| /* |
| * soc-core.c -- ALSA SoC Audio Layer |
| * |
| * Copyright 2005 Wolfson Microelectronics PLC. |
| * Copyright 2005 Openedhand Ltd. |
| * |
| * Author: Liam Girdwood <lrg@slimlogic.co.uk> |
| * with code, comments and ideas from :- |
| * Richard Purdie <richard@openedhand.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; either version 2 of the License, or (at your |
| * option) any later version. |
| * |
| * TODO: |
| * o Add hw rules to enforce rates, etc. |
| * o More testing with other codecs/machines. |
| * o Add more codecs and platforms to ensure good API coverage. |
| * o Support TDM on PCM and I2S |
| */ |
| |
| #include <linux/module.h> |
| #include <linux/moduleparam.h> |
| #include <linux/init.h> |
| #include <linux/delay.h> |
| #include <linux/pm.h> |
| #include <linux/bitops.h> |
| #include <linux/debugfs.h> |
| #include <linux/platform_device.h> |
| #include <sound/ac97_codec.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> |
| |
| static DEFINE_MUTEX(pcm_mutex); |
| static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); |
| |
| #ifdef CONFIG_DEBUG_FS |
| static struct dentry *debugfs_root; |
| #endif |
| |
| static DEFINE_MUTEX(client_mutex); |
| static LIST_HEAD(card_list); |
| static LIST_HEAD(dai_list); |
| static LIST_HEAD(platform_list); |
| static LIST_HEAD(codec_list); |
| |
| static int snd_soc_register_card(struct snd_soc_card *card); |
| static int snd_soc_unregister_card(struct snd_soc_card *card); |
| |
| /* |
| * 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; |
| module_param(pmdown_time, int, 0); |
| MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); |
| |
| /* |
| * This function forces any delayed work to be queued and run. |
| */ |
| static int run_delayed_work(struct delayed_work *dwork) |
| { |
| int ret; |
| |
| /* cancel any work waiting to be queued. */ |
| ret = cancel_delayed_work(dwork); |
| |
| /* if there was any work waiting then we run it now and |
| * wait for it's completion */ |
| if (ret) { |
| schedule_delayed_work(dwork, 0); |
| flush_scheduled_work(); |
| } |
| return ret; |
| } |
| |
| /* codec register dump */ |
| static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) |
| { |
| int i, step = 1, count = 0; |
| |
| if (!codec->reg_cache_size) |
| return 0; |
| |
| if (codec->reg_cache_step) |
| step = codec->reg_cache_step; |
| |
| count += sprintf(buf, "%s registers\n", codec->name); |
| for (i = 0; i < codec->reg_cache_size; i += step) { |
| if (codec->readable_register && !codec->readable_register(i)) |
| continue; |
| |
| count += sprintf(buf + count, "%2x: ", i); |
| if (count >= PAGE_SIZE - 1) |
| break; |
| |
| if (codec->display_register) |
| count += codec->display_register(codec, buf + count, |
| PAGE_SIZE - count, i); |
| else |
| count += snprintf(buf + count, PAGE_SIZE - count, |
| "%4x", codec->read(codec, i)); |
| |
| if (count >= PAGE_SIZE - 1) |
| break; |
| |
| count += snprintf(buf + count, PAGE_SIZE - count, "\n"); |
| if (count >= PAGE_SIZE - 1) |
| break; |
| } |
| |
| /* Truncate count; min() would cause a warning */ |
| if (count >= PAGE_SIZE) |
| count = PAGE_SIZE - 1; |
| |
| return count; |
| } |
| static ssize_t codec_reg_show(struct device *dev, |
| struct device_attribute *attr, char *buf) |
| { |
| struct snd_soc_device *devdata = dev_get_drvdata(dev); |
| return soc_codec_reg_show(devdata->card->codec, buf); |
| } |
| |
| static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); |
| |
| static ssize_t pmdown_time_show(struct device *dev, |
| struct device_attribute *attr, char *buf) |
| { |
| struct snd_soc_device *socdev = dev_get_drvdata(dev); |
| struct snd_soc_card *card = socdev->card; |
| |
| return sprintf(buf, "%ld\n", card->pmdown_time); |
| } |
| |
| static ssize_t pmdown_time_set(struct device *dev, |
| struct device_attribute *attr, |
| const char *buf, size_t count) |
| { |
| struct snd_soc_device *socdev = dev_get_drvdata(dev); |
| struct snd_soc_card *card = socdev->card; |
| |
| strict_strtol(buf, 10, &card->pmdown_time); |
| |
| return count; |
| } |
| |
| 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) |
| { |
| ssize_t ret; |
| struct snd_soc_codec *codec = file->private_data; |
| char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); |
| if (!buf) |
| return -ENOMEM; |
| ret = soc_codec_reg_show(codec, buf); |
| if (ret >= 0) |
| ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); |
| kfree(buf); |
| return ret; |
| } |
| |
| static ssize_t codec_reg_write_file(struct file *file, |
| const char __user *user_buf, size_t count, loff_t *ppos) |
| { |
| char buf[32]; |
| int buf_size; |
| char *start = buf; |
| unsigned long reg, value; |
| int step = 1; |
| struct snd_soc_codec *codec = file->private_data; |
| |
| buf_size = min(count, (sizeof(buf)-1)); |
| if (copy_from_user(buf, user_buf, buf_size)) |
| return -EFAULT; |
| buf[buf_size] = 0; |
| |
| if (codec->reg_cache_step) |
| step = codec->reg_cache_step; |
| |
| while (*start == ' ') |
| start++; |
| reg = simple_strtoul(start, &start, 16); |
| if ((reg >= codec->reg_cache_size) || (reg % step)) |
| return -EINVAL; |
| while (*start == ' ') |
| start++; |
| if (strict_strtoul(start, 16, &value)) |
| return -EINVAL; |
| codec->write(codec, reg, value); |
| return buf_size; |
| } |
| |
| static const struct file_operations codec_reg_fops = { |
| .open = codec_reg_open_file, |
| .read = codec_reg_read_file, |
| .write = codec_reg_write_file, |
| }; |
| |
| static void soc_init_codec_debugfs(struct snd_soc_codec *codec) |
| { |
| char codec_root[128]; |
| |
| if (codec->dev) |
| snprintf(codec_root, sizeof(codec_root), |
| "%s.%s", codec->name, dev_name(codec->dev)); |
| else |
| snprintf(codec_root, sizeof(codec_root), |
| "%s", codec->name); |
| |
| codec->debugfs_codec_root = debugfs_create_dir(codec_root, |
| debugfs_root); |
| if (!codec->debugfs_codec_root) { |
| printk(KERN_WARNING |
| "ASoC: Failed to create codec debugfs directory\n"); |
| return; |
| } |
| |
| codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, |
| codec->debugfs_codec_root, |
| codec, &codec_reg_fops); |
| if (!codec->debugfs_reg) |
| printk(KERN_WARNING |
| "ASoC: Failed to create codec register debugfs file\n"); |
| |
| codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, |
| codec->debugfs_codec_root, |
| &codec->pop_time); |
| if (!codec->debugfs_pop_time) |
| printk(KERN_WARNING |
| "Failed to create pop time debugfs file\n"); |
| |
| codec->debugfs_dapm = debugfs_create_dir("dapm", |
| codec->debugfs_codec_root); |
| if (!codec->debugfs_dapm) |
| printk(KERN_WARNING |
| "Failed to create DAPM debugfs directory\n"); |
| |
| snd_soc_dapm_debugfs_init(codec); |
| } |
| |
| static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) |
| { |
| debugfs_remove_recursive(codec->debugfs_codec_root); |
| } |
| |
| #else |
| |
| static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) |
| { |
| } |
| |
| static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) |
| { |
| } |
| #endif |
| |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| /* unregister ac97 codec */ |
| static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) |
| { |
| if (codec->ac97->dev.bus) |
| device_unregister(&codec->ac97->dev); |
| return 0; |
| } |
| |
| /* stop no dev release warning */ |
| static void soc_ac97_device_release(struct device *dev){} |
| |
| /* register ac97 codec to bus */ |
| static int soc_ac97_dev_register(struct snd_soc_codec *codec) |
| { |
| int err; |
| |
| codec->ac97->dev.bus = &ac97_bus_type; |
| codec->ac97->dev.parent = codec->card->dev; |
| codec->ac97->dev.release = soc_ac97_device_release; |
| |
| dev_set_name(&codec->ac97->dev, "%d-%d:%s", |
| codec->card->number, 0, codec->name); |
| err = device_register(&codec->ac97->dev); |
| if (err < 0) { |
| snd_printk(KERN_ERR "Can't register ac97 bus\n"); |
| codec->ac97->dev.bus = NULL; |
| return err; |
| } |
| return 0; |
| } |
| #endif |
| |
| static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_device *socdev = rtd->socdev; |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret; |
| |
| if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || |
| machine->symmetric_rates) { |
| dev_dbg(card->dev, "Symmetry forces %dHz rate\n", |
| machine->rate); |
| |
| ret = snd_pcm_hw_constraint_minmax(substream->runtime, |
| SNDRV_PCM_HW_PARAM_RATE, |
| machine->rate, |
| machine->rate); |
| if (ret < 0) { |
| dev_err(card->dev, |
| "Unable to apply rate symmetry constraint: %d\n", ret); |
| return ret; |
| } |
| } |
| |
| 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. |
| */ |
| static int soc_pcm_open(struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_device *socdev = rtd->socdev; |
| struct snd_soc_card *card = socdev->card; |
| struct snd_pcm_runtime *runtime = substream->runtime; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret = 0; |
| |
| mutex_lock(&pcm_mutex); |
| |
| /* startup the audio subsystem */ |
| if (cpu_dai->ops->startup) { |
| ret = cpu_dai->ops->startup(substream, cpu_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't open interface %s\n", |
| cpu_dai->name); |
| goto out; |
| } |
| } |
| |
| if (platform->pcm_ops->open) { |
| ret = platform->pcm_ops->open(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); |
| goto platform_err; |
| } |
| } |
| |
| if (codec_dai->ops->startup) { |
| ret = codec_dai->ops->startup(substream, codec_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't open codec %s\n", |
| codec_dai->name); |
| goto codec_dai_err; |
| } |
| } |
| |
| if (machine->ops && machine->ops->startup) { |
| ret = machine->ops->startup(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: %s startup failed\n", machine->name); |
| goto machine_err; |
| } |
| } |
| |
| /* Check that the codec and cpu DAI's are compatible */ |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { |
| runtime->hw.rate_min = |
| max(codec_dai->playback.rate_min, |
| cpu_dai->playback.rate_min); |
| runtime->hw.rate_max = |
| min(codec_dai->playback.rate_max, |
| cpu_dai->playback.rate_max); |
| runtime->hw.channels_min = |
| max(codec_dai->playback.channels_min, |
| cpu_dai->playback.channels_min); |
| runtime->hw.channels_max = |
| min(codec_dai->playback.channels_max, |
| cpu_dai->playback.channels_max); |
| runtime->hw.formats = |
| codec_dai->playback.formats & cpu_dai->playback.formats; |
| runtime->hw.rates = |
| codec_dai->playback.rates & cpu_dai->playback.rates; |
| } else { |
| runtime->hw.rate_min = |
| max(codec_dai->capture.rate_min, |
| cpu_dai->capture.rate_min); |
| runtime->hw.rate_max = |
| min(codec_dai->capture.rate_max, |
| cpu_dai->capture.rate_max); |
| runtime->hw.channels_min = |
| max(codec_dai->capture.channels_min, |
| cpu_dai->capture.channels_min); |
| runtime->hw.channels_max = |
| min(codec_dai->capture.channels_max, |
| cpu_dai->capture.channels_max); |
| runtime->hw.formats = |
| codec_dai->capture.formats & cpu_dai->capture.formats; |
| runtime->hw.rates = |
| codec_dai->capture.rates & cpu_dai->capture.rates; |
| } |
| |
| snd_pcm_limit_hw_rates(runtime); |
| if (!runtime->hw.rates) { |
| printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", |
| codec_dai->name, cpu_dai->name); |
| goto machine_err; |
| } |
| if (!runtime->hw.formats) { |
| printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", |
| codec_dai->name, cpu_dai->name); |
| goto machine_err; |
| } |
| if (!runtime->hw.channels_min || !runtime->hw.channels_max) { |
| printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", |
| codec_dai->name, cpu_dai->name); |
| goto machine_err; |
| } |
| |
| /* Symmetry only applies if we've already got an active stream. */ |
| if (cpu_dai->active || codec_dai->active) { |
| ret = soc_pcm_apply_symmetry(substream); |
| if (ret != 0) |
| goto machine_err; |
| } |
| |
| pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); |
| pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); |
| pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, |
| runtime->hw.channels_max); |
| pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, |
| runtime->hw.rate_max); |
| |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| cpu_dai->playback.active = codec_dai->playback.active = 1; |
| else |
| cpu_dai->capture.active = codec_dai->capture.active = 1; |
| cpu_dai->active = codec_dai->active = 1; |
| cpu_dai->runtime = runtime; |
| card->codec->active++; |
| mutex_unlock(&pcm_mutex); |
| return 0; |
| |
| machine_err: |
| if (machine->ops && machine->ops->shutdown) |
| machine->ops->shutdown(substream); |
| |
| codec_dai_err: |
| if (platform->pcm_ops->close) |
| platform->pcm_ops->close(substream); |
| |
| platform_err: |
| if (cpu_dai->ops->shutdown) |
| cpu_dai->ops->shutdown(substream, cpu_dai); |
| out: |
| mutex_unlock(&pcm_mutex); |
| return ret; |
| } |
| |
| /* |
| * Power down the audio subsystem pmdown_time msecs after close is called. |
| * This is to ensure there are no pops or clicks in between any music tracks |
| * due to DAPM power cycling. |
| */ |
| static void close_delayed_work(struct work_struct *work) |
| { |
| struct snd_soc_card *card = container_of(work, struct snd_soc_card, |
| delayed_work.work); |
| struct snd_soc_codec *codec = card->codec; |
| struct snd_soc_dai *codec_dai; |
| int i; |
| |
| mutex_lock(&pcm_mutex); |
| for (i = 0; i < codec->num_dai; i++) { |
| codec_dai = &codec->dai[i]; |
| |
| pr_debug("pop wq checking: %s status: %s waiting: %s\n", |
| codec_dai->playback.stream_name, |
| codec_dai->playback.active ? "active" : "inactive", |
| codec_dai->pop_wait ? "yes" : "no"); |
| |
| /* are we waiting on this codec DAI stream */ |
| if (codec_dai->pop_wait == 1) { |
| codec_dai->pop_wait = 0; |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->playback.stream_name, |
| SND_SOC_DAPM_STREAM_STOP); |
| } |
| } |
| mutex_unlock(&pcm_mutex); |
| } |
| |
| /* |
| * Called by ALSA when a PCM substream is closed. Private data can be |
| * freed here. The cpu DAI, codec DAI, machine and platform are also |
| * shutdown. |
| */ |
| static int soc_codec_close(struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_device *socdev = rtd->socdev; |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| struct snd_soc_codec *codec = card->codec; |
| |
| mutex_lock(&pcm_mutex); |
| |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| cpu_dai->playback.active = codec_dai->playback.active = 0; |
| else |
| cpu_dai->capture.active = codec_dai->capture.active = 0; |
| |
| if (codec_dai->playback.active == 0 && |
| codec_dai->capture.active == 0) { |
| cpu_dai->active = codec_dai->active = 0; |
| } |
| codec->active--; |
| |
| /* Muting the DAC suppresses artifacts caused during digital |
| * shutdown, for example from stopping clocks. |
| */ |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| snd_soc_dai_digital_mute(codec_dai, 1); |
| |
| if (cpu_dai->ops->shutdown) |
| cpu_dai->ops->shutdown(substream, cpu_dai); |
| |
| if (codec_dai->ops->shutdown) |
| codec_dai->ops->shutdown(substream, codec_dai); |
| |
| if (machine->ops && machine->ops->shutdown) |
| machine->ops->shutdown(substream); |
| |
| if (platform->pcm_ops->close) |
| platform->pcm_ops->close(substream); |
| cpu_dai->runtime = NULL; |
| |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { |
| /* start delayed pop wq here for playback streams */ |
| codec_dai->pop_wait = 1; |
| schedule_delayed_work(&card->delayed_work, |
| msecs_to_jiffies(card->pmdown_time)); |
| } else { |
| /* capture streams can be powered down now */ |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->capture.stream_name, |
| SND_SOC_DAPM_STREAM_STOP); |
| } |
| |
| mutex_unlock(&pcm_mutex); |
| return 0; |
| } |
| |
| /* |
| * Called by ALSA when the PCM substream is prepared, can set format, sample |
| * rate, etc. This function is non atomic and can be called multiple times, |
| * it can refer to the runtime info. |
| */ |
| static int soc_pcm_prepare(struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_device *socdev = rtd->socdev; |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| struct snd_soc_codec *codec = card->codec; |
| int ret = 0; |
| |
| mutex_lock(&pcm_mutex); |
| |
| if (machine->ops && machine->ops->prepare) { |
| ret = machine->ops->prepare(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: machine prepare error\n"); |
| goto out; |
| } |
| } |
| |
| if (platform->pcm_ops->prepare) { |
| ret = platform->pcm_ops->prepare(substream); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: platform prepare error\n"); |
| goto out; |
| } |
| } |
| |
| if (codec_dai->ops->prepare) { |
| ret = codec_dai->ops->prepare(substream, codec_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: codec DAI prepare error\n"); |
| goto out; |
| } |
| } |
| |
| if (cpu_dai->ops->prepare) { |
| ret = cpu_dai->ops->prepare(substream, cpu_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: cpu DAI prepare error\n"); |
| goto out; |
| } |
| } |
| |
| /* cancel any delayed stream shutdown that is pending */ |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && |
| codec_dai->pop_wait) { |
| codec_dai->pop_wait = 0; |
| cancel_delayed_work(&card->delayed_work); |
| } |
| |
| if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->playback.stream_name, |
| SND_SOC_DAPM_STREAM_START); |
| else |
| snd_soc_dapm_stream_event(codec, |
| codec_dai->capture.stream_name, |
| SND_SOC_DAPM_STREAM_START); |
| |
| snd_soc_dai_digital_mute(codec_dai, 0); |
| |
| out: |
| mutex_unlock(&pcm_mutex); |
| return ret; |
| } |
| |
| /* |
| * Called by ALSA when the hardware params are set by application. This |
| * function can also be called multiple times and can allocate buffers |
| * (using snd_pcm_lib_* ). It's non-atomic. |
| */ |
| static int soc_pcm_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_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret = 0; |
| |
| mutex_lock(&pcm_mutex); |
| |
| if (machine->ops && machine->ops->hw_params) { |
| ret = machine->ops->hw_params(substream, params); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: machine hw_params failed\n"); |
| goto out; |
| } |
| } |
| |
| if (codec_dai->ops->hw_params) { |
| ret = codec_dai->ops->hw_params(substream, params, codec_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't set codec %s hw params\n", |
| codec_dai->name); |
| goto codec_err; |
| } |
| } |
| |
| if (cpu_dai->ops->hw_params) { |
| ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: interface %s hw params failed\n", |
| cpu_dai->name); |
| goto interface_err; |
| } |
| } |
| |
| if (platform->pcm_ops->hw_params) { |
| ret = platform->pcm_ops->hw_params(substream, params); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: platform %s hw params failed\n", |
| platform->name); |
| goto platform_err; |
| } |
| } |
| |
| machine->rate = params_rate(params); |
| |
| out: |
| mutex_unlock(&pcm_mutex); |
| return ret; |
| |
| platform_err: |
| if (cpu_dai->ops->hw_free) |
| cpu_dai->ops->hw_free(substream, cpu_dai); |
| |
| interface_err: |
| if (codec_dai->ops->hw_free) |
| codec_dai->ops->hw_free(substream, codec_dai); |
| |
| codec_err: |
| if (machine->ops && machine->ops->hw_free) |
| machine->ops->hw_free(substream); |
| |
| mutex_unlock(&pcm_mutex); |
| return ret; |
| } |
| |
| /* |
| * Free's resources allocated by hw_params, can be called multiple times |
| */ |
| static int soc_pcm_hw_free(struct snd_pcm_substream *substream) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_device *socdev = rtd->socdev; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| struct snd_soc_codec *codec = card->codec; |
| |
| mutex_lock(&pcm_mutex); |
| |
| /* apply codec digital mute */ |
| if (!codec->active) |
| snd_soc_dai_digital_mute(codec_dai, 1); |
| |
| /* free any machine hw params */ |
| if (machine->ops && machine->ops->hw_free) |
| machine->ops->hw_free(substream); |
| |
| /* free any DMA resources */ |
| if (platform->pcm_ops->hw_free) |
| platform->pcm_ops->hw_free(substream); |
| |
| /* now free hw params for the DAI's */ |
| if (codec_dai->ops->hw_free) |
| codec_dai->ops->hw_free(substream, codec_dai); |
| |
| if (cpu_dai->ops->hw_free) |
| cpu_dai->ops->hw_free(substream, cpu_dai); |
| |
| mutex_unlock(&pcm_mutex); |
| return 0; |
| } |
| |
| static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) |
| { |
| struct snd_soc_pcm_runtime *rtd = substream->private_data; |
| struct snd_soc_device *socdev = rtd->socdev; |
| struct snd_soc_card *card= socdev->card; |
| struct snd_soc_dai_link *machine = rtd->dai; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *cpu_dai = machine->cpu_dai; |
| struct snd_soc_dai *codec_dai = machine->codec_dai; |
| int ret; |
| |
| if (codec_dai->ops->trigger) { |
| ret = codec_dai->ops->trigger(substream, cmd, codec_dai); |
| if (ret < 0) |
| return ret; |
| } |
| |
| if (platform->pcm_ops->trigger) { |
| ret = platform->pcm_ops->trigger(substream, cmd); |
| if (ret < 0) |
| return ret; |
| } |
| |
| if (cpu_dai->ops->trigger) { |
| ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); |
| if (ret < 0) |
| return ret; |
| } |
| return 0; |
| } |
| |
| /* ASoC PCM operations */ |
| static struct snd_pcm_ops soc_pcm_ops = { |
| .open = soc_pcm_open, |
| .close = soc_codec_close, |
| .hw_params = soc_pcm_hw_params, |
| .hw_free = soc_pcm_hw_free, |
| .prepare = soc_pcm_prepare, |
| .trigger = soc_pcm_trigger, |
| }; |
| |
| #ifdef CONFIG_PM |
| /* powers down audio subsystem for suspend */ |
| static int soc_suspend(struct device *dev) |
| { |
| struct platform_device *pdev = to_platform_device(dev); |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| struct snd_soc_codec *codec = card->codec; |
| int i; |
| |
| /* If the initialization of this soc device failed, there is no codec |
| * associated with it. Just bail out in this case. |
| */ |
| if (!codec) |
| return 0; |
| |
| /* Due to the resume being scheduled into a workqueue we could |
| * suspend before that's finished - wait for it to complete. |
| */ |
| snd_power_lock(codec->card); |
| snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); |
| snd_power_unlock(codec->card); |
| |
| /* we're going to block userspace touching us until resume completes */ |
| snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); |
| |
| /* mute any active DAC's */ |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *dai = card->dai_link[i].codec_dai; |
| if (dai->ops->digital_mute && dai->playback.active) |
| dai->ops->digital_mute(dai, 1); |
| } |
| |
| /* suspend all pcms */ |
| for (i = 0; i < card->num_links; i++) |
| snd_pcm_suspend_all(card->dai_link[i].pcm); |
| |
| if (card->suspend_pre) |
| card->suspend_pre(pdev, PMSG_SUSPEND); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->suspend && !cpu_dai->ac97_control) |
| cpu_dai->suspend(cpu_dai); |
| if (platform->suspend) |
| platform->suspend(&card->dai_link[i]); |
| } |
| |
| /* close any waiting streams and save state */ |
| run_delayed_work(&card->delayed_work); |
| codec->suspend_bias_level = codec->bias_level; |
| |
| for (i = 0; i < codec->num_dai; i++) { |
| char *stream = codec->dai[i].playback.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_SUSPEND); |
| stream = codec->dai[i].capture.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_SUSPEND); |
| } |
| |
| if (codec_dev->suspend) |
| codec_dev->suspend(pdev, PMSG_SUSPEND); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->suspend && cpu_dai->ac97_control) |
| cpu_dai->suspend(cpu_dai); |
| } |
| |
| if (card->suspend_post) |
| card->suspend_post(pdev, PMSG_SUSPEND); |
| |
| return 0; |
| } |
| |
| /* deferred resume work, so resume can complete before we finished |
| * setting our codec back up, which can be very slow on I2C |
| */ |
| static void soc_resume_deferred(struct work_struct *work) |
| { |
| struct snd_soc_card *card = container_of(work, |
| struct snd_soc_card, |
| deferred_resume_work); |
| struct snd_soc_device *socdev = card->socdev; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| struct snd_soc_codec *codec = card->codec; |
| struct platform_device *pdev = to_platform_device(socdev->dev); |
| int i; |
| |
| /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, |
| * so userspace apps are blocked from touching us |
| */ |
| |
| dev_dbg(socdev->dev, "starting resume work\n"); |
| |
| if (card->resume_pre) |
| card->resume_pre(pdev); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->resume && cpu_dai->ac97_control) |
| cpu_dai->resume(cpu_dai); |
| } |
| |
| if (codec_dev->resume) |
| codec_dev->resume(pdev); |
| |
| for (i = 0; i < codec->num_dai; i++) { |
| char *stream = codec->dai[i].playback.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_RESUME); |
| stream = codec->dai[i].capture.stream_name; |
| if (stream != NULL) |
| snd_soc_dapm_stream_event(codec, stream, |
| SND_SOC_DAPM_STREAM_RESUME); |
| } |
| |
| /* unmute any active DACs */ |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *dai = card->dai_link[i].codec_dai; |
| if (dai->ops->digital_mute && dai->playback.active) |
| dai->ops->digital_mute(dai, 0); |
| } |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->resume && !cpu_dai->ac97_control) |
| cpu_dai->resume(cpu_dai); |
| if (platform->resume) |
| platform->resume(&card->dai_link[i]); |
| } |
| |
| if (card->resume_post) |
| card->resume_post(pdev); |
| |
| dev_dbg(socdev->dev, "resume work completed\n"); |
| |
| /* userspace can access us now we are back as we were before */ |
| snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); |
| } |
| |
| /* powers up audio subsystem after a suspend */ |
| static int soc_resume(struct device *dev) |
| { |
| struct platform_device *pdev = to_platform_device(dev); |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; |
| |
| /* AC97 devices might have other drivers hanging off them so |
| * need to resume immediately. Other drivers don't have that |
| * problem and may take a substantial amount of time to resume |
| * due to I/O costs and anti-pop so handle them out of line. |
| */ |
| if (cpu_dai->ac97_control) { |
| dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); |
| soc_resume_deferred(&card->deferred_resume_work); |
| } else { |
| dev_dbg(socdev->dev, "Scheduling resume work\n"); |
| if (!schedule_work(&card->deferred_resume_work)) |
| dev_err(socdev->dev, "resume work item may be lost\n"); |
| } |
| |
| return 0; |
| } |
| #else |
| #define soc_suspend NULL |
| #define soc_resume NULL |
| #endif |
| |
| static struct snd_soc_dai_ops null_dai_ops = { |
| }; |
| |
| static void snd_soc_instantiate_card(struct snd_soc_card *card) |
| { |
| struct platform_device *pdev = container_of(card->dev, |
| struct platform_device, |
| dev); |
| struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; |
| struct snd_soc_codec *codec; |
| struct snd_soc_platform *platform; |
| struct snd_soc_dai *dai; |
| int i, found, ret, ac97; |
| |
| if (card->instantiated) |
| return; |
| |
| found = 0; |
| list_for_each_entry(platform, &platform_list, list) |
| if (card->platform == platform) { |
| found = 1; |
| break; |
| } |
| if (!found) { |
| dev_dbg(card->dev, "Platform %s not registered\n", |
| card->platform->name); |
| return; |
| } |
| |
| ac97 = 0; |
| for (i = 0; i < card->num_links; i++) { |
| found = 0; |
| list_for_each_entry(dai, &dai_list, list) |
| if (card->dai_link[i].cpu_dai == dai) { |
| found = 1; |
| break; |
| } |
| if (!found) { |
| dev_dbg(card->dev, "DAI %s not registered\n", |
| card->dai_link[i].cpu_dai->name); |
| return; |
| } |
| |
| if (card->dai_link[i].cpu_dai->ac97_control) |
| ac97 = 1; |
| } |
| |
| for (i = 0; i < card->num_links; i++) { |
| if (!card->dai_link[i].codec_dai->ops) |
| card->dai_link[i].codec_dai->ops = &null_dai_ops; |
| } |
| |
| /* If we have AC97 in the system then don't wait for the |
| * codec. This will need revisiting if we have to handle |
| * systems with mixed AC97 and non-AC97 parts. Only check for |
| * DAIs currently; we can't do this per link since some AC97 |
| * codecs have non-AC97 DAIs. |
| */ |
| if (!ac97) |
| for (i = 0; i < card->num_links; i++) { |
| found = 0; |
| list_for_each_entry(dai, &dai_list, list) |
| if (card->dai_link[i].codec_dai == dai) { |
| found = 1; |
| break; |
| } |
| if (!found) { |
| dev_dbg(card->dev, "DAI %s not registered\n", |
| card->dai_link[i].codec_dai->name); |
| return; |
| } |
| } |
| |
| /* Note that we do not current check for codec components */ |
| |
| dev_dbg(card->dev, "All components present, instantiating\n"); |
| |
| /* Found everything, bring it up */ |
| card->pmdown_time = pmdown_time; |
| |
| if (card->probe) { |
| ret = card->probe(pdev); |
| if (ret < 0) |
| return; |
| } |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->probe) { |
| ret = cpu_dai->probe(pdev, cpu_dai); |
| if (ret < 0) |
| goto cpu_dai_err; |
| } |
| } |
| |
| if (codec_dev->probe) { |
| ret = codec_dev->probe(pdev); |
| if (ret < 0) |
| goto cpu_dai_err; |
| } |
| codec = card->codec; |
| |
| if (platform->probe) { |
| ret = platform->probe(pdev); |
| if (ret < 0) |
| goto platform_err; |
| } |
| |
| /* DAPM stream work */ |
| INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work); |
| #ifdef CONFIG_PM |
| /* deferred resume work */ |
| INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); |
| #endif |
| |
| for (i = 0; i < card->num_links; i++) { |
| if (card->dai_link[i].init) { |
| ret = card->dai_link[i].init(codec); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: failed to init %s\n", |
| card->dai_link[i].stream_name); |
| continue; |
| } |
| } |
| if (card->dai_link[i].codec_dai->ac97_control) |
| ac97 = 1; |
| } |
| |
| snprintf(codec->card->shortname, sizeof(codec->card->shortname), |
| "%s", card->name); |
| snprintf(codec->card->longname, sizeof(codec->card->longname), |
| "%s (%s)", card->name, codec->name); |
| |
| /* Make sure all DAPM widgets are instantiated */ |
| snd_soc_dapm_new_widgets(codec); |
| |
| ret = snd_card_register(codec->card); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: failed to register soundcard for %s\n", |
| codec->name); |
| goto card_err; |
| } |
| |
| mutex_lock(&codec->mutex); |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| /* Only instantiate AC97 if not already done by the adaptor |
| * for the generic AC97 subsystem. |
| */ |
| if (ac97 && strcmp(codec->name, "AC97") != 0) { |
| ret = soc_ac97_dev_register(codec); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: AC97 device register failed\n"); |
| snd_card_free(codec->card); |
| mutex_unlock(&codec->mutex); |
| goto card_err; |
| } |
| } |
| #endif |
| |
| ret = snd_soc_dapm_sys_add(card->socdev->dev); |
| if (ret < 0) |
| printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); |
| |
| ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time); |
| if (ret < 0) |
| printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); |
| |
| ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); |
| if (ret < 0) |
| printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); |
| |
| soc_init_codec_debugfs(codec); |
| mutex_unlock(&codec->mutex); |
| |
| card->instantiated = 1; |
| |
| return; |
| |
| card_err: |
| if (platform->remove) |
| platform->remove(pdev); |
| |
| platform_err: |
| if (codec_dev->remove) |
| codec_dev->remove(pdev); |
| |
| cpu_dai_err: |
| for (i--; i >= 0; i--) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->remove) |
| cpu_dai->remove(pdev, cpu_dai); |
| } |
| |
| if (card->remove) |
| card->remove(pdev); |
| } |
| |
| /* |
| * Attempt to initialise any uninitalised cards. Must be called with |
| * client_mutex. |
| */ |
| static void snd_soc_instantiate_cards(void) |
| { |
| struct snd_soc_card *card; |
| list_for_each_entry(card, &card_list, list) |
| snd_soc_instantiate_card(card); |
| } |
| |
| /* probes a new socdev */ |
| static int soc_probe(struct platform_device *pdev) |
| { |
| int ret = 0; |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| |
| /* Bodge while we push things out of socdev */ |
| card->socdev = socdev; |
| |
| /* Bodge while we unpick instantiation */ |
| card->dev = &pdev->dev; |
| ret = snd_soc_register_card(card); |
| if (ret != 0) { |
| dev_err(&pdev->dev, "Failed to register card\n"); |
| return ret; |
| } |
| |
| return 0; |
| } |
| |
| /* removes a socdev */ |
| static int soc_remove(struct platform_device *pdev) |
| { |
| int i; |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_codec_device *codec_dev = socdev->codec_dev; |
| |
| if (!card->instantiated) |
| return 0; |
| |
| run_delayed_work(&card->delayed_work); |
| |
| if (platform->remove) |
| platform->remove(pdev); |
| |
| if (codec_dev->remove) |
| codec_dev->remove(pdev); |
| |
| for (i = 0; i < card->num_links; i++) { |
| struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; |
| if (cpu_dai->remove) |
| cpu_dai->remove(pdev, cpu_dai); |
| } |
| |
| if (card->remove) |
| card->remove(pdev); |
| |
| snd_soc_unregister_card(card); |
| |
| return 0; |
| } |
| |
| static int soc_poweroff(struct device *dev) |
| { |
| struct platform_device *pdev = to_platform_device(dev); |
| struct snd_soc_device *socdev = platform_get_drvdata(pdev); |
| struct snd_soc_card *card = socdev->card; |
| |
| if (!card->instantiated) |
| return 0; |
| |
| /* Flush out pmdown_time work - we actually do want to run it |
| * now, we're shutting down so no imminent restart. */ |
| run_delayed_work(&card->delayed_work); |
| |
| snd_soc_dapm_shutdown(socdev); |
| |
| return 0; |
| } |
| |
| static const struct dev_pm_ops soc_pm_ops = { |
| .suspend = soc_suspend, |
| .resume = soc_resume, |
| .poweroff = soc_poweroff, |
| }; |
| |
| /* ASoC platform driver */ |
| static struct platform_driver soc_driver = { |
| .driver = { |
| .name = "soc-audio", |
| .owner = THIS_MODULE, |
| .pm = &soc_pm_ops, |
| }, |
| .probe = soc_probe, |
| .remove = soc_remove, |
| }; |
| |
| /* create a new pcm */ |
| static int soc_new_pcm(struct snd_soc_device *socdev, |
| struct snd_soc_dai_link *dai_link, int num) |
| { |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_codec *codec = card->codec; |
| struct snd_soc_platform *platform = card->platform; |
| struct snd_soc_dai *codec_dai = dai_link->codec_dai; |
| struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; |
| struct snd_soc_pcm_runtime *rtd; |
| struct snd_pcm *pcm; |
| char new_name[64]; |
| int ret = 0, playback = 0, capture = 0; |
| |
| rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); |
| if (rtd == NULL) |
| return -ENOMEM; |
| |
| rtd->dai = dai_link; |
| rtd->socdev = socdev; |
| codec_dai->codec = card->codec; |
| |
| /* check client and interface hw capabilities */ |
| snprintf(new_name, sizeof(new_name), "%s %s-%d", |
| dai_link->stream_name, codec_dai->name, num); |
| |
| if (codec_dai->playback.channels_min) |
| playback = 1; |
| if (codec_dai->capture.channels_min) |
| capture = 1; |
| |
| ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, |
| capture, &pcm); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't create pcm for codec %s\n", |
| codec->name); |
| kfree(rtd); |
| return ret; |
| } |
| |
| dai_link->pcm = pcm; |
| pcm->private_data = rtd; |
| soc_pcm_ops.mmap = platform->pcm_ops->mmap; |
| soc_pcm_ops.pointer = platform->pcm_ops->pointer; |
| soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; |
| soc_pcm_ops.copy = platform->pcm_ops->copy; |
| soc_pcm_ops.silence = platform->pcm_ops->silence; |
| soc_pcm_ops.ack = platform->pcm_ops->ack; |
| soc_pcm_ops.page = platform->pcm_ops->page; |
| |
| if (playback) |
| snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); |
| |
| if (capture) |
| snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); |
| |
| ret = platform->pcm_new(codec->card, codec_dai, pcm); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: platform pcm constructor failed\n"); |
| kfree(rtd); |
| return ret; |
| } |
| |
| pcm->private_free = platform->pcm_free; |
| printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, |
| cpu_dai->name); |
| return ret; |
| } |
| |
| /** |
| * snd_soc_codec_volatile_register: Report if a register is volatile. |
| * |
| * @codec: CODEC to query. |
| * @reg: Register to query. |
| * |
| * Boolean function indiciating if a CODEC register is volatile. |
| */ |
| int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) |
| { |
| if (codec->volatile_register) |
| return codec->volatile_register(reg); |
| else |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); |
| |
| /** |
| * snd_soc_new_ac97_codec - initailise AC97 device |
| * @codec: audio codec |
| * @ops: AC97 bus operations |
| * @num: AC97 codec number |
| * |
| * Initialises AC97 codec resources for use by ad-hoc devices only. |
| */ |
| int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, |
| struct snd_ac97_bus_ops *ops, int num) |
| { |
| mutex_lock(&codec->mutex); |
| |
| codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); |
| if (codec->ac97 == NULL) { |
| mutex_unlock(&codec->mutex); |
| return -ENOMEM; |
| } |
| |
| codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); |
| if (codec->ac97->bus == NULL) { |
| kfree(codec->ac97); |
| codec->ac97 = NULL; |
| mutex_unlock(&codec->mutex); |
| return -ENOMEM; |
| } |
| |
| codec->ac97->bus->ops = ops; |
| codec->ac97->num = num; |
| codec->dev = &codec->ac97->dev; |
| mutex_unlock(&codec->mutex); |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); |
| |
| /** |
| * snd_soc_free_ac97_codec - free AC97 codec device |
| * @codec: audio codec |
| * |
| * Frees AC97 codec device resources. |
| */ |
| void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) |
| { |
| mutex_lock(&codec->mutex); |
| kfree(codec->ac97->bus); |
| kfree(codec->ac97); |
| codec->ac97 = NULL; |
| mutex_unlock(&codec->mutex); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); |
| |
| /** |
| * snd_soc_update_bits - update codec register bits |
| * @codec: audio codec |
| * @reg: codec register |
| * @mask: register mask |
| * @value: new value |
| * |
| * Writes new register value. |
| * |
| * Returns 1 for change else 0. |
| */ |
| int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, |
| unsigned int mask, unsigned int value) |
| { |
| int change; |
| unsigned int old, new; |
| |
| old = snd_soc_read(codec, reg); |
| new = (old & ~mask) | value; |
| change = old != new; |
| if (change) |
| snd_soc_write(codec, reg, new); |
| |
| return change; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_update_bits); |
| |
| /** |
| * snd_soc_update_bits_locked - update codec register bits |
| * @codec: audio codec |
| * @reg: codec register |
| * @mask: register mask |
| * @value: new value |
| * |
| * Writes new register value, and takes the codec mutex. |
| * |
| * Returns 1 for change else 0. |
| */ |
| int snd_soc_update_bits_locked(struct snd_soc_codec *codec, |
| unsigned short reg, unsigned int mask, |
| unsigned int value) |
| { |
| int change; |
| |
| mutex_lock(&codec->mutex); |
| change = snd_soc_update_bits(codec, reg, mask, value); |
| mutex_unlock(&codec->mutex); |
| |
| return change; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked); |
| |
| /** |
| * snd_soc_test_bits - test register for change |
| * @codec: audio codec |
| * @reg: codec register |
| * @mask: register mask |
| * @value: new value |
| * |
| * Tests a register with a new value and checks if the new value is |
| * different from the old value. |
| * |
| * Returns 1 for change else 0. |
| */ |
| int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, |
| unsigned int mask, unsigned int value) |
| { |
| int change; |
| unsigned int old, new; |
| |
| old = snd_soc_read(codec, reg); |
| new = (old & ~mask) | value; |
| change = old != new; |
| |
| return change; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_test_bits); |
| |
| /** |
| * snd_soc_new_pcms - create new sound card and pcms |
| * @socdev: the SoC audio device |
| * @idx: ALSA card index |
| * @xid: card identification |
| * |
| * Create a new sound card based upon the codec and interface pcms. |
| * |
| * Returns 0 for success, else error. |
| */ |
| int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) |
| { |
| struct snd_soc_card *card = socdev->card; |
| struct snd_soc_codec *codec = card->codec; |
| int ret, i; |
| |
| mutex_lock(&codec->mutex); |
| |
| /* register a sound card */ |
| ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't create sound card for codec %s\n", |
| codec->name); |
| mutex_unlock(&codec->mutex); |
| return ret; |
| } |
| |
| codec->socdev = socdev; |
| codec->card->dev = socdev->dev; |
| codec->card->private_data = codec; |
| strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); |
| |
| /* create the pcms */ |
| for (i = 0; i < card->num_links; i++) { |
| ret = soc_new_pcm(socdev, &card->dai_link[i], i); |
| if (ret < 0) { |
| printk(KERN_ERR "asoc: can't create pcm %s\n", |
| card->dai_link[i].stream_name); |
| mutex_unlock(&codec->mutex); |
| return ret; |
| } |
| if (card->dai_link[i].codec_dai->ac97_control) { |
| snd_ac97_dev_add_pdata(codec->ac97, |
| card->dai_link[i].cpu_dai->ac97_pdata); |
| } |
| } |
| |
| mutex_unlock(&codec->mutex); |
| return ret; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_new_pcms); |
| |
| /** |
| * snd_soc_free_pcms - free sound card and pcms |
| * @socdev: the SoC audio device |
| * |
| * Frees sound card and pcms associated with the socdev. |
| * Also unregister the codec if it is an AC97 device. |
| */ |
| void snd_soc_free_pcms(struct snd_soc_device *socdev) |
| { |
| struct snd_soc_codec *codec = socdev->card->codec; |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| struct snd_soc_dai *codec_dai; |
| int i; |
| #endif |
| |
| mutex_lock(&codec->mutex); |
| soc_cleanup_codec_debugfs(codec); |
| #ifdef CONFIG_SND_SOC_AC97_BUS |
| for (i = 0; i < codec->num_dai; i++) { |
| codec_dai = &codec->dai[i]; |
| if (codec_dai->ac97_control && codec->ac97 && |
| strcmp(codec->name, "AC97") != 0) { |
| soc_ac97_dev_unregister(codec); |
| goto free_card; |
| } |
| } |
| free_card: |
| #endif |
| |
| if (codec->card) |
| snd_card_free(codec->card); |
| device_remove_file(socdev->dev, &dev_attr_codec_reg); |
| mutex_unlock(&codec->mutex); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_free_pcms); |
| |
| /** |
| * snd_soc_set_runtime_hwparams - set the runtime hardware parameters |
| * @substream: the pcm substream |
| * @hw: the hardware parameters |
| * |
| * Sets the substream runtime hardware parameters. |
| */ |
| int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, |
| const struct snd_pcm_hardware *hw) |
| { |
| struct snd_pcm_runtime *runtime = substream->runtime; |
| runtime->hw.info = hw->info; |
| runtime->hw.formats = hw->formats; |
| runtime->hw.period_bytes_min = hw->period_bytes_min; |
| runtime->hw.period_bytes_max = hw->period_bytes_max; |
| runtime->hw.periods_min = hw->periods_min; |
| runtime->hw.periods_max = hw->periods_max; |
| runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; |
| runtime->hw.fifo_size = hw->fifo_size; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); |
| |
| /** |
| * snd_soc_cnew - create new control |
| * @_template: control template |
| * @data: control private data |
| * @long_name: control long name |
| * |
| * Create a new mixer control from a template control. |
| * |
| * Returns 0 for success, else error. |
| */ |
| struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, |
| void *data, char *long_name) |
| { |
| struct snd_kcontrol_new template; |
| |
| memcpy(&template, _template, sizeof(template)); |
| if (long_name) |
| template.name = long_name; |
| template.index = 0; |
| |
| return snd_ctl_new1(&template, data); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_cnew); |
| |
| /** |
| * snd_soc_add_controls - add an array of controls to a codec. |
| * Convienience function to add a list of controls. Many codecs were |
| * duplicating this code. |
| * |
| * @codec: codec to add controls to |
| * @controls: array of controls to add |
| * @num_controls: number of elements in the array |
| * |
| * Return 0 for success, else error. |
| */ |
| int snd_soc_add_controls(struct snd_soc_codec *codec, |
| const struct snd_kcontrol_new *controls, int num_controls) |
| { |
| struct snd_card *card = codec->card; |
| int err, i; |
| |
| for (i = 0; i < num_controls; i++) { |
| const struct snd_kcontrol_new *control = &controls[i]; |
| err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); |
| if (err < 0) { |
| dev_err(codec->dev, "%s: Failed to add %s\n", |
| codec->name, control->name); |
| return err; |
| } |
| } |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_add_controls); |
| |
| /** |
| * snd_soc_info_enum_double - enumerated double mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a double enumerated |
| * mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_enum_double(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 = e->shift_l == e->shift_r ? 1 : 2; |
| 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, |
| e->texts[uinfo->value.enumerated.item]); |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); |
| |
| /** |
| * snd_soc_get_enum_double - enumerated double mixer get callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to get the value of a double enumerated mixer. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| unsigned int val, bitmask; |
| |
| for (bitmask = 1; bitmask < e->max; bitmask <<= 1) |
| ; |
| val = snd_soc_read(codec, e->reg); |
| ucontrol->value.enumerated.item[0] |
| = (val >> e->shift_l) & (bitmask - 1); |
| if (e->shift_l != e->shift_r) |
| ucontrol->value.enumerated.item[1] = |
| (val >> e->shift_r) & (bitmask - 1); |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); |
| |
| /** |
| * snd_soc_put_enum_double - enumerated double mixer put callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to set the value of a double enumerated mixer. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| unsigned int val; |
| unsigned int mask, bitmask; |
| |
| for (bitmask = 1; bitmask < e->max; bitmask <<= 1) |
| ; |
| if (ucontrol->value.enumerated.item[0] > e->max - 1) |
| return -EINVAL; |
| val = ucontrol->value.enumerated.item[0] << e->shift_l; |
| mask = (bitmask - 1) << e->shift_l; |
| if (e->shift_l != e->shift_r) { |
| if (ucontrol->value.enumerated.item[1] > e->max - 1) |
| return -EINVAL; |
| val |= ucontrol->value.enumerated.item[1] << e->shift_r; |
| mask |= (bitmask - 1) << e->shift_r; |
| } |
| |
| return snd_soc_update_bits_locked(codec, e->reg, mask, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); |
| |
| /** |
| * snd_soc_get_value_enum_double - semi enumerated double mixer get callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to get the value of a double semi enumerated mixer. |
| * |
| * Semi enumerated mixer: the enumerated items are referred as values. Can be |
| * used for handling bitfield coded enumeration for example. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| unsigned int reg_val, val, mux; |
| |
| reg_val = snd_soc_read(codec, e->reg); |
| val = (reg_val >> e->shift_l) & e->mask; |
| for (mux = 0; mux < e->max; mux++) { |
| if (val == e->values[mux]) |
| break; |
| } |
| ucontrol->value.enumerated.item[0] = mux; |
| if (e->shift_l != e->shift_r) { |
| val = (reg_val >> e->shift_r) & e->mask; |
| for (mux = 0; mux < e->max; mux++) { |
| if (val == e->values[mux]) |
| break; |
| } |
| ucontrol->value.enumerated.item[1] = mux; |
| } |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double); |
| |
| /** |
| * snd_soc_put_value_enum_double - semi enumerated double mixer put callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to set the value of a double semi enumerated mixer. |
| * |
| * Semi enumerated mixer: the enumerated items are referred as values. Can be |
| * used for handling bitfield coded enumeration for example. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; |
| unsigned int val; |
| unsigned int mask; |
| |
| if (ucontrol->value.enumerated.item[0] > e->max - 1) |
| return -EINVAL; |
| val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; |
| mask = e->mask << e->shift_l; |
| if (e->shift_l != e->shift_r) { |
| if (ucontrol->value.enumerated.item[1] > e->max - 1) |
| return -EINVAL; |
| val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; |
| mask |= e->mask << e->shift_r; |
| } |
| |
| return snd_soc_update_bits_locked(codec, e->reg, mask, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); |
| |
| /** |
| * snd_soc_info_enum_ext - external enumerated single mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about an external enumerated |
| * single mixer. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_enum_ext(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, |
| e->texts[uinfo->value.enumerated.item]); |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); |
| |
| /** |
| * snd_soc_info_volsw_ext - external single mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a single external mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| int max = kcontrol->private_value; |
| |
| if (max == 1 && !strstr(kcontrol->id.name, " Volume")) |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; |
| else |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| |
| uinfo->count = 1; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); |
| |
| /** |
| * snd_soc_info_volsw - single mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a single mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| int max = mc->max; |
| unsigned int shift = mc->shift; |
| unsigned int rshift = mc->rshift; |
| |
| if (max == 1 && !strstr(kcontrol->id.name, " Volume")) |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; |
| else |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| |
| uinfo->count = shift == rshift ? 1 : 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw); |
| |
| /** |
| * snd_soc_get_volsw - single mixer get callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to get the value of a single mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (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 max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| |
| ucontrol->value.integer.value[0] = |
| (snd_soc_read(codec, reg) >> shift) & mask; |
| if (shift != rshift) |
| ucontrol->value.integer.value[1] = |
| (snd_soc_read(codec, reg) >> rshift) & mask; |
| if (invert) { |
| ucontrol->value.integer.value[0] = |
| max - ucontrol->value.integer.value[0]; |
| if (shift != rshift) |
| ucontrol->value.integer.value[1] = |
| max - ucontrol->value.integer.value[1]; |
| } |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_volsw); |
| |
| /** |
| * snd_soc_put_volsw - single mixer put callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to set the value of a single mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (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 max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| unsigned int val, val2, val_mask; |
| |
| val = (ucontrol->value.integer.value[0] & mask); |
| if (invert) |
| val = max - val; |
| val_mask = mask << shift; |
| val = val << shift; |
| if (shift != rshift) { |
| val2 = (ucontrol->value.integer.value[1] & mask); |
| if (invert) |
| val2 = max - val2; |
| val_mask |= mask << rshift; |
| val |= val2 << rshift; |
| } |
| return snd_soc_update_bits_locked(codec, reg, val_mask, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_volsw); |
| |
| /** |
| * snd_soc_info_volsw_2r - double mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a double mixer control that |
| * spans 2 codec registers. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| int max = mc->max; |
| |
| if (max == 1 && !strstr(kcontrol->id.name, " Volume")) |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; |
| else |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| |
| uinfo->count = 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); |
| |
| /** |
| * snd_soc_get_volsw_2r - double mixer get callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to get the value of a double mixer control that spans 2 registers. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| unsigned int reg2 = mc->rreg; |
| unsigned int shift = mc->shift; |
| int max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| |
| ucontrol->value.integer.value[0] = |
| (snd_soc_read(codec, reg) >> shift) & mask; |
| ucontrol->value.integer.value[1] = |
| (snd_soc_read(codec, reg2) >> shift) & mask; |
| if (invert) { |
| ucontrol->value.integer.value[0] = |
| max - ucontrol->value.integer.value[0]; |
| ucontrol->value.integer.value[1] = |
| max - ucontrol->value.integer.value[1]; |
| } |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); |
| |
| /** |
| * snd_soc_put_volsw_2r - double mixer set callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to set the value of a double mixer control that spans 2 registers. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| unsigned int reg2 = mc->rreg; |
| unsigned int shift = mc->shift; |
| int max = mc->max; |
| unsigned int mask = (1 << fls(max)) - 1; |
| unsigned int invert = mc->invert; |
| int err; |
| unsigned int val, val2, val_mask; |
| |
| val_mask = mask << shift; |
| val = (ucontrol->value.integer.value[0] & mask); |
| val2 = (ucontrol->value.integer.value[1] & mask); |
| |
| if (invert) { |
| val = max - val; |
| val2 = max - val2; |
| } |
| |
| val = val << shift; |
| val2 = val2 << shift; |
| |
| err = snd_soc_update_bits_locked(codec, reg, val_mask, val); |
| if (err < 0) |
| return err; |
| |
| err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); |
| return err; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); |
| |
| /** |
| * snd_soc_info_volsw_s8 - signed mixer info callback |
| * @kcontrol: mixer control |
| * @uinfo: control element information |
| * |
| * Callback to provide information about a signed mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_info *uinfo) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| int max = mc->max; |
| int min = mc->min; |
| |
| uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; |
| uinfo->count = 2; |
| uinfo->value.integer.min = 0; |
| uinfo->value.integer.max = max-min; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8); |
| |
| /** |
| * snd_soc_get_volsw_s8 - signed mixer get callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to get the value of a signed mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| 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; |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); |
| |
| /** |
| * snd_soc_put_volsw_sgn - signed mixer put callback |
| * @kcontrol: mixer control |
| * @ucontrol: control element information |
| * |
| * Callback to set the value of a signed mixer control. |
| * |
| * Returns 0 for success. |
| */ |
| int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, |
| struct snd_ctl_elem_value *ucontrol) |
| { |
| struct soc_mixer_control *mc = |
| (struct soc_mixer_control *)kcontrol->private_value; |
| struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); |
| unsigned int reg = mc->reg; |
| int min = mc->min; |
| unsigned int val; |
| |
| val = (ucontrol->value.integer.value[0]+min) & 0xff; |
| val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; |
| |
| return snd_soc_update_bits_locked(codec, reg, 0xffff, val); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); |
| |
| /** |
| * snd_soc_dai_set_sysclk - configure DAI system or master clock. |
| * @dai: DAI |
| * @clk_id: DAI specific clock ID |
| * @freq: new clock frequency in Hz |
| * @dir: new clock direction - input/output. |
| * |
| * Configures the DAI master (MCLK) or system (SYSCLK) clocking. |
| */ |
| int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, |
| unsigned int freq, int dir) |
| { |
| if (dai->ops && dai->ops->set_sysclk) |
| return dai->ops->set_sysclk(dai, clk_id, freq, dir); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); |
| |
| /** |
| * snd_soc_dai_set_clkdiv - configure DAI clock dividers. |
| * @dai: DAI |
| * @div_id: DAI specific clock divider ID |
| * @div: new clock divisor. |
| * |
| * Configures the clock dividers. This is used to derive the best DAI bit and |
| * frame clocks from the system or master clock. It's best to set the DAI bit |
| * and frame clocks as low as possible to save system power. |
| */ |
| int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, |
| int div_id, int div) |
| { |
| if (dai->ops && dai->ops->set_clkdiv) |
| return dai->ops->set_clkdiv(dai, div_id, div); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); |
| |
| /** |
| * snd_soc_dai_set_pll - configure DAI PLL. |
| * @dai: DAI |
| * @pll_id: DAI specific PLL ID |
| * @source: DAI specific source for the PLL |
| * @freq_in: PLL input clock frequency in Hz |
| * @freq_out: requested PLL output clock frequency in Hz |
| * |
| * Configures and enables PLL to generate output clock based on input clock. |
| */ |
| int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, |
| unsigned int freq_in, unsigned int freq_out) |
| { |
| if (dai->ops && dai->ops->set_pll) |
| return dai->ops->set_pll(dai, pll_id, source, |
| freq_in, freq_out); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); |
| |
| /** |
| * snd_soc_dai_set_fmt - configure DAI hardware audio format. |
| * @dai: DAI |
| * @fmt: SND_SOC_DAIFMT_ format value. |
| * |
| * Configures the DAI hardware format and clocking. |
| */ |
| int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) |
| { |
| if (dai->ops && dai->ops->set_fmt) |
| return dai->ops->set_fmt(dai, fmt); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); |
| |
| /** |
| * snd_soc_dai_set_tdm_slot - configure DAI TDM. |
| * @dai: DAI |
| * @tx_mask: bitmask representing active TX slots. |
| * @rx_mask: bitmask representing active RX slots. |
| * @slots: Number of slots in use. |
| * @slot_width: Width in bits for each slot. |
| * |
| * Configures a DAI for TDM operation. Both mask and slots are codec and DAI |
| * specific. |
| */ |
| int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, |
| unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) |
| { |
| if (dai->ops && dai->ops->set_tdm_slot) |
| return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, |
| slots, slot_width); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); |
| |
| /** |
| * snd_soc_dai_set_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_set_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->ops && dai->ops->set_channel_map) |
| return dai->ops->set_channel_map(dai, tx_num, tx_slot, |
| rx_num, rx_slot); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); |
| |
| /** |
| * snd_soc_dai_set_tristate - configure DAI system or master clock. |
| * @dai: DAI |
| * @tristate: tristate enable |
| * |
| * Tristates the DAI so that others can use it. |
| */ |
| int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) |
| { |
| if (dai->ops && dai->ops->set_tristate) |
| return dai->ops->set_tristate(dai, tristate); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); |
| |
| /** |
| * snd_soc_dai_digital_mute - configure DAI system or master clock. |
| * @dai: DAI |
| * @mute: mute enable |
| * |
| * Mutes the DAI DAC. |
| */ |
| int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) |
| { |
| if (dai->ops && dai->ops->digital_mute) |
| return dai->ops->digital_mute(dai, mute); |
| else |
| return -EINVAL; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); |
| |
| /** |
| * snd_soc_register_card - Register a card with the ASoC core |
| * |
| * @card: Card to register |
| * |
| * Note that currently this is an internal only function: it will be |
| * exposed to machine drivers after further backporting of ASoC v2 |
| * registration APIs. |
| */ |
| static int snd_soc_register_card(struct snd_soc_card *card) |
| { |
| if (!card->name || !card->dev) |
| return -EINVAL; |
| |
| INIT_LIST_HEAD(&card->list); |
| card->instantiated = 0; |
| |
| mutex_lock(&client_mutex); |
| list_add(&card->list, &card_list); |
| snd_soc_instantiate_cards(); |
| mutex_unlock(&client_mutex); |
| |
| dev_dbg(card->dev, "Registered card '%s'\n", card->name); |
| |
| return 0; |
| } |
| |
| /** |
| * snd_soc_unregister_card - Unregister a card with the ASoC core |
| * |
| * @card: Card to unregister |
| * |
| * Note that currently this is an internal only function: it will be |
| * exposed to machine drivers after further backporting of ASoC v2 |
| * registration APIs. |
| */ |
| static int snd_soc_unregister_card(struct snd_soc_card *card) |
| { |
| mutex_lock(&client_mutex); |
| list_del(&card->list); |
| mutex_unlock(&client_mutex); |
| |
| dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); |
| |
| return 0; |
| } |
| |
| /** |
| * snd_soc_register_dai - Register a DAI with the ASoC core |
| * |
| * @dai: DAI to register |
| */ |
| int snd_soc_register_dai(struct snd_soc_dai *dai) |
| { |
| if (!dai->name) |
| return -EINVAL; |
| |
| /* The device should become mandatory over time */ |
| if (!dai->dev) |
| printk(KERN_WARNING "No device for DAI %s\n", dai->name); |
| |
| if (!dai->ops) |
| dai->ops = &null_dai_ops; |
| |
| INIT_LIST_HEAD(&dai->list); |
| |
| mutex_lock(&client_mutex); |
| list_add(&dai->list, &dai_list); |
| snd_soc_instantiate_cards(); |
| mutex_unlock(&client_mutex); |
| |
| pr_debug("Registered DAI '%s'\n", dai->name); |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_register_dai); |
| |
| /** |
| * snd_soc_unregister_dai - Unregister a DAI from the ASoC core |
| * |
| * @dai: DAI to unregister |
| */ |
| void snd_soc_unregister_dai(struct snd_soc_dai *dai) |
| { |
| mutex_lock(&client_mutex); |
| list_del(&dai->list); |
| mutex_unlock(&client_mutex); |
| |
| pr_debug("Unregistered DAI '%s'\n", dai->name); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); |
| |
| /** |
| * snd_soc_register_dais - Register multiple DAIs with the ASoC core |
| * |
| * @dai: Array of DAIs to register |
| * @count: Number of DAIs |
| */ |
| int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) |
| { |
| int i, ret; |
| |
| for (i = 0; i < count; i++) { |
| ret = snd_soc_register_dai(&dai[i]); |
| if (ret != 0) |
| goto err; |
| } |
| |
| return 0; |
| |
| err: |
| for (i--; i >= 0; i--) |
| snd_soc_unregister_dai(&dai[i]); |
| |
| return ret; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_register_dais); |
| |
| /** |
| * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core |
| * |
| * @dai: Array of DAIs to unregister |
| * @count: Number of DAIs |
| */ |
| void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count) |
| { |
| int i; |
| |
| for (i = 0; i < count; i++) |
| snd_soc_unregister_dai(&dai[i]); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); |
| |
| /** |
| * snd_soc_register_platform - Register a platform with the ASoC core |
| * |
| * @platform: platform to register |
| */ |
| int snd_soc_register_platform(struct snd_soc_platform *platform) |
| { |
| if (!platform->name) |
| return -EINVAL; |
| |
| INIT_LIST_HEAD(&platform->list); |
| |
| mutex_lock(&client_mutex); |
| list_add(&platform->list, &platform_list); |
| snd_soc_instantiate_cards(); |
| mutex_unlock(&client_mutex); |
| |
| pr_debug("Registered platform '%s'\n", platform->name); |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_register_platform); |
| |
| /** |
| * snd_soc_unregister_platform - Unregister a platform from the ASoC core |
| * |
| * @platform: platform to unregister |
| */ |
| void snd_soc_unregister_platform(struct snd_soc_platform *platform) |
| { |
| mutex_lock(&client_mutex); |
| list_del(&platform->list); |
| mutex_unlock(&client_mutex); |
| |
| pr_debug("Unregistered platform '%s'\n", platform->name); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); |
| |
| static u64 codec_format_map[] = { |
| SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE, |
| SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE, |
| SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE, |
| SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, |
| SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE, |
| SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE, |
| SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE, |
| SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE, |
| SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE, |
| SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE, |
| SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE, |
| SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE, |
| SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE, |
| SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE, |
| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE |
| | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE, |
| }; |
| |
| /* Fix up the DAI formats for endianness: codecs don't actually see |
| * the endianness of the data but we're using the CPU format |
| * definitions which do need to include endianness so we ensure that |
| * codec DAIs always have both big and little endian variants set. |
| */ |
| static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) |
| { |
| int i; |
| |
| for (i = 0; i < ARRAY_SIZE(codec_format_map); i++) |
| if (stream->formats & codec_format_map[i]) |
| stream->formats |= codec_format_map[i]; |
| } |
| |
| /** |
| * snd_soc_register_codec - Register a codec with the ASoC core |
| * |
| * @codec: codec to register |
| */ |
| int snd_soc_register_codec(struct snd_soc_codec *codec) |
| { |
| int i; |
| |
| if (!codec->name) |
| return -EINVAL; |
| |
| /* The device should become mandatory over time */ |
| if (!codec->dev) |
| printk(KERN_WARNING "No device for codec %s\n", codec->name); |
| |
| INIT_LIST_HEAD(&codec->list); |
| |
| for (i = 0; i < codec->num_dai; i++) { |
| fixup_codec_formats(&codec->dai[i].playback); |
| fixup_codec_formats(&codec->dai[i].capture); |
| } |
| |
| mutex_lock(&client_mutex); |
| list_add(&codec->list, &codec_list); |
| snd_soc_instantiate_cards(); |
| mutex_unlock(&client_mutex); |
| |
| pr_debug("Registered codec '%s'\n", codec->name); |
| |
| return 0; |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_register_codec); |
| |
| /** |
| * snd_soc_unregister_codec - Unregister a codec from the ASoC core |
| * |
| * @codec: codec to unregister |
| */ |
| void snd_soc_unregister_codec(struct snd_soc_codec *codec) |
| { |
| mutex_lock(&client_mutex); |
| list_del(&codec->list); |
| mutex_unlock(&client_mutex); |
| |
| pr_debug("Unregistered codec '%s'\n", codec->name); |
| } |
| EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); |
| |
| static int __init snd_soc_init(void) |
| { |
| #ifdef CONFIG_DEBUG_FS |
| debugfs_root = debugfs_create_dir("asoc", NULL); |
| if (IS_ERR(debugfs_root) || !debugfs_root) { |
| printk(KERN_WARNING |
| "ASoC: Failed to create debugfs directory\n"); |
| debugfs_root = NULL; |
| } |
| #endif |
| |
| return platform_driver_register(&soc_driver); |
| } |
| |
| static void __exit snd_soc_exit(void) |
| { |
| #ifdef CONFIG_DEBUG_FS |
| debugfs_remove_recursive(debugfs_root); |
| #endif |
| platform_driver_unregister(&soc_driver); |
| } |
| |
| module_init(snd_soc_init); |
| module_exit(snd_soc_exit); |
| |
| /* Module information */ |
| MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); |
| MODULE_DESCRIPTION("ALSA SoC Core"); |
| MODULE_LICENSE("GPL"); |
| MODULE_ALIAS("platform:soc-audio"); |