
+ continue;
+
+ /* is the BE already in the trigger START state ? */
+ if (dsp_params->state == SND_SOC_DSP_LINK_STATE_START)
+ continue;
+
+ be_substream = snd_soc_dsp_get_substream(dsp_params->be, stream);
+
+ /* mute BE port */
+ mute_be(be, dai, stream);
+
+ /* enabled BE port */
+ enable_be_port(be, dai, stream);
+
+ /* DAI work must be started/stopped at least 250us after ABE */
+ udelay(250);
+
+ /* trigger BE port */
+ snd_soc_dai_trigger(be_substream, cmd, be->cpu_dai);
+
+ /* unmute the BE port */
+ unmute_be(be, dai, stream);
+ }
+
+ /* does this trigger() apply to the FE ? */
+ if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
+
+ /* Enable Frontend sDMA */
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+ enable_fe_port(substream, dai, stream);
+
+ /* unmute FE port */
+ unmute_fe_port(substream, dai, stream);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /* Enable Frontend sDMA */
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+ enable_fe_port(substream, dai, stream);
+
+ /* unmute FE port */
+ unmute_fe_port(substream, dai, stream);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ /* disable Frontend sDMA */
+ disable_fe_port(substream, dai, stream);
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+
+ /* mute FE port */
+ mute_fe_port(substream, dai, stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+
+ /* does this trigger() apply to the FE ? */
+ if (snd_soc_dsp_is_trigger_for_fe(fe, stream)) {
+
+ /* disable the transfer */
+ disable_fe_port(substream, dai, stream);
+ snd_soc_dsp_platform_trigger(substream, cmd, fe->platform);
+
+ /* mute FE port */
+ mute_fe_port(substream, dai, stream);
+ }
+
+ /* disable BE ports */
+ list_for_each_entry_safe(dsp_params, tmp, &fe->dsp[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dsp_params->be;
+
+ /* does this trigger() apply to this BE and stream ? */
+ if (!snd_soc_dsp_is_trigger_for_be(fe, be, stream))
+ continue;
+
+ /* only STOP BE in FREE state */
+ if (dsp_params->state != SND_SOC_DSP_LINK_STATE_FREE)
+ continue;
+
+ be_substream = snd_soc_dsp_get_substream(dsp_params->be, stream);
+
+ /* disable the BE */
+ disable_be_port(be, dai, stream);
+
+ /* DAI work must be started/stopped at least 250us after ABE */
+ udelay(250);
+
+ /* trigger the BE port */
+ snd_soc_dai_trigger(be_substream, cmd, be->cpu_dai);
+ }
+ break;
+ default:
+ break;
+ }
+}
+
+static int omap_abe_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+
+ ret = modem_get_dai(substream, dai);
+ if (ret < 0) {
+ dev_err(dai->dev, "failed to get MODEM DAI\n");
+ return ret;
+ }
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d\n",
+ __func__, substream->stream);
+
+ ret = snd_soc_dai_startup(abe_priv->modem_substream[substream->stream],
+ abe_priv->modem_dai);
+ if (ret < 0) {
+ dev_err(abe_priv->modem_dai->dev, "failed to open DAI %d\n", ret);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static int omap_abe_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ abe_data_format_t format;
+ abe_dma_t dma_sink;
+ abe_dma_t dma_params;
+ int ret;
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ switch (params_channels(params)) {
+ case 1:
+ if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE)
+ format.samp_format = MONO_RSHIFTED_16;
+ else
+ format.samp_format = MONO_MSB;
+ break;
+ case 2:
+ if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE)
+ format.samp_format = STEREO_16_16;
+ else
+ format.samp_format = STEREO_MSB;
+ break;
+ case 3:
+ format.samp_format = THREE_MSB;
+ break;
+ case 4:
+ format.samp_format = FOUR_MSB;
+ break;
+ case 5:
+ format.samp_format = FIVE_MSB;
+ break;
+ case 6 :
+ format.samp_format = SIX_MSB;
+ break;
+ case 7 :
+ format.samp_format = SEVEN_MSB;
+ break;
+ case 8:
+ format.samp_format = EIGHT_MSB;
+ break;
+ default:
+ dev_err(dai->dev, "%d channels not supported",
+ params_channels(params));
+ return -EINVAL;
+ }
+
+ format.f = params_rate(params);
+
+ switch (dai->id) {
+ case ABE_FRONTEND_DAI_MEDIA:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ abe_connect_cbpr_dmareq_port(MM_DL_PORT, &format, ABE_CBPR0_IDX,
+ &dma_sink);
+ abe_read_port_address(MM_DL_PORT, &dma_params);
+ } else {
+ abe_connect_cbpr_dmareq_port(MM_UL_PORT, &format, ABE_CBPR3_IDX,
+ &dma_sink);
+ abe_read_port_address(MM_UL_PORT, &dma_params);
+ }
+ break;
+ case ABE_FRONTEND_DAI_LP_MEDIA:
+ return 0;
+ break;
+ case ABE_FRONTEND_DAI_MEDIA_CAPTURE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return -EINVAL;
+ else {
+ abe_connect_cbpr_dmareq_port(MM_UL2_PORT, &format, ABE_CBPR4_IDX,
+ &dma_sink);
+ abe_read_port_address(MM_UL2_PORT, &dma_params);
+ }
+ break;
+ case ABE_FRONTEND_DAI_VOICE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ abe_connect_cbpr_dmareq_port(VX_DL_PORT, &format, ABE_CBPR1_IDX,
+ &dma_sink);
+ abe_read_port_address(VX_DL_PORT, &dma_params);
+ } else {
+ abe_connect_cbpr_dmareq_port(VX_UL_PORT, &format, ABE_CBPR2_IDX,
+ &dma_sink);
+ abe_read_port_address(VX_UL_PORT, &dma_params);
+ }
+ break;
+ case ABE_FRONTEND_DAI_TONES:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ abe_connect_cbpr_dmareq_port(TONES_DL_PORT, &format, ABE_CBPR5_IDX,
+ &dma_sink);
+ abe_read_port_address(TONES_DL_PORT, &dma_params);
+ } else
+ return -EINVAL;
+ break;
+ case ABE_FRONTEND_DAI_VIBRA:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ abe_connect_cbpr_dmareq_port(VIB_DL_PORT, &format, ABE_CBPR6_IDX,
+ &dma_sink);
+ abe_read_port_address(VIB_DL_PORT, &dma_params);
+ } else
+ return -EINVAL;
+ break;
+ case ABE_FRONTEND_DAI_MODEM:
+ /* MODEM is special case where data IO is performed by McBSP2
+ * directly onto VX_DL and VX_UL (instead of SDMA).
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Vx_DL connection to McBSP 2 ports */
+ format.samp_format = STEREO_RSHIFTED_16;
+ abe_connect_serial_port(VX_DL_PORT, &format, MCBSP2_RX);
+ abe_read_port_address(VX_DL_PORT, &dma_params);
+ } else {
+ /* Vx_UL connection to McBSP 2 ports */
+ format.samp_format = STEREO_RSHIFTED_16;
+ abe_connect_serial_port(VX_UL_PORT, &format, MCBSP2_TX);
+ abe_read_port_address(VX_UL_PORT, &dma_params);
+ }
+ break;
+ }
+
+ /* configure frontend SDMA data */
+ omap_abe_dai_dma_params[dai->id][substream->stream].port_addr =
+ (unsigned long)dma_params.data;
+ omap_abe_dai_dma_params[dai->id][substream->stream].packet_size =
+ dma_params.iter;
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+ /* call hw_params on McBSP with correct DMA data */
+ snd_soc_dai_set_dma_data(abe_priv->modem_dai, substream,
+ &omap_abe_dai_dma_params[dai->id][substream->stream]);
+
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d\n",
+ __func__, substream->stream);
+
+ ret = snd_soc_dai_hw_params(abe_priv->modem_substream[substream->stream],
+ params, abe_priv->modem_dai);
+ if (ret < 0)
+ dev_err(abe_priv->modem_dai->dev, "MODEM hw_params failed\n");
+ return ret;
+ }
+
+ snd_soc_dai_set_dma_data(dai, substream,
+ &omap_abe_dai_dma_params[dai->id][substream->stream]);
+
+ return 0;
+}
+
+static int omap_abe_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+ ret = snd_soc_dai_prepare(abe_priv->modem_substream[substream->stream],
+ abe_priv->modem_dai);
+
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d\n",
+ __func__, substream->stream);
+
+ if (ret < 0) {
+ dev_err(abe_priv->modem_dai->dev, "MODEM prepare failed\n");
+ return ret;
+ }
+ }
+ return ret;
+}
+
+static int omap_abe_dai_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s: %s cmd %d\n", __func__, dai->name, cmd);
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d cmd %d\n",
+ __func__, substream->stream, cmd);
+
+ ret = snd_soc_dai_trigger(abe_priv->modem_substream[substream->stream],
+ cmd, abe_priv->modem_dai);
+ if (ret < 0) {
+ dev_err(abe_priv->modem_dai->dev, "MODEM trigger failed\n");
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static int omap_abe_dai_bespoke_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s: %s cmd %d\n", __func__, dai->name, cmd);
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d cmd %d\n",
+ __func__, substream->stream, cmd);
+
+ ret = snd_soc_dai_trigger(abe_priv->modem_substream[substream->stream],
+ cmd, abe_priv->modem_dai);
+ if (ret < 0) {
+ dev_err(abe_priv->modem_dai->dev, "MODEM trigger failed\n");
+ return ret;
+ }
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ playback_trigger(substream, dai, cmd);
+ else
+ capture_trigger(substream, dai, cmd);
+
+ return ret;
+}
+
+static int omap_abe_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+ int ret = 0;
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d\n",
+ __func__, substream->stream);
+
+ ret = snd_soc_dai_hw_free(abe_priv->modem_substream[substream->stream],
+ abe_priv->modem_dai);
+ if (ret < 0) {
+ dev_err(abe_priv->modem_dai->dev, "MODEM hw_free failed\n");
+ return ret;
+ }
+ }
+ return ret;
+}
+
+static void omap_abe_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+ if (dai->id == ABE_FRONTEND_DAI_MODEM) {
+ dev_dbg(abe_priv->modem_dai->dev, "%s: MODEM stream %d\n",
+ __func__, substream->stream);
+
+ snd_soc_dai_shutdown(abe_priv->modem_substream[substream->stream],
+ abe_priv->modem_dai);
+ }
+}
+
+static int omap_abe_dai_probe(struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv;
+ int i;
+
+ abe_priv = kzalloc(sizeof(struct omap_abe_data), GFP_KERNEL);
+ if (abe_priv == NULL)
+ return -ENOMEM;
+
+ abe_priv->abe = omap_abe_port_mgr_get();
+ if (!abe_priv->abe)
+ goto err;
+
+ for (i = 0; i <= OMAP_ABE_MAX_PORT_ID; i++) {
+
+ abe_priv->port[i] = omap_abe_port_open(abe_priv->abe, i);
+ if (abe_priv->port[i] == NULL) {
+ for (--i; i >= 0; i--)
+ omap_abe_port_close(abe_priv->abe, abe_priv->port[i]);
+
+ goto err_port;
+ }
+ }
+
+ snd_soc_dai_set_drvdata(dai, abe_priv);
+ return 0;
+
+err_port:
+ omap_abe_port_mgr_put(abe_priv->abe);
+err:
+ kfree(abe_priv);
+ return -ENOMEM;
+}
+
+static int omap_abe_dai_remove(struct snd_soc_dai *dai)
+{
+ struct omap_abe_data *abe_priv = snd_soc_dai_get_drvdata(dai);
+
+ omap_abe_port_mgr_put(abe_priv->abe);
+ kfree(abe_priv);
+ return 0;
+}
+
+static struct snd_soc_dai_ops omap_abe_dai_ops = {
+ .startup = omap_abe_dai_startup,
+ .shutdown = omap_abe_dai_shutdown,
+ .hw_params = omap_abe_dai_hw_params,
+ .hw_free = omap_abe_dai_hw_free,
+ .prepare = omap_abe_dai_prepare,
+ .trigger = omap_abe_dai_trigger,
+ .bespoke_trigger = omap_abe_dai_bespoke_trigger,
+};
+
+static struct snd_soc_dai_driver omap_abe_dai[] = {
+ { /* Multimedia Playback and Capture */
+ .name = "MultiMedia1",
+ .probe = omap_abe_dai_probe,
+ .remove = omap_abe_dai_remove,
+ .playback = {
+ .stream_name = "MultiMedia1 Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .capture = {
+ .stream_name = "MultiMedia1 Capture",
+ .channels_min = 2,
+ .channels_max = 8,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+ { /* Multimedia Capture */
+ .name = "MultiMedia2",
+ .capture = {
+ .stream_name = "MultiMedia2 Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+ { /* Voice Playback and Capture */
+ .name = "Voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+ { /* Tones Playback */
+ .name = "Tones",
+ .playback = {
+ .stream_name = "Tones Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+ { /* Vibra */
+ .name = "Vibra",
+ .playback = {
+ .stream_name = "Vibra Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .formats = OMAP_ABE_FORMATS,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+ { /* MODEM Voice Playback and Capture */
+ .name = "MODEM",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = OMAP_ABE_FORMATS | SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = OMAP_ABE_FORMATS | SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+ { /* Low Power HiFi Playback */
+ .name = "MultiMedia1 LP",
+ .playback = {
+ .stream_name = "MultiMedia1 LP Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = OMAP_ABE_FORMATS | SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &omap_abe_dai_ops,
+ },
+};
+
+static int __devinit omap_abe_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_dais(&pdev->dev, omap_abe_dai,
+ ARRAY_SIZE(omap_abe_dai));
+}
+
+static int __devexit omap_abe_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(omap_abe_dai));
+ return 0;
+}
+
+static struct platform_driver omap_abe_driver = {
+ .driver = {
+ .name = "omap-abe-dai",
+ .owner = THIS_MODULE,
+ },
+ .probe = omap_abe_probe,
+ .remove = __devexit_p(omap_abe_remove),
+};
+
+static int __init omap_abe_init(void)
+{
+ return platform_driver_register(&omap_abe_driver);
+}
+module_init(omap_abe_init);
+
+static void __exit omap_abe_exit(void)
+{
+ platform_driver_unregister(&omap_abe_driver);
+}
+module_exit(omap_abe_exit);
+
+MODULE_AUTHOR("Liam Girdwood <lrg@ti.com>");
+MODULE_DESCRIPTION("OMAP ABE SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-abe.h b/sound/soc/omap/omap-abe.h
new file mode 100644
index 0000000..f6fad97
--- /dev/null
+++ b/sound/soc/omap/omap-abe.h
@@ -0,0 +1,59 @@
+/*
+ * omap-abe.h
+ *
+ * Copyright (C) 2010 Texas Instruments
+ *
+ * Contact: Liam Girdwood <lrg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_ABE_H__
+#define __OMAP_ABE_H__
+
+#define ABE_FRONTEND_DAI_MEDIA 0
+#define ABE_FRONTEND_DAI_MEDIA_CAPTURE 1
+#define ABE_FRONTEND_DAI_VOICE 2
+#define ABE_FRONTEND_DAI_TONES 3
+#define ABE_FRONTEND_DAI_VIBRA 4
+#define ABE_FRONTEND_DAI_MODEM 5
+#define ABE_FRONTEND_DAI_LP_MEDIA 6
+
+/* This must currently match the BE order in DSP */
+#define OMAP_ABE_DAI_PDM_UL 0
+#define OMAP_ABE_DAI_PDM_DL1 1
+#define OMAP_ABE_DAI_PDM_DL2 2
+#define OMAP_ABE_DAI_PDM_VIB 3
+#define OMAP_ABE_DAI_BT_VX 4
+#define OMAP_ABE_DAI_MM_FM 5
+#define OMAP_ABE_DAI_MODEM 6
+#define OMAP_ABE_DAI_DMIC0 7
+#define OMAP_ABE_DAI_DMIC1 8
+#define OMAP_ABE_DAI_DMIC2 9
+
+#define OMAP_ABE_BE_PDM_DL1 "PDM-DL1"
+#define OMAP_ABE_BE_PDM_UL1 "PDM-UL1"
+#define OMAP_ABE_BE_PDM_DL2 "PDM-DL2"
+#define OMAP_ABE_BE_PDM_VIB "PDM-VIB"
+#define OMAP_ABE_BE_BT_VX "BT-VX"
+#define OMAP_ABE_BE_MM_EXT0 "FM-EXT"
+#define OMAP_ABE_BE_MM_EXT1 "MODEM-EXT"
+#define OMAP_ABE_BE_DMIC0 "DMIC0"
+#define OMAP_ABE_BE_DMIC1 "DMIC1"
+#define OMAP_ABE_BE_DMIC2 "DMIC2"
+
+
+#endif /* End of __OMAP_MCPDM_H__ */
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 5cfcc65..1a970a6 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -6,8 +6,7 @@
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
+ * the Free Software Foundation; only version 2 of the License.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c88d974..4522d15 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -39,6 +39,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/soc-dpcm.h>
#include <sound/initval.h>
#define CREATE_TRACE_POINTS
@@ -59,12 +60,22 @@
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
+int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
+int soc_dpcm_be_ac97_cpu_dai_suspend(struct snd_soc_pcm_runtime *fe);
+int soc_dpcm_be_ac97_cpu_dai_resume(struct snd_soc_pcm_runtime *fe);
+int soc_dpcm_be_cpu_dai_resume(struct snd_soc_pcm_runtime *fe);
+int soc_dpcm_be_cpu_dai_suspend(struct snd_soc_pcm_runtime *fe);
+int soc_dpcm_be_platform_suspend(struct snd_soc_pcm_runtime *fe);
+int soc_dpcm_be_platform_resume(struct snd_soc_pcm_runtime *fe);
+
/*
* This is a timeout to do a DAPM powerdown after a stream is closed().
* It can be used to eliminate pops between different playback streams, e.g.
* between two audio tracks.
*/
-static int pmdown_time = 5000;
+static int pmdown_time;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
@@ -201,6 +212,12 @@
static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set);
#ifdef CONFIG_DEBUG_FS
+static int codec_reg_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
size_t count, loff_t *ppos)
{
@@ -258,7 +275,7 @@
}
static const struct file_operations codec_reg_fops = {
- .open = simple_open,
+ .open = codec_reg_open_file,
.read = codec_reg_read_file,
.write = codec_reg_write_file,
.llseek = default_llseek,
@@ -271,7 +288,8 @@
codec->debugfs_codec_root = debugfs_create_dir(codec->name,
debugfs_card_root);
if (!codec->debugfs_codec_root) {
- dev_warn(codec->dev, "Failed to create codec debugfs directory\n");
+ printk(KERN_WARNING
+ "ASoC: Failed to create codec debugfs directory\n");
return;
}
@@ -284,7 +302,8 @@
codec->debugfs_codec_root,
codec, &codec_reg_fops);
if (!codec->debugfs_reg)
- dev_warn(codec->dev, "Failed to create codec register debugfs file\n");
+ printk(KERN_WARNING
+ "ASoC: Failed to create codec register debugfs file\n");
snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root);
}
@@ -301,13 +320,12 @@
platform->debugfs_platform_root = debugfs_create_dir(platform->name,
debugfs_card_root);
if (!platform->debugfs_platform_root) {
- dev_warn(platform->dev,
- "Failed to create platform debugfs directory\n");
+ printk(KERN_WARNING
+ "ASoC: Failed to create platform debugfs directory\n");
return;
}
- snd_soc_dapm_debugfs_init(&platform->dapm,
- platform->debugfs_platform_root);
+ snd_soc_dapm_debugfs_init(&platform->dapm, platform->debugfs_platform_root);
}
static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform)
@@ -465,6 +483,35 @@
}
#endif
+struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
+ const char *dai_link, int stream)
+{
+ int i;
+
+ for (i = 0; i < card->num_links; i++) {
+ if (card->rtd[i].dai_link->no_pcm &&
+ !strcmp(card->rtd[i].dai_link->name, dai_link))
+ return card->rtd[i].pcm->streams[stream].substream;
+ }
+ dev_dbg(card->dev, "failed to find dai link %s\n", dai_link);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream);
+
+struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
+ const char *dai_link)
+{
+ int i;
+
+ for (i = 0; i < card->num_links; i++) {
+ if (!strcmp(card->rtd[i].dai_link->name, dai_link))
+ return &card->rtd[i];
+ }
+ dev_dbg(card->dev, "failed to find rtd %s\n", dai_link);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
+
#ifdef CONFIG_SND_SOC_AC97_BUS
/* unregister ac97 codec */
static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
@@ -527,16 +574,22 @@
struct snd_soc_dai *dai = card->rtd[i].codec_dai;
struct snd_soc_dai_driver *drv = dai->driver;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- if (drv->ops->digital_mute && dai->playback_active)
- drv->ops->digital_mute(dai, 1);
+ if (card->rtd[i].dai_link->dynamic)
+ soc_dpcm_be_digital_mute(&card->rtd[i], 1);
+ else {
+ if (drv->ops->digital_mute && dai->playback_active)
+ drv->ops->digital_mute(dai, 1);
+ }
}
/* suspend all pcms */
for (i = 0; i < card->num_rtd; i++) {
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
snd_pcm_suspend_all(card->rtd[i].pcm);
@@ -549,14 +602,20 @@
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
struct snd_soc_platform *platform = card->rtd[i].platform;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control)
- cpu_dai->driver->suspend(cpu_dai);
- if (platform->driver->suspend && !platform->suspended) {
- platform->driver->suspend(cpu_dai);
- platform->suspended = 1;
+ if (card->rtd[i].dai_link->dynamic) {
+ soc_dpcm_be_cpu_dai_suspend(&card->rtd[i]);
+ soc_dpcm_be_platform_suspend(&card->rtd[i]);
+ } else {
+ if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control)
+ cpu_dai->driver->suspend(cpu_dai);
+ if (platform->driver->suspend && !platform->suspended) {
+ platform->driver->suspend(cpu_dai);
+ platform->suspended = 1;
+ }
}
}
@@ -567,20 +626,19 @@
}
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
+ struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- snd_soc_dapm_stream_event(&card->rtd[i],
- SNDRV_PCM_STREAM_PLAYBACK,
- codec_dai,
- SND_SOC_DAPM_STREAM_SUSPEND);
+ if (driver->playback.stream_name != NULL)
+ snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_SUSPEND);
- snd_soc_dapm_stream_event(&card->rtd[i],
- SNDRV_PCM_STREAM_CAPTURE,
- codec_dai,
- SND_SOC_DAPM_STREAM_SUSPEND);
+ if (driver->capture.stream_name != NULL)
+ snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_SUSPEND);
}
/* suspend all CODECs */
@@ -590,17 +648,6 @@
if (!codec->suspended && codec->driver->suspend) {
switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_STANDBY:
- /*
- * If the CODEC is capable of idle
- * bias off then being in STANDBY
- * means it's doing something,
- * otherwise fall through.
- */
- if (codec->dapm.idle_bias_off) {
- dev_dbg(codec->dev,
- "idle_bias_off CODEC on over suspend\n");
- break;
- }
case SND_SOC_BIAS_OFF:
codec->driver->suspend(codec);
codec->suspended = 1;
@@ -616,11 +663,15 @@
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control)
- cpu_dai->driver->suspend(cpu_dai);
+ if (card->rtd[i].dai_link->dynamic)
+ soc_dpcm_be_ac97_cpu_dai_suspend(&card->rtd[i]);
+ else
+ if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control)
+ cpu_dai->driver->suspend(cpu_dai);
}
if (card->suspend_post)
@@ -656,11 +707,15 @@
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- if (cpu_dai->driver->resume && cpu_dai->driver->ac97_control)
- cpu_dai->driver->resume(cpu_dai);
+ if (card->rtd[i].dai_link->dynamic)
+ soc_dpcm_be_ac97_cpu_dai_resume(&card->rtd[i]);
+ else
+ if (cpu_dai->driver->resume && cpu_dai->driver->ac97_control)
+ cpu_dai->driver->resume(cpu_dai);
}
list_for_each_entry(codec, &card->codec_dev_list, card_list) {
@@ -683,18 +738,19 @@
}
for (i = 0; i < card->num_rtd; i++) {
- struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai;
+ struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- snd_soc_dapm_stream_event(&card->rtd[i],
- SNDRV_PCM_STREAM_PLAYBACK, codec_dai,
- SND_SOC_DAPM_STREAM_RESUME);
+ if (driver->playback.stream_name != NULL)
+ snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_RESUME);
- snd_soc_dapm_stream_event(&card->rtd[i],
- SNDRV_PCM_STREAM_CAPTURE, codec_dai,
- SND_SOC_DAPM_STREAM_RESUME);
+ if (driver->capture.stream_name != NULL)
+ snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_RESUME);
}
/* unmute any active DACs */
@@ -702,25 +758,36 @@
struct snd_soc_dai *dai = card->rtd[i].codec_dai;
struct snd_soc_dai_driver *drv = dai->driver;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- if (drv->ops->digital_mute && dai->playback_active)
- drv->ops->digital_mute(dai, 0);
+ if (card->rtd[i].dai_link->dynamic)
+ soc_dpcm_be_digital_mute(&card->rtd[i], 0);
+ else {
+ if (drv->ops->digital_mute && dai->playback_active)
+ drv->ops->digital_mute(dai, 0);
+ }
}
for (i = 0; i < card->num_rtd; i++) {
struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai;
struct snd_soc_platform *platform = card->rtd[i].platform;
- if (card->rtd[i].dai_link->ignore_suspend)
+ if (card->rtd[i].dai_link->ignore_suspend ||
+ card->rtd[i].dai_link->no_pcm)
continue;
- if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control)
- cpu_dai->driver->resume(cpu_dai);
- if (platform->driver->resume && platform->suspended) {
- platform->driver->resume(cpu_dai);
- platform->suspended = 0;
+ if (card->rtd[i].dai_link->dynamic) {
+ soc_dpcm_be_cpu_dai_resume(&card->rtd[i]);
+ soc_dpcm_be_platform_resume(&card->rtd[i]);
+ } else {
+ if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control)
+ cpu_dai->driver->resume(cpu_dai);
+ if (platform->driver->resume && platform->suspended) {
+ platform->driver->resume(cpu_dai);
+ platform->suspended = 0;
+ }
}
}
@@ -927,8 +994,7 @@
if (codec_dai->driver->remove) {
err = codec_dai->driver->remove(codec_dai);
if (err < 0)
- pr_err("asoc: failed to remove %s: %d\n",
- codec_dai->name, err);
+ printk(KERN_ERR "asoc: failed to remove %s\n", codec_dai->name);
}
codec_dai->probed = 0;
list_del(&codec_dai->card_list);
@@ -940,10 +1006,8 @@
if (platform->driver->remove) {
err = platform->driver->remove(platform);
if (err < 0)
- pr_err("asoc: failed to remove %s: %d\n",
- platform->name, err);
+ printk(KERN_ERR "asoc: failed to remove %s\n", platform->name);
}
-
/* Make sure all DAPM widgets are freed */
snd_soc_dapm_free(&platform->dapm);
@@ -964,8 +1028,7 @@
if (cpu_dai->driver->remove) {
err = cpu_dai->driver->remove(cpu_dai);
if (err < 0)
- pr_err("asoc: failed to remove %s: %d\n",
- cpu_dai->name, err);
+ printk(KERN_ERR "asoc: failed to remove %s\n", cpu_dai->name);
}
cpu_dai->probed = 0;
list_del(&cpu_dai->card_list);
@@ -1007,7 +1070,6 @@
{
int ret = 0;
const struct snd_soc_codec_driver *driver = codec->driver;
- struct snd_soc_dai *dai;
codec->card = card;
codec->dapm.card = card;
@@ -1022,14 +1084,6 @@
snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets,
driver->num_dapm_widgets);
- /* Create DAPM widgets for each DAI stream */
- list_for_each_entry(dai, &dai_list, list) {
- if (dai->dev != codec->dev)
- continue;
-
- snd_soc_dapm_new_dai_widgets(&codec->dapm, dai);
- }
-
codec->dapm.idle_bias_off = driver->idle_bias_off;
if (driver->probe) {
@@ -1081,8 +1135,6 @@
snd_soc_dapm_new_controls(&platform->dapm,
driver->dapm_widgets, driver->num_dapm_widgets);
- platform->dapm.idle_bias_off = 1;
-
if (driver->probe) {
ret = driver->probe(platform);
if (ret < 0) {
@@ -1108,7 +1160,6 @@
return 0;
err_probe:
- soc_cleanup_platform_debugfs(platform);
module_put(platform->dev->driver->owner);
return ret;
@@ -1170,6 +1221,10 @@
rtd->dev->init_name = name;
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
+ INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
+ INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
+ INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
+ INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
ret = device_add(rtd->dev);
if (ret < 0) {
dev_err(card->dev,
@@ -1191,6 +1246,17 @@
dev_err(codec->dev,
"asoc: failed to add codec sysfs files: %d\n", ret);
+#ifdef CONFIG_DEBUG_FS
+ /* add DSP sysfs entries */
+ if (!dai_link->dynamic)
+ goto out;
+
+ ret = soc_dpcm_debugfs_add(rtd);
+ if (ret < 0)
+ dev_err(rtd->dev, "asoc: failed to add dpcm sysfs entries: %d\n", ret);
+
+out:
+#endif
return 0;
}
@@ -1224,8 +1290,8 @@
if (cpu_dai->driver->probe) {
ret = cpu_dai->driver->probe(cpu_dai);
if (ret < 0) {
- pr_err("asoc: failed to probe CPU DAI %s: %d\n",
- cpu_dai->name, ret);
+ printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n",
+ cpu_dai->name);
module_put(cpu_dai->dev->driver->owner);
return ret;
}
@@ -1256,8 +1322,8 @@
if (codec_dai->driver->probe) {
ret = codec_dai->driver->probe(codec_dai);
if (ret < 0) {
- pr_err("asoc: failed to probe CODEC DAI %s: %d\n",
- codec_dai->name, ret);
+ printk(KERN_ERR "asoc: failed to probe CODEC DAI %s\n",
+ codec_dai->name);
return ret;
}
}
@@ -1277,13 +1343,12 @@
ret = device_create_file(rtd->dev, &dev_attr_pmdown_time);
if (ret < 0)
- pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret);
+ printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n");
/* create the pcm */
ret = soc_new_pcm(rtd, num);
if (ret < 0) {
- pr_err("asoc: can't create pcm %s :%d\n",
- dai_link->stream_name, ret);
+ printk(KERN_ERR "asoc: can't create pcm %s\n", dai_link->stream_name);
return ret;
}
@@ -1316,7 +1381,7 @@
ret = soc_ac97_dev_register(rtd->codec);
if (ret < 0) {
- pr_err("asoc: AC97 device register failed:%d\n", ret);
+ printk(KERN_ERR "asoc: AC97 device register failed\n");
return ret;
}
@@ -1404,6 +1469,96 @@
return 0;
}
+static void soc_init_dai_aif_channel_map(struct snd_soc_card *card,
+ struct snd_soc_dai *dai, int stream)
+{
+ struct snd_soc_dapm_widget *w = NULL;
+ const char *aif_name;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aif_name = dai->driver->playback.aif_name;
+ else
+ aif_name = dai->driver->capture.aif_name;
+
+ if (dai->codec) {
+ struct snd_soc_codec *codec;
+
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+ w = snd_soc_get_codec_widget(card, codec, aif_name);
+ if (w)
+ break;
+ }
+ } else if (dai->platform) {
+ struct snd_soc_platform *platform;
+
+ list_for_each_entry(platform, &card->platform_dev_list, card_list) {
+ w = snd_soc_get_platform_widget(card, platform, aif_name);
+ if (w)
+ break;
+ }
+ }
+
+ if (w) {
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback_aif = w;
+ else
+ dai->capture_aif = w;
+ } else
+ dev_err(dai->dev, "unable to find %s DAI AIF %s\n",
+ stream ? "capture" : "playback", aif_name);
+
+ dai->channel_map_instanciated = 1;
+}
+
+static int soc_is_dai_pcm(struct snd_soc_card *card, struct snd_soc_dai *dai)
+{
+ int i;
+
+ for (i = 0; i < card->num_rtd; i++) {
+ if (card->rtd[i].cpu_dai == dai && !card->rtd[i].dai_link->no_pcm)
+ return 1;
+ }
+ return 0;
+}
+
+static void soc_init_card_aif_channel_map(struct snd_soc_card *card)
+{
+ struct snd_soc_dai *dai;
+
+ list_for_each_entry(dai, &card->dai_dev_list, card_list) {
+
+ /* only process DAIs that use the new API until
+ * the old "stream name" API is fully deprecated */
+ if (!dai->driver->playback.aif_name && !dai->driver->capture.aif_name)
+ continue;
+
+ /* channels are only mapped from PCM DAIs */
+ if (!soc_is_dai_pcm(card, dai))
+ continue;
+
+ /* skip if already instanciated */
+ if (dai->channel_map_instanciated)
+ continue;
+
+ /* create unique channels masks for each DAI in the sound card */
+ dai->playback_channel_map =
+ ((1 << dai->driver->playback.channels_max) - 1)
+ << card->num_playback_channels;
+ card->num_playback_channels += dai->driver->playback.channels_max;
+
+ dai->capture_channel_map =
+ ((1 << dai->driver->capture.channels_max) - 1)
+ << card->num_capture_channels;
+ card->num_capture_channels += dai->driver->capture.channels_max;
+
+ if (dai->driver->playback.channels_max)
+ soc_init_dai_aif_channel_map(card, dai, SNDRV_PCM_STREAM_PLAYBACK);
+ if (dai->driver->capture.channels_max)
+ soc_init_dai_aif_channel_map(card, dai, SNDRV_PCM_STREAM_CAPTURE);
+ }
+}
+
+
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct snd_soc_codec *codec;
@@ -1456,8 +1611,8 @@
ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
card->owner, 0, &card->snd_card);
if (ret < 0) {
- pr_err("asoc: can't create sound card for card %s: %d\n",
- card->name, ret);
+ printk(KERN_ERR "asoc: can't create sound card for card %s\n",
+ card->name);
mutex_unlock(&card->mutex);
return;
}
@@ -1510,8 +1665,6 @@
}
}
- snd_soc_dapm_link_dai_widgets(card);
-
if (card->controls)
snd_soc_add_card_controls(card, card->controls, card->num_controls);
@@ -1527,14 +1680,14 @@
if (dai_link->dai_fmt) {
ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai,
dai_link->dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP)
+ if (ret != 0)
dev_warn(card->rtd[i].codec_dai->dev,
"Failed to set DAI format: %d\n",
ret);
ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai,
dai_link->dai_fmt);
- if (ret != 0 && ret != -ENOTSUPP)
+ if (ret != 0)
dev_warn(card->rtd[i].cpu_dai->dev,
"Failed to set DAI format: %d\n",
ret);
@@ -1570,6 +1723,7 @@
}
snd_soc_dapm_new_widgets(&card->dapm);
+ soc_init_card_aif_channel_map(card);
if (card->fully_routed)
list_for_each_entry(codec, &card->codec_dev_list, card_list)
@@ -1577,8 +1731,7 @@
ret = snd_card_register(card->snd_card);
if (ret < 0) {
- pr_err("asoc: failed to register soundcard for %s: %d\n",
- card->name, ret);
+ printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name);
goto probe_aux_dev_err;
}
@@ -1587,8 +1740,7 @@
for (i = 0; i < card->num_rtd; i++) {
ret = soc_register_ac97_dai_link(&card->rtd[i]);
if (ret < 0) {
- pr_err("asoc: failed to register AC97 %s: %d\n",
- card->name, ret);
+ printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name);
while (--i >= 0)
soc_unregister_ac97_dai_link(card->rtd[i].codec);
goto probe_aux_dev_err;
@@ -1641,10 +1793,6 @@
if (!card)
return -EINVAL;
- dev_warn(&pdev->dev,
- "ASoC machine %s should use snd_soc_register_card()\n",
- card->name);
-
/* Bodge while we unpick instantiation */
card->dev = &pdev->dev;
@@ -1682,6 +1830,7 @@
snd_soc_dapm_free(&card->dapm);
+ kfree(card->rtd);
snd_card_free(card->snd_card);
return 0;
@@ -1720,10 +1869,7 @@
const struct dev_pm_ops snd_soc_pm_ops = {
.suspend = snd_soc_suspend,
.resume = snd_soc_resume,
- .freeze = snd_soc_suspend,
- .thaw = snd_soc_resume,
.poweroff = snd_soc_poweroff,
- .restore = snd_soc_resume,
};
EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
@@ -2015,6 +2161,8 @@
const struct snd_pcm_hardware *hw)
{
struct snd_pcm_runtime *runtime = substream->runtime;
+ if (!runtime)
+ return 0;
runtime->hw.info = hw->info;
runtime->hw.formats = hw->formats;
runtime->hw.period_bytes_min = hw->period_bytes_min;
@@ -2039,7 +2187,7 @@
* Returns 0 for success, else error.
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
- void *data, const char *long_name,
+ void *data, char *long_name,
const char *prefix)
{
struct snd_kcontrol_new template;
@@ -2196,7 +2344,8 @@
if (uinfo->value.enumerated.item > e->max - 1)
uinfo->value.enumerated.item = e->max - 1;
strcpy(uinfo->value.enumerated.name,
- e->texts[uinfo->value.enumerated.item]);
+ snd_soc_get_enum_text(e, uinfo->value.enumerated.item));
+
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
@@ -2360,7 +2509,7 @@
if (uinfo->value.enumerated.item > e->max - 1)
uinfo->value.enumerated.item = e->max - 1;
strcpy(uinfo->value.enumerated.name,
- e->texts[uinfo->value.enumerated.item]);
+ snd_soc_get_enum_text(e, uinfo->value.enumerated.item));
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
@@ -2392,6 +2541,39 @@
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
/**
+ * snd_soc_info_multi_ext - external single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single external mixer control.
+ * that accepts multiple input.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_multi_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct soc_multi_mixer_control *mc =
+ (struct soc_multi_mixer_control *)kcontrol->private_value;
+ int platform_max;
+
+ if (!mc->platform_max)
+ mc->platform_max = mc->max;
+ platform_max = mc->platform_max;
+
+ if (platform_max == 1 && !strnstr(kcontrol->id.name, " Volume", 30))
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ else
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+
+ uinfo->count = mc->count;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = platform_max;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_multi_ext);
+
+/**
* snd_soc_info_volsw - single mixer info callback
* @kcontrol: mixer control
* @uinfo: control element information
@@ -2542,13 +2724,15 @@
(struct soc_mixer_control *)kcontrol->private_value;
int platform_max;
int min = mc->min;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
if (!mc->platform_max)
mc->platform_max = mc->max;
platform_max = mc->platform_max;
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 2;
+ uinfo->count = shift == rshift ? 1 : 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = platform_max - min;
return 0;
@@ -2571,13 +2755,16 @@
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
int min = mc->min;
int val = snd_soc_read(codec, reg);
ucontrol->value.integer.value[0] =
- ((signed char)(val & 0xff))-min;
- ucontrol->value.integer.value[1] =
- ((signed char)((val >> 8) & 0xff))-min;
+ ((signed char)((val >> shift) & 0xff))-min;
+ if (shift != rshift)
+ ucontrol->value.integer.value[1] =
+ ((signed char)((val >> rshift) & 0xff))-min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);
@@ -2598,13 +2785,20 @@
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int rshift = mc->rshift;
int min = mc->min;
- unsigned int val;
+ unsigned int val, val2, val_mask;
- val = (ucontrol->value.integer.value[0]+min) & 0xff;
- val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
+ val = ((ucontrol->value.integer.value[0]+min) & 0xff) << shift;
+ val_mask = 0xff << shift;
+ if (shift != rshift) {
+ val2 = (ucontrol->value.integer.value[1]+min) & 0xff;
+ val |= val2 << rshift;
+ val_mask |= 0xff << rshift;
+ }
- return snd_soc_update_bits_locked(codec, reg, 0xffff, val);
+ return snd_soc_update_bits_locked(codec, reg, val_mask, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
@@ -2740,115 +2934,6 @@
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx);
-int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct soc_bytes *params = (void *)kcontrol->private_value;
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
- uinfo->count = params->num_regs * codec->val_bytes;
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_bytes_info);
-
-int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_bytes *params = (void *)kcontrol->private_value;
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int ret;
-
- if (codec->using_regmap)
- ret = regmap_raw_read(codec->control_data, params->base,
- ucontrol->value.bytes.data,
- params->num_regs * codec->val_bytes);
- else
- ret = -EINVAL;
-
- /* Hide any masked bytes to ensure consistent data reporting */
- if (ret == 0 && params->mask) {
- switch (codec->val_bytes) {
- case 1:
- ucontrol->value.bytes.data[0] &= ~params->mask;
- break;
- case 2:
- ((u16 *)(&ucontrol->value.bytes.data))[0]
- &= ~params->mask;
- break;
- case 4:
- ((u32 *)(&ucontrol->value.bytes.data))[0]
- &= ~params->mask;
- break;
- default:
- return -EINVAL;
- }
- }
-
- return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_bytes_get);
-
-int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_bytes *params = (void *)kcontrol->private_value;
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int ret, len;
- unsigned int val;
- void *data;
-
- if (!codec->using_regmap)
- return -EINVAL;
-
- data = ucontrol->value.bytes.data;
- len = params->num_regs * codec->val_bytes;
-
- /*
- * If we've got a mask then we need to preserve the register
- * bits. We shouldn't modify the incoming data so take a
- * copy.
- */
- if (params->mask) {
- ret = regmap_read(codec->control_data, params->base, &val);
- if (ret != 0)
- return ret;
-
- val &= params->mask;
-
- data = kmemdup(data, len, GFP_KERNEL);
- if (!data)
- return -ENOMEM;
-
- switch (codec->val_bytes) {
- case 1:
- ((u8 *)data)[0] &= ~params->mask;
- ((u8 *)data)[0] |= val;
- break;
- case 2:
- ((u16 *)data)[0] &= cpu_to_be16(~params->mask);
- ((u16 *)data)[0] |= cpu_to_be16(val);
- break;
- case 4:
- ((u32 *)data)[0] &= cpu_to_be32(~params->mask);
- ((u32 *)data)[0] |= cpu_to_be32(val);
- break;
- default:
- return -EINVAL;
- }
- }
-
- ret = regmap_raw_write(codec->control_data, params->base,
- data, len);
-
- if (params->mask)
- kfree(data);
-
- return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_bytes_put);
-
/**
* snd_soc_dai_set_sysclk - configure DAI system or master clock.
* @dai: DAI
@@ -2966,11 +3051,10 @@
*/
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
- if (dai->driver == NULL)
+ if (dai->driver && dai->driver->ops->set_fmt)
+ return dai->driver->ops->set_fmt(dai, fmt);
+ else
return -EINVAL;
- if (dai->driver->ops->set_fmt == NULL)
- return -ENOTSUPP;
- return dai->driver->ops->set_fmt(dai, fmt);
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
@@ -3021,6 +3105,29 @@
EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
/**
+ * snd_soc_dai_get_channel_map - configure DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ * 0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ * 0~num-1 uses
+ *
+ * configure the relationship between channel number and TDM slot number.
+ */
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+ unsigned int *tx_num, unsigned int *tx_slot,
+ unsigned int *rx_num, unsigned int *rx_slot)
+{
+ if (dai->driver && dai->driver->ops->get_channel_map)
+ return dai->driver->ops->get_channel_map(dai, tx_num, tx_slot,
+ rx_num, rx_slot);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_get_channel_map);
+/**
* snd_soc_dai_set_tristate - configure DAI system or master clock.
* @dai: DAI
* @tristate: tristate enable
@@ -3061,6 +3168,7 @@
int snd_soc_register_card(struct snd_soc_card *card)
{
int i;
+ int ret = 0;
if (!card->name || !card->dev)
return -EINVAL;
@@ -3107,10 +3215,9 @@
soc_init_card_debugfs(card);
- card->rtd = devm_kzalloc(card->dev,
- sizeof(struct snd_soc_pcm_runtime) *
- (card->num_links + card->num_aux_devs),
- GFP_KERNEL);
+ card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) *
+ (card->num_links + card->num_aux_devs),
+ GFP_KERNEL);
if (card->rtd == NULL)
return -ENOMEM;
card->num_rtd = 0;
@@ -3123,6 +3230,7 @@
INIT_LIST_HEAD(&card->dapm_dirty);
card->instantiated = 0;
mutex_init(&card->mutex);
+ mutex_init(&card->dpcm_mutex);
mutex_lock(&client_mutex);
list_add(&card->list, &card_list);
@@ -3131,7 +3239,7 @@
dev_dbg(card->dev, "Registered card '%s'\n", card->name);
- return 0;
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_register_card);
@@ -3205,7 +3313,7 @@
struct snd_soc_dai_driver *dai_drv)
{
if (dai_drv->name == NULL) {
- pr_err("asoc: error - multiple DAI %s registered with no name\n",
+ printk(KERN_ERR "asoc: error - multiple DAI %s registered with no name\n",
dev_name(dev));
return NULL;
}
@@ -3310,6 +3418,7 @@
dai->dev = dev;
dai->driver = &dai_drv[i];
+
if (dai->driver->id)
dai->id = dai->driver->id;
else
@@ -3380,7 +3489,6 @@
platform->dapm.dev = dev;
platform->dapm.platform = platform;
platform->dapm.stream_event = platform_drv->stream_event;
- mutex_init(&platform->mutex);
mutex_lock(&client_mutex);
list_add(&platform->list, &platform_list);
@@ -3489,7 +3597,6 @@
codec->volatile_register = codec_drv->volatile_register;
codec->readable_register = codec_drv->readable_register;
codec->writable_register = codec_drv->writable_register;
- codec->ignore_pmdown_time = codec_drv->ignore_pmdown_time;
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
codec->dapm.dev = dev;
codec->dapm.codec = codec;
@@ -3678,7 +3785,8 @@
#ifdef CONFIG_DEBUG_FS
snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL);
if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) {
- pr_warn("ASoC: Failed to create debugfs directory\n");
+ printk(KERN_WARNING
+ "ASoC: Failed to create debugfs directory\n");
snd_soc_debugfs_root = NULL;
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1bb6d4a..729142f 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -14,13 +14,19 @@
* dynamic configuration of codec internal audio paths and active
* DACs/ADCs.
* o Platform power domain - can support external components i.e. amps and
- * mic/headphone insertion events.
+ * mic/meadphone insertion events.
* o Automatic Mic Bias support
* o Jack insertion power event initiation - e.g. hp insertion will enable
* sinks, dacs, etc
- * o Delayed power down of audio subsystem to reduce pops between a quick
+ * o Delayed powerdown of audio susbsystem to reduce pops between a quick
* device reopen.
*
+ * Todo:
+ * o DAPM power change sequencing - allow for configurable per
+ * codec sequences.
+ * o Support for analogue bias optimisation.
+ * o Support for reduced codec oversampling rates.
+ * o Support for reduced codec bias currents.
*/
#include <linux/module.h>
@@ -34,7 +40,6 @@
#include <linux/jiffies.h>
#include <linux/debugfs.h>
#include <linux/pm_runtime.h>
-#include <linux/regulator/consumer.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -43,6 +48,7 @@
#include <sound/initval.h>
#include <trace/events/asoc.h>
+int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *);
#define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++;
@@ -50,11 +56,8 @@
static int dapm_up_seq[] = {
[snd_soc_dapm_pre] = 0,
[snd_soc_dapm_supply] = 1,
- [snd_soc_dapm_regulator_supply] = 1,
[snd_soc_dapm_micbias] = 2,
- [snd_soc_dapm_dai] = 3,
- [snd_soc_dapm_aif_in] = 3,
- [snd_soc_dapm_aif_out] = 3,
+ [snd_soc_dapm_adc] = 3,
[snd_soc_dapm_mic] = 4,
[snd_soc_dapm_mux] = 5,
[snd_soc_dapm_virt_mux] = 5,
@@ -63,20 +66,21 @@
[snd_soc_dapm_mixer] = 7,
[snd_soc_dapm_mixer_named_ctl] = 7,
[snd_soc_dapm_pga] = 8,
- [snd_soc_dapm_adc] = 9,
+ [snd_soc_dapm_aif_in] = 8,
+ [snd_soc_dapm_aif_out] = 8,
[snd_soc_dapm_out_drv] = 10,
[snd_soc_dapm_hp] = 10,
[snd_soc_dapm_spk] = 10,
- [snd_soc_dapm_line] = 10,
[snd_soc_dapm_post] = 11,
};
static int dapm_down_seq[] = {
[snd_soc_dapm_pre] = 0,
+ [snd_soc_dapm_aif_in] = 1,
+ [snd_soc_dapm_aif_out] = 1,
[snd_soc_dapm_adc] = 1,
[snd_soc_dapm_hp] = 2,
[snd_soc_dapm_spk] = 2,
- [snd_soc_dapm_line] = 2,
[snd_soc_dapm_out_drv] = 2,
[snd_soc_dapm_pga] = 4,
[snd_soc_dapm_mixer_named_ctl] = 5,
@@ -87,10 +91,6 @@
[snd_soc_dapm_mux] = 9,
[snd_soc_dapm_virt_mux] = 9,
[snd_soc_dapm_value_mux] = 9,
- [snd_soc_dapm_aif_in] = 10,
- [snd_soc_dapm_aif_out] = 10,
- [snd_soc_dapm_dai] = 10,
- [snd_soc_dapm_regulator_supply] = 11,
[snd_soc_dapm_supply] = 11,
[snd_soc_dapm_post] = 12,
};
@@ -173,19 +173,6 @@
return NULL;
}
-static void dapm_reset(struct snd_soc_card *card)
-{
- struct snd_soc_dapm_widget *w;
-
- memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
-
- list_for_each_entry(w, &card->widgets, list) {
- w->power_checked = false;
- w->inputs = -1;
- w->outputs = -1;
- }
-}
-
static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg)
{
if (w->codec)
@@ -316,7 +303,7 @@
p->connect = 0;
for (i = 0; i < e->max; i++) {
- if (!(strcmp(p->name, e->texts[i])) && item == i)
+ if (!(strcmp(p->name, snd_soc_get_enum_text(e, i))) && item == i)
p->connect = 1;
}
}
@@ -332,7 +319,7 @@
* that the default mux choice (the first) will be
* correctly powered up during initialization.
*/
- if (!strcmp(p->name, e->texts[0]))
+ if (!strcmp(p->name, snd_soc_get_enum_text(e, 0)))
p->connect = 1;
}
break;
@@ -350,7 +337,7 @@
p->connect = 0;
for (i = 0; i < e->max; i++) {
- if (!(strcmp(p->name, e->texts[i])) && item == i)
+ if (!(strcmp(p->name, snd_soc_get_enum_text(e, i))) && item == i)
p->connect = 1;
}
}
@@ -366,10 +353,8 @@
case snd_soc_dapm_micbias:
case snd_soc_dapm_vmid:
case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_spk:
@@ -394,11 +379,11 @@
int i;
for (i = 0; i < e->max; i++) {
- if (!(strcmp(control_name, e->texts[i]))) {
+ if (!(strcmp(control_name, snd_soc_get_enum_text(e, i)))) {
list_add(&path->list, &dapm->card->paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
- path->name = (char*)e->texts[i];
+ path->name = (char*)snd_soc_get_enum_text(e, i);
dapm_set_path_status(dest, path, 0);
return 0;
}
@@ -527,17 +512,17 @@
* for widgets so cut the prefix off
* the front of the widget name.
*/
- snprintf((char *)path->long_name, name_len,
- "%s %s", w->name + prefix_len,
+ snprintf(path->long_name, name_len, "%s %s",
+ w->name + prefix_len,
w->kcontrol_news[i].name);
break;
case snd_soc_dapm_mixer_named_ctl:
- snprintf((char *)path->long_name, name_len,
- "%s", w->kcontrol_news[i].name);
+ snprintf(path->long_name, name_len, "%s",
+ w->kcontrol_news[i].name);
break;
}
- ((char *)path->long_name)[name_len - 1] = '\0';
+ path->long_name[name_len - 1] = '\0';
path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i],
wlist, path->long_name,
@@ -571,7 +556,7 @@
struct snd_soc_dapm_widget_list *wlist;
int shared, wlistentries;
size_t wlistsize;
- const char *name;
+ char *name;
if (w->num_kcontrols != 1) {
dev_err(dapm->dev,
@@ -653,15 +638,6 @@
return 0;
}
-/* reset 'walked' bit for each dapm path */
-static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm)
-{
- struct snd_soc_dapm_path *p;
-
- list_for_each_entry(p, &dapm->card->paths, list)
- p->walked = 0;
-}
-
/* We implement power down on suspend by checking the power state of
* the ALSA card - when we are suspending the ALSA state for the card
* is set to D3.
@@ -682,11 +658,60 @@
}
}
+/* reset 'walked' bit for each dapm path */
+static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm)
+{
+ struct snd_soc_dapm_path *p;
+
+ list_for_each_entry(p, &dapm->card->paths, list)
+ p->walked = 0;
+}
+
+/* add widget to list if it's not already in the list */
+static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list,
+ struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_widget_list *wlist;
+ int wlistsize, wlistentries, i;
+
+ if (*list == NULL)
+ return -EINVAL;
+
+ wlist = *list;
+
+ /* is this widget already in the list */
+ for (i = 0; i < wlist->num_widgets; i++) {
+ if (wlist->widgets[i] == w)
+ return 0;
+ }
+
+ /* allocate some new space */
+ wlistentries = wlist->num_widgets + 1;
+ wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
+ wlistentries * sizeof(struct snd_soc_dapm_widget *);
+ *list = krealloc(wlist, wlistsize, GFP_KERNEL);
+ if (*list == NULL) {
+ dev_err(w->dapm->dev, "can't allocate widget list for %s\n",
+ w->name);
+ return -ENOMEM;
+ }
+ wlist = *list;
+
+ /* insert the widget */
+ dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n",
+ w->name, wlist->num_widgets);
+
+ wlist->widgets[wlist->num_widgets] = w;
+ wlist->num_widgets++;
+ return 1;
+}
+
/*
* Recursively check for a completed path to an active or physically connected
* output widget. Returns number of complete paths.
*/
-static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
+static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
+ struct snd_soc_dapm_widget_list **list)
{
struct snd_soc_dapm_path *path;
int con = 0;
@@ -696,18 +721,12 @@
DAPM_UPDATE_STAT(widget, path_checks);
- switch (widget->id) {
- case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
+ if (widget->id == snd_soc_dapm_supply)
return 0;
- default:
- break;
- }
switch (widget->id) {
case snd_soc_dapm_adc:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
if (widget->active) {
widget->outputs = snd_soc_dapm_suspend_check(widget);
return widget->outputs;
@@ -742,9 +761,26 @@
if (path->walked)
continue;
+ dev_vdbg(widget->dapm->dev," %c : %s -> %s -> %s : %c\n",
+ path->sink && path->connect ? '*' : ' ',
+ widget->name, path->name, path->sink->name,
+ path->weak ? 'w': ' ');
+
if (path->sink && path->connect) {
path->walked = 1;
- con += is_connected_output_ep(path->sink);
+
+ /* do we need to add this widget to the list ? */
+ if (list) {
+ int err;
+ err = dapm_list_add_widget(list, path->sink);
+ if (err < 0) {
+ dev_err(widget->dapm->dev, "could not add widget %s\n",
+ widget->name);
+ return con;
+ }
+ }
+
+ con += is_connected_output_ep(path->sink, list);
}
}
@@ -757,7 +793,8 @@
* Recursively check for a completed path to an active or physically connected
* input widget. Returns number of complete paths.
*/
-static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
+static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
+ struct snd_soc_dapm_widget_list **list)
{
struct snd_soc_dapm_path *path;
int con = 0;
@@ -767,19 +804,13 @@
DAPM_UPDATE_STAT(widget, path_checks);
- switch (widget->id) {
- case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
+ if (widget->id == snd_soc_dapm_supply)
return 0;
- default:
- break;
- }
/* active stream ? */
switch (widget->id) {
case snd_soc_dapm_dac:
case snd_soc_dapm_aif_in:
- case snd_soc_dapm_dai:
if (widget->active) {
widget->inputs = snd_soc_dapm_suspend_check(widget);
return widget->inputs;
@@ -819,6 +850,12 @@
list_for_each_entry(path, &widget->sources, list_sink) {
DAPM_UPDATE_STAT(widget, neighbour_checks);
+ if (path->source)
+ dev_vdbg(widget->dapm->dev," %c : %s <- %s <- %s : %c\n",
+ path->source && path->connect ? '*' : ' ',
+ widget->name, path->name, path->source->name,
+ path->weak ? 'w': ' ');
+
if (path->weak)
continue;
@@ -827,7 +864,19 @@
if (path->source && path->connect) {
path->walked = 1;
- con += is_connected_input_ep(path->source);
+
+ /* do we need to add this widget to the list ? */
+ if (list) {
+ int err;
+ err = dapm_list_add_widget(list, path->sink);
+ if (err < 0) {
+ dev_err(widget->dapm->dev, "could not add widget %s\n",
+ widget->name);
+ return con;
+ }
+ }
+
+ con += is_connected_input_ep(path->source, list);
}
}
@@ -836,6 +885,120 @@
return con;
}
+/* Get the DAPM AIF widget for thie DAI stream */
+struct snd_soc_dapm_widget *snd_soc_get_codec_widget(struct snd_soc_card *card,
+ struct snd_soc_codec *codec, const char *name)
+{
+ struct snd_soc_dapm_widget *w;
+
+ /* get stream root widget AIF from stream string and direction */
+ list_for_each_entry(w, &card->widgets, list) {
+
+ /* make sure the widget belongs the DAI codec or platform */
+ if (w->codec && w->codec != codec)
+ continue;
+
+ if (!strcmp(w->name, name))
+ return w;
+
+ }
+ dev_err(card->dapm.dev, "DAI AIF widget for %s not found\n", name);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_codec_widget);
+
+struct snd_soc_dapm_widget *snd_soc_get_platform_widget(struct snd_soc_card *card,
+ struct snd_soc_platform *platform, const char *name)
+{
+ struct snd_soc_dapm_widget *w;
+
+ /* get stream root widget AIF from stream string and direction */
+ list_for_each_entry(w, &card->widgets, list) {
+
+ /* make sure the widget belongs the DAI codec or platform */
+ if (w->platform && w->platform != platform)
+ continue;
+
+ if (!strcmp(w->name, name))
+ return w;
+ }
+ dev_err(card->dapm.dev, "DAI AIF widget for %s not found\n", name);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_platform_widget);
+
+static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dapm_widget *root,
+ struct snd_soc_dapm_widget_list **list)
+{
+ int paths;
+
+ if (!root) {
+ dev_err(dapm->dev, "no root widget for path discovery\n");
+ return 0;
+ }
+
+ dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name);
+ paths = is_connected_output_ep(root, list);
+ dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name);
+
+ return paths;
+}
+
+static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dapm_widget *root,
+ struct snd_soc_dapm_widget_list **list)
+{
+ int paths;
+
+ if (!root) {
+ dev_err(dapm->dev, "no root widget for path discovery\n");
+ return 0;
+ }
+
+ dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name);
+ paths = is_connected_input_ep(root, list);
+ dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name);
+
+ return paths;
+}
+
+/**
+ * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets.
+ * @dai: the soc DAI.
+ * @stream: stream direction.
+ * @list: list of active widgets for this stream.
+ *
+ * Queries DAPM graph as to whether an valid audio stream path exists for
+ * the initial stream specified by name. This takes into account
+ * current mixer and mux kcontrol settings. Creates list of valid widgets.
+ *
+ * Returns the number of valid paths or negative error.
+ */
+int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
+ struct snd_soc_dapm_widget_list **list)
+{
+ struct snd_soc_card *card = dai->card;
+ struct snd_soc_dapm_widget *w;
+ int paths;
+
+ memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
+
+ list_for_each_entry(w, &card->widgets, list) {
+ w->power_checked = false;
+ w->inputs = -1;
+ w->outputs = -1;
+ }
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ paths = dapm_get_playback_paths(&card->dapm, dai->playback_aif, list);
+ else
+ paths = dapm_get_capture_paths(&card->dapm, dai->capture_aif, list);
+
+ dapm_clear_walk(&card->dapm);
+ return paths;
+}
+
/*
* Handler for generic register modifier widget.
*/
@@ -856,19 +1019,6 @@
}
EXPORT_SYMBOL_GPL(dapm_reg_event);
-/*
- * Handler for regulator supply widget.
- */
-int dapm_regulator_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (SND_SOC_DAPM_EVENT_ON(event))
- return regulator_enable(w->priv);
- else
- return regulator_disable_deferred(w->priv, w->shift);
-}
-EXPORT_SYMBOL_GPL(dapm_regulator_event);
-
static int dapm_widget_power_check(struct snd_soc_dapm_widget *w)
{
if (w->power_checked)
@@ -892,20 +1042,13 @@
DAPM_UPDATE_STAT(w, power_checks);
- in = is_connected_input_ep(w);
+ in = is_connected_input_ep(w, NULL);
dapm_clear_walk(w->dapm);
- out = is_connected_output_ep(w);
+ out = is_connected_output_ep(w, NULL);
dapm_clear_walk(w->dapm);
return out != 0 && in != 0;
}
-static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
-{
- DAPM_UPDATE_STAT(w, power_checks);
-
- return w->active;
-}
-
/* Check to see if an ADC has power */
static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
{
@@ -914,7 +1057,7 @@
DAPM_UPDATE_STAT(w, power_checks);
if (w->active) {
- in = is_connected_input_ep(w);
+ in = is_connected_input_ep(w, NULL);
dapm_clear_walk(w->dapm);
return in != 0;
} else {
@@ -930,7 +1073,7 @@
DAPM_UPDATE_STAT(w, power_checks);
if (w->active) {
- out = is_connected_output_ep(w);
+ out = is_connected_output_ep(w, NULL);
dapm_clear_walk(w->dapm);
return out != 0;
} else {
@@ -1306,7 +1449,7 @@
dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
if (d->dev)
- pm_runtime_put(d->dev);
+ pm_runtime_put_sync(d->dev);
}
/* If we just powered up then move to active bias */
@@ -1356,7 +1499,6 @@
}
switch (w->id) {
case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
/* Supplies can't affect their outputs, only their inputs */
break;
default:
@@ -1369,10 +1511,15 @@
break;
}
- if (power)
+ if (power) {
dapm_seq_insert(w, up_list, true);
- else
+ dev_dbg(w->dapm->dev,
+ "dapm: power up widget %s\n", w->name);
+ } else {
dapm_seq_insert(w, down_list, false);
+ dev_dbg(w->dapm->dev,
+ "dapm: power down widget %s\n", w->name);
+ }
w->power = power;
}
@@ -1429,7 +1576,13 @@
}
}
- dapm_reset(card);
+ memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
+
+ list_for_each_entry(w, &card->widgets, list) {
+ w->power_checked = false;
+ w->inputs = -1;
+ w->outputs = -1;
+ }
/* Check which widgets we need to power and store them in
* lists indicating if they should be powered up or down. We
@@ -1450,15 +1603,10 @@
/* Supplies and micbiases only bring the
* context up to STANDBY as unless something
* else is active and passing audio they
- * generally don't require full power. Signal
- * generators are virtual pins and have no
- * power impact themselves.
+ * generally don't require full power.
*/
switch (w->id) {
- case snd_soc_dapm_siggen:
- break;
case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
case snd_soc_dapm_micbias:
if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
d->target_bias_level = SND_SOC_BIAS_STANDBY;
@@ -1530,12 +1678,6 @@
&async_domain);
async_synchronize_full_domain(&async_domain);
- /* do we need to notify any clients that DAPM event is complete */
- list_for_each_entry(d, &card->dapm_list, list) {
- if (d->stream_event)
- d->stream_event(d, event);
- }
-
pop_dbg(dapm->dev, card->pop_time,
"DAPM sequencing finished, waiting %dms\n", card->pop_time);
pop_wait(card->pop_time);
@@ -1546,6 +1688,12 @@
}
#ifdef CONFIG_DEBUG_FS
+static int dapm_widget_power_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
static ssize_t dapm_widget_power_read_file(struct file *file,
char __user *user_buf,
size_t count, loff_t *ppos)
@@ -1560,14 +1708,13 @@
if (!buf)
return -ENOMEM;
- in = is_connected_input_ep(w);
+ in = is_connected_input_ep(w, NULL);
dapm_clear_walk(w->dapm);
- out = is_connected_output_ep(w);
+ out = is_connected_output_ep(w, NULL);
dapm_clear_walk(w->dapm);
- ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d",
- w->name, w->power ? "On" : "Off",
- w->force ? " (forced)" : "", in, out);
+ ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d",
+ w->name, w->power ? "On" : "Off", in, out);
if (w->reg >= 0)
ret += snprintf(buf + ret, PAGE_SIZE - ret,
@@ -1609,11 +1756,17 @@
}
static const struct file_operations dapm_widget_power_fops = {
- .open = simple_open,
+ .open = dapm_widget_power_open_file,
.read = dapm_widget_power_read_file,
.llseek = default_llseek,
};
+static int dapm_bias_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
static ssize_t dapm_bias_read_file(struct file *file, char __user *user_buf,
size_t count, loff_t *ppos)
{
@@ -1644,7 +1797,7 @@
}
static const struct file_operations dapm_bias_fops = {
- .open = simple_open,
+ .open = dapm_bias_open_file,
.read = dapm_bias_read_file,
.llseek = default_llseek,
};
@@ -1657,7 +1810,7 @@
dapm->debugfs_dapm = debugfs_create_dir("dapm", parent);
if (!dapm->debugfs_dapm) {
- dev_warn(dapm->dev,
+ printk(KERN_WARNING
"Failed to create DAPM debugfs directory\n");
return;
}
@@ -1710,7 +1863,8 @@
/* test and update the power status of a mux widget */
int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
- struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e)
+ struct snd_kcontrol *kcontrol, int change,
+ int mux, struct soc_enum *e)
{
struct snd_soc_dapm_path *path;
int found = 0;
@@ -1720,17 +1874,21 @@
widget->id != snd_soc_dapm_value_mux)
return -ENODEV;
+ if (!change)
+ return 0;
+
/* find dapm widget path assoc with kcontrol */
list_for_each_entry(path, &widget->dapm->card->paths, list) {
+
if (path->kcontrol != kcontrol)
continue;
- if (!path->name || !e->texts[mux])
+ if (!path->name || !snd_soc_get_enum_text(e, mux))
continue;
found = 1;
/* we now need to match the string in the enum to the path */
- if (!(strcmp(path->name, e->texts[mux]))) {
+ if (!(strcmp(path->name, snd_soc_get_enum_text(e, mux)))) {
path->connect = 1; /* new connection */
dapm_mark_dirty(path->source, "mux connection");
} else {
@@ -1742,8 +1900,13 @@
}
if (found) {
- dapm_mark_dirty(widget, "mux change");
- dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
+ if (widget->platform) {
+ soc_dpcm_runtime_update(widget);
+ } else {
+ dapm_mark_dirty(widget, "mux change");
+ dapm_power_widgets(widget->dapm,
+ SND_SOC_DAPM_STREAM_NOP);
+ }
}
return 0;
@@ -1774,8 +1937,13 @@
}
if (found) {
+ if (widget->platform) {
+ soc_dpcm_runtime_update(widget);
+ } else {
dapm_mark_dirty(widget, "mixer update");
dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP);
+ }
+
}
return 0;
@@ -1810,7 +1978,6 @@
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
if (w->name)
count += sprintf(buf + count, "%s: %s\n",
w->name, w->power ? "On":"Off");
@@ -1918,12 +2085,10 @@
return -EINVAL;
}
- if (w->connected != status)
- dapm_mark_dirty(w, "pin configuration");
-
w->connected = status;
if (status == 0)
w->force = 0;
+ dapm_mark_dirty(w, "pin configuration");
return 0;
}
@@ -2051,10 +2216,8 @@
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
case snd_soc_dapm_aif_in:
case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai:
list_add(&path->list, &dapm->card->paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
@@ -2459,7 +2622,7 @@
update.val = val;
widget->dapm->update = &update;
- snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e);
+ snd_soc_dapm_mux_update_power(widget, kcontrol, change, mux, e);
widget->dapm->update = NULL;
}
@@ -2520,7 +2683,8 @@
widget->value = ucontrol->value.enumerated.item[0];
- snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e);
+ snd_soc_dapm_mux_update_power(widget, kcontrol, change,
+ widget->value, e);
}
}
@@ -2623,7 +2787,7 @@
update.val = val;
widget->dapm->update = &update;
- snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e);
+ snd_soc_dapm_mux_update_power(widget, kcontrol, change, mux, e);
widget->dapm->update = NULL;
}
@@ -2663,15 +2827,15 @@
int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
const char *pin = (const char *)kcontrol->private_value;
- mutex_lock(&card->mutex);
+ mutex_lock(&codec->mutex);
ucontrol->value.integer.value[0] =
- snd_soc_dapm_get_pin_status(&card->dapm, pin);
+ snd_soc_dapm_get_pin_status(&codec->dapm, pin);
- mutex_unlock(&card->mutex);
+ mutex_unlock(&codec->mutex);
return 0;
}
@@ -2686,48 +2850,41 @@
int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
const char *pin = (const char *)kcontrol->private_value;
- mutex_lock(&card->mutex);
+ mutex_lock(&codec->mutex);
if (ucontrol->value.integer.value[0])
- snd_soc_dapm_enable_pin(&card->dapm, pin);
+ snd_soc_dapm_enable_pin(&codec->dapm, pin);
else
- snd_soc_dapm_disable_pin(&card->dapm, pin);
+ snd_soc_dapm_disable_pin(&codec->dapm, pin);
- snd_soc_dapm_sync(&card->dapm);
+ snd_soc_dapm_sync(&codec->dapm);
- mutex_unlock(&card->mutex);
+ mutex_unlock(&codec->mutex);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
-static struct snd_soc_dapm_widget *
-snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
- const struct snd_soc_dapm_widget *widget)
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @dapm: DAPM context
+ * @widget: widget template
+ *
+ * Creates a new dapm control based upon the template.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_widget *widget)
{
struct snd_soc_dapm_widget *w;
size_t name_len;
- int ret;
if ((w = dapm_cnew_widget(widget)) == NULL)
- return NULL;
-
- switch (w->id) {
- case snd_soc_dapm_regulator_supply:
- w->priv = devm_regulator_get(dapm->dev, w->name);
- if (IS_ERR(w->priv)) {
- ret = PTR_ERR(w->priv);
- dev_err(dapm->dev, "Failed to request %s: %d\n",
- w->name, ret);
- return NULL;
- }
- break;
- default:
- break;
- }
+ return -ENOMEM;
name_len = strlen(widget->name) + 1;
if (dapm->codec && dapm->codec->name_prefix)
@@ -2735,13 +2892,13 @@
w->name = kmalloc(name_len, GFP_KERNEL);
if (w->name == NULL) {
kfree(w);
- return NULL;
+ return -ENOMEM;
}
if (dapm->codec && dapm->codec->name_prefix)
- snprintf((char *)w->name, name_len, "%s %s",
+ snprintf(w->name, name_len, "%s %s",
dapm->codec->name_prefix, widget->name);
else
- snprintf((char *)w->name, name_len, "%s", widget->name);
+ snprintf(w->name, name_len, "%s", widget->name);
switch (w->id) {
case snd_soc_dapm_switch:
@@ -2774,12 +2931,8 @@
w->power_check = dapm_generic_check_power;
break;
case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
w->power_check = dapm_supply_check_power;
break;
- case snd_soc_dapm_dai:
- w->power_check = dapm_dai_check_power;
- break;
default:
w->power_check = dapm_always_on_check_power;
break;
@@ -2789,6 +2942,7 @@
w->dapm = dapm;
w->codec = dapm->codec;
w->platform = dapm->platform;
+ w->dai = dapm->dai;
INIT_LIST_HEAD(&w->sources);
INIT_LIST_HEAD(&w->sinks);
INIT_LIST_HEAD(&w->list);
@@ -2797,8 +2951,9 @@
/* machine layer set ups unconnected pins and insertions */
w->connected = 1;
- return w;
+ return 0;
}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
/**
* snd_soc_dapm_new_controls - create new dapm controls
@@ -2814,16 +2969,15 @@
const struct snd_soc_dapm_widget *widget,
int num)
{
- struct snd_soc_dapm_widget *w;
- int i;
+ int i, ret;
for (i = 0; i < num; i++) {
- w = snd_soc_dapm_new_control(dapm, widget);
- if (!w) {
+ ret = snd_soc_dapm_new_control(dapm, widget);
+ if (ret < 0) {
dev_err(dapm->dev,
- "ASoC: Failed to create DAPM control %s\n",
- widget->name);
- return -ENOMEM;
+ "ASoC: Failed to create DAPM control %s: %d\n",
+ widget->name, ret);
+ return ret;
}
widget++;
}
@@ -2831,126 +2985,54 @@
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
-int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_dapm_widget template;
- struct snd_soc_dapm_widget *w;
-
- WARN_ON(dapm->dev != dai->dev);
-
- memset(&template, 0, sizeof(template));
- template.reg = SND_SOC_NOPM;
-
- if (dai->driver->playback.stream_name) {
- template.id = snd_soc_dapm_dai;
- template.name = dai->driver->playback.stream_name;
- template.sname = dai->driver->playback.stream_name;
-
- dev_dbg(dai->dev, "adding %s widget\n",
- template.name);
-
- w = snd_soc_dapm_new_control(dapm, &template);
- if (!w) {
- dev_err(dapm->dev, "Failed to create %s widget\n",
- dai->driver->playback.stream_name);
- }
-
- w->priv = dai;
- dai->playback_widget = w;
- }
-
- if (dai->driver->capture.stream_name) {
- template.id = snd_soc_dapm_dai;
- template.name = dai->driver->capture.stream_name;
- template.sname = dai->driver->capture.stream_name;
-
- dev_dbg(dai->dev, "adding %s widget\n",
- template.name);
-
- w = snd_soc_dapm_new_control(dapm, &template);
- if (!w) {
- dev_err(dapm->dev, "Failed to create %s widget\n",
- dai->driver->capture.stream_name);
- }
-
- w->priv = dai;
- dai->capture_widget = w;
- }
-
- return 0;
-}
-
-int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
-{
- struct snd_soc_dapm_widget *dai_w, *w;
- struct snd_soc_dai *dai;
- struct snd_soc_dapm_route r;
-
- memset(&r, 0, sizeof(r));
-
- /* For each DAI widget... */
- list_for_each_entry(dai_w, &card->widgets, list) {
- if (dai_w->id != snd_soc_dapm_dai)
- continue;
-
- dai = dai_w->priv;
-
- /* ...find all widgets with the same stream and link them */
- list_for_each_entry(w, &card->widgets, list) {
- if (w->dapm != dai_w->dapm)
- continue;
-
- if (w->id == snd_soc_dapm_dai)
- continue;
-
- if (!w->sname)
- continue;
-
- if (dai->driver->playback.stream_name &&
- strstr(w->sname,
- dai->driver->playback.stream_name)) {
- r.source = dai->playback_widget->name;
- r.sink = w->name;
- dev_dbg(dai->dev, "%s -> %s\n",
- r.source, r.sink);
-
- snd_soc_dapm_add_route(w->dapm, &r);
- }
-
- if (dai->driver->capture.stream_name &&
- strstr(w->sname,
- dai->driver->capture.stream_name)) {
- r.source = w->name;
- r.sink = dai->capture_widget->name;
- dev_dbg(dai->dev, "%s -> %s\n",
- r.source, r.sink);
-
- snd_soc_dapm_add_route(w->dapm, &r);
- }
- }
- }
-
- return 0;
-}
-
static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
- int stream, struct snd_soc_dai *dai,
- int event)
+ const char *stream, int event)
{
struct snd_soc_dapm_widget *w;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
- w = dai->playback_widget;
- else
- w = dai->capture_widget;
+ if (!dapm)
+ return;
+ list_for_each_entry(w, &dapm->card->widgets, list)
+ {
+ if (!w->sname || w->dapm != dapm)
+ continue;
+ dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
+ w->name, w->sname, stream, event);
+ if (strstr(w->sname, stream)) {
+ dapm_mark_dirty(w, "stream event");
+ switch(event) {
+ case SND_SOC_DAPM_STREAM_START:
+ w->active = 1;
+ break;
+ case SND_SOC_DAPM_STREAM_STOP:
+ w->active = 0;
+ break;
+ case SND_SOC_DAPM_STREAM_SUSPEND:
+ case SND_SOC_DAPM_STREAM_RESUME:
+ case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
+ case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
+ break;
+ }
+ }
+ }
+
+ dapm_power_widgets(dapm, event);
+
+ /* do we need to notify any clients that DAPM stream is complete */
+ if (dapm->stream_event)
+ dapm->stream_event(dapm, event);
+}
+
+static void widget_stream_event(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dapm_widget *w, int event)
+{
if (!w)
return;
dapm_mark_dirty(w, "stream event");
- switch (event) {
+ switch(event) {
case SND_SOC_DAPM_STREAM_START:
w->active = 1;
break;
@@ -2967,6 +3049,30 @@
dapm_power_widgets(dapm, event);
}
+void snd_soc_dapm_rtd_stream_event(struct snd_soc_pcm_runtime *rtd,
+ int stream, int event)
+{
+ struct snd_soc_dapm_context *pdapm = &rtd->platform->dapm;
+ struct snd_soc_dapm_context *cdapm = &rtd->codec->dapm;
+
+ dev_dbg(rtd->dev, "rtd stream %d event %d\n", stream, event);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ widget_stream_event(pdapm, rtd->cpu_dai->playback_aif, event);
+ widget_stream_event(cdapm, rtd->codec_dai->playback_aif, event);
+ } else {
+ widget_stream_event(pdapm, rtd->cpu_dai->capture_aif, event);
+ widget_stream_event(cdapm, rtd->codec_dai->capture_aif, event);
+ }
+
+ /* do we need to notify any clients that DAPM stream is complete */
+ if (pdapm->stream_event)
+ pdapm->stream_event(pdapm, event);
+ if (cdapm->stream_event)
+ cdapm->stream_event(cdapm, event);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_rtd_stream_event);
+
/**
* snd_soc_dapm_stream_event - send a stream event to the dapm core
* @rtd: PCM runtime data
@@ -2978,17 +3084,30 @@
*
* Returns 0 for success else error.
*/
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
- struct snd_soc_dai *dai, int event)
+int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
+ const char *stream, int event)
{
struct snd_soc_codec *codec = rtd->codec;
- mutex_lock(&codec->mutex);
- soc_dapm_stream_event(&codec->dapm, stream, dai, event);
- mutex_unlock(&codec->mutex);
+ if (stream == NULL)
+ return 0;
+
+ //mutex_lock(&codec->mutex);
+ soc_dapm_stream_event(&codec->dapm, stream, event);
+ //mutex_unlock(&codec->mutex);
+
return 0;
}
+void snd_soc_dapm_codec_stream_event(struct snd_soc_codec *codec,
+ const char *stream, int event)
+{
+// mutex_lock(&codec->card->dapm_mutex);
+ soc_dapm_stream_event(&codec->dapm, stream, event);
+// mutex_unlock(&codec->card->dapm_mutex);
+}
+EXPORT_SYMBOL(snd_soc_dapm_codec_stream_event);
+
/**
* snd_soc_dapm_enable_pin - enable pin.
* @dapm: DAPM context
@@ -3228,13 +3347,9 @@
* standby.
*/
if (powerdown) {
- if (dapm->bias_level == SND_SOC_BIAS_ON)
- snd_soc_dapm_set_bias_level(dapm,
- SND_SOC_BIAS_PREPARE);
+ snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- if (dapm->bias_level == SND_SOC_BIAS_PREPARE)
- snd_soc_dapm_set_bias_level(dapm,
- SND_SOC_BIAS_STANDBY);
+ snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
}
}
@@ -3247,9 +3362,7 @@
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
- snd_soc_dapm_set_bias_level(&codec->dapm,
- SND_SOC_BIAS_OFF);
+ snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
}
}
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index ee4353f..9eb96e5 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -114,6 +114,23 @@
EXPORT_SYMBOL_GPL(snd_soc_jack_report);
/**
+ * snd_soc_jack_report_no_dapm - Report the current status for a jack
+ * without DAPM sync
+ * @jack: the jack
+ * @status: a bitmask of enum snd_jack_type values that are currently detected.
+ * @mask: a bitmask of enum snd_jack_type values that being reported.
+ */
+void snd_soc_jack_report_no_dapm(struct snd_soc_jack *jack, int status,
+ int mask)
+{
+ jack->status &= ~mask;
+ jack->status |= status & mask;
+
+ snd_jack_report(jack->jack, jack->status);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_report_no_dapm);
+
+/**
* snd_soc_jack_add_zones - Associate voltage zones with jack
*
* @jack: ASoC jack
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 0ad8dca..3a9fbe1 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -22,12 +22,84 @@
#include <linux/pm_runtime.h>
#include <linux/slab.h>
#include <linux/workqueue.h>
+#include <linux/debugfs.h>
+#include <linux/dma-mapping.h>
+#include <linux/export.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/soc-dpcm.h>
#include <sound/initval.h>
+#define MAX_BE_USERS 8 /* adjust if too low for everday use */
+
+static int soc_dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream);
+
+/* ASoC no host IO hardware.
+ * TODO: fine tune these values for all host less transfers.
+ */
+static const struct snd_pcm_hardware no_host_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = PAGE_SIZE >> 2,
+ .period_bytes_max = PAGE_SIZE >> 1,
+ .periods_min = 2,
+ .periods_max = 4,
+ .buffer_bytes_max = PAGE_SIZE,
+};
+
+/*
+ * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
+ * are not running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ list_for_each_entry(dpcm_params, &be->dpcm[stream].fe_clients, list_fe) {
+
+ if (dpcm_params->fe == fe)
+ continue;
+
+ if (dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_START ||
+ dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PAUSED ||
+ dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_SUSPEND)
+ return 0;
+ }
+ return 1;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
+
+/*
+ * We can only change hw params a BE DAI if any of it's FE are not prepared,
+ * running, paused or suspended for the specified stream direction.
+ */
+static int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ list_for_each_entry(dpcm_params, &be->dpcm[stream].fe_clients, list_fe) {
+
+ if (dpcm_params->fe == fe)
+ continue;
+
+ if (dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_START ||
+ dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PAUSED ||
+ dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_SUSPEND ||
+ dpcm_params->fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PREPARE)
+ return 0;
+ }
+ return 1;
+}
+
static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
struct snd_soc_dai *soc_dai)
{
@@ -68,7 +140,7 @@
* like the DAC/ADC resolution to use but there isn't right now.
*/
static int sample_sizes[] = {
- 24, 32,
+ 8, 16, 24, 32,
};
static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
@@ -98,6 +170,29 @@
}
/*
+ * stream event, send event to FE and all active BEs.
+ */
+int soc_dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe,
+ int dir, const char *stream, int event)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ snd_soc_dapm_rtd_stream_event(fe, dir, event);
+
+ list_for_each_entry(dpcm_params, &fe->dpcm[dir].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+
+ dev_dbg(be->dev, "pm: BE %s stream %s event %d dir %d\n",
+ be->dai_link->name, stream, event, dir);
+
+ snd_soc_dapm_rtd_stream_event(be, dir, event);
+ }
+
+ return 0;
+}
+
+/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
* startup for the cpu DAI, platform, machine and codec DAI.
@@ -119,12 +214,15 @@
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
+ if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST)
+ snd_soc_set_runtime_hwparams(substream, &no_host_hardware);
+
/* startup the audio subsystem */
if (cpu_dai->driver->ops->startup) {
ret = cpu_dai->driver->ops->startup(substream, cpu_dai);
if (ret < 0) {
- dev_err(cpu_dai->dev, "can't open interface %s: %d\n",
- cpu_dai->name, ret);
+ printk(KERN_ERR "asoc: can't open interface %s\n",
+ cpu_dai->name);
goto out;
}
}
@@ -132,8 +230,7 @@
if (platform->driver->ops && platform->driver->ops->open) {
ret = platform->driver->ops->open(substream);
if (ret < 0) {
- dev_err(platform->dev, "can't open platform %s: %d\n",
- platform->name, ret);
+ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
goto platform_err;
}
}
@@ -141,8 +238,8 @@
if (codec_dai->driver->ops->startup) {
ret = codec_dai->driver->ops->startup(substream, codec_dai);
if (ret < 0) {
- dev_err(codec_dai->dev, "can't open codec %s: %d\n",
- codec_dai->name, ret);
+ printk(KERN_ERR "asoc: can't open codec %s\n",
+ codec_dai->name);
goto codec_dai_err;
}
}
@@ -150,12 +247,15 @@
if (rtd->dai_link->ops && rtd->dai_link->ops->startup) {
ret = rtd->dai_link->ops->startup(substream);
if (ret < 0) {
- pr_err("asoc: %s startup failed: %d\n",
- rtd->dai_link->name, ret);
+ printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name);
goto machine_err;
}
}
+ /* DSP DAI links compat checks are different */
+ if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm)
+ goto dynamic;
+
/* Check that the codec and cpu DAIs are compatible */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw.rate_min =
@@ -248,6 +348,7 @@
pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
runtime->hw.rate_max);
+dynamic:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
cpu_dai->playback_active++;
codec_dai->playback_active++;
@@ -307,8 +408,9 @@
/* are we waiting on this codec DAI stream */
if (codec_dai->pop_wait == 1) {
codec_dai->pop_wait = 0;
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
- codec_dai, SND_SOC_DAPM_STREAM_STOP);
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
}
mutex_unlock(&rtd->pcm_mutex);
@@ -368,13 +470,13 @@
cpu_dai->runtime = NULL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
- rtd->dai_link->ignore_pmdown_time) {
+ if (codec->ignore_pmdown_time ||
+ rtd->dai_link->ignore_pmdown_time ||
+ !rtd->pmdown_time) {
/* powered down playback stream now */
snd_soc_dapm_stream_event(rtd,
- SNDRV_PCM_STREAM_PLAYBACK,
- codec_dai,
- SND_SOC_DAPM_STREAM_STOP);
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
} else {
/* start delayed pop wq here for playback streams */
codec_dai->pop_wait = 1;
@@ -383,8 +485,9 @@
}
} else {
/* capture streams can be powered down now */
- snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
- codec_dai, SND_SOC_DAPM_STREAM_STOP);
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
}
mutex_unlock(&rtd->pcm_mutex);
@@ -414,7 +517,7 @@
if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
ret = rtd->dai_link->ops->prepare(substream);
if (ret < 0) {
- pr_err("asoc: machine prepare error: %d\n", ret);
+ printk(KERN_ERR "asoc: machine prepare error\n");
goto out;
}
}
@@ -422,8 +525,7 @@
if (platform->driver->ops && platform->driver->ops->prepare) {
ret = platform->driver->ops->prepare(substream);
if (ret < 0) {
- dev_err(platform->dev, "platform prepare error: %d\n",
- ret);
+ printk(KERN_ERR "asoc: platform prepare error\n");
goto out;
}
}
@@ -431,8 +533,7 @@
if (codec_dai->driver->ops->prepare) {
ret = codec_dai->driver->ops->prepare(substream, codec_dai);
if (ret < 0) {
- dev_err(codec_dai->dev, "DAI prepare error: %d\n",
- ret);
+ printk(KERN_ERR "asoc: codec DAI prepare error\n");
goto out;
}
}
@@ -440,8 +541,7 @@
if (cpu_dai->driver->ops->prepare) {
ret = cpu_dai->driver->ops->prepare(substream, cpu_dai);
if (ret < 0) {
- dev_err(cpu_dai->dev, "DAI prepare error: %d\n",
- ret);
+ printk(KERN_ERR "asoc: cpu DAI prepare error\n");
goto out;
}
}
@@ -453,8 +553,14 @@
cancel_delayed_work(&rtd->delayed_work);
}
- snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai,
- SND_SOC_DAPM_STREAM_START);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(rtd,
+ codec_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
snd_soc_dai_digital_mute(codec_dai, 0);
@@ -482,7 +588,7 @@
if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) {
ret = rtd->dai_link->ops->hw_params(substream, params);
if (ret < 0) {
- pr_err("asoc: machine hw_params failed: %d\n", ret);
+ printk(KERN_ERR "asoc: machine hw_params failed\n");
goto out;
}
}
@@ -490,8 +596,8 @@
if (codec_dai->driver->ops->hw_params) {
ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai);
if (ret < 0) {
- dev_err(codec_dai->dev, "can't set %s hw params: %d\n",
- codec_dai->name, ret);
+ printk(KERN_ERR "asoc: can't set codec %s hw params\n",
+ codec_dai->name);
goto codec_err;
}
}
@@ -499,8 +605,8 @@
if (cpu_dai->driver->ops->hw_params) {
ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai);
if (ret < 0) {
- dev_err(cpu_dai->dev, "%s hw params failed: %d\n",
- cpu_dai->name, ret);
+ printk(KERN_ERR "asoc: interface %s hw params failed\n",
+ cpu_dai->name);
goto interface_err;
}
}
@@ -508,8 +614,8 @@
if (platform->driver->ops && platform->driver->ops->hw_params) {
ret = platform->driver->ops->hw_params(substream, params);
if (ret < 0) {
- dev_err(platform->dev, "%s hw params failed: %d\n",
- platform->name, ret);
+ printk(KERN_ERR "asoc: platform %s hw params failed\n",
+ platform->name);
goto platform_err;
}
}
@@ -518,6 +624,20 @@
cpu_dai->rate = params_rate(params);
codec_dai->rate = params_rate(params);
+ /* malloc a page for hostless IO.
+ * FIXME: rework with alsa-lib changes so that this malloc is not required.
+ */
+ if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST) {
+ substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV;
+ substream->dma_buffer.dev.dev = rtd->dev;
+ substream->dma_buffer.dev.dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ substream->dma_buffer.private_data = NULL;
+
+ ret = snd_pcm_lib_malloc_pages(substream, PAGE_SIZE);
+ if (ret < 0)
+ goto platform_err;
+ }
+
out:
mutex_unlock(&rtd->pcm_mutex);
return ret;
@@ -570,6 +690,9 @@
if (cpu_dai->driver->ops->hw_free)
cpu_dai->driver->ops->hw_free(substream, cpu_dai);
+ if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST)
+ snd_pcm_lib_free_pages(substream);
+
mutex_unlock(&rtd->pcm_mutex);
return 0;
}
@@ -602,6 +725,34 @@
return 0;
}
+int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ if (codec_dai->driver->ops->bespoke_trigger) {
+ ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (platform->driver->bespoke_trigger) {
+ ret = platform->driver->bespoke_trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (cpu_dai->driver->ops->bespoke_trigger) {
+ ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai);
+ if (ret < 0)
+ return ret;
+ }
+ return 0;
+}
+
/*
* soc level wrapper for pointer callback
* If cpu_dai, codec_dai, platform driver has the delay callback, than
@@ -634,6 +785,1648 @@
return offset;
}
+static inline int be_connect(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ if (!fe->dpcm[stream].runtime) {
+ dev_err(fe->dev, "%s no runtime\n", fe->dai_link->name);
+ return -ENODEV;
+ }
+
+ /* only add new dpcm_paramss */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+ if (dpcm_params->be == be && dpcm_params->fe == fe)
+ return 0;
+ }
+
+ dpcm_params = kzalloc(sizeof(struct snd_soc_dpcm_params), GFP_KERNEL);
+ if (!dpcm_params)
+ return -ENOMEM;
+
+ dpcm_params->be = be;
+ dpcm_params->fe = fe;
+ be->dpcm[stream].runtime = fe->dpcm[stream].runtime;
+ dpcm_params->state = SND_SOC_DPCM_LINK_STATE_NEW;
+ list_add(&dpcm_params->list_be, &fe->dpcm[stream].be_clients);
+ list_add(&dpcm_params->list_fe, &be->dpcm[stream].fe_clients);
+
+ dev_dbg(fe->dev, " connected new DSP %s path %s %s %s\n",
+ stream ? "capture" : "playback", fe->dai_link->name,
+ stream ? "<-" : "->", be->dai_link->name);
+
+#ifdef CONFIG_DEBUG_FS
+ dpcm_params->debugfs_state = debugfs_create_u32(be->dai_link->name, 0644,
+ fe->debugfs_dpcm_root, &dpcm_params->state);
+#endif
+
+ return 1;
+}
+
+static inline void be_reparent(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+ struct snd_pcm_substream *fe_substream, *be_substream;
+
+ /* reparent if BE is connected to other FEs */
+ if (!be->dpcm[stream].users)
+ return;
+
+ be_substream = snd_soc_dpcm_get_substream(be, stream);
+
+ list_for_each_entry(dpcm_params, &be->dpcm[stream].fe_clients, list_fe) {
+ if (dpcm_params->fe != fe) {
+
+ dev_dbg(fe->dev, " reparent %s path %s %s %s\n",
+ stream ? "capture" : "playback",
+ dpcm_params->fe->dai_link->name,
+ stream ? "<-" : "->", dpcm_params->be->dai_link->name);
+
+ fe_substream = snd_soc_dpcm_get_substream(dpcm_params->fe,
+ stream);
+ be_substream->runtime = fe_substream->runtime;
+ break;
+ }
+ }
+}
+
+static inline void be_disconnect(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params, *d;
+
+ list_for_each_entry_safe(dpcm_params, d, &fe->dpcm[stream].be_clients, list_be) {
+ dev_dbg(fe->dev, "BE %s disconnect check for %s\n",
+ stream ? "capture" : "playback",
+ dpcm_params->be->dai_link->name);
+
+ if (dpcm_params->state == SND_SOC_DPCM_LINK_STATE_FREE) {
+ dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n",
+ stream ? "capture" : "playback", fe->dai_link->name,
+ stream ? "<-" : "->", dpcm_params->be->dai_link->name);
+
+ /* BEs still alive need new FE */
+ be_reparent(fe, dpcm_params->be, stream);
+
+#ifdef CONFIG_DEBUG_FS
+ debugfs_remove(dpcm_params->debugfs_state);
+#endif
+
+ list_del(&dpcm_params->list_be);
+ list_del(&dpcm_params->list_fe);
+ kfree(dpcm_params);
+ }
+ }
+}
+
+static struct snd_soc_pcm_runtime *be_get_rtd(struct snd_soc_card *card,
+ struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_pcm_runtime *be;
+ int i;
+
+ if (!widget->sname) {
+ dev_err(card->dev, "widget %s has no stream\n", widget->name);
+ return NULL;
+ }
+
+ for (i = 0; i < card->num_links; i++) {
+ be = &card->rtd[i];
+
+ if (!strcmp(widget->sname, be->dai_link->stream_name))
+ return be;
+ }
+
+ dev_err(card->dev, "can't get BE for %s\n", widget->name);
+ return NULL;
+}
+
+static struct snd_soc_dapm_widget *be_get_widget(struct snd_soc_card *card,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dapm_widget *widget;
+
+ list_for_each_entry(widget, &card->widgets, list) {
+
+ if (!widget->sname)
+ continue;
+
+ if (!strcmp(widget->sname, rtd->dai_link->stream_name))
+ return widget;
+ }
+
+ dev_err(card->dev, "can't get widget for %s\n",
+ rtd->dai_link->stream_name);
+ return NULL;
+}
+
+static int widget_in_list(struct snd_soc_dapm_widget_list *list,
+ struct snd_soc_dapm_widget *widget)
+{
+ int i;
+
+ for (i = 0; i < list->num_widgets; i++) {
+ if (widget == list->widgets[i])
+ return 1;
+ }
+
+ return 0;
+}
+
+static int fe_path_get(struct snd_soc_pcm_runtime *fe,
+ int stream, struct snd_soc_dapm_widget_list **list_)
+{
+ struct snd_soc_dai *cpu_dai = fe->cpu_dai;
+ struct snd_soc_dapm_widget_list *list;
+ int paths;
+
+ list = kzalloc(sizeof(struct snd_soc_dapm_widget_list) +
+ sizeof(struct snd_soc_dapm_widget *), GFP_KERNEL);
+ if (list == NULL)
+ return -ENOMEM;
+
+ /* get number of valid DAI paths and their widgets */
+ paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, &list);
+
+ dev_dbg(fe->dev, "found %d audio %s paths\n", paths,
+ stream ? "capture" : "playback");
+
+ *list_ = list;
+ return paths;
+}
+
+static inline void fe_path_put(struct snd_soc_dapm_widget_list **list)
+{
+ kfree(*list);
+}
+
+static int be_prune_old(struct snd_soc_pcm_runtime *fe, int stream,
+ struct snd_soc_dapm_widget_list **list_)
+{
+ struct snd_soc_card *card = fe->card;
+ struct snd_soc_dpcm_params *dpcm_params;
+ struct snd_soc_dapm_widget_list *list = *list_;
+ struct snd_soc_dapm_widget *widget;
+ int old = 0;
+
+ /* Destroy any old FE <--> BE connections */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ /* is there a valid widget for this BE */
+ widget = be_get_widget(card, dpcm_params->be);
+ if (!widget) {
+ dev_err(fe->dev, "no widget found for %s\n",
+ dpcm_params->be->dai_link->name);
+ continue;
+ }
+
+ /* prune the BE if it's no longer in our active list */
+ if (widget_in_list(list, widget))
+ continue;
+
+ dev_dbg(fe->dev, "pruning %s BE %s for %s\n",
+ stream ? "capture" : "playback", dpcm_params->be->dai_link->name,
+ fe->dai_link->name);
+ dpcm_params->state = SND_SOC_DPCM_LINK_STATE_FREE;
+ dpcm_params->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+ old++;
+ }
+
+ dev_dbg(fe->dev, "found %d old BEs\n", old);
+ return old;
+}
+
+static int be_add_new(struct snd_soc_pcm_runtime *fe, int stream,
+ struct snd_soc_dapm_widget_list **list_)
+{
+ struct snd_soc_card *card = fe->card;
+ struct snd_soc_dapm_widget_list *list = *list_;
+ enum snd_soc_dapm_type be_type;
+ int i, new = 0, err;
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ be_type = snd_soc_dapm_aif_out;
+ else
+ be_type = snd_soc_dapm_aif_in;
+
+ /* Create any new FE <--> BE connections */
+ for (i = 0; i < list->num_widgets; i++) {
+
+ if (list->widgets[i]->id == be_type) {
+ struct snd_soc_pcm_runtime *be;
+
+ /* is there a valid BE rtd for this widget */
+ be = be_get_rtd(card, list->widgets[i]);
+ if (!be) {
+ dev_err(fe->dev, "no BE found for %s\n",
+ list->widgets[i]->name);
+ continue;
+ }
+
+ /* don't connect if FE is not running */
+ if (!fe->dpcm[stream].runtime)
+ continue;
+
+ /* newly connected FE and BE */
+ err = be_connect(fe, be, stream);
+ if (err < 0) {
+ dev_err(fe->dev, "can't connect %s\n", list->widgets[i]->name);
+ break;
+ } else if (err == 0) /* already connected */
+ continue;
+
+ /* new */
+ be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+ new++;
+ }
+ }
+
+ dev_dbg(fe->dev, "found %d new BEs\n", new);
+ return new;
+}
+
+/*
+ * Find the corresponding BE DAIs that source or sink audio to this
+ * FE substream.
+ */
+static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe,
+ int stream, struct snd_soc_dapm_widget_list **list, int new)
+{
+ if (new)
+ return be_add_new(fe, stream, list);
+ else
+ return be_prune_old(fe, stream, list);
+ return 0;
+}
+
+/*
+ * Clear the runtime pending state of all BE's.
+ */
+static void fe_clear_pending(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be)
+ dpcm_params->be->dpcm[stream].runtime_update =
+ SND_SOC_DPCM_UPDATE_NO;
+}
+
+/* Unwind the BE startup */
+static void soc_dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* disable any enabled and non active backends */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ if (be->dpcm[stream].users == 0)
+ dev_err(be->dev, "no users %s at close - state %d\n",
+ stream ? "capture" : "playback", be->dpcm[stream].state);
+
+ if (--be->dpcm[stream].users != 0)
+ continue;
+
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)
+ continue;
+
+ soc_pcm_close(be_substream);
+ be_substream->runtime = NULL;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+ }
+}
+
+/* Startup all new BE */
+static int soc_dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+ int err, count = 0;
+
+ /* only startup BE DAIs that are either sinks or sources to this FE DAI */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ /* is this op for this BE ? */
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ /* first time the dpcm_params is open ? */
+ if (be->dpcm[stream].users == MAX_BE_USERS)
+ dev_err(be->dev, "too many users %s at open - state %d\n",
+ stream ? "capture" : "playback", be->dpcm[stream].state);
+
+ if (be->dpcm[stream].users++ != 0)
+ continue;
+
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE))
+ continue;
+
+ dev_dbg(be->dev, "dpcm: open BE %s\n", be->dai_link->name);
+
+ be_substream->runtime = be->dpcm[stream].runtime;
+ err = soc_pcm_open(be_substream);
+ if (err < 0) {
+ dev_err(be->dev, "BE open failed %d\n", err);
+ be->dpcm[stream].users--;
+ if (be->dpcm[stream].users < 0)
+ dev_err(be->dev, "no users %s at unwind - state %d\n",
+ stream ? "capture" : "playback",
+ be->dpcm[stream].state);
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+ goto unwind;
+ }
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
+ count++;
+ }
+
+ return count;
+
+unwind:
+ /* disable any enabled and non active backends */
+ list_for_each_entry_continue_reverse(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ if (be->dpcm[stream].users == 0)
+ dev_err(be->dev, "no users %s at close - state %d\n",
+ stream ? "capture" : "playback", be->dpcm[stream].state);
+
+ if (--be->dpcm[stream].users != 0)
+ continue;
+
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN)
+ continue;
+
+ soc_pcm_close(be_substream);
+ be_substream->runtime = NULL;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+ }
+
+ return err;
+}
+
+void soc_dpcm_set_dynamic_runtime(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw.rate_min = cpu_dai_drv->playback.rate_min;
+ runtime->hw.rate_max = cpu_dai_drv->playback.rate_max;
+ runtime->hw.channels_min = cpu_dai_drv->playback.channels_min;
+ runtime->hw.channels_max = cpu_dai_drv->playback.channels_max;
+ runtime->hw.formats &= cpu_dai_drv->playback.formats;
+ runtime->hw.rates = cpu_dai_drv->playback.rates;
+ } else {
+ runtime->hw.rate_min = cpu_dai_drv->capture.rate_min;
+ runtime->hw.rate_max = cpu_dai_drv->capture.rate_max;
+ runtime->hw.channels_min = cpu_dai_drv->capture.channels_min;
+ runtime->hw.channels_max = cpu_dai_drv->capture.channels_max;
+ runtime->hw.formats &= cpu_dai_drv->capture.formats;
+ runtime->hw.rates = cpu_dai_drv->capture.rates;
+ }
+}
+
+static int soc_dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
+{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+ struct snd_pcm_runtime *runtime = fe_substream->runtime;
+ int stream = fe_substream->stream, ret = 0;
+
+ mutex_lock(&fe->card->dpcm_mutex);
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ ret = soc_dpcm_be_dai_startup(fe, fe_substream->stream);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: failed to start some BEs %d\n", ret);
+ goto be_err;
+ }
+
+ dev_dbg(fe->dev, "dpcm: open FE %s\n", fe->dai_link->name);
+
+ /* start the DAI frontend */
+ ret = soc_pcm_open(fe_substream);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: failed to start FE %d\n", ret);
+ goto unwind;
+ }
+
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
+
+ soc_dpcm_set_dynamic_runtime(fe_substream);
+ snd_pcm_limit_hw_rates(runtime);
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ mutex_unlock(&fe->card->dpcm_mutex);
+ return 0;
+
+unwind:
+ soc_dpcm_be_dai_startup_unwind(fe, fe_substream->stream);
+be_err:
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ mutex_unlock(&fe->card->dpcm_mutex);
+ return ret;
+}
+
+/* BE shutdown - called on DAPM sync updates (i.e. FE is already running)*/
+static int soc_dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* only shutdown backends that are either sinks or sources to this frontend DAI */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ /* is this op for this BE ? */
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ if (be->dpcm[stream].users == 0)
+ dev_err(be->dev, "no users %s at close - state %d\n",
+ stream ? "capture" : "playback", be->dpcm[stream].state);
+
+ if (--be->dpcm[stream].users != 0)
+ continue;
+
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN))
+ continue;
+
+ dev_dbg(be->dev, "dpcm: close BE %s\n",
+ dpcm_params->fe->dai_link->name);
+
+ soc_pcm_close(be_substream);
+ be_substream->runtime = NULL;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+ }
+ return 0;
+}
+
+/* FE +BE shutdown - called on FE PCM ops */
+static int soc_dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int stream = substream->stream;
+
+ mutex_lock(&fe->card->dpcm_mutex);
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ dev_dbg(fe->dev, "dpcm: close FE %s\n", fe->dai_link->name);
+
+ /* now shutdown the frontend */
+ soc_pcm_close(substream);
+
+ /* shutdown the BEs */
+ soc_dpcm_be_dai_shutdown(fe, substream->stream);
+ /* run the stream event for each BE */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+ else
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_STOP);
+
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+ mutex_unlock(&fe->card->dpcm_mutex);
+ return 0;
+}
+
+static int soc_dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+ int ret;
+
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ /* is this op for this BE ? */
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ /* only allow hw_params() if no connected FEs are running */
+ if (!snd_soc_dpcm_can_be_params(fe, be, stream))
+ continue;
+
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE))
+ continue;
+
+ dev_dbg(be->dev, "dpcm: hw_params BE %s\n",
+ dpcm_params->fe->dai_link->name);
+
+ /* copy params for each dpcm_params */
+ memcpy(&dpcm_params->hw_params, &fe->dpcm[stream].hw_params,
+ sizeof(struct snd_pcm_hw_params));
+
+ /* perform any hw_params fixups */
+ if (be->dai_link->be_hw_params_fixup) {
+ ret = be->dai_link->be_hw_params_fixup(be,
+ &dpcm_params->hw_params);
+ if (ret < 0) {
+ dev_err(be->dev,
+ "dpcm: hw_params BE fixup failed %d\n",
+ ret);
+ goto unwind;
+ }
+ }
+
+ ret = soc_pcm_hw_params(be_substream, &dpcm_params->hw_params);
+ if (ret < 0) {
+ dev_err(dpcm_params->be->dev, "dpcm: hw_params BE failed %d\n", ret);
+ goto unwind;
+ }
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS;
+ }
+ return 0;
+
+unwind:
+ /* disable any enabled and non active backends */
+ list_for_each_entry_continue_reverse(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ /* only allow hw_free() if no connected FEs are running */
+ if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+ continue;
+
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_OPEN) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ continue;
+
+ soc_pcm_hw_free(be_substream);
+ }
+
+ return ret;
+}
+
+int soc_dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int ret, stream = substream->stream;
+
+ mutex_lock(&fe->card->dpcm_mutex);
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ memcpy(&fe->dpcm[substream->stream].hw_params, params,
+ sizeof(struct snd_pcm_hw_params));
+ ret = soc_dpcm_be_dai_hw_params(fe, substream->stream);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: hw_params failed for some BEs %d\n", ret);
+ goto out;
+ }
+
+ dev_dbg(fe->dev, "dpcm: hw_params FE %s rate %d chan %x fmt %d\n",
+ fe->dai_link->name, params_rate(params), params_channels(params),
+ params_format(params));
+
+ /* call hw_params on the frontend */
+ ret = soc_pcm_hw_params(substream, params);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: hw_params FE failed %d\n", ret);
+ soc_dpcm_be_dai_hw_free(fe, stream);
+ } else
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS;
+
+out:
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ mutex_unlock(&fe->card->dpcm_mutex);
+ return ret;
+}
+
+static int dpcm_do_trigger(struct snd_soc_dpcm_params *dpcm_params,
+ struct snd_pcm_substream *substream, int cmd)
+{
+ int ret;
+
+ dev_dbg(dpcm_params->be->dev, "dpcm: trigger BE %s cmd %d\n",
+ dpcm_params->fe->dai_link->name, cmd);
+
+ ret = soc_pcm_trigger(substream, cmd);
+ if (ret < 0)
+ dev_err(dpcm_params->be->dev,"dpcm: trigger BE failed %d\n", ret);
+
+ return ret;
+}
+
+int soc_dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+ int ret = 0;
+
+ if ((cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) ||
+ (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH))
+ return ret;
+
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ /* is this op for this BE ? */
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ continue;
+
+ ret = dpcm_do_trigger(dpcm_params, be_substream, cmd);
+ if (ret)
+ return ret;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND))
+ continue;
+
+ ret = dpcm_do_trigger(dpcm_params, be_substream, cmd);
+ if (ret)
+ return ret;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
+ continue;
+
+ ret = dpcm_do_trigger(dpcm_params, be_substream, cmd);
+ if (ret)
+ return ret;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+ continue;
+
+ if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+ continue;
+
+ ret = dpcm_do_trigger(dpcm_params, be_substream, cmd);
+ if (ret)
+ return ret;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)
+ continue;
+
+ if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+ continue;
+
+ ret = dpcm_do_trigger(dpcm_params, be_substream, cmd);
+ if (ret)
+ return ret;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND;
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+ continue;
+
+ if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+ continue;
+
+ ret = dpcm_do_trigger(dpcm_params, be_substream, cmd);
+ if (ret)
+ return ret;
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED;
+ break;
+ }
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(soc_dpcm_be_dai_trigger);
+
+int soc_dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int stream = substream->stream, ret;
+ enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ switch (trigger) {
+ case SND_SOC_DPCM_TRIGGER_PRE:
+ /* call trigger on the frontend before the backend. */
+
+ dev_dbg(fe->dev, "dpcm: pre trigger FE %s cmd %d\n",
+ fe->dai_link->name, cmd);
+
+ ret = soc_pcm_trigger(substream, cmd);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+ goto out;
+ }
+
+ ret = soc_dpcm_be_dai_trigger(fe, substream->stream, cmd);
+ break;
+ case SND_SOC_DPCM_TRIGGER_POST:
+ /* call trigger on the frontend after the backend. */
+
+ ret = soc_dpcm_be_dai_trigger(fe, substream->stream, cmd);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+ goto out;
+ }
+
+ dev_dbg(fe->dev, "dpcm: post trigger FE %s cmd %d\n",
+ fe->dai_link->name, cmd);
+
+ ret = soc_pcm_trigger(substream, cmd);
+ break;
+ case SND_SOC_DPCM_TRIGGER_BESPOKE:
+ /* bespoke trigger() - handles both FE and BEs */
+
+ dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd %d\n",
+ fe->dai_link->name, cmd);
+
+ ret = soc_pcm_bespoke_trigger(substream, cmd);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+ goto out;
+ }
+ break;
+ default:
+ dev_err(fe->dev, "dpcm: invalid trigger cmd %d for %s\n", cmd,
+ fe->dai_link->name);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
+ break;
+ }
+
+out:
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ return ret;
+}
+
+static int soc_dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+ int ret = 0;
+
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ /* is this op for this BE ? */
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ continue;
+
+ dev_dbg(be->dev, "dpcm: prepare BE %s\n",
+ dpcm_params->fe->dai_link->name);
+
+ ret = soc_pcm_prepare(be_substream);
+ if (ret < 0) {
+ dev_err(be->dev, "dpcm: backend prepare failed %d\n",
+ ret);
+ break;
+ }
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
+ }
+ return ret;
+}
+
+int soc_dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int stream = substream->stream, ret = 0;
+
+ mutex_lock(&fe->card->dpcm_mutex);
+
+ dev_dbg(fe->dev, "dpcm: prepare FE %s\n", fe->dai_link->name);
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ /* there is no point preparing this FE if there are no BEs */
+ if (list_empty(&fe->dpcm[stream].be_clients)) {
+ dev_err(fe->dev, "dpcm: no backend DAIs enabled for %s\n",
+ fe->dai_link->name);
+ ret = -EINVAL;
+ goto out;
+ }
+
+ ret = soc_dpcm_be_dai_prepare(fe, substream->stream);
+ if (ret < 0)
+ goto out;
+
+ /* call prepare on the frontend */
+ ret = soc_pcm_prepare(substream);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: prepare FE %s failed\n", fe->dai_link->name);
+ goto out;
+ }
+
+ /* run the stream event for each BE */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
+
+out:
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ mutex_unlock(&fe->card->dpcm_mutex);
+ return ret;
+}
+
+static int soc_dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* only hw_params backends that are either sinks or sources
+ * to this frontend DAI */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_pcm_substream *be_substream =
+ snd_soc_dpcm_get_substream(be, stream);
+
+ /* is this op for this BE ? */
+ if (!snd_soc_dpcm_be_can_update(fe, be, stream))
+ continue;
+
+ /* only free hw when no longer used - check all FEs */
+ if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
+ continue;
+
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ continue;
+
+ dev_dbg(be->dev, "dpcm: hw_free BE %s\n",
+ dpcm_params->fe->dai_link->name);
+
+ soc_pcm_hw_free(be_substream);
+
+ be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE;
+ }
+
+ return 0;
+}
+
+int soc_dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int err, stream = substream->stream;
+
+ mutex_lock(&fe->card->dpcm_mutex);
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+
+ dev_dbg(fe->dev, "dpcm: hw_free FE %s\n", fe->dai_link->name);
+
+ /* call hw_free on the frontend */
+ err = soc_pcm_hw_free(substream);
+ if (err < 0)
+ dev_err(fe->dev,"dpcm: hw_free FE %s failed\n", fe->dai_link->name);
+
+ /* only hw_params backends that are either sinks or sources
+ * to this frontend DAI */
+ err = soc_dpcm_be_dai_hw_free(fe, stream);
+
+ fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE;
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+ mutex_unlock(&fe->card->dpcm_mutex);
+ return 0;
+}
+
+static int soc_pcm_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_platform *platform = rtd->platform;
+
+ if (platform->driver->ops->ioctl)
+ return platform->driver->ops->ioctl(substream, cmd, arg);
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream);
+ enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
+ int err;
+
+ dev_dbg(fe->dev, "runtime %s close on FE %s\n",
+ stream ? "capture" : "playback", fe->dai_link->name);
+
+ if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) {
+ /* call bespoke trigger - FE takes care of all BE triggers */
+ dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd stop\n",
+ fe->dai_link->name);
+
+ err = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP);
+ if (err < 0)
+ dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err);
+ } else {
+ dev_dbg(fe->dev, "dpcm: trigger FE %s cmd stop\n",
+ fe->dai_link->name);
+
+ err = soc_dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP);
+ if (err < 0)
+ dev_err(fe->dev,"dpcm: trigger FE failed %d\n", err);
+ }
+
+ err = soc_dpcm_be_dai_hw_free(fe, stream);
+ if (err < 0)
+ dev_err(fe->dev,"dpcm: hw_free FE failed %d\n", err);
+
+ err = soc_dpcm_be_dai_shutdown(fe, stream);
+ if (err < 0)
+ dev_err(fe->dev,"dpcm: shutdown FE failed %d\n", err);
+
+ /* run the stream event for each BE */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_NOP);
+ else
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_NOP);
+
+ return 0;
+}
+
+static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream);
+ struct snd_soc_dpcm_params *dpcm_params;
+ enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
+ int ret;
+
+ dev_dbg(fe->dev, "runtime %s open on FE %s\n",
+ stream ? "capture" : "playback", fe->dai_link->name);
+
+ /* Only start the BE if the FE is ready */
+ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE ||
+ fe->dpcm[stream].state == SND_SOC_DPCM_STATE_CLOSE)
+ return -EINVAL;
+
+ /* startup must always be called for new BEs */
+ ret = soc_dpcm_be_dai_startup(fe, stream);
+ if (ret < 0) {
+ goto disconnect;
+ return ret;
+ }
+
+ /* keep going if FE state is > open */
+ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN)
+ return 0;
+
+ ret = soc_dpcm_be_dai_hw_params(fe, stream);
+ if (ret < 0) {
+ goto close;
+ return ret;
+ }
+
+ /* keep going if FE state is > hw_params */
+ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS)
+ return 0;
+
+
+ ret = soc_dpcm_be_dai_prepare(fe, stream);
+ if (ret < 0) {
+ goto hw_free;
+ return ret;
+ }
+
+ /* run the stream event for each BE */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->playback.stream_name,
+ SND_SOC_DAPM_STREAM_NOP);
+ else
+ soc_dpcm_dapm_stream_event(fe, stream,
+ fe->cpu_dai->driver->capture.stream_name,
+ SND_SOC_DAPM_STREAM_NOP);
+
+ /* keep going if FE state is > prepare */
+ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_PREPARE ||
+ fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP)
+ return 0;
+
+ if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) {
+ /* call trigger on the frontend - FE takes care of all BE triggers */
+ dev_dbg(fe->dev, "dpcm: bespoke trigger FE %s cmd start\n",
+ fe->dai_link->name);
+
+ ret = soc_pcm_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: bespoke trigger FE failed %d\n", ret);
+ goto hw_free;
+ }
+ } else {
+ dev_dbg(fe->dev, "dpcm: trigger FE %s cmd start\n",
+ fe->dai_link->name);
+
+ ret = soc_dpcm_be_dai_trigger(fe, stream,
+ SNDRV_PCM_TRIGGER_START);
+ if (ret < 0) {
+ dev_err(fe->dev,"dpcm: trigger FE failed %d\n", ret);
+ goto hw_free;
+ }
+ }
+
+ return 0;
+
+hw_free:
+ soc_dpcm_be_dai_hw_free(fe, stream);
+close:
+ soc_dpcm_be_dai_shutdown(fe, stream);
+disconnect:
+ /* disconnect any non started BEs */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+ dpcm_params->state = SND_SOC_DPCM_LINK_STATE_FREE;
+ }
+
+ return ret;
+}
+
+static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ int ret;
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+ ret = dpcm_run_update_startup(fe, stream);
+ if (ret < 0)
+ dev_err(fe->dev, "failed to startup some BEs\n");
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+ return ret;
+}
+
+static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
+{
+ int ret;
+
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+ ret = dpcm_run_update_shutdown(fe, stream);
+ if (ret < 0)
+ dev_err(fe->dev, "failed to shutdown some BEs\n");
+ fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+
+ return ret;
+}
+
+/* called when any mixer updates change FE -> BE the stream */
+int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *widget)
+{
+ struct snd_soc_card *card;
+ int i, ret = 0, old, new, paths;
+
+ if (widget->codec)
+ card = widget->codec->card;
+ else if (widget->platform)
+ card = widget->platform->card;
+ else
+ return -EINVAL;
+
+ mutex_lock(&card->dpcm_mutex);
+
+ for (i = 0; i < card->num_rtd; i++) {
+ struct snd_soc_dapm_widget_list *list;
+ struct snd_soc_pcm_runtime *fe = &card->rtd[i];
+
+ /* make sure link is FE */
+ if (!fe->dai_link->dynamic)
+ continue;
+
+ /* only check active links */
+ if (!fe->cpu_dai->active)
+ continue;
+
+ /* DAPM sync will call this to update DSP paths */
+ dev_dbg(fe->dev, "DPCM runtime update for FE %s\n", fe->dai_link->name);
+
+ /* skip if FE doesn't have playback capability */
+ if (!fe->cpu_dai->driver->playback.channels_min)
+ goto capture;
+
+ paths = fe_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
+ if (paths < 0) {
+ dev_warn(fe->dev, "%s no valid %s route from source to sink\n",
+ fe->dai_link->name, "playback");
+ ret = paths;
+ goto out;
+ }
+
+ /* update any new playback paths */
+ new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1);
+ if (new) {
+ dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ fe_clear_pending(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+
+ /* update any old playback paths */
+ old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0);
+ if (old) {
+ dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ fe_clear_pending(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
+ }
+
+capture:
+ /* skip if FE doesn't have capture capability */
+ if (!fe->cpu_dai->driver->capture.channels_min)
+ continue;
+
+ paths = fe_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
+ if (paths < 0) {
+ dev_warn(fe->dev, "%s no valid %s route from source to sink\n",
+ fe->dai_link->name, "capture");
+ ret = paths;
+ goto out;
+ }
+
+ /* update any new capture paths */
+ new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1);
+ if (new) {
+ dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+ fe_clear_pending(fe, SNDRV_PCM_STREAM_CAPTURE);
+ be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+ }
+
+ /* update any old capture paths */
+ old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0);
+ if (old) {
+ dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+ fe_clear_pending(fe, SNDRV_PCM_STREAM_CAPTURE);
+ be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+ }
+
+ fe_path_put(&list);
+ }
+
+out:
+ mutex_unlock(&card->dpcm_mutex);
+ return ret;
+}
+
+int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->codec_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "BE digital mute %s\n", be->dai_link->name);
+
+ if (drv->ops->digital_mute && dai->playback_active)
+ drv->ops->digital_mute(dai, mute);
+ }
+
+ return 0;
+}
+
+int soc_dpcm_be_cpu_dai_suspend(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* suspend for playback */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI playback suspend %s\n",
+ be->dai_link->name);
+
+ if (drv->suspend && !drv->ac97_control)
+ drv->suspend(dai);
+ }
+
+ /* suspend for capture */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI capture suspend %s\n",
+ be->dai_link->name);
+
+ if (drv->suspend && !drv->ac97_control)
+ drv->suspend(dai);
+ }
+
+ return 0;
+}
+
+int soc_dpcm_be_ac97_cpu_dai_suspend(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* suspend for playback */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI playback suspend %s\n",
+ be->dai_link->name);
+
+ if (drv->suspend && drv->ac97_control)
+ drv->suspend(dai);
+ }
+
+ /* suspend for capture */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI capture suspend %s\n",
+ be->dai_link->name);
+
+ if (drv->suspend && drv->ac97_control)
+ drv->suspend(dai);
+ }
+
+ return 0;
+}
+
+int soc_dpcm_be_platform_suspend(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* suspend for playback */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_platform *platform = be->platform;
+ struct snd_soc_platform_driver *drv = platform->driver;
+ struct snd_soc_dai *dai = be->cpu_dai;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE platform playback suspend %s\n",
+ be->dai_link->name);
+
+ if (drv->suspend && !platform->suspended) {
+ drv->suspend(dai);
+ platform->suspended = 1;
+ }
+ }
+
+ /* suspend for capture */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_platform *platform = be->platform;
+ struct snd_soc_platform_driver *drv = platform->driver;
+ struct snd_soc_dai *dai = be->cpu_dai;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE platform capture suspend %s\n",
+ be->dai_link->name);
+
+ if (drv->suspend && !platform->suspended) {
+ drv->suspend(dai);
+ platform->suspended = 1;
+ }
+ }
+ return 0;
+}
+
+int soc_dpcm_fe_suspend(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dai *dai = fe->cpu_dai;
+ struct snd_soc_dai_driver *dai_drv = dai->driver;
+ struct snd_soc_platform *platform = fe->platform;
+ struct snd_soc_platform_driver *plat_drv = platform->driver;
+
+ if (dai_drv->suspend && !dai_drv->ac97_control)
+ dai_drv->suspend(dai);
+
+ if (plat_drv->suspend && !platform->suspended) {
+ plat_drv->suspend(dai);
+ platform->suspended = 1;
+ }
+
+ soc_dpcm_be_cpu_dai_suspend(fe);
+ soc_dpcm_be_platform_suspend(fe);
+
+ return 0;
+}
+
+int soc_dpcm_be_cpu_dai_resume(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* resume for playback */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI playback resume %s\n",
+ be->dai_link->name);
+
+ if (drv->resume && !drv->ac97_control)
+ drv->resume(dai);
+ }
+
+ /* suspend for capture */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI capture resume %s\n",
+ be->dai_link->name);
+
+ if (drv->resume && !drv->ac97_control)
+ drv->resume(dai);
+ }
+
+ return 0;
+}
+
+int soc_dpcm_be_ac97_cpu_dai_resume(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* resume for playback */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI playback resume %s\n",
+ be->dai_link->name);
+
+ if (drv->resume && drv->ac97_control)
+ drv->resume(dai);
+ }
+
+ /* suspend for capture */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_dai *dai = be->cpu_dai;
+ struct snd_soc_dai_driver *drv = dai->driver;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE CPU DAI capture resume %s\n",
+ be->dai_link->name);
+
+ if (drv->resume && drv->ac97_control)
+ drv->resume(dai);
+ }
+
+ return 0;
+}
+
+int soc_dpcm_be_platform_resume(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dpcm_params *dpcm_params;
+
+ /* resume for playback */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_platform *platform = be->platform;
+ struct snd_soc_platform_driver *drv = platform->driver;
+ struct snd_soc_dai *dai = be->cpu_dai;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE platform playback resume %s\n",
+ be->dai_link->name);
+
+ if (drv->resume && platform->suspended) {
+ drv->resume(dai);
+ platform->suspended = 0;
+ }
+ }
+
+ /* resume for capture */
+ list_for_each_entry(dpcm_params,
+ &fe->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients, list_be) {
+
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+ struct snd_soc_platform *platform = be->platform;
+ struct snd_soc_platform_driver *drv = platform->driver;
+ struct snd_soc_dai *dai = be->cpu_dai;
+
+ if (be->dai_link->ignore_suspend)
+ continue;
+
+ dev_dbg(be->dev, "pm: BE platform capture resume %s\n",
+ be->dai_link->name);
+
+ if (drv->resume && platform->suspended) {
+ drv->resume(dai);
+ platform->suspended = 0;
+ }
+ }
+
+ return 0;
+}
+
+int soc_dpcm_fe_resume(struct snd_soc_pcm_runtime *fe)
+{
+ struct snd_soc_dai *dai = fe->cpu_dai;
+ struct snd_soc_dai_driver *dai_drv = dai->driver;
+ struct snd_soc_platform *platform = fe->platform;
+ struct snd_soc_platform_driver *plat_drv = platform->driver;
+
+ soc_dpcm_be_cpu_dai_resume(fe);
+ soc_dpcm_be_platform_resume(fe);
+
+ if (dai_drv->resume && !dai_drv->ac97_control)
+ dai_drv->resume(dai);
+
+ if (plat_drv->resume && platform->suspended) {
+ plat_drv->resume(dai);
+ platform->suspended = 0;
+ }
+
+ return 0;
+}
+
+/* called when opening FE stream */
+int soc_dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
+{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+ struct snd_soc_dpcm_params *dpcm_params;
+ struct snd_soc_dapm_widget_list *list;
+ int ret;
+ int stream = fe_substream->stream;
+
+ fe->dpcm[stream].runtime = fe_substream->runtime;
+
+ if (fe_path_get(fe, stream, &list) <= 0) {
+ dev_warn(fe->dev, "asoc: %s no valid %s route from source to sink\n",
+ fe->dai_link->name, stream ? "capture" : "playback");
+ return -EINVAL;
+ }
+
+ /* calculate valid and active FE <-> BE dpcm_paramss */
+ dpcm_process_paths(fe, stream, &list, 1);
+
+ ret = soc_dpcm_fe_dai_startup(fe_substream);
+ if (ret < 0) {
+ /* clean up all links */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be)
+ dpcm_params->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+ be_disconnect(fe, stream);
+ fe->dpcm[stream].runtime = NULL;
+ }
+
+ fe_clear_pending(fe, stream);
+ fe_path_put(&list);
+ return ret;
+}
+
+/* called when closing FE stream */
+int soc_dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
+{
+ struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
+ struct snd_soc_dpcm_params *dpcm_params;
+ int stream = fe_substream->stream, ret;
+
+ ret = soc_dpcm_fe_dai_shutdown(fe_substream);
+
+ /* mark FE's links ready to prune */
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be)
+ dpcm_params->state = SND_SOC_DPCM_LINK_STATE_FREE;
+
+ be_disconnect(fe, stream);
+
+ fe->dpcm[stream].runtime = NULL;
+
+ return ret;
+}
+
/* create a new pcm */
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
{
@@ -641,56 +2434,114 @@
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_pcm_ops *soc_pcm_ops = &rtd->ops;
+ struct snd_pcm_substream *substream[2];
struct snd_pcm *pcm;
char new_name[64];
int ret = 0, playback = 0, capture = 0;
- soc_pcm_ops->open = soc_pcm_open;
- soc_pcm_ops->close = soc_pcm_close;
- soc_pcm_ops->hw_params = soc_pcm_hw_params;
- soc_pcm_ops->hw_free = soc_pcm_hw_free;
- soc_pcm_ops->prepare = soc_pcm_prepare;
- soc_pcm_ops->trigger = soc_pcm_trigger;
- soc_pcm_ops->pointer = soc_pcm_pointer;
+ if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
+ if (cpu_dai->driver->playback.channels_min)
+ playback = 1;
+ if (cpu_dai->driver->capture.channels_min)
+ capture = 1;
+ } else {
+ if (codec_dai->driver->playback.channels_min)
+ playback = 1;
+ if (codec_dai->driver->capture.channels_min)
+ capture = 1;
+ }
- /* check client and interface hw capabilities */
- snprintf(new_name, sizeof(new_name), "%s %s-%d",
- rtd->dai_link->stream_name, codec_dai->name, num);
+ /* create the PCM */
+ if (rtd->dai_link->no_pcm) {
+ snprintf(new_name, sizeof(new_name), "(%s)",
+ rtd->dai_link->stream_name);
- if (codec_dai->driver->playback.channels_min)
- playback = 1;
- if (codec_dai->driver->capture.channels_min)
- capture = 1;
+ ret = snd_pcm_new_soc_be(rtd->card->snd_card, new_name, num,
+ playback, capture, &pcm);
+ } else {
+ if (rtd->dai_link->dynamic)
+ snprintf(new_name, sizeof(new_name), "%s (*)",
+ rtd->dai_link->stream_name);
+ else
+ snprintf(new_name, sizeof(new_name), "%s %s-%d",
+ rtd->dai_link->stream_name, codec_dai->name, num);
- dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name);
- ret = snd_pcm_new(rtd->card->snd_card, new_name,
- num, playback, capture, &pcm);
+ ret = snd_pcm_new(rtd->card->snd_card, new_name, num, playback,
+ capture, &pcm);
+ }
if (ret < 0) {
printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
return ret;
}
-
- /* DAPM dai link stream work */
- INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+ dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name);
rtd->pcm = pcm;
pcm->private_data = rtd;
+ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+
+ substream[SNDRV_PCM_STREAM_PLAYBACK] =
+ pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ substream[SNDRV_PCM_STREAM_CAPTURE] =
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+
+ if (rtd->dai_link->no_pcm) {
+ if (playback)
+ substream[SNDRV_PCM_STREAM_PLAYBACK]->private_data = rtd;
+ if (capture)
+ substream[SNDRV_PCM_STREAM_CAPTURE]->private_data = rtd;
+ goto out;
+ }
+
+ /* setup any hostless PCMs - i.e. no host IO is performed */
+ if (rtd->dai_link->no_host_mode) {
+ if (substream[SNDRV_PCM_STREAM_PLAYBACK]) {
+ substream[SNDRV_PCM_STREAM_PLAYBACK]->hw_no_buffer = 1;
+ snd_soc_set_runtime_hwparams(
+ substream[SNDRV_PCM_STREAM_PLAYBACK],
+ &no_host_hardware);
+ }
+ if (substream[SNDRV_PCM_STREAM_CAPTURE]) {
+ substream[SNDRV_PCM_STREAM_CAPTURE]->hw_no_buffer = 1;
+ snd_soc_set_runtime_hwparams(
+ substream[SNDRV_PCM_STREAM_CAPTURE],
+ &no_host_hardware);
+ }
+ }
+
+ /* ASoC PCM operations */
+ if (rtd->dai_link->dynamic) {
+ rtd->ops.open = soc_dpcm_fe_dai_open;
+ rtd->ops.hw_params = soc_dpcm_fe_dai_hw_params;
+ rtd->ops.prepare = soc_dpcm_fe_dai_prepare;
+ rtd->ops.trigger = soc_dpcm_fe_dai_trigger;
+ rtd->ops.hw_free = soc_dpcm_fe_dai_hw_free;
+ rtd->ops.close = soc_dpcm_fe_dai_close;
+ rtd->ops.pointer = soc_pcm_pointer;
+ rtd->ops.ioctl = soc_pcm_ioctl;
+ } else {
+ rtd->ops.open = soc_pcm_open;
+ rtd->ops.hw_params = soc_pcm_hw_params;
+ rtd->ops.prepare = soc_pcm_prepare;
+ rtd->ops.trigger = soc_pcm_trigger;
+ rtd->ops.hw_free = soc_pcm_hw_free;
+ rtd->ops.close = soc_pcm_close;
+ rtd->ops.pointer = soc_pcm_pointer;
+ rtd->ops.ioctl = soc_pcm_ioctl;
+ }
+
if (platform->driver->ops) {
- soc_pcm_ops->mmap = platform->driver->ops->mmap;
- soc_pcm_ops->pointer = platform->driver->ops->pointer;
- soc_pcm_ops->ioctl = platform->driver->ops->ioctl;
- soc_pcm_ops->copy = platform->driver->ops->copy;
- soc_pcm_ops->silence = platform->driver->ops->silence;
- soc_pcm_ops->ack = platform->driver->ops->ack;
- soc_pcm_ops->page = platform->driver->ops->page;
+ rtd->ops.ack = platform->driver->ops->ack;
+ rtd->ops.copy = platform->driver->ops->copy;
+ rtd->ops.silence = platform->driver->ops->silence;
+ rtd->ops.page = platform->driver->ops->page;
+ rtd->ops.mmap = platform->driver->ops->mmap;
}
if (playback)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &rtd->ops);
if (capture)
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops);
if (platform->driver->pcm_new) {
ret = platform->driver->pcm_new(rtd);
@@ -701,7 +2552,151 @@
}
pcm->private_free = platform->driver->pcm_free;
+out:
printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
cpu_dai->name);
return ret;
}
+
+#ifdef CONFIG_DEBUG_FS
+static char *dpcm_state_string(enum snd_soc_dpcm_state state)
+{
+ switch (state) {
+ case SND_SOC_DPCM_STATE_NEW:
+ return "new";
+ case SND_SOC_DPCM_STATE_OPEN:
+ return "open";
+ case SND_SOC_DPCM_STATE_HW_PARAMS:
+ return "hw_params";
+ case SND_SOC_DPCM_STATE_PREPARE:
+ return "prepare";
+ case SND_SOC_DPCM_STATE_START:
+ return "start";
+ case SND_SOC_DPCM_STATE_STOP:
+ return "stop";
+ case SND_SOC_DPCM_STATE_SUSPEND:
+ return "suspend";
+ case SND_SOC_DPCM_STATE_PAUSED:
+ return "paused";
+ case SND_SOC_DPCM_STATE_HW_FREE:
+ return "hw_free";
+ case SND_SOC_DPCM_STATE_CLOSE:
+ return "close";
+ }
+
+ return "unknown";
+}
+
+static int soc_dpcm_state_open_file(struct inode *inode, struct file *file)
+{
+ file->private_data = inode->i_private;
+ return 0;
+}
+
+static ssize_t soc_dpcm_show_state(struct snd_soc_pcm_runtime *fe,
+ int stream, char *buf, size_t size)
+{
+ struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params;
+ struct snd_soc_dpcm_params *dpcm_params;
+ ssize_t offset = 0;
+
+ /* FE state */
+ offset += snprintf(buf + offset, size - offset,
+ "[%s - %s]\n", fe->dai_link->name,
+ stream ? "Capture" : "Playback");
+
+ offset += snprintf(buf + offset, size - offset, "State: %s\n",
+ dpcm_state_string(fe->dpcm[stream].state));
+
+ if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+ offset += snprintf(buf + offset, size - offset,
+ "Hardware Params: "
+ "Format = %s, Channels = %d, Rate = %d\n",
+ snd_pcm_format_name(params_format(params)),
+ params_channels(params),
+ params_rate(params));
+
+ /* BEs state */
+ offset += snprintf(buf + offset, size - offset, "Backends:\n");
+
+ if (list_empty(&fe->dpcm[stream].be_clients)) {
+ offset += snprintf(buf + offset, size - offset,
+ " No active DSP links\n");
+ goto out;
+ }
+
+ list_for_each_entry(dpcm_params, &fe->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm_params->be;
+
+ offset += snprintf(buf + offset, size - offset,
+ "- %s\n", be->dai_link->name);
+
+ offset += snprintf(buf + offset, size - offset,
+ " State: %s\n",
+ dpcm_state_string(fe->dpcm[stream].state));
+
+ if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
+ (be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
+ offset += snprintf(buf + offset, size - offset,
+ " Hardware Params: "
+ "Format = %s, Channels = %d, Rate = %d\n",
+ snd_pcm_format_name(params_format(params)),
+ params_channels(params),
+ params_rate(params));
+ }
+
+out:
+ return offset;
+}
+
+static ssize_t soc_dpcm_state_read_file(struct file *file, char __user *user_buf,
+ size_t count, loff_t *ppos)
+{
+ struct snd_soc_pcm_runtime *fe = file->private_data;
+ ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
+ char *buf;
+
+ buf = kmalloc(out_count, GFP_KERNEL);
+ if (!buf)
+ return -ENOMEM;
+
+ if (fe->cpu_dai->driver->playback.channels_min)
+ offset += soc_dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
+ buf + offset, out_count - offset);
+
+ if (fe->cpu_dai->driver->capture.channels_min)
+ offset += soc_dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
+ buf + offset, out_count - offset);
+
+ ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
+
+ kfree(buf);
+
+ return ret;
+}
+
+static const struct file_operations soc_dpcm_state_fops = {
+ .open = soc_dpcm_state_open_file,
+ .read = soc_dpcm_state_read_file,
+ .llseek = default_llseek,
+};
+
+int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd)
+{
+ rtd->debugfs_dpcm_root = debugfs_create_dir(rtd->dai_link->name,
+ rtd->card->debugfs_card_root);
+ if (!rtd->debugfs_dpcm_root) {
+ dev_dbg(rtd->dev,
+ "ASoC: Failed to create dpcm debugfs directory %s\n",
+ rtd->dai_link->name);
+ return -EINVAL;
+ }
+
+ rtd->debugfs_dpcm_state = debugfs_create_file("state", 0644,
+ rtd->debugfs_dpcm_root,
+ rtd, &soc_dpcm_state_fops);
+
+ return 0;
+}
+#endif
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 4a7be7b..1452312 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -48,6 +48,7 @@
#include <linux/usb/audio.h>
#include <linux/usb/audio-v2.h>
#include <linux/module.h>
+#include <linux/switch.h>
#include <sound/control.h>
#include <sound/core.h>
@@ -86,6 +87,7 @@
static bool async_unlink = 1;
static int device_setup[SNDRV_CARDS]; /* device parameter for this card */
static bool ignore_ctl_error;
+struct switch_dev *usbaudiosdev;
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the USB audio adapter.");
@@ -419,6 +421,7 @@
}
snd_usb_audio_create_proc(chip);
+ switch_set_state(usbaudiosdev, 1);
*rchip = chip;
return 0;
@@ -581,6 +584,7 @@
mutex_unlock(&chip->shutdown_mutex);
mutex_unlock(®ister_mutex);
}
+ switch_set_state(usbaudiosdev, 0);
}
/*
@@ -713,16 +717,27 @@
static int __init snd_usb_audio_init(void)
{
+ int err;
if (nrpacks < 1 || nrpacks > MAX_PACKS) {
printk(KERN_WARNING "invalid nrpacks value.\n");
return -EINVAL;
}
+
+ usbaudiosdev = kzalloc(sizeof(usbaudiosdev), GFP_KERNEL);
+ usbaudiosdev->name = "usb_audio";
+
+ err = switch_dev_register(usbaudiosdev);
+ if (err)
+ pr_err("Usb-audio switch registration failed\n");
+ else
+ pr_debug("usb hs_detected\n");
return usb_register(&usb_audio_driver);
}
static void __exit snd_usb_audio_cleanup(void)
{
usb_deregister(&usb_audio_driver);
+ kfree(usbaudiosdev);
}
module_init(snd_usb_audio_init);