Merge tag 'asoc-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: A few bug fixes for the merge window
Small driver-specific updates.
diff --git a/Documentation/devicetree/bindings/sound/omap-dmic.txt b/Documentation/devicetree/bindings/sound/omap-dmic.txt
new file mode 100644
index 0000000..fd8105f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-dmic.txt
@@ -0,0 +1,21 @@
+* Texas Instruments OMAP4+ Digital Microphone Module
+
+Required properties:
+- compatible: "ti,omap4-dmic"
+- reg: Register location and size as an array:
+ <MPU access base address, size>,
+ <L3 interconnect address, size>;
+- interrupts: Interrupt number for DMIC
+- interrupt-parent: The parent interrupt controller
+- ti,hwmods: Name of the hwmod associated with OMAP dmic IP
+
+Example:
+
+dmic: dmic@4012e000 {
+ compatible = "ti,omap4-dmic";
+ reg = <0x4012e000 0x7f>, /* MPU private access */
+ <0x4902e000 0x7f>; /* L3 Interconnect */
+ interrupts = <0 114 0x4>;
+ interrupt-parent = <&gic>;
+ ti,hwmods = "dmic";
+};
diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
new file mode 100644
index 0000000..0741dff
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt
@@ -0,0 +1,21 @@
+* Texas Instruments OMAP4+ McPDM
+
+Required properties:
+- compatible: "ti,omap4-mcpdm"
+- reg: Register location and size as an array:
+ <MPU access base address, size>,
+ <L3 interconnect address, size>;
+- interrupts: Interrupt number for McPDM
+- interrupt-parent: The parent interrupt controller
+- ti,hwmods: Name of the hwmod associated to the McPDM
+
+Example:
+
+mcpdm: mcpdm@40132000 {
+ compatible = "ti,omap4-mcpdm";
+ reg = <0x40132000 0x7f>, /* MPU private access */
+ <0x49032000 0x7f>; /* L3 Interconnect */
+ interrupts = <0 112 0x4>;
+ interrupt-parent = <&gic>;
+ ti,hwmods = "mcpdm";
+};
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 8c16d50..221b810 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -1545,7 +1545,7 @@
Module for sound cards based on the C-Media CMI8786/8787/8788 chip:
* Asound A-8788
- * Asus Xonar DG
+ * Asus Xonar DG/DGX
* AuzenTech X-Meridian
* AuzenTech X-Meridian 2G
* Bgears b-Enspirer
diff --git a/include/sound/asound.h b/include/sound/asound.h
index a2e4ff5..0876a1e 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -70,6 +70,20 @@
/****************************************************************************
* *
+ * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort *
+ * *
+ ****************************************************************************/
+
+struct snd_cea_861_aud_if {
+ unsigned char db1_ct_cc; /* coding type and channel count */
+ unsigned char db2_sf_ss; /* sample frequency and size */
+ unsigned char db3; /* not used, all zeros */
+ unsigned char db4_ca; /* channel allocation code */
+ unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */
+};
+
+/****************************************************************************
+ * *
* Section for driver hardware dependent interface - /dev/snd/hw? *
* *
****************************************************************************/
diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h
index 20ebf32..bb05c02 100644
--- a/include/sound/asoundef.h
+++ b/include/sound/asoundef.h
@@ -170,6 +170,47 @@
#define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0) /* condition not be used */
#define IEC958_AES5_CON_CGMSA_COPYNEVER (3<<0) /* no copying is permitted */
+/****************************************************************************
+ * *
+ * CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort *
+ * *
+ ****************************************************************************/
+#define CEA861_AUDIO_INFOFRAME_DB1CC (7<<0) /* mask - channel count */
+#define CEA861_AUDIO_INFOFRAME_DB1CT (0xf<<4) /* mask - coding type */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958 (1<<4) /* IEC-60958 L-PCM */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3 (2<<4) /* AC-3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1 (3<<4) /* MPEG1 Layers 1 & 2 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3 (4<<4) /* MPEG1 Layer 3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC (6<<4) /* AAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS (7<<4) /* DTS */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC (8<<4) /* ATRAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT (9<<4) /* One Bit Audio */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD (11<<4) /* DTS-HD */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT (12<<4) /* MAT (MLP) */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DST (13<<4) /* DST */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO (14<<4) /* WMA Pro */
+#define CEA861_AUDIO_INFOFRAME_DB2SF (7<<2) /* mask - sample frequency */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_32000 (1<<2) /* 32kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_44100 (2<<2) /* 44.1kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_48000 (3<<2) /* 48kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_88200 (4<<2) /* 88.2kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_96000 (5<<2) /* 96kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_176400 (6<<2) /* 176.4kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_192000 (7<<2) /* 192kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SS (3<<0) /* mask - sample size */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT (1<<0) /* 16 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT (2<<0) /* 20 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT (3<<0) /* 24 bits */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH (1<<7) /* mask - inhibit downmixing */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */
+#define CEA861_AUDIO_INFOFRAME_DB5_LSV (0xf<<3) /* mask - level-shift values */
+
/*****************************************************************************
* *
* MIDI v1.0 interface *
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 115313e..f5ded64 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -991,6 +991,8 @@
gpio_direction_output(pdata->reset_pin, 1);
chip->reset_pin = pdata->reset_pin;
}
+ } else {
+ chip->reset_pin = -EINVAL;
}
snd_card_set_dev(card, &pdev->dev);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 471e1e3..a06b165 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -155,7 +155,7 @@
* @jack: The jack to configure
* @parent: The device to set as parent for the jack.
*
- * Set the parent for the jack input device in the device tree. This
+ * Set the parent for the jack devices in the device tree. This
* function is only valid prior to registration of the jack. If no
* parent is configured then the parent device will be the sound card.
*/
@@ -179,6 +179,9 @@
* mapping is provided but keys are enabled in the jack type then
* BTN_n numeric buttons will be reported.
*
+ * If jacks are not reporting via the input API this call will have no
+ * effect.
+ *
* Note that this is intended to be use by simple devices with small
* numbers of keys that can be reported. It is also possible to
* access the input device directly - devices with complex input
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 4d18941..8f312fa 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -313,9 +313,22 @@
snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base;
snd_pcm_sframes_t hdelta, delta;
unsigned long jdelta;
+ unsigned long curr_jiffies;
+ struct timespec curr_tstamp;
old_hw_ptr = runtime->status->hw_ptr;
+
+ /*
+ * group pointer, time and jiffies reads to allow for more
+ * accurate correlations/corrections.
+ * The values are stored at the end of this routine after
+ * corrections for hw_ptr position
+ */
pos = substream->ops->pointer(substream);
+ curr_jiffies = jiffies;
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp);
+
if (pos == SNDRV_PCM_POS_XRUN) {
xrun(substream);
return -EPIPE;
@@ -343,7 +356,7 @@
delta = runtime->hw_ptr_interrupt + runtime->period_size;
if (delta > new_hw_ptr) {
/* check for double acknowledged interrupts */
- hdelta = jiffies - runtime->hw_ptr_jiffies;
+ hdelta = curr_jiffies - runtime->hw_ptr_jiffies;
if (hdelta > runtime->hw_ptr_buffer_jiffies/2) {
hw_base += runtime->buffer_size;
if (hw_base >= runtime->boundary)
@@ -388,7 +401,7 @@
* Without regular period interrupts, we have to check
* the elapsed time to detect xruns.
*/
- jdelta = jiffies - runtime->hw_ptr_jiffies;
+ jdelta = curr_jiffies - runtime->hw_ptr_jiffies;
if (jdelta < runtime->hw_ptr_buffer_jiffies / 2)
goto no_delta_check;
hdelta = jdelta - delta * HZ / runtime->rate;
@@ -430,7 +443,7 @@
if (hdelta < runtime->delay)
goto no_jiffies_check;
hdelta -= runtime->delay;
- jdelta = jiffies - runtime->hw_ptr_jiffies;
+ jdelta = curr_jiffies - runtime->hw_ptr_jiffies;
if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
delta = jdelta /
(((runtime->period_size * HZ) / runtime->rate)
@@ -492,9 +505,9 @@
}
runtime->hw_ptr_base = hw_base;
runtime->status->hw_ptr = new_hw_ptr;
- runtime->hw_ptr_jiffies = jiffies;
+ runtime->hw_ptr_jiffies = curr_jiffies;
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
- snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
+ runtime->status->tstamp = curr_tstamp;
return snd_pcm_update_state(substream, runtime);
}
@@ -1894,6 +1907,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t xfer = 0;
snd_pcm_uframes_t offset = 0;
+ snd_pcm_uframes_t avail;
int err = 0;
if (size == 0)
@@ -1917,13 +1931,12 @@
}
runtime->twake = runtime->control->avail_min ? : 1;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+ avail = snd_pcm_playback_avail(runtime);
while (size > 0) {
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
- snd_pcm_uframes_t avail;
snd_pcm_uframes_t cont;
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- snd_pcm_update_hw_ptr(substream);
- avail = snd_pcm_playback_avail(runtime);
if (!avail) {
if (nonblock) {
err = -EAGAIN;
@@ -1971,6 +1984,7 @@
offset += frames;
size -= frames;
xfer += frames;
+ avail -= frames;
if (runtime->status->state == SNDRV_PCM_STATE_PREPARED &&
snd_pcm_playback_hw_avail(runtime) >= (snd_pcm_sframes_t)runtime->start_threshold) {
err = snd_pcm_start(substream);
@@ -2111,6 +2125,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_uframes_t xfer = 0;
snd_pcm_uframes_t offset = 0;
+ snd_pcm_uframes_t avail;
int err = 0;
if (size == 0)
@@ -2141,13 +2156,12 @@
}
runtime->twake = runtime->control->avail_min ? : 1;
+ if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
+ snd_pcm_update_hw_ptr(substream);
+ avail = snd_pcm_capture_avail(runtime);
while (size > 0) {
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
- snd_pcm_uframes_t avail;
snd_pcm_uframes_t cont;
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- snd_pcm_update_hw_ptr(substream);
- avail = snd_pcm_capture_avail(runtime);
if (!avail) {
if (runtime->status->state ==
SNDRV_PCM_STATE_DRAINING) {
@@ -2202,6 +2216,7 @@
offset += frames;
size -= frames;
xfer += frames;
+ avail -= frames;
}
_end_unlock:
runtime->twake = 0;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 3fe99e6..53b5ada 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1360,7 +1360,14 @@
static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state)
{
- substream->runtime->trigger_master = substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ switch (runtime->status->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_DISCONNECTED:
+ case SNDRV_PCM_STATE_SUSPENDED:
+ return -EBADFD;
+ }
+ runtime->trigger_master = substream;
return 0;
}
@@ -1379,6 +1386,9 @@
case SNDRV_PCM_STATE_RUNNING:
runtime->status->state = SNDRV_PCM_STATE_DRAINING;
break;
+ case SNDRV_PCM_STATE_XRUN:
+ runtime->status->state = SNDRV_PCM_STATE_SETUP;
+ break;
default:
break;
}
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index c700920..e952833 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -35,7 +35,7 @@
#include <linux/sound.h>
#include <linux/mutex.h>
-#define SNDRV_OSS_MINORS 128
+#define SNDRV_OSS_MINORS 256
static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS];
static DEFINE_MUTEX(sound_oss_mutex);
@@ -111,7 +111,7 @@
int register1 = -1, register2 = -1;
struct device *carddev = snd_card_get_device_link(card);
- if (card && card->number >= 8)
+ if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
return 0; /* ignore silently */
if (minor < 0)
return minor;
@@ -170,7 +170,7 @@
int track2 = -1;
struct snd_minor *mptr;
- if (card && card->number >= 8)
+ if (card && card->number >= SNDRV_MINOR_OSS_DEVICES)
return 0;
if (minor < 0)
return minor;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index ad079b6..8b5c36f 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -117,6 +117,7 @@
/* timer stuff */
unsigned int irq_pos; /* fractional IRQ position */
unsigned int period_size_frac;
+ unsigned int last_drift;
unsigned long last_jiffies;
struct timer_list timer;
};
@@ -264,6 +265,7 @@
return err;
dpcm->last_jiffies = jiffies;
dpcm->pcm_rate_shift = 0;
+ dpcm->last_drift = 0;
spin_lock(&cable->lock);
cable->running |= stream;
cable->pause &= ~stream;
@@ -444,34 +446,30 @@
}
}
-#define BYTEPOS_UPDATE_POSONLY 0
-#define BYTEPOS_UPDATE_CLEAR 1
-#define BYTEPOS_UPDATE_COPY 2
-
-static void loopback_bytepos_update(struct loopback_pcm *dpcm,
- unsigned int delta,
- unsigned int cmd)
+static inline unsigned int bytepos_delta(struct loopback_pcm *dpcm,
+ unsigned int jiffies_delta)
{
- unsigned int count;
unsigned long last_pos;
+ unsigned int delta;
last_pos = byte_pos(dpcm, dpcm->irq_pos);
- dpcm->irq_pos += delta * dpcm->pcm_bps;
- count = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
- if (!count)
- return;
- if (cmd == BYTEPOS_UPDATE_CLEAR)
- clear_capture_buf(dpcm, count);
- else if (cmd == BYTEPOS_UPDATE_COPY)
- copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK],
- dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE],
- count);
- dpcm->buf_pos += count;
- dpcm->buf_pos %= dpcm->pcm_buffer_size;
+ dpcm->irq_pos += jiffies_delta * dpcm->pcm_bps;
+ delta = byte_pos(dpcm, dpcm->irq_pos) - last_pos;
+ if (delta >= dpcm->last_drift)
+ delta -= dpcm->last_drift;
+ dpcm->last_drift = 0;
if (dpcm->irq_pos >= dpcm->period_size_frac) {
dpcm->irq_pos %= dpcm->period_size_frac;
dpcm->period_update_pending = 1;
}
+ return delta;
+}
+
+static inline void bytepos_finish(struct loopback_pcm *dpcm,
+ unsigned int delta)
+{
+ dpcm->buf_pos += delta;
+ dpcm->buf_pos %= dpcm->pcm_buffer_size;
}
static unsigned int loopback_pos_update(struct loopback_cable *cable)
@@ -481,7 +479,7 @@
struct loopback_pcm *dpcm_capt =
cable->streams[SNDRV_PCM_STREAM_CAPTURE];
unsigned long delta_play = 0, delta_capt = 0;
- unsigned int running;
+ unsigned int running, count1, count2;
unsigned long flags;
spin_lock_irqsave(&cable->lock, flags);
@@ -500,12 +498,13 @@
goto unlock;
if (delta_play > delta_capt) {
- loopback_bytepos_update(dpcm_play, delta_play - delta_capt,
- BYTEPOS_UPDATE_POSONLY);
+ count1 = bytepos_delta(dpcm_play, delta_play - delta_capt);
+ bytepos_finish(dpcm_play, count1);
delta_play = delta_capt;
} else if (delta_play < delta_capt) {
- loopback_bytepos_update(dpcm_capt, delta_capt - delta_play,
- BYTEPOS_UPDATE_CLEAR);
+ count1 = bytepos_delta(dpcm_capt, delta_capt - delta_play);
+ clear_capture_buf(dpcm_capt, count1);
+ bytepos_finish(dpcm_capt, count1);
delta_capt = delta_play;
}
@@ -513,8 +512,17 @@
goto unlock;
/* note delta_capt == delta_play at this moment */
- loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY);
- loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY);
+ count1 = bytepos_delta(dpcm_play, delta_play);
+ count2 = bytepos_delta(dpcm_capt, delta_capt);
+ if (count1 < count2) {
+ dpcm_capt->last_drift = count2 - count1;
+ count1 = count2;
+ } else if (count1 > count2) {
+ dpcm_play->last_drift = count1 - count2;
+ }
+ copy_play_buf(dpcm_play, dpcm_capt, count1);
+ bytepos_finish(dpcm_play, count1);
+ bytepos_finish(dpcm_capt, count1);
unlock:
spin_unlock_irqrestore(&cable->lock, flags);
return running;
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c
index 87657dd..ea995af 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp.c
@@ -31,6 +31,8 @@
#define INTERRUPT_INTERVAL 16
#define QUEUE_LENGTH 48
+static void pcm_period_tasklet(unsigned long data);
+
/**
* amdtp_out_stream_init - initialize an AMDTP output stream structure
* @s: the AMDTP output stream to initialize
@@ -47,6 +49,7 @@
s->flags = flags;
s->context = ERR_PTR(-1);
mutex_init(&s->mutex);
+ tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s);
s->packet_index = 0;
return 0;
@@ -164,6 +167,21 @@
}
EXPORT_SYMBOL(amdtp_out_stream_set_pcm_format);
+/**
+ * amdtp_out_stream_pcm_prepare - prepare PCM device for running
+ * @s: the AMDTP output stream
+ *
+ * This function should be called from the PCM device's .prepare callback.
+ */
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
+{
+ tasklet_kill(&s->period_tasklet);
+ s->pcm_buffer_pointer = 0;
+ s->pcm_period_pointer = 0;
+ s->pointer_flush = true;
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_prepare);
+
static unsigned int calculate_data_blocks(struct amdtp_out_stream *s)
{
unsigned int phase, data_blocks;
@@ -376,11 +394,21 @@
s->pcm_period_pointer += data_blocks;
if (s->pcm_period_pointer >= pcm->runtime->period_size) {
s->pcm_period_pointer -= pcm->runtime->period_size;
- snd_pcm_period_elapsed(pcm);
+ s->pointer_flush = false;
+ tasklet_hi_schedule(&s->period_tasklet);
}
}
}
+static void pcm_period_tasklet(unsigned long data)
+{
+ struct amdtp_out_stream *s = (void *)data;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+
+ if (pcm)
+ snd_pcm_period_elapsed(pcm);
+}
+
static void out_packet_callback(struct fw_iso_context *context, u32 cycle,
size_t header_length, void *header, void *data)
{
@@ -506,6 +534,24 @@
EXPORT_SYMBOL(amdtp_out_stream_start);
/**
+ * amdtp_out_stream_pcm_pointer - get the PCM buffer position
+ * @s: the AMDTP output stream that transports the PCM data
+ *
+ * Returns the current buffer position, in frames.
+ */
+unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
+{
+ /* this optimization is allowed to be racy */
+ if (s->pointer_flush)
+ fw_iso_context_flush_completions(s->context);
+ else
+ s->pointer_flush = true;
+
+ return ACCESS_ONCE(s->pcm_buffer_pointer);
+}
+EXPORT_SYMBOL(amdtp_out_stream_pcm_pointer);
+
+/**
* amdtp_out_stream_update - update the stream after a bus reset
* @s: the AMDTP output stream
*/
@@ -532,6 +578,7 @@
return;
}
+ tasklet_kill(&s->period_tasklet);
fw_iso_context_stop(s->context);
fw_iso_context_destroy(s->context);
s->context = ERR_PTR(-1);
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h
index 537a9cb..b680c5e 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp.h
@@ -1,6 +1,7 @@
#ifndef SOUND_FIREWIRE_AMDTP_H_INCLUDED
#define SOUND_FIREWIRE_AMDTP_H_INCLUDED
+#include <linux/interrupt.h>
#include <linux/mutex.h>
#include <linux/spinlock.h>
#include "packets-buffer.h"
@@ -55,6 +56,7 @@
struct iso_packets_buffer buffer;
struct snd_pcm_substream *pcm;
+ struct tasklet_struct period_tasklet;
int packet_index;
unsigned int data_block_counter;
@@ -66,6 +68,7 @@
unsigned int pcm_buffer_pointer;
unsigned int pcm_period_pointer;
+ bool pointer_flush;
};
int amdtp_out_stream_init(struct amdtp_out_stream *s, struct fw_unit *unit,
@@ -81,6 +84,8 @@
void amdtp_out_stream_set_pcm_format(struct amdtp_out_stream *s,
snd_pcm_format_t format);
+void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s);
+unsigned long amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s);
void amdtp_out_stream_pcm_abort(struct amdtp_out_stream *s);
/**
@@ -123,18 +128,6 @@
}
/**
- * amdtp_out_stream_pcm_prepare - prepare PCM device for running
- * @s: the AMDTP output stream
- *
- * This function should be called from the PCM device's .prepare callback.
- */
-static inline void amdtp_out_stream_pcm_prepare(struct amdtp_out_stream *s)
-{
- s->pcm_buffer_pointer = 0;
- s->pcm_period_pointer = 0;
-}
-
-/**
* amdtp_out_stream_pcm_trigger - start/stop playback from a PCM device
* @s: the AMDTP output stream
* @pcm: the PCM device to be started, or %NULL to stop the current device
@@ -149,18 +142,6 @@
ACCESS_ONCE(s->pcm) = pcm;
}
-/**
- * amdtp_out_stream_pcm_pointer - get the PCM buffer position
- * @s: the AMDTP output stream that transports the PCM data
- *
- * Returns the current buffer position, in frames.
- */
-static inline unsigned long
-amdtp_out_stream_pcm_pointer(struct amdtp_out_stream *s)
-{
- return ACCESS_ONCE(s->pcm_buffer_pointer);
-}
-
static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
{
return sfc & 1;
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 5ca0939..ff3af6e 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -228,7 +228,7 @@
Say Y here to include support for sound cards based on the
C-Media CMI8788 (Oxygen HD Audio) chip:
* Asound A-8788
- * Asus Xonar DG
+ * Asus Xonar DG/DGX
* AuzenTech X-Meridian
* AuzenTech X-Meridian 2G
* Bgears b-Enspirer
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 9d91d61..e672ff4 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -1062,17 +1062,4 @@
.remove = __devexit_p(snd_ad1889_remove),
};
-static int __init
-alsa_ad1889_init(void)
-{
- return pci_register_driver(&ad1889_pci_driver);
-}
-
-static void __exit
-alsa_ad1889_fini(void)
-{
- pci_unregister_driver(&ad1889_pci_driver);
-}
-
-module_init(alsa_ad1889_init);
-module_exit(alsa_ad1889_fini);
+module_pci_driver(ad1889_pci_driver);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index bdd6164..9dfc27b 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2294,7 +2294,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ali5451_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ali_ids,
.probe = snd_ali_probe,
@@ -2305,15 +2305,4 @@
#endif
};
-static int __init alsa_card_ali_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ali_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ali_init)
-module_exit(alsa_card_ali_exit)
+module_pci_driver(ali5451_driver);
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8196e22..59d6538 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -852,7 +852,7 @@
return 0;
}
-static struct pci_driver driver = {
+static struct pci_driver als300_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_als300_ids,
.probe = snd_als300_probe,
@@ -863,15 +863,4 @@
#endif
};
-static int __init alsa_card_als300_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_als300_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_als300_init)
-module_exit(alsa_card_als300_exit)
+module_pci_driver(als300_driver);
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 3269b80..7d7f259 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -1036,7 +1036,7 @@
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver als4000_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_als4000_ids,
.probe = snd_card_als4000_probe,
@@ -1047,15 +1047,4 @@
#endif
};
-static int __init alsa_card_als4000_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_als4000_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_als4000_init)
-module_exit(alsa_card_als4000_exit)
+module_pci_driver(als4000_driver);
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 590682f..156a94f 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1700,7 +1700,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver atiixp_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
@@ -1711,16 +1711,4 @@
#endif
};
-
-static int __init alsa_card_atiixp_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_atiixp_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_atiixp_init)
-module_exit(alsa_card_atiixp_exit)
+module_pci_driver(atiixp_driver);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 524d35f..30a4fd9 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1331,7 +1331,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver atiixp_modem_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_atiixp_ids,
.probe = snd_atiixp_probe,
@@ -1342,16 +1342,4 @@
#endif
};
-
-static int __init alsa_card_atiixp_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_atiixp_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_atiixp_init)
-module_exit(alsa_card_atiixp_exit)
+module_pci_driver(atiixp_modem_driver);
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index f13ad53..ffc376f 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -375,24 +375,11 @@
}
// pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver vortex_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_vortex_ids,
.probe = snd_vortex_probe,
.remove = __devexit_p(snd_vortex_remove),
};
-// initialization of the module
-static int __init alsa_card_vortex_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_vortex_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_vortex_init)
-module_exit(alsa_card_vortex_exit)
+module_pci_driver(vortex_driver);
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 1c523193..0f80474 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -112,8 +112,6 @@
/*********************************
* FUNCTION DECLARATIONS
********************************/
-static int __init alsa_card_aw2_init(void);
-static void __exit alsa_card_aw2_exit(void);
static int snd_aw2_dev_free(struct snd_device *device);
static int __devinit snd_aw2_create(struct snd_card *card,
struct pci_dev *pci, struct aw2 **rchip);
@@ -171,13 +169,15 @@
MODULE_DEVICE_TABLE(pci, snd_aw2_ids);
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver aw2_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_aw2_ids,
.probe = snd_aw2_probe,
.remove = __devexit_p(snd_aw2_remove),
};
+module_pci_driver(aw2_driver);
+
/* operators for playback PCM alsa interface */
static struct snd_pcm_ops snd_aw2_playback_ops = {
.open = snd_aw2_pcm_playback_open,
@@ -217,23 +217,6 @@
* FUNCTION IMPLEMENTATIONS
********************************/
-/* initialization of the module */
-static int __init alsa_card_aw2_init(void)
-{
- snd_printdd(KERN_DEBUG "aw2: Load aw2 module\n");
- return pci_register_driver(&driver);
-}
-
-/* clean up the module */
-static void __exit alsa_card_aw2_exit(void)
-{
- snd_printdd(KERN_DEBUG "aw2: Unload aw2 module\n");
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_aw2_init);
-module_exit(alsa_card_aw2_exit);
-
/* component-destructor */
static int snd_aw2_dev_free(struct snd_device *device)
{
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 496f14c..f0b4d74 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2862,7 +2862,7 @@
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver azf3328_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_azf3328_ids,
.probe = snd_azf3328_probe,
@@ -2873,23 +2873,4 @@
#endif
};
-static int __init
-alsa_card_azf3328_init(void)
-{
- int err;
- snd_azf3328_dbgcallenter();
- err = pci_register_driver(&driver);
- snd_azf3328_dbgcallleave();
- return err;
-}
-
-static void __exit
-alsa_card_azf3328_exit(void)
-{
- snd_azf3328_dbgcallenter();
- pci_unregister_driver(&driver);
- snd_azf3328_dbgcallleave();
-}
-
-module_init(alsa_card_azf3328_init)
-module_exit(alsa_card_azf3328_exit)
+module_pci_driver(azf3328_driver);
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 62d6163..b6a95ee 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -836,8 +836,6 @@
{0x7063, 0x2000}, /* pcHDTV HD-2000 TV */
};
-static struct pci_driver driver;
-
/* return the id of the card, or a negative value if it's blacklisted */
static int __devinit snd_bt87x_detect_card(struct pci_dev *pci)
{
@@ -964,24 +962,11 @@
{ }
};
-static struct pci_driver driver = {
+static struct pci_driver bt87x_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_bt87x_ids,
.probe = snd_bt87x_probe,
.remove = __devexit_p(snd_bt87x_remove),
};
-static int __init alsa_card_bt87x_init(void)
-{
- if (load_all)
- driver.id_table = snd_bt87x_default_ids;
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_bt87x_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_bt87x_init)
-module_exit(alsa_card_bt87x_exit)
+module_pci_driver(bt87x_driver);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 08d6ebf..e76d68a 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1932,7 +1932,7 @@
MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
// pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver ca0106_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ca0106_ids,
.probe = snd_ca0106_probe,
@@ -1943,17 +1943,4 @@
#endif
};
-// initialization of the module
-static int __init alsa_card_ca0106_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_ca0106_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ca0106_init)
-module_exit(alsa_card_ca0106_exit)
+module_pci_driver(ca0106_driver);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 19b0626..3815bd4 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3398,7 +3398,7 @@
}
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver cmipci_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cmipci_ids,
.probe = snd_cmipci_probe,
@@ -3409,15 +3409,4 @@
#endif
};
-static int __init alsa_card_cmipci_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cmipci_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cmipci_init)
-module_exit(alsa_card_cmipci_exit)
+module_pci_driver(cmipci_driver);
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index a9f368f..33506ee 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -2084,7 +2084,7 @@
}
#endif /* CONFIG_PM */
-static struct pci_driver driver = {
+static struct pci_driver cs4281_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs4281_ids,
.probe = snd_cs4281_probe,
@@ -2095,15 +2095,4 @@
#endif
};
-static int __init alsa_card_cs4281_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs4281_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs4281_init)
-module_exit(alsa_card_cs4281_exit)
+module_pci_driver(cs4281_driver);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 819d79d..6cc7404 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -161,7 +161,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver cs46xx_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs46xx_ids,
.probe = snd_card_cs46xx_probe,
@@ -172,15 +172,4 @@
#endif
};
-static int __init alsa_card_cs46xx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs46xx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs46xx_init)
-module_exit(alsa_card_cs46xx_exit)
+module_pci_driver(cs46xx_driver);
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index c47cabf..f1e4229 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -291,23 +291,11 @@
return 0;
}
-static struct pci_driver driver = {
+static struct pci_driver cs5530_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs5530_ids,
.probe = snd_cs5530_probe,
.remove = __devexit_p(snd_cs5530_remove),
};
-static int __init alsa_card_cs5530_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs5530_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs5530_init)
-module_exit(alsa_card_cs5530_exit)
-
+module_pci_driver(cs5530_driver);
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index a2fb217..2c9697c 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -394,7 +394,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver cs5535audio_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_cs5535audio_ids,
.probe = snd_cs5535audio_probe,
@@ -405,18 +405,7 @@
#endif
};
-static int __init alsa_card_cs5535audio_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_cs5535audio_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_cs5535audio_init)
-module_exit(alsa_card_cs5535audio_exit)
+module_pci_driver(cs5535audio_driver);
MODULE_AUTHOR("Jaya Kumar");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index 15d95d2..75aa2c3 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -154,15 +154,4 @@
#endif
};
-static int __init ct_card_init(void)
-{
- return pci_register_driver(&ct_driver);
-}
-
-static void __exit ct_card_exit(void)
-{
- pci_unregister_driver(&ct_driver);
-}
-
-module_init(ct_card_init)
-module_exit(ct_card_exit)
+module_pci_driver(ct_driver);
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 595c11f..0f8eda1 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2328,7 +2328,7 @@
******************************************************************************/
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver echo_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_echo_ids,
.probe = snd_echo_probe,
@@ -2339,22 +2339,4 @@
#endif /* CONFIG_PM */
};
-
-
-/* initialization of the module */
-static int __init alsa_card_echo_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-
-
-/* clean up the module */
-static void __exit alsa_card_echo_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-
-module_init(alsa_card_echo_init)
-module_exit(alsa_card_echo_exit)
+module_pci_driver(echo_driver);
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 790c65d..7fdbbe4 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -263,7 +263,7 @@
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver emu10k1_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_emu10k1_ids,
.probe = snd_card_emu10k1_probe,
@@ -274,15 +274,4 @@
#endif
};
-static int __init alsa_card_emu10k1_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_emu10k1_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_emu10k1_init)
-module_exit(alsa_card_emu10k1_exit)
+module_pci_driver(emu10k1_driver);
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 47a651c..5c8978b 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1612,24 +1612,11 @@
MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids);
// pci_driver definition
-static struct pci_driver driver = {
+static struct pci_driver emu10k1x_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_emu10k1x_ids,
.probe = snd_emu10k1x_probe,
.remove = __devexit_p(snd_emu10k1x_remove),
};
-// initialization of the module
-static int __init alsa_card_emu10k1x_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-// clean up the module
-static void __exit alsa_card_emu10k1x_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_emu10k1x_init)
-module_exit(alsa_card_emu10k1x_exit)
+module_pci_driver(emu10k1x_driver);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 47a245e..3821c81 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2488,7 +2488,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ens137x_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_audiopci_ids,
.probe = snd_audiopci_probe,
@@ -2499,15 +2499,4 @@
#endif
};
-static int __init alsa_card_ens137x_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ens137x_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ens137x_init)
-module_exit(alsa_card_ens137x_exit)
+module_pci_driver(ens137x_driver);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 53eb76b..82c8d8c 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1882,7 +1882,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver es1938_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_es1938_ids,
.probe = snd_es1938_probe,
@@ -1893,15 +1893,4 @@
#endif
};
-static int __init alsa_card_es1938_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_es1938_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_es1938_init)
-module_exit(alsa_card_es1938_exit)
+module_pci_driver(es1938_driver);
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index a8faae1..67f47d8 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2898,7 +2898,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver es1968_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_es1968_ids,
.probe = snd_es1968_probe,
@@ -2909,15 +2909,4 @@
#endif
};
-static int __init alsa_card_es1968_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_es1968_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_es1968_init)
-module_exit(alsa_card_es1968_exit)
+module_pci_driver(es1968_driver);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index a416ea8..f696623 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1416,7 +1416,7 @@
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver fm801_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_fm801_ids,
.probe = snd_card_fm801_probe,
@@ -1427,15 +1427,4 @@
#endif
};
-static int __init alsa_card_fm801_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_fm801_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_fm801_init)
-module_exit(alsa_card_fm801_exit)
+module_pci_driver(fm801_driver);
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index ace157c..bd4149f 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,6 +1,6 @@
snd-hda-intel-objs := hda_intel.o
-snd-hda-codec-y := hda_codec.o hda_jack.o
+snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o
snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o
snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o
snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
new file mode 100644
index 0000000..6e9ef3e
--- /dev/null
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -0,0 +1,760 @@
+/*
+ * BIOS auto-parser helper functions for HD-audio
+ *
+ * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver 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.
+ */
+
+#include <linux/slab.h>
+#include <linux/export.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+#include "hda_auto_parser.h"
+
+#define SFX "hda_codec: "
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
+{
+ for (; *list; list++)
+ if (*list == nid)
+ return 1;
+ return 0;
+}
+
+
+/*
+ * Sort an associated group of pins according to their sequence numbers.
+ */
+static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences,
+ int num_pins)
+{
+ int i, j;
+ short seq;
+ hda_nid_t nid;
+
+ for (i = 0; i < num_pins; i++) {
+ for (j = i + 1; j < num_pins; j++) {
+ if (sequences[i] > sequences[j]) {
+ seq = sequences[i];
+ sequences[i] = sequences[j];
+ sequences[j] = seq;
+ nid = pins[i];
+ pins[i] = pins[j];
+ pins[j] = nid;
+ }
+ }
+ }
+}
+
+
+/* add the found input-pin to the cfg->inputs[] table */
+static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid,
+ int type)
+{
+ if (cfg->num_inputs < AUTO_CFG_MAX_INS) {
+ cfg->inputs[cfg->num_inputs].pin = nid;
+ cfg->inputs[cfg->num_inputs].type = type;
+ cfg->num_inputs++;
+ }
+}
+
+/* sort inputs in the order of AUTO_PIN_* type */
+static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
+{
+ int i, j;
+
+ for (i = 0; i < cfg->num_inputs; i++) {
+ for (j = i + 1; j < cfg->num_inputs; j++) {
+ if (cfg->inputs[i].type > cfg->inputs[j].type) {
+ struct auto_pin_cfg_item tmp;
+ tmp = cfg->inputs[i];
+ cfg->inputs[i] = cfg->inputs[j];
+ cfg->inputs[j] = tmp;
+ }
+ }
+ }
+}
+
+/* Reorder the surround channels
+ * ALSA sequence is front/surr/clfe/side
+ * HDA sequence is:
+ * 4-ch: front/surr => OK as it is
+ * 6-ch: front/clfe/surr
+ * 8-ch: front/clfe/rear/side|fc
+ */
+static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
+{
+ hda_nid_t nid;
+
+ switch (nums) {
+ case 3:
+ case 4:
+ nid = pins[1];
+ pins[1] = pins[2];
+ pins[2] = nid;
+ break;
+ }
+}
+
+/*
+ * Parse all pin widgets and store the useful pin nids to cfg
+ *
+ * The number of line-outs or any primary output is stored in line_outs,
+ * and the corresponding output pins are assigned to line_out_pins[],
+ * in the order of front, rear, CLFE, side, ...
+ *
+ * If more extra outputs (speaker and headphone) are found, the pins are
+ * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack
+ * is detected, one of speaker of HP pins is assigned as the primary
+ * output, i.e. to line_out_pins[0]. So, line_outs is always positive
+ * if any analog output exists.
+ *
+ * The analog input pins are assigned to inputs array.
+ * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
+ * respectively.
+ */
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ const hda_nid_t *ignore_nids,
+ unsigned int cond_flags)
+{
+ hda_nid_t nid, end_nid;
+ short seq, assoc_line_out;
+ short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
+ short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
+ short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
+ int i;
+
+ memset(cfg, 0, sizeof(*cfg));
+
+ memset(sequences_line_out, 0, sizeof(sequences_line_out));
+ memset(sequences_speaker, 0, sizeof(sequences_speaker));
+ memset(sequences_hp, 0, sizeof(sequences_hp));
+ assoc_line_out = 0;
+
+ codec->ignore_misc_bit = true;
+ end_nid = codec->start_nid + codec->num_nodes;
+ for (nid = codec->start_nid; nid < end_nid; nid++) {
+ unsigned int wid_caps = get_wcaps(codec, nid);
+ unsigned int wid_type = get_wcaps_type(wid_caps);
+ unsigned int def_conf;
+ short assoc, loc, conn, dev;
+
+ /* read all default configuration for pin complex */
+ if (wid_type != AC_WID_PIN)
+ continue;
+ /* ignore the given nids (e.g. pc-beep returns error) */
+ if (ignore_nids && is_in_nid_list(nid, ignore_nids))
+ continue;
+
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+ AC_DEFCFG_MISC_NO_PRESENCE))
+ codec->ignore_misc_bit = false;
+ conn = get_defcfg_connect(def_conf);
+ if (conn == AC_JACK_PORT_NONE)
+ continue;
+ loc = get_defcfg_location(def_conf);
+ dev = get_defcfg_device(def_conf);
+
+ /* workaround for buggy BIOS setups */
+ if (dev == AC_JACK_LINE_OUT) {
+ if (conn == AC_JACK_PORT_FIXED)
+ dev = AC_JACK_SPEAKER;
+ }
+
+ switch (dev) {
+ case AC_JACK_LINE_OUT:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+
+ if (!(wid_caps & AC_WCAP_STEREO))
+ if (!cfg->mono_out_pin)
+ cfg->mono_out_pin = nid;
+ if (!assoc)
+ continue;
+ if (!assoc_line_out)
+ assoc_line_out = assoc;
+ else if (assoc_line_out != assoc)
+ continue;
+ if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
+ continue;
+ cfg->line_out_pins[cfg->line_outs] = nid;
+ sequences_line_out[cfg->line_outs] = seq;
+ cfg->line_outs++;
+ break;
+ case AC_JACK_SPEAKER:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
+ continue;
+ cfg->speaker_pins[cfg->speaker_outs] = nid;
+ sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
+ cfg->speaker_outs++;
+ break;
+ case AC_JACK_HP_OUT:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
+ continue;
+ cfg->hp_pins[cfg->hp_outs] = nid;
+ sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
+ cfg->hp_outs++;
+ break;
+ case AC_JACK_MIC_IN:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC);
+ break;
+ case AC_JACK_LINE_IN:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN);
+ break;
+ case AC_JACK_CD:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD);
+ break;
+ case AC_JACK_AUX:
+ add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX);
+ break;
+ case AC_JACK_SPDIF_OUT:
+ case AC_JACK_DIG_OTHER_OUT:
+ if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
+ continue;
+ cfg->dig_out_pins[cfg->dig_outs] = nid;
+ cfg->dig_out_type[cfg->dig_outs] =
+ (loc == AC_JACK_LOC_HDMI) ?
+ HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
+ cfg->dig_outs++;
+ break;
+ case AC_JACK_SPDIF_IN:
+ case AC_JACK_DIG_OTHER_IN:
+ cfg->dig_in_pin = nid;
+ if (loc == AC_JACK_LOC_HDMI)
+ cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
+ else
+ cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+ break;
+ }
+ }
+
+ /* FIX-UP:
+ * If no line-out is defined but multiple HPs are found,
+ * some of them might be the real line-outs.
+ */
+ if (!cfg->line_outs && cfg->hp_outs > 1 &&
+ !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
+ int i = 0;
+ while (i < cfg->hp_outs) {
+ /* The real HPs should have the sequence 0x0f */
+ if ((sequences_hp[i] & 0x0f) == 0x0f) {
+ i++;
+ continue;
+ }
+ /* Move it to the line-out table */
+ cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
+ sequences_line_out[cfg->line_outs] = sequences_hp[i];
+ cfg->line_outs++;
+ cfg->hp_outs--;
+ memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
+ sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
+ memmove(sequences_hp + i, sequences_hp + i + 1,
+ sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
+ }
+ memset(cfg->hp_pins + cfg->hp_outs, 0,
+ sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs));
+ if (!cfg->hp_outs)
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+
+ }
+
+ /* sort by sequence */
+ sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
+ cfg->line_outs);
+ sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
+ cfg->speaker_outs);
+ sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
+ cfg->hp_outs);
+
+ /*
+ * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
+ * as a primary output
+ */
+ if (!cfg->line_outs &&
+ !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
+ if (cfg->speaker_outs) {
+ cfg->line_outs = cfg->speaker_outs;
+ memcpy(cfg->line_out_pins, cfg->speaker_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->speaker_outs = 0;
+ memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
+ cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
+ } else if (cfg->hp_outs) {
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins,
+ sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+ }
+ }
+
+ reorder_outputs(cfg->line_outs, cfg->line_out_pins);
+ reorder_outputs(cfg->hp_outs, cfg->hp_pins);
+ reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
+
+ sort_autocfg_input_pins(cfg);
+
+ /*
+ * debug prints of the parsed results
+ */
+ snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
+ cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
+ cfg->line_out_pins[2], cfg->line_out_pins[3],
+ cfg->line_out_pins[4],
+ cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
+ (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
+ "speaker" : "line"));
+ snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->speaker_outs, cfg->speaker_pins[0],
+ cfg->speaker_pins[1], cfg->speaker_pins[2],
+ cfg->speaker_pins[3], cfg->speaker_pins[4]);
+ snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+ cfg->hp_outs, cfg->hp_pins[0],
+ cfg->hp_pins[1], cfg->hp_pins[2],
+ cfg->hp_pins[3], cfg->hp_pins[4]);
+ snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin);
+ if (cfg->dig_outs)
+ snd_printd(" dig-out=0x%x/0x%x\n",
+ cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
+ snd_printd(" inputs:");
+ for (i = 0; i < cfg->num_inputs; i++) {
+ snd_printd(" %s=0x%x",
+ hda_get_autocfg_input_label(codec, cfg, i),
+ cfg->inputs[i].pin);
+ }
+ snd_printd("\n");
+ if (cfg->dig_in_pin)
+ snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin);
+
+ return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
+
+int snd_hda_get_input_pin_attr(unsigned int def_conf)
+{
+ unsigned int loc = get_defcfg_location(def_conf);
+ unsigned int conn = get_defcfg_connect(def_conf);
+ if (conn == AC_JACK_PORT_NONE)
+ return INPUT_PIN_ATTR_UNUSED;
+ /* Windows may claim the internal mic to be BOTH, too */
+ if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH)
+ return INPUT_PIN_ATTR_INT;
+ if ((loc & 0x30) == AC_JACK_LOC_INTERNAL)
+ return INPUT_PIN_ATTR_INT;
+ if ((loc & 0x30) == AC_JACK_LOC_SEPARATE)
+ return INPUT_PIN_ATTR_DOCK;
+ if (loc == AC_JACK_LOC_REAR)
+ return INPUT_PIN_ATTR_REAR;
+ if (loc == AC_JACK_LOC_FRONT)
+ return INPUT_PIN_ATTR_FRONT;
+ return INPUT_PIN_ATTR_NORMAL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
+
+/**
+ * hda_get_input_pin_label - Give a label for the given input pin
+ *
+ * When check_location is true, the function checks the pin location
+ * for mic and line-in pins, and set an appropriate prefix like "Front",
+ * "Rear", "Internal".
+ */
+
+static const char *hda_get_input_pin_label(struct hda_codec *codec,
+ hda_nid_t pin, bool check_location)
+{
+ unsigned int def_conf;
+ static const char * const mic_names[] = {
+ "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
+ };
+ int attr;
+
+ def_conf = snd_hda_codec_get_pincfg(codec, pin);
+
+ switch (get_defcfg_device(def_conf)) {
+ case AC_JACK_MIC_IN:
+ if (!check_location)
+ return "Mic";
+ attr = snd_hda_get_input_pin_attr(def_conf);
+ if (!attr)
+ return "None";
+ return mic_names[attr - 1];
+ case AC_JACK_LINE_IN:
+ if (!check_location)
+ return "Line";
+ attr = snd_hda_get_input_pin_attr(def_conf);
+ if (!attr)
+ return "None";
+ if (attr == INPUT_PIN_ATTR_DOCK)
+ return "Dock Line";
+ return "Line";
+ case AC_JACK_AUX:
+ return "Aux";
+ case AC_JACK_CD:
+ return "CD";
+ case AC_JACK_SPDIF_IN:
+ return "SPDIF In";
+ case AC_JACK_DIG_OTHER_IN:
+ return "Digital In";
+ default:
+ return "Misc";
+ }
+}
+
+/* Check whether the location prefix needs to be added to the label.
+ * If all mic-jacks are in the same location (e.g. rear panel), we don't
+ * have to put "Front" prefix to each label. In such a case, returns false.
+ */
+static int check_mic_location_need(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ int input)
+{
+ unsigned int defc;
+ int i, attr, attr2;
+
+ defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin);
+ attr = snd_hda_get_input_pin_attr(defc);
+ /* for internal or docking mics, we need locations */
+ if (attr <= INPUT_PIN_ATTR_NORMAL)
+ return 1;
+
+ attr = 0;
+ for (i = 0; i < cfg->num_inputs; i++) {
+ defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin);
+ attr2 = snd_hda_get_input_pin_attr(defc);
+ if (attr2 >= INPUT_PIN_ATTR_NORMAL) {
+ if (attr && attr != attr2)
+ return 1; /* different locations found */
+ attr = attr2;
+ }
+ }
+ return 0;
+}
+
+/**
+ * hda_get_autocfg_input_label - Get a label for the given input
+ *
+ * Get a label for the given input pin defined by the autocfg item.
+ * Unlike hda_get_input_pin_label(), this function checks all inputs
+ * defined in autocfg and avoids the redundant mic/line prefix as much as
+ * possible.
+ */
+const char *hda_get_autocfg_input_label(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ int input)
+{
+ int type = cfg->inputs[input].type;
+ int has_multiple_pins = 0;
+
+ if ((input > 0 && cfg->inputs[input - 1].type == type) ||
+ (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type))
+ has_multiple_pins = 1;
+ if (has_multiple_pins && type == AUTO_PIN_MIC)
+ has_multiple_pins &= check_mic_location_need(codec, cfg, input);
+ return hda_get_input_pin_label(codec, cfg->inputs[input].pin,
+ has_multiple_pins);
+}
+EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
+
+/* return the position of NID in the list, or -1 if not found */
+static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+ int i;
+ for (i = 0; i < nums; i++)
+ if (list[i] == nid)
+ return i;
+ return -1;
+}
+
+/* get a unique suffix or an index number */
+static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins,
+ int num_pins, int *indexp)
+{
+ static const char * const channel_sfx[] = {
+ " Front", " Surround", " CLFE", " Side"
+ };
+ int i;
+
+ i = find_idx_in_nid_list(nid, pins, num_pins);
+ if (i < 0)
+ return NULL;
+ if (num_pins == 1)
+ return "";
+ if (num_pins > ARRAY_SIZE(channel_sfx)) {
+ if (indexp)
+ *indexp = i;
+ return "";
+ }
+ return channel_sfx[i];
+}
+
+static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
+ const struct auto_pin_cfg *cfg,
+ const char *name, char *label, int maxlen,
+ int *indexp)
+{
+ unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ int attr = snd_hda_get_input_pin_attr(def_conf);
+ const char *pfx = "", *sfx = "";
+
+ /* handle as a speaker if it's a fixed line-out */
+ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
+ name = "Speaker";
+ /* check the location */
+ switch (attr) {
+ case INPUT_PIN_ATTR_DOCK:
+ pfx = "Dock ";
+ break;
+ case INPUT_PIN_ATTR_FRONT:
+ pfx = "Front ";
+ break;
+ }
+ if (cfg) {
+ /* try to give a unique suffix if needed */
+ sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs,
+ indexp);
+ if (!sfx)
+ sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs,
+ indexp);
+ if (!sfx) {
+ /* don't add channel suffix for Headphone controls */
+ int idx = find_idx_in_nid_list(nid, cfg->hp_pins,
+ cfg->hp_outs);
+ if (idx >= 0)
+ *indexp = idx;
+ sfx = "";
+ }
+ }
+ snprintf(label, maxlen, "%s%s%s", pfx, name, sfx);
+ return 1;
+}
+
+/**
+ * snd_hda_get_pin_label - Get a label for the given I/O pin
+ *
+ * Get a label for the given pin. This function works for both input and
+ * output pins. When @cfg is given as non-NULL, the function tries to get
+ * an optimized label using hda_get_autocfg_input_label().
+ *
+ * This function tries to give a unique label string for the pin as much as
+ * possible. For example, when the multiple line-outs are present, it adds
+ * the channel suffix like "Front", "Surround", etc (only when @cfg is given).
+ * If no unique name with a suffix is available and @indexp is non-NULL, the
+ * index number is stored in the pointer.
+ */
+int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
+ const struct auto_pin_cfg *cfg,
+ char *label, int maxlen, int *indexp)
+{
+ unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ const char *name = NULL;
+ int i;
+
+ if (indexp)
+ *indexp = 0;
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+ return 0;
+
+ switch (get_defcfg_device(def_conf)) {
+ case AC_JACK_LINE_OUT:
+ return fill_audio_out_name(codec, nid, cfg, "Line Out",
+ label, maxlen, indexp);
+ case AC_JACK_SPEAKER:
+ return fill_audio_out_name(codec, nid, cfg, "Speaker",
+ label, maxlen, indexp);
+ case AC_JACK_HP_OUT:
+ return fill_audio_out_name(codec, nid, cfg, "Headphone",
+ label, maxlen, indexp);
+ case AC_JACK_SPDIF_OUT:
+ case AC_JACK_DIG_OTHER_OUT:
+ if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
+ name = "HDMI";
+ else
+ name = "SPDIF";
+ if (cfg && indexp) {
+ i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
+ cfg->dig_outs);
+ if (i >= 0)
+ *indexp = i;
+ }
+ break;
+ default:
+ if (cfg) {
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].pin != nid)
+ continue;
+ name = hda_get_autocfg_input_label(codec, cfg, i);
+ if (name)
+ break;
+ }
+ }
+ if (!name)
+ name = hda_get_input_pin_label(codec, nid, true);
+ break;
+ }
+ if (!name)
+ return 0;
+ strlcpy(label, name, maxlen);
+ return 1;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+
+int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
+ const struct hda_verb *list)
+{
+ const struct hda_verb **v;
+ snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
+ v = snd_array_new(&spec->verbs);
+ if (!v)
+ return -ENOMEM;
+ *v = list;
+ return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_gen_add_verbs);
+
+void snd_hda_gen_apply_verbs(struct hda_codec *codec)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ int i;
+ for (i = 0; i < spec->verbs.used; i++) {
+ struct hda_verb **v = snd_array_elem(&spec->verbs, i);
+ snd_hda_sequence_write(codec, *v);
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_gen_apply_verbs);
+
+void snd_hda_apply_pincfgs(struct hda_codec *codec,
+ const struct hda_pintbl *cfg)
+{
+ for (; cfg->nid; cfg++)
+ snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+EXPORT_SYMBOL_HDA(snd_hda_apply_pincfgs);
+
+void snd_hda_apply_fixup(struct hda_codec *codec, int action)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ const char *modelname = spec->fixup_name;
+#endif
+ int depth = 0;
+
+ if (!spec->fixup_list)
+ return;
+
+ while (id >= 0) {
+ const struct hda_fixup *fix = spec->fixup_list + id;
+
+ switch (fix->type) {
+ case HDA_FIXUP_PINS:
+ if (action != HDA_FIXUP_ACT_PRE_PROBE || !fix->v.pins)
+ break;
+ snd_printdd(KERN_INFO SFX
+ "%s: Apply pincfg for %s\n",
+ codec->chip_name, modelname);
+ snd_hda_apply_pincfgs(codec, fix->v.pins);
+ break;
+ case HDA_FIXUP_VERBS:
+ if (action != HDA_FIXUP_ACT_PROBE || !fix->v.verbs)
+ break;
+ snd_printdd(KERN_INFO SFX
+ "%s: Apply fix-verbs for %s\n",
+ codec->chip_name, modelname);
+ snd_hda_gen_add_verbs(codec->spec, fix->v.verbs);
+ break;
+ case HDA_FIXUP_FUNC:
+ if (!fix->v.func)
+ break;
+ snd_printdd(KERN_INFO SFX
+ "%s: Apply fix-func for %s\n",
+ codec->chip_name, modelname);
+ fix->v.func(codec, fix, action);
+ break;
+ default:
+ snd_printk(KERN_ERR SFX
+ "%s: Invalid fixup type %d\n",
+ codec->chip_name, fix->type);
+ break;
+ }
+ if (!fix->chained)
+ break;
+ if (++depth > 10)
+ break;
+ id = fix->chain_id;
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_apply_fixup);
+
+void snd_hda_pick_fixup(struct hda_codec *codec,
+ const struct hda_model_fixup *models,
+ const struct snd_pci_quirk *quirk,
+ const struct hda_fixup *fixlist)
+{
+ struct hda_gen_spec *spec = codec->spec;
+ const struct snd_pci_quirk *q;
+ int id = -1;
+ const char *name = NULL;
+
+ /* when model=nofixup is given, don't pick up any fixups */
+ if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
+ spec->fixup_list = NULL;
+ spec->fixup_id = -1;
+ return;
+ }
+
+ if (codec->modelname && models) {
+ while (models->name) {
+ if (!strcmp(codec->modelname, models->name)) {
+ id = models->id;
+ name = models->name;
+ break;
+ }
+ models++;
+ }
+ }
+ if (id < 0) {
+ q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+ if (q) {
+ id = q->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ name = q->name;
+#endif
+ }
+ }
+ if (id < 0) {
+ for (q = quirk; q->subvendor; q++) {
+ unsigned int vendorid =
+ q->subdevice | (q->subvendor << 16);
+ if (vendorid == codec->subsystem_id) {
+ id = q->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ name = q->name;
+#endif
+ break;
+ }
+ }
+ }
+
+ spec->fixup_id = id;
+ if (id >= 0) {
+ spec->fixup_list = fixlist;
+ spec->fixup_name = name;
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_pick_fixup);
diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h
new file mode 100644
index 0000000..2a7889d
--- /dev/null
+++ b/sound/pci/hda/hda_auto_parser.h
@@ -0,0 +1,160 @@
+/*
+ * BIOS auto-parser helper functions for HD-audio
+ *
+ * Copyright (c) 2012 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver 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.
+ */
+
+#ifndef __SOUND_HDA_AUTO_PARSER_H
+#define __SOUND_HDA_AUTO_PARSER_H
+
+/*
+ * Helper for automatic pin configuration
+ */
+
+enum {
+ AUTO_PIN_MIC,
+ AUTO_PIN_LINE_IN,
+ AUTO_PIN_CD,
+ AUTO_PIN_AUX,
+ AUTO_PIN_LAST
+};
+
+enum {
+ AUTO_PIN_LINE_OUT,
+ AUTO_PIN_SPEAKER_OUT,
+ AUTO_PIN_HP_OUT
+};
+
+#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS
+#define AUTO_CFG_MAX_INS 8
+
+struct auto_pin_cfg_item {
+ hda_nid_t pin;
+ int type;
+};
+
+struct auto_pin_cfg;
+const char *hda_get_autocfg_input_label(struct hda_codec *codec,
+ const struct auto_pin_cfg *cfg,
+ int input);
+int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
+ const struct auto_pin_cfg *cfg,
+ char *label, int maxlen, int *indexp);
+
+enum {
+ INPUT_PIN_ATTR_UNUSED, /* pin not connected */
+ INPUT_PIN_ATTR_INT, /* internal mic/line-in */
+ INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */
+ INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */
+ INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */
+ INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */
+};
+
+int snd_hda_get_input_pin_attr(unsigned int def_conf);
+
+struct auto_pin_cfg {
+ int line_outs;
+ /* sorted in the order of Front/Surr/CLFE/Side */
+ hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
+ int speaker_outs;
+ hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
+ int hp_outs;
+ int line_out_type; /* AUTO_PIN_XXX_OUT */
+ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
+ int num_inputs;
+ struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS];
+ int dig_outs;
+ hda_nid_t dig_out_pins[2];
+ hda_nid_t dig_in_pin;
+ hda_nid_t mono_out_pin;
+ int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
+ int dig_in_type; /* HDA_PCM_TYPE_XXX */
+};
+
+/* bit-flags for snd_hda_parse_pin_def_config() behavior */
+#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */
+#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */
+
+int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
+ struct auto_pin_cfg *cfg,
+ const hda_nid_t *ignore_nids,
+ unsigned int cond_flags);
+
+/* older function */
+#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
+ snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
+
+/*
+ */
+
+struct hda_gen_spec {
+ /* fix-up list */
+ int fixup_id;
+ const struct hda_fixup *fixup_list;
+ const char *fixup_name;
+
+ /* additional init verbs */
+ struct snd_array verbs;
+};
+
+
+/*
+ * Fix-up pin default configurations and add default verbs
+ */
+
+struct hda_pintbl {
+ hda_nid_t nid;
+ u32 val;
+};
+
+struct hda_model_fixup {
+ const int id;
+ const char *name;
+};
+
+struct hda_fixup {
+ int type;
+ bool chained;
+ int chain_id;
+ union {
+ const struct hda_pintbl *pins;
+ const struct hda_verb *verbs;
+ void (*func)(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action);
+ } v;
+};
+
+/* fixup types */
+enum {
+ HDA_FIXUP_INVALID,
+ HDA_FIXUP_PINS,
+ HDA_FIXUP_VERBS,
+ HDA_FIXUP_FUNC,
+};
+
+/* fixup action definitions */
+enum {
+ HDA_FIXUP_ACT_PRE_PROBE,
+ HDA_FIXUP_ACT_PROBE,
+ HDA_FIXUP_ACT_INIT,
+ HDA_FIXUP_ACT_BUILD,
+};
+
+int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
+ const struct hda_verb *list);
+void snd_hda_gen_apply_verbs(struct hda_codec *codec);
+void snd_hda_apply_pincfgs(struct hda_codec *codec,
+ const struct hda_pintbl *cfg);
+void snd_hda_apply_fixup(struct hda_codec *codec, int action);
+void snd_hda_pick_fixup(struct hda_codec *codec,
+ const struct hda_model_fixup *models,
+ const struct snd_pci_quirk *quirk,
+ const struct hda_fixup *fixlist);
+
+#endif /* __SOUND_HDA_AUTO_PARSER_H */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 841475c..eb09a33 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -334,54 +334,23 @@
return NULL;
}
-/**
- * snd_hda_get_conn_list - get connection list
- * @codec: the HDA codec
- * @nid: NID to parse
- * @listp: the pointer to store NID list
- *
- * Parses the connection list of the given widget and stores the list
- * of NIDs.
- *
- * Returns the number of connections, or a negative error code.
- */
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
- const hda_nid_t **listp)
+/* read the connection and add to the cache */
+static int read_and_add_raw_conns(struct hda_codec *codec, hda_nid_t nid)
{
- struct snd_array *array = &codec->conn_lists;
- int len, err;
hda_nid_t list[HDA_MAX_CONNECTIONS];
- hda_nid_t *p;
- bool added = false;
+ int len;
- again:
- /* if the connection-list is already cached, read it */
- p = lookup_conn_list(array, nid);
- if (p) {
- if (listp)
- *listp = p + 2;
- return p[1];
- }
- if (snd_BUG_ON(added))
- return -EINVAL;
-
- /* read the connection and add to the cache */
- len = snd_hda_get_raw_connections(codec, nid, list, HDA_MAX_CONNECTIONS);
+ len = snd_hda_get_raw_connections(codec, nid, list, ARRAY_SIZE(list));
if (len < 0)
return len;
- err = snd_hda_override_conn_list(codec, nid, len, list);
- if (err < 0)
- return err;
- added = true;
- goto again;
+ return snd_hda_override_conn_list(codec, nid, len, list);
}
-EXPORT_SYMBOL_HDA(snd_hda_get_conn_list);
/**
* snd_hda_get_connections - copy connection list
* @codec: the HDA codec
* @nid: NID to parse
- * @conn_list: connection list array
+ * @conn_list: connection list array; when NULL, checks only the size
* @max_conns: max. number of connections to store
*
* Parses the connection list of the given widget and stores the list
@@ -390,21 +359,41 @@
* Returns the number of connections, or a negative error code.
*/
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
- hda_nid_t *conn_list, int max_conns)
+ hda_nid_t *conn_list, int max_conns)
{
- const hda_nid_t *list;
- int len = snd_hda_get_conn_list(codec, nid, &list);
+ struct snd_array *array = &codec->conn_lists;
+ int len;
+ hda_nid_t *p;
+ bool added = false;
- if (len <= 0)
- return len;
- if (len > max_conns) {
- snd_printk(KERN_ERR "hda_codec: "
- "Too many connections %d for NID 0x%x\n",
- len, nid);
- return -EINVAL;
+ again:
+ mutex_lock(&codec->hash_mutex);
+ len = -1;
+ /* if the connection-list is already cached, read it */
+ p = lookup_conn_list(array, nid);
+ if (p) {
+ len = p[1];
+ if (conn_list && len > max_conns) {
+ snd_printk(KERN_ERR "hda_codec: "
+ "Too many connections %d for NID 0x%x\n",
+ len, nid);
+ mutex_unlock(&codec->hash_mutex);
+ return -EINVAL;
+ }
+ if (conn_list && len)
+ memcpy(conn_list, p + 2, len * sizeof(hda_nid_t));
}
- memcpy(conn_list, list, len * sizeof(hda_nid_t));
- return len;
+ mutex_unlock(&codec->hash_mutex);
+ if (len >= 0)
+ return len;
+ if (snd_BUG_ON(added))
+ return -EINVAL;
+
+ len = read_and_add_raw_conns(codec, nid);
+ if (len < 0)
+ return len;
+ added = true;
+ goto again;
}
EXPORT_SYMBOL_HDA(snd_hda_get_connections);
@@ -543,6 +532,7 @@
hda_nid_t *p;
int i, old_used;
+ mutex_lock(&codec->hash_mutex);
p = lookup_conn_list(array, nid);
if (p)
*p = -1; /* invalidate the old entry */
@@ -553,10 +543,12 @@
for (i = 0; i < len; i++)
if (!add_conn_list(array, list[i]))
goto error_add;
+ mutex_unlock(&codec->hash_mutex);
return 0;
error_add:
array->used = old_used;
+ mutex_unlock(&codec->hash_mutex);
return -ENOMEM;
}
EXPORT_SYMBOL_HDA(snd_hda_override_conn_list);
@@ -1255,6 +1247,7 @@
codec->addr = codec_addr;
mutex_init(&codec->spdif_mutex);
mutex_init(&codec->control_mutex);
+ mutex_init(&codec->hash_mutex);
init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32);
@@ -1264,15 +1257,9 @@
snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8);
snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64);
snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16);
- if (codec->bus->modelname) {
- codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
- if (!codec->modelname) {
- snd_hda_codec_free(codec);
- return -ENODEV;
- }
- }
#ifdef CONFIG_SND_HDA_POWER_SAVE
+ spin_lock_init(&codec->power_lock);
INIT_DELAYED_WORK(&codec->power_work, hda_power_work);
/* snd_hda_codec_new() marks the codec as power-up, and leave it as is.
* the caller has to power down appropriatley after initialization
@@ -1281,6 +1268,14 @@
hda_keep_power_on(codec);
#endif
+ if (codec->bus->modelname) {
+ codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
+ if (!codec->modelname) {
+ snd_hda_codec_free(codec);
+ return -ENODEV;
+ }
+ }
+
list_add_tail(&codec->list, &bus->codec_list);
bus->caddr_tbl[codec_addr] = codec;
@@ -1603,6 +1598,60 @@
return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key);
}
+/* overwrite the value with the key in the caps hash */
+static int write_caps_hash(struct hda_codec *codec, u32 key, unsigned int val)
+{
+ struct hda_amp_info *info;
+
+ mutex_lock(&codec->hash_mutex);
+ info = get_alloc_amp_hash(codec, key);
+ if (!info) {
+ mutex_unlock(&codec->hash_mutex);
+ return -EINVAL;
+ }
+ info->amp_caps = val;
+ info->head.val |= INFO_AMP_CAPS;
+ mutex_unlock(&codec->hash_mutex);
+ return 0;
+}
+
+/* query the value from the caps hash; if not found, fetch the current
+ * value from the given function and store in the hash
+ */
+static unsigned int
+query_caps_hash(struct hda_codec *codec, hda_nid_t nid, int dir, u32 key,
+ unsigned int (*func)(struct hda_codec *, hda_nid_t, int))
+{
+ struct hda_amp_info *info;
+ unsigned int val;
+
+ mutex_lock(&codec->hash_mutex);
+ info = get_alloc_amp_hash(codec, key);
+ if (!info) {
+ mutex_unlock(&codec->hash_mutex);
+ return 0;
+ }
+ if (!(info->head.val & INFO_AMP_CAPS)) {
+ mutex_unlock(&codec->hash_mutex); /* for reentrance */
+ val = func(codec, nid, dir);
+ write_caps_hash(codec, key, val);
+ } else {
+ val = info->amp_caps;
+ mutex_unlock(&codec->hash_mutex);
+ }
+ return val;
+}
+
+static unsigned int read_amp_cap(struct hda_codec *codec, hda_nid_t nid,
+ int direction)
+{
+ if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
+ nid = codec->afg;
+ return snd_hda_param_read(codec, nid,
+ direction == HDA_OUTPUT ?
+ AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
+}
+
/**
* query_amp_caps - query AMP capabilities
* @codec: the HD-auio codec
@@ -1617,22 +1666,9 @@
*/
u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
{
- struct hda_amp_info *info;
-
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
- if (!info)
- return 0;
- if (!(info->head.val & INFO_AMP_CAPS)) {
- if (!(get_wcaps(codec, nid) & AC_WCAP_AMP_OVRD))
- nid = codec->afg;
- info->amp_caps = snd_hda_param_read(codec, nid,
- direction == HDA_OUTPUT ?
- AC_PAR_AMP_OUT_CAP :
- AC_PAR_AMP_IN_CAP);
- if (info->amp_caps)
- info->head.val |= INFO_AMP_CAPS;
- }
- return info->amp_caps;
+ return query_caps_hash(codec, nid, direction,
+ HDA_HASH_KEY(nid, direction, 0),
+ read_amp_cap);
}
EXPORT_SYMBOL_HDA(query_amp_caps);
@@ -1652,34 +1688,12 @@
int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int caps)
{
- struct hda_amp_info *info;
-
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, dir, 0));
- if (!info)
- return -EINVAL;
- info->amp_caps = caps;
- info->head.val |= INFO_AMP_CAPS;
- return 0;
+ return write_caps_hash(codec, HDA_HASH_KEY(nid, dir, 0), caps);
}
EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
-static unsigned int
-query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key,
- unsigned int (*func)(struct hda_codec *, hda_nid_t))
-{
- struct hda_amp_info *info;
-
- info = get_alloc_amp_hash(codec, key);
- if (!info)
- return 0;
- if (!info->head.val) {
- info->head.val |= INFO_AMP_CAPS;
- info->amp_caps = func(codec, nid);
- }
- return info->amp_caps;
-}
-
-static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid,
+ int dir)
{
return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
}
@@ -1697,7 +1711,7 @@
*/
u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
{
- return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid),
+ return query_caps_hash(codec, nid, 0, HDA_HASH_PINCAP_KEY(nid),
read_pin_cap);
}
EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
@@ -1715,41 +1729,47 @@
int snd_hda_override_pin_caps(struct hda_codec *codec, hda_nid_t nid,
unsigned int caps)
{
- struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
- if (!info)
- return -ENOMEM;
- info->amp_caps = caps;
- info->head.val |= INFO_AMP_CAPS;
- return 0;
+ return write_caps_hash(codec, HDA_HASH_PINCAP_KEY(nid), caps);
}
EXPORT_SYMBOL_HDA(snd_hda_override_pin_caps);
-/*
- * read the current volume to info
- * if the cache exists, read the cache value.
+/* read or sync the hash value with the current value;
+ * call within hash_mutex
*/
-static unsigned int get_vol_mute(struct hda_codec *codec,
- struct hda_amp_info *info, hda_nid_t nid,
- int ch, int direction, int index)
+static struct hda_amp_info *
+update_amp_hash(struct hda_codec *codec, hda_nid_t nid, int ch,
+ int direction, int index)
{
- u32 val, parm;
+ struct hda_amp_info *info;
+ unsigned int parm, val = 0;
+ bool val_read = false;
- if (info->head.val & INFO_AMP_VOL(ch))
- return info->vol[ch];
-
- parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
- parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
- parm |= index;
- val = snd_hda_codec_read(codec, nid, 0,
+ retry:
+ info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
+ if (!info)
+ return NULL;
+ if (!(info->head.val & INFO_AMP_VOL(ch))) {
+ if (!val_read) {
+ mutex_unlock(&codec->hash_mutex);
+ parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
+ parm |= direction == HDA_OUTPUT ?
+ AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
+ parm |= index;
+ val = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_AMP_GAIN_MUTE, parm);
- info->vol[ch] = val & 0xff;
- info->head.val |= INFO_AMP_VOL(ch);
- return info->vol[ch];
+ val &= 0xff;
+ val_read = true;
+ mutex_lock(&codec->hash_mutex);
+ goto retry;
+ }
+ info->vol[ch] = val;
+ info->head.val |= INFO_AMP_VOL(ch);
+ }
+ return info;
}
/*
- * write the current volume in info to the h/w and update the cache
+ * write the current volume in info to the h/w
*/
static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
hda_nid_t nid, int ch, int direction, int index,
@@ -1766,7 +1786,6 @@
else
parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
- info->vol[ch] = val;
}
/**
@@ -1783,10 +1802,14 @@
int direction, int index)
{
struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
- if (!info)
- return 0;
- return get_vol_mute(codec, info, nid, ch, direction, index);
+ unsigned int val = 0;
+
+ mutex_lock(&codec->hash_mutex);
+ info = update_amp_hash(codec, nid, ch, direction, index);
+ if (info)
+ val = info->vol[ch];
+ mutex_unlock(&codec->hash_mutex);
+ return val;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read);
@@ -1808,15 +1831,23 @@
{
struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx));
- if (!info)
- return 0;
if (snd_BUG_ON(mask & ~0xff))
mask &= 0xff;
val &= mask;
- val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
- if (info->vol[ch] == val)
+
+ mutex_lock(&codec->hash_mutex);
+ info = update_amp_hash(codec, nid, ch, direction, idx);
+ if (!info) {
+ mutex_unlock(&codec->hash_mutex);
return 0;
+ }
+ val |= info->vol[ch] & ~mask;
+ if (info->vol[ch] == val) {
+ mutex_unlock(&codec->hash_mutex);
+ return 0;
+ }
+ info->vol[ch] = val;
+ mutex_unlock(&codec->hash_mutex);
put_vol_mute(codec, info, nid, ch, direction, idx, val);
return 1;
}
@@ -2263,7 +2294,10 @@
/* OK, let it free */
#ifdef CONFIG_SND_HDA_POWER_SAVE
- cancel_delayed_work(&codec->power_work);
+ cancel_delayed_work_sync(&codec->power_work);
+ codec->power_on = 0;
+ codec->power_transition = 0;
+ codec->power_jiffies = jiffies;
flush_workqueue(codec->bus->workq);
#endif
snd_hda_ctls_clear(codec);
@@ -2859,12 +2893,15 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
+ mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.iec958.status[0] = spdif->status & 0xff;
ucontrol->value.iec958.status[1] = (spdif->status >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (spdif->status >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (spdif->status >> 24) & 0xff;
+ mutex_unlock(&codec->spdif_mutex);
return 0;
}
@@ -2950,12 +2987,14 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
- hda_nid_t nid = spdif->nid;
+ struct hda_spdif_out *spdif;
+ hda_nid_t nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
+ nid = spdif->nid;
spdif->status = ucontrol->value.iec958.status[0] |
((unsigned int)ucontrol->value.iec958.status[1] << 8) |
((unsigned int)ucontrol->value.iec958.status[2] << 16) |
@@ -2977,9 +3016,12 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
+ mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
+ mutex_unlock(&codec->spdif_mutex);
return 0;
}
@@ -2999,12 +3041,14 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
- hda_nid_t nid = spdif->nid;
+ struct hda_spdif_out *spdif;
+ hda_nid_t nid;
unsigned short val;
int change;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
+ nid = spdif->nid;
val = spdif->ctls & ~AC_DIG1_ENABLE;
if (ucontrol->value.integer.value[0])
val |= AC_DIG1_ENABLE;
@@ -3092,6 +3136,9 @@
}
EXPORT_SYMBOL_HDA(snd_hda_create_spdif_out_ctls);
+/* get the hda_spdif_out entry from the given NID
+ * call within spdif_mutex lock
+ */
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid)
{
@@ -3108,9 +3155,10 @@
void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
{
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
spdif->nid = (u16)-1;
mutex_unlock(&codec->spdif_mutex);
}
@@ -3118,10 +3166,11 @@
void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
{
- struct hda_spdif_out *spdif = snd_array_elem(&codec->spdif_out, idx);
+ struct hda_spdif_out *spdif;
unsigned short val;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_array_elem(&codec->spdif_out, idx);
if (spdif->nid != nid) {
spdif->nid = nid;
val = spdif->ctls;
@@ -3486,11 +3535,14 @@
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D3);
#ifdef CONFIG_SND_HDA_POWER_SAVE
- snd_hda_update_power_acct(codec);
cancel_delayed_work(&codec->power_work);
+ spin_lock(&codec->power_lock);
+ snd_hda_update_power_acct(codec);
+ trace_hda_power_down(codec);
codec->power_on = 0;
codec->power_transition = 0;
codec->power_jiffies = jiffies;
+ spin_unlock(&codec->power_lock);
#endif
}
@@ -3499,6 +3551,10 @@
*/
static void hda_call_codec_resume(struct hda_codec *codec)
{
+ /* set as if powered on for avoiding re-entering the resume
+ * in the resume / power-save sequence
+ */
+ hda_keep_power_on(codec);
hda_set_power_state(codec,
codec->afg ? codec->afg : codec->mfg,
AC_PWRST_D0);
@@ -3514,6 +3570,7 @@
snd_hda_codec_resume_amp(codec);
snd_hda_codec_resume_cache(codec);
}
+ snd_hda_power_down(codec); /* flag down before returning */
}
#endif /* CONFIG_PM */
@@ -3665,7 +3722,8 @@
}
EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
-static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid,
+ int dir)
{
unsigned int val = 0;
if (nid != codec->afg &&
@@ -3680,11 +3738,12 @@
static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid)
{
- return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid),
+ return query_caps_hash(codec, nid, 0, HDA_HASH_PARPCM_KEY(nid),
get_pcm_param);
}
-static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid,
+ int dir)
{
unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
if (!streams || streams == -1)
@@ -3696,7 +3755,7 @@
static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
{
- return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid),
+ return query_caps_hash(codec, nid, 0, HDA_HASH_PARSTR_KEY(nid),
get_stream_param);
}
@@ -3775,11 +3834,13 @@
bps = 20;
}
}
+#if 0 /* FIXME: CS4206 doesn't work, which is the only codec supporting float */
if (streams & AC_SUPFMT_FLOAT32) {
formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
if (!bps)
bps = 32;
}
+#endif
if (streams == AC_SUPFMT_AC3) {
/* should be exclusive */
/* temporary hack: we have still no proper support
@@ -4283,12 +4344,18 @@
container_of(work, struct hda_codec, power_work.work);
struct hda_bus *bus = codec->bus;
- if (!codec->power_on || codec->power_count) {
- codec->power_transition = 0;
+ spin_lock(&codec->power_lock);
+ if (codec->power_transition > 0) { /* during power-up sequence? */
+ spin_unlock(&codec->power_lock);
return;
}
+ if (!codec->power_on || codec->power_count) {
+ codec->power_transition = 0;
+ spin_unlock(&codec->power_lock);
+ return;
+ }
+ spin_unlock(&codec->power_lock);
- trace_hda_power_down(codec);
hda_call_codec_suspend(codec);
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
@@ -4296,9 +4363,11 @@
static void hda_keep_power_on(struct hda_codec *codec)
{
+ spin_lock(&codec->power_lock);
codec->power_count++;
codec->power_on = 1;
codec->power_jiffies = jiffies;
+ spin_unlock(&codec->power_lock);
}
/* update the power on/off account with the current jiffies */
@@ -4323,19 +4392,31 @@
{
struct hda_bus *bus = codec->bus;
+ spin_lock(&codec->power_lock);
codec->power_count++;
- if (codec->power_on || codec->power_transition)
+ if (codec->power_on || codec->power_transition > 0) {
+ spin_unlock(&codec->power_lock);
return;
+ }
+ spin_unlock(&codec->power_lock);
+ cancel_delayed_work_sync(&codec->power_work);
+
+ spin_lock(&codec->power_lock);
trace_hda_power_up(codec);
snd_hda_update_power_acct(codec);
codec->power_on = 1;
codec->power_jiffies = jiffies;
+ codec->power_transition = 1; /* avoid reentrance */
+ spin_unlock(&codec->power_lock);
+
if (bus->ops.pm_notify)
bus->ops.pm_notify(bus);
hda_call_codec_resume(codec);
- cancel_delayed_work(&codec->power_work);
+
+ spin_lock(&codec->power_lock);
codec->power_transition = 0;
+ spin_unlock(&codec->power_lock);
}
EXPORT_SYMBOL_HDA(snd_hda_power_up);
@@ -4351,14 +4432,18 @@
*/
void snd_hda_power_down(struct hda_codec *codec)
{
+ spin_lock(&codec->power_lock);
--codec->power_count;
- if (!codec->power_on || codec->power_count || codec->power_transition)
+ if (!codec->power_on || codec->power_count || codec->power_transition) {
+ spin_unlock(&codec->power_lock);
return;
+ }
if (power_save(codec)) {
- codec->power_transition = 1; /* avoid reentrance */
+ codec->power_transition = -1; /* avoid reentrance */
queue_delayed_work(codec->bus->workq, &codec->power_work,
msecs_to_jiffies(power_save(codec) * 1000));
}
+ spin_unlock(&codec->power_lock);
}
EXPORT_SYMBOL_HDA(snd_hda_power_down);
@@ -4710,11 +4795,11 @@
{
const hda_nid_t *nids = mout->dac_nids;
int chs = substream->runtime->channels;
- struct hda_spdif_out *spdif =
- snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
+ struct hda_spdif_out *spdif;
int i;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
if (mout->dig_out_nid && mout->share_spdif &&
mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
if (chs == 2 &&
@@ -4795,601 +4880,58 @@
}
EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup);
-/*
- * Helper for automatic pin configuration
- */
-
-static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list)
-{
- for (; *list; list++)
- if (*list == nid)
- return 1;
- return 0;
-}
-
-
-/*
- * Sort an associated group of pins according to their sequence numbers.
- */
-static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences,
- int num_pins)
-{
- int i, j;
- short seq;
- hda_nid_t nid;
-
- for (i = 0; i < num_pins; i++) {
- for (j = i + 1; j < num_pins; j++) {
- if (sequences[i] > sequences[j]) {
- seq = sequences[i];
- sequences[i] = sequences[j];
- sequences[j] = seq;
- nid = pins[i];
- pins[i] = pins[j];
- pins[j] = nid;
- }
- }
- }
-}
-
-
-/* add the found input-pin to the cfg->inputs[] table */
-static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid,
- int type)
-{
- if (cfg->num_inputs < AUTO_CFG_MAX_INS) {
- cfg->inputs[cfg->num_inputs].pin = nid;
- cfg->inputs[cfg->num_inputs].type = type;
- cfg->num_inputs++;
- }
-}
-
-/* sort inputs in the order of AUTO_PIN_* type */
-static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg)
-{
- int i, j;
-
- for (i = 0; i < cfg->num_inputs; i++) {
- for (j = i + 1; j < cfg->num_inputs; j++) {
- if (cfg->inputs[i].type > cfg->inputs[j].type) {
- struct auto_pin_cfg_item tmp;
- tmp = cfg->inputs[i];
- cfg->inputs[i] = cfg->inputs[j];
- cfg->inputs[j] = tmp;
- }
- }
- }
-}
-
-/* Reorder the surround channels
- * ALSA sequence is front/surr/clfe/side
- * HDA sequence is:
- * 4-ch: front/surr => OK as it is
- * 6-ch: front/clfe/surr
- * 8-ch: front/clfe/rear/side|fc
- */
-static void reorder_outputs(unsigned int nums, hda_nid_t *pins)
-{
- hda_nid_t nid;
-
- switch (nums) {
- case 3:
- case 4:
- nid = pins[1];
- pins[1] = pins[2];
- pins[2] = nid;
- break;
- }
-}
-
-/*
- * Parse all pin widgets and store the useful pin nids to cfg
- *
- * The number of line-outs or any primary output is stored in line_outs,
- * and the corresponding output pins are assigned to line_out_pins[],
- * in the order of front, rear, CLFE, side, ...
- *
- * If more extra outputs (speaker and headphone) are found, the pins are
- * assisnged to hp_pins[] and speaker_pins[], respectively. If no line-out jack
- * is detected, one of speaker of HP pins is assigned as the primary
- * output, i.e. to line_out_pins[0]. So, line_outs is always positive
- * if any analog output exists.
- *
- * The analog input pins are assigned to inputs array.
- * The digital input/output pins are assigned to dig_in_pin and dig_out_pin,
- * respectively.
- */
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids,
- unsigned int cond_flags)
-{
- hda_nid_t nid, end_nid;
- short seq, assoc_line_out;
- short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)];
- short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)];
- short sequences_hp[ARRAY_SIZE(cfg->hp_pins)];
- int i;
-
- memset(cfg, 0, sizeof(*cfg));
-
- memset(sequences_line_out, 0, sizeof(sequences_line_out));
- memset(sequences_speaker, 0, sizeof(sequences_speaker));
- memset(sequences_hp, 0, sizeof(sequences_hp));
- assoc_line_out = 0;
-
- codec->ignore_misc_bit = true;
- end_nid = codec->start_nid + codec->num_nodes;
- for (nid = codec->start_nid; nid < end_nid; nid++) {
- unsigned int wid_caps = get_wcaps(codec, nid);
- unsigned int wid_type = get_wcaps_type(wid_caps);
- unsigned int def_conf;
- short assoc, loc, conn, dev;
-
- /* read all default configuration for pin complex */
- if (wid_type != AC_WID_PIN)
- continue;
- /* ignore the given nids (e.g. pc-beep returns error) */
- if (ignore_nids && is_in_nid_list(nid, ignore_nids))
- continue;
-
- def_conf = snd_hda_codec_get_pincfg(codec, nid);
- if (!(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
- AC_DEFCFG_MISC_NO_PRESENCE))
- codec->ignore_misc_bit = false;
- conn = get_defcfg_connect(def_conf);
- if (conn == AC_JACK_PORT_NONE)
- continue;
- loc = get_defcfg_location(def_conf);
- dev = get_defcfg_device(def_conf);
-
- /* workaround for buggy BIOS setups */
- if (dev == AC_JACK_LINE_OUT) {
- if (conn == AC_JACK_PORT_FIXED)
- dev = AC_JACK_SPEAKER;
- }
-
- switch (dev) {
- case AC_JACK_LINE_OUT:
- seq = get_defcfg_sequence(def_conf);
- assoc = get_defcfg_association(def_conf);
-
- if (!(wid_caps & AC_WCAP_STEREO))
- if (!cfg->mono_out_pin)
- cfg->mono_out_pin = nid;
- if (!assoc)
- continue;
- if (!assoc_line_out)
- assoc_line_out = assoc;
- else if (assoc_line_out != assoc)
- continue;
- if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
- continue;
- cfg->line_out_pins[cfg->line_outs] = nid;
- sequences_line_out[cfg->line_outs] = seq;
- cfg->line_outs++;
- break;
- case AC_JACK_SPEAKER:
- seq = get_defcfg_sequence(def_conf);
- assoc = get_defcfg_association(def_conf);
- if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins))
- continue;
- cfg->speaker_pins[cfg->speaker_outs] = nid;
- sequences_speaker[cfg->speaker_outs] = (assoc << 4) | seq;
- cfg->speaker_outs++;
- break;
- case AC_JACK_HP_OUT:
- seq = get_defcfg_sequence(def_conf);
- assoc = get_defcfg_association(def_conf);
- if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
- continue;
- cfg->hp_pins[cfg->hp_outs] = nid;
- sequences_hp[cfg->hp_outs] = (assoc << 4) | seq;
- cfg->hp_outs++;
- break;
- case AC_JACK_MIC_IN:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC);
- break;
- case AC_JACK_LINE_IN:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN);
- break;
- case AC_JACK_CD:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD);
- break;
- case AC_JACK_AUX:
- add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX);
- break;
- case AC_JACK_SPDIF_OUT:
- case AC_JACK_DIG_OTHER_OUT:
- if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
- continue;
- cfg->dig_out_pins[cfg->dig_outs] = nid;
- cfg->dig_out_type[cfg->dig_outs] =
- (loc == AC_JACK_LOC_HDMI) ?
- HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
- cfg->dig_outs++;
- break;
- case AC_JACK_SPDIF_IN:
- case AC_JACK_DIG_OTHER_IN:
- cfg->dig_in_pin = nid;
- if (loc == AC_JACK_LOC_HDMI)
- cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
- else
- cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
- break;
- }
- }
-
- /* FIX-UP:
- * If no line-out is defined but multiple HPs are found,
- * some of them might be the real line-outs.
- */
- if (!cfg->line_outs && cfg->hp_outs > 1 &&
- !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
- int i = 0;
- while (i < cfg->hp_outs) {
- /* The real HPs should have the sequence 0x0f */
- if ((sequences_hp[i] & 0x0f) == 0x0f) {
- i++;
- continue;
- }
- /* Move it to the line-out table */
- cfg->line_out_pins[cfg->line_outs] = cfg->hp_pins[i];
- sequences_line_out[cfg->line_outs] = sequences_hp[i];
- cfg->line_outs++;
- cfg->hp_outs--;
- memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1,
- sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i));
- memmove(sequences_hp + i, sequences_hp + i + 1,
- sizeof(sequences_hp[0]) * (cfg->hp_outs - i));
- }
- memset(cfg->hp_pins + cfg->hp_outs, 0,
- sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs));
- if (!cfg->hp_outs)
- cfg->line_out_type = AUTO_PIN_HP_OUT;
-
- }
-
- /* sort by sequence */
- sort_pins_by_sequence(cfg->line_out_pins, sequences_line_out,
- cfg->line_outs);
- sort_pins_by_sequence(cfg->speaker_pins, sequences_speaker,
- cfg->speaker_outs);
- sort_pins_by_sequence(cfg->hp_pins, sequences_hp,
- cfg->hp_outs);
-
- /*
- * FIX-UP: if no line-outs are detected, try to use speaker or HP pin
- * as a primary output
- */
- if (!cfg->line_outs &&
- !(cond_flags & HDA_PINCFG_NO_LO_FIXUP)) {
- if (cfg->speaker_outs) {
- cfg->line_outs = cfg->speaker_outs;
- memcpy(cfg->line_out_pins, cfg->speaker_pins,
- sizeof(cfg->speaker_pins));
- cfg->speaker_outs = 0;
- memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
- cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
- } else if (cfg->hp_outs) {
- cfg->line_outs = cfg->hp_outs;
- memcpy(cfg->line_out_pins, cfg->hp_pins,
- sizeof(cfg->hp_pins));
- cfg->hp_outs = 0;
- memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
- cfg->line_out_type = AUTO_PIN_HP_OUT;
- }
- }
-
- reorder_outputs(cfg->line_outs, cfg->line_out_pins);
- reorder_outputs(cfg->hp_outs, cfg->hp_pins);
- reorder_outputs(cfg->speaker_outs, cfg->speaker_pins);
-
- sort_autocfg_input_pins(cfg);
-
- /*
- * debug prints of the parsed results
- */
- snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n",
- cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1],
- cfg->line_out_pins[2], cfg->line_out_pins[3],
- cfg->line_out_pins[4],
- cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" :
- (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ?
- "speaker" : "line"));
- snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
- cfg->speaker_outs, cfg->speaker_pins[0],
- cfg->speaker_pins[1], cfg->speaker_pins[2],
- cfg->speaker_pins[3], cfg->speaker_pins[4]);
- snd_printd(" hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
- cfg->hp_outs, cfg->hp_pins[0],
- cfg->hp_pins[1], cfg->hp_pins[2],
- cfg->hp_pins[3], cfg->hp_pins[4]);
- snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin);
- if (cfg->dig_outs)
- snd_printd(" dig-out=0x%x/0x%x\n",
- cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
- snd_printd(" inputs:");
- for (i = 0; i < cfg->num_inputs; i++) {
- snd_printd(" %s=0x%x",
- hda_get_autocfg_input_label(codec, cfg, i),
- cfg->inputs[i].pin);
- }
- snd_printd("\n");
- if (cfg->dig_in_pin)
- snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin);
-
- return 0;
-}
-EXPORT_SYMBOL_HDA(snd_hda_parse_pin_defcfg);
-
-int snd_hda_get_input_pin_attr(unsigned int def_conf)
-{
- unsigned int loc = get_defcfg_location(def_conf);
- unsigned int conn = get_defcfg_connect(def_conf);
- if (conn == AC_JACK_PORT_NONE)
- return INPUT_PIN_ATTR_UNUSED;
- /* Windows may claim the internal mic to be BOTH, too */
- if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH)
- return INPUT_PIN_ATTR_INT;
- if ((loc & 0x30) == AC_JACK_LOC_INTERNAL)
- return INPUT_PIN_ATTR_INT;
- if ((loc & 0x30) == AC_JACK_LOC_SEPARATE)
- return INPUT_PIN_ATTR_DOCK;
- if (loc == AC_JACK_LOC_REAR)
- return INPUT_PIN_ATTR_REAR;
- if (loc == AC_JACK_LOC_FRONT)
- return INPUT_PIN_ATTR_FRONT;
- return INPUT_PIN_ATTR_NORMAL;
-}
-EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr);
-
/**
- * hda_get_input_pin_label - Give a label for the given input pin
+ * snd_hda_get_default_vref - Get the default (mic) VREF pin bits
*
- * When check_location is true, the function checks the pin location
- * for mic and line-in pins, and set an appropriate prefix like "Front",
- * "Rear", "Internal".
+ * Guess the suitable VREF pin bits to be set as the pin-control value.
+ * Note: the function doesn't set the AC_PINCTL_IN_EN bit.
*/
-
-static const char *hda_get_input_pin_label(struct hda_codec *codec,
- hda_nid_t pin, bool check_location)
+unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin)
{
- unsigned int def_conf;
- static const char * const mic_names[] = {
- "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
- };
- int attr;
-
- def_conf = snd_hda_codec_get_pincfg(codec, pin);
-
- switch (get_defcfg_device(def_conf)) {
- case AC_JACK_MIC_IN:
- if (!check_location)
- return "Mic";
- attr = snd_hda_get_input_pin_attr(def_conf);
- if (!attr)
- return "None";
- return mic_names[attr - 1];
- case AC_JACK_LINE_IN:
- if (!check_location)
- return "Line";
- attr = snd_hda_get_input_pin_attr(def_conf);
- if (!attr)
- return "None";
- if (attr == INPUT_PIN_ATTR_DOCK)
- return "Dock Line";
- return "Line";
- case AC_JACK_AUX:
- return "Aux";
- case AC_JACK_CD:
- return "CD";
- case AC_JACK_SPDIF_IN:
- return "SPDIF In";
- case AC_JACK_DIG_OTHER_IN:
- return "Digital In";
- default:
- return "Misc";
- }
+ unsigned int pincap;
+ unsigned int oldval;
+ oldval = snd_hda_codec_read(codec, pin, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ pincap = snd_hda_query_pin_caps(codec, pin);
+ pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+ /* Exception: if the default pin setup is vref50, we give it priority */
+ if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
+ return AC_PINCTL_VREF_80;
+ else if (pincap & AC_PINCAP_VREF_50)
+ return AC_PINCTL_VREF_50;
+ else if (pincap & AC_PINCAP_VREF_100)
+ return AC_PINCTL_VREF_100;
+ else if (pincap & AC_PINCAP_VREF_GRD)
+ return AC_PINCTL_VREF_GRD;
+ return AC_PINCTL_VREF_HIZ;
}
+EXPORT_SYMBOL_HDA(snd_hda_get_default_vref);
-/* Check whether the location prefix needs to be added to the label.
- * If all mic-jacks are in the same location (e.g. rear panel), we don't
- * have to put "Front" prefix to each label. In such a case, returns false.
- */
-static int check_mic_location_need(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- int input)
+int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int val, bool cached)
{
- unsigned int defc;
- int i, attr, attr2;
-
- defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin);
- attr = snd_hda_get_input_pin_attr(defc);
- /* for internal or docking mics, we need locations */
- if (attr <= INPUT_PIN_ATTR_NORMAL)
- return 1;
-
- attr = 0;
- for (i = 0; i < cfg->num_inputs; i++) {
- defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin);
- attr2 = snd_hda_get_input_pin_attr(defc);
- if (attr2 >= INPUT_PIN_ATTR_NORMAL) {
- if (attr && attr != attr2)
- return 1; /* different locations found */
- attr = attr2;
+ if (val) {
+ unsigned int cap = snd_hda_query_pin_caps(codec, pin);
+ if (cap && (val & AC_PINCTL_OUT_EN)) {
+ if (!(cap & AC_PINCAP_OUT))
+ val &= ~(AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+ else if ((val & AC_PINCTL_HP_EN) &&
+ !(cap & AC_PINCAP_HP_DRV))
+ val &= ~AC_PINCTL_HP_EN;
+ }
+ if (cap && (val & AC_PINCTL_IN_EN)) {
+ if (!(cap & AC_PINCAP_IN))
+ val &= ~(AC_PINCTL_IN_EN | AC_PINCTL_VREFEN);
}
}
- return 0;
+ if (cached)
+ return snd_hda_codec_update_cache(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ else
+ return snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
}
-
-/**
- * hda_get_autocfg_input_label - Get a label for the given input
- *
- * Get a label for the given input pin defined by the autocfg item.
- * Unlike hda_get_input_pin_label(), this function checks all inputs
- * defined in autocfg and avoids the redundant mic/line prefix as much as
- * possible.
- */
-const char *hda_get_autocfg_input_label(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- int input)
-{
- int type = cfg->inputs[input].type;
- int has_multiple_pins = 0;
-
- if ((input > 0 && cfg->inputs[input - 1].type == type) ||
- (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type))
- has_multiple_pins = 1;
- if (has_multiple_pins && type == AUTO_PIN_MIC)
- has_multiple_pins &= check_mic_location_need(codec, cfg, input);
- return hda_get_input_pin_label(codec, cfg->inputs[input].pin,
- has_multiple_pins);
-}
-EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label);
-
-/* return the position of NID in the list, or -1 if not found */
-static int find_idx_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
-{
- int i;
- for (i = 0; i < nums; i++)
- if (list[i] == nid)
- return i;
- return -1;
-}
-
-/* get a unique suffix or an index number */
-static const char *check_output_sfx(hda_nid_t nid, const hda_nid_t *pins,
- int num_pins, int *indexp)
-{
- static const char * const channel_sfx[] = {
- " Front", " Surround", " CLFE", " Side"
- };
- int i;
-
- i = find_idx_in_nid_list(nid, pins, num_pins);
- if (i < 0)
- return NULL;
- if (num_pins == 1)
- return "";
- if (num_pins > ARRAY_SIZE(channel_sfx)) {
- if (indexp)
- *indexp = i;
- return "";
- }
- return channel_sfx[i];
-}
-
-static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg,
- const char *name, char *label, int maxlen,
- int *indexp)
-{
- unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
- int attr = snd_hda_get_input_pin_attr(def_conf);
- const char *pfx = "", *sfx = "";
-
- /* handle as a speaker if it's a fixed line-out */
- if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
- name = "Speaker";
- /* check the location */
- switch (attr) {
- case INPUT_PIN_ATTR_DOCK:
- pfx = "Dock ";
- break;
- case INPUT_PIN_ATTR_FRONT:
- pfx = "Front ";
- break;
- }
- if (cfg) {
- /* try to give a unique suffix if needed */
- sfx = check_output_sfx(nid, cfg->line_out_pins, cfg->line_outs,
- indexp);
- if (!sfx)
- sfx = check_output_sfx(nid, cfg->speaker_pins, cfg->speaker_outs,
- indexp);
- if (!sfx) {
- /* don't add channel suffix for Headphone controls */
- int idx = find_idx_in_nid_list(nid, cfg->hp_pins,
- cfg->hp_outs);
- if (idx >= 0)
- *indexp = idx;
- sfx = "";
- }
- }
- snprintf(label, maxlen, "%s%s%s", pfx, name, sfx);
- return 1;
-}
-
-/**
- * snd_hda_get_pin_label - Get a label for the given I/O pin
- *
- * Get a label for the given pin. This function works for both input and
- * output pins. When @cfg is given as non-NULL, the function tries to get
- * an optimized label using hda_get_autocfg_input_label().
- *
- * This function tries to give a unique label string for the pin as much as
- * possible. For example, when the multiple line-outs are present, it adds
- * the channel suffix like "Front", "Surround", etc (only when @cfg is given).
- * If no unique name with a suffix is available and @indexp is non-NULL, the
- * index number is stored in the pointer.
- */
-int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg,
- char *label, int maxlen, int *indexp)
-{
- unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid);
- const char *name = NULL;
- int i;
-
- if (indexp)
- *indexp = 0;
- if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
- return 0;
-
- switch (get_defcfg_device(def_conf)) {
- case AC_JACK_LINE_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Line Out",
- label, maxlen, indexp);
- case AC_JACK_SPEAKER:
- return fill_audio_out_name(codec, nid, cfg, "Speaker",
- label, maxlen, indexp);
- case AC_JACK_HP_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Headphone",
- label, maxlen, indexp);
- case AC_JACK_SPDIF_OUT:
- case AC_JACK_DIG_OTHER_OUT:
- if (get_defcfg_location(def_conf) == AC_JACK_LOC_HDMI)
- name = "HDMI";
- else
- name = "SPDIF";
- if (cfg && indexp) {
- i = find_idx_in_nid_list(nid, cfg->dig_out_pins,
- cfg->dig_outs);
- if (i >= 0)
- *indexp = i;
- }
- break;
- default:
- if (cfg) {
- for (i = 0; i < cfg->num_inputs; i++) {
- if (cfg->inputs[i].pin != nid)
- continue;
- name = hda_get_autocfg_input_label(codec, cfg, i);
- if (name)
- break;
- }
- }
- if (!name)
- name = hda_get_input_pin_label(codec, nid, true);
- break;
- }
- if (!name)
- return 0;
- strlcpy(label, name, maxlen);
- return 1;
-}
-EXPORT_SYMBOL_HDA(snd_hda_get_pin_label);
+EXPORT_SYMBOL_HDA(_snd_hda_set_pin_ctl);
/**
* snd_hda_add_imux_item - Add an item to input_mux
@@ -5444,8 +4986,6 @@
list_for_each_entry(codec, &bus->codec_list, list) {
if (hda_codec_is_power_on(codec))
hda_call_codec_suspend(codec);
- if (codec->patch_ops.post_suspend)
- codec->patch_ops.post_suspend(codec);
}
return 0;
}
@@ -5465,10 +5005,7 @@
struct hda_codec *codec;
list_for_each_entry(codec, &bus->codec_list, list) {
- if (codec->patch_ops.pre_resume)
- codec->patch_ops.pre_resume(codec);
- if (snd_hda_codec_needs_resume(codec))
- hda_call_codec_resume(codec);
+ hda_call_codec_resume(codec);
}
return 0;
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 56b4f74..54b5281 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -704,8 +704,6 @@
unsigned int power_state);
#ifdef CONFIG_PM
int (*suspend)(struct hda_codec *codec, pm_message_t state);
- int (*post_suspend)(struct hda_codec *codec);
- int (*pre_resume)(struct hda_codec *codec);
int (*resume)(struct hda_codec *codec);
#endif
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -829,6 +827,7 @@
struct mutex spdif_mutex;
struct mutex control_mutex;
+ struct mutex hash_mutex;
struct snd_array spdif_out;
unsigned int spdif_in_enable; /* SPDIF input enable? */
const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
@@ -861,12 +860,13 @@
unsigned int no_jack_detect:1; /* Machine has no jack-detection */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
- unsigned int power_transition :1; /* power-state in transition */
+ int power_transition; /* power-state in transition */
int power_count; /* current (global) power refcount */
struct delayed_work power_work; /* delayed task for powerdown */
unsigned long power_on_acct;
unsigned long power_off_acct;
unsigned long power_jiffies;
+ spinlock_t power_lock;
#endif
/* codec-specific additional proc output */
@@ -911,10 +911,13 @@
hda_nid_t *start_id);
int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
+static inline int
+snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid)
+{
+ return snd_hda_get_connections(codec, nid, NULL, 0);
+}
int snd_hda_get_raw_connections(struct hda_codec *codec, hda_nid_t nid,
hda_nid_t *conn_list, int max_conns);
-int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid,
- const hda_nid_t **listp);
int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
const hda_nid_t *list);
int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
@@ -1051,12 +1054,10 @@
#ifdef CONFIG_SND_HDA_POWER_SAVE
void snd_hda_power_up(struct hda_codec *codec);
void snd_hda_power_down(struct hda_codec *codec);
-#define snd_hda_codec_needs_resume(codec) codec->power_count
void snd_hda_update_power_acct(struct hda_codec *codec);
#else
static inline void snd_hda_power_up(struct hda_codec *codec) {}
static inline void snd_hda_power_down(struct hda_codec *codec) {}
-#define snd_hda_codec_needs_resume(codec) 1
#endif
#ifdef CONFIG_SND_HDA_PATCH_LOADER
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1f35052..4ab8102 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -497,6 +497,7 @@
AZX_DRIVER_NVIDIA,
AZX_DRIVER_TERA,
AZX_DRIVER_CTX,
+ AZX_DRIVER_CTHDA,
AZX_DRIVER_GENERIC,
AZX_NUM_DRIVERS, /* keep this as last entry */
};
@@ -518,6 +519,7 @@
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */
+#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -533,6 +535,9 @@
(AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
AZX_DCAPS_ALIGN_BUFSIZE)
+#define AZX_DCAPS_PRESET_CTHDA \
+ (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY)
+
static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ICH] = "HDA Intel",
[AZX_DRIVER_PCH] = "HDA Intel PCH",
@@ -546,6 +551,7 @@
[AZX_DRIVER_NVIDIA] = "HDA NVidia",
[AZX_DRIVER_TERA] = "HDA Teradici",
[AZX_DRIVER_CTX] = "HDA Creative",
+ [AZX_DRIVER_CTHDA] = "HDA Creative",
[AZX_DRIVER_GENERIC] = "HD-Audio Generic",
};
@@ -1285,7 +1291,8 @@
/*
* set up a BDL entry
*/
-static int setup_bdle(struct snd_pcm_substream *substream,
+static int setup_bdle(struct azx *chip,
+ struct snd_pcm_substream *substream,
struct azx_dev *azx_dev, u32 **bdlp,
int ofs, int size, int with_ioc)
{
@@ -1304,6 +1311,12 @@
bdl[1] = cpu_to_le32(upper_32_bits(addr));
/* program the size field of the BDL entry */
chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size);
+ /* one BDLE cannot cross 4K boundary on CTHDA chips */
+ if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) {
+ u32 remain = 0x1000 - (ofs & 0xfff);
+ if (chunk > remain)
+ chunk = remain;
+ }
bdl[2] = cpu_to_le32(chunk);
/* program the IOC to enable interrupt
* only when the whole fragment is processed
@@ -1356,7 +1369,7 @@
bdl_pos_adj[chip->dev_index]);
pos_adj = 0;
} else {
- ofs = setup_bdle(substream, azx_dev,
+ ofs = setup_bdle(chip, substream, azx_dev,
&bdl, ofs, pos_adj,
!substream->runtime->no_period_wakeup);
if (ofs < 0)
@@ -1366,10 +1379,10 @@
pos_adj = 0;
for (i = 0; i < periods; i++) {
if (i == periods - 1 && pos_adj)
- ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+ ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs,
period_bytes - pos_adj, 0);
else
- ofs = setup_bdle(substream, azx_dev, &bdl, ofs,
+ ofs = setup_bdle(chip, substream, azx_dev, &bdl, ofs,
period_bytes,
!substream->runtime->no_period_wakeup);
if (ofs < 0)
@@ -2353,17 +2366,6 @@
* power management
*/
-static int snd_hda_codecs_inuse(struct hda_bus *bus)
-{
- struct hda_codec *codec;
-
- list_for_each_entry(codec, &bus->codec_list, list) {
- if (snd_hda_codec_needs_resume(codec))
- return 1;
- }
- return 0;
-}
-
static int azx_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
@@ -2410,8 +2412,7 @@
return -EIO;
azx_init_pci(chip);
- if (snd_hda_codecs_inuse(chip->bus))
- azx_init_chip(chip, 1);
+ azx_init_chip(chip, 1);
snd_hda_resume(chip->bus);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
@@ -2565,6 +2566,8 @@
/* forced codec slots */
SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
+ /* WinFast VP200 H (Teradici) user reported broken communication */
+ SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101),
{}
};
@@ -3130,6 +3133,11 @@
.driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND |
AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB },
#endif
+ /* CTHDA chips */
+ { PCI_DEVICE(0x1102, 0x0010),
+ .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
+ { PCI_DEVICE(0x1102, 0x0012),
+ .driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
/* Vortex86MX */
{ PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC },
/* VMware HDAudio */
@@ -3148,7 +3156,7 @@
MODULE_DEVICE_TABLE(pci, azx_ids);
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver azx_driver = {
.name = KBUILD_MODNAME,
.id_table = azx_ids,
.probe = azx_probe,
@@ -3159,15 +3167,4 @@
#endif
};
-static int __init alsa_card_azx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_azx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_azx_init)
-module_exit(alsa_card_azx_exit)
+module_pci_driver(azx_driver);
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d689484..2dd1c11 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -17,6 +17,7 @@
#include <sound/jack.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_jack.h"
bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index c66655c..8ae5246 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -12,6 +12,8 @@
#ifndef __SOUND_HDA_JACK_H
#define __SOUND_HDA_JACK_H
+struct auto_pin_cfg;
+
struct hda_jack_tbl {
hda_nid_t nid;
unsigned char action; /* event action (0 = none) */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 0ec9248..9a096a8 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -262,6 +262,8 @@
const struct hda_input_mux *imux,
struct snd_ctl_elem_value *ucontrol, hda_nid_t nid,
unsigned int *cur_val);
+int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
+ int index, int *type_index_ret);
/*
* Channel mode helper
@@ -393,72 +395,7 @@
struct hda_bus *bus;
};
-/*
- * Helper for automatic pin configuration
- */
-
-enum {
- AUTO_PIN_MIC,
- AUTO_PIN_LINE_IN,
- AUTO_PIN_CD,
- AUTO_PIN_AUX,
- AUTO_PIN_LAST
-};
-
-enum {
- AUTO_PIN_LINE_OUT,
- AUTO_PIN_SPEAKER_OUT,
- AUTO_PIN_HP_OUT
-};
-
-#define AUTO_CFG_MAX_OUTS HDA_MAX_OUTS
-#define AUTO_CFG_MAX_INS 8
-
-struct auto_pin_cfg_item {
- hda_nid_t pin;
- int type;
-};
-
-struct auto_pin_cfg;
-const char *hda_get_autocfg_input_label(struct hda_codec *codec,
- const struct auto_pin_cfg *cfg,
- int input);
-int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg,
- char *label, int maxlen, int *indexp);
-int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label,
- int index, int *type_index_ret);
-
-enum {
- INPUT_PIN_ATTR_UNUSED, /* pin not connected */
- INPUT_PIN_ATTR_INT, /* internal mic/line-in */
- INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */
- INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */
- INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */
- INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */
-};
-
-int snd_hda_get_input_pin_attr(unsigned int def_conf);
-
-struct auto_pin_cfg {
- int line_outs;
- /* sorted in the order of Front/Surr/CLFE/Side */
- hda_nid_t line_out_pins[AUTO_CFG_MAX_OUTS];
- int speaker_outs;
- hda_nid_t speaker_pins[AUTO_CFG_MAX_OUTS];
- int hp_outs;
- int line_out_type; /* AUTO_PIN_XXX_OUT */
- hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
- int num_inputs;
- struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS];
- int dig_outs;
- hda_nid_t dig_out_pins[2];
- hda_nid_t dig_in_pin;
- hda_nid_t mono_out_pin;
- int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
- int dig_in_type; /* HDA_PCM_TYPE_XXX */
-};
-
+/* helper macros to retrieve pin default-config values */
#define get_defcfg_connect(cfg) \
((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
#define get_defcfg_association(cfg) \
@@ -472,19 +409,6 @@
#define get_defcfg_misc(cfg) \
((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT)
-/* bit-flags for snd_hda_parse_pin_def_config() behavior */
-#define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */
-#define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */
-
-int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
- struct auto_pin_cfg *cfg,
- const hda_nid_t *ignore_nids,
- unsigned int cond_flags);
-
-/* older function */
-#define snd_hda_parse_pin_def_config(codec, cfg, ignore) \
- snd_hda_parse_pin_defcfg(codec, cfg, ignore, 0)
-
/* amp values */
#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8))
@@ -502,6 +426,46 @@
#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN)
#define PIN_HP_AMP (AC_PINCTL_HP_EN)
+unsigned int snd_hda_get_default_vref(struct hda_codec *codec, hda_nid_t pin);
+int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int val, bool cached);
+
+/**
+ * _snd_hda_set_pin_ctl - Set a pin-control value safely
+ * @codec: the codec instance
+ * @pin: the pin NID to set the control
+ * @val: the pin-control value (AC_PINCTL_* bits)
+ *
+ * This function sets the pin-control value to the given pin, but
+ * filters out the invalid pin-control bits when the pin has no such
+ * capabilities. For example, when PIN_HP is passed but the pin has no
+ * HP-drive capability, the HP bit is omitted.
+ *
+ * The function doesn't check the input VREF capability bits, though.
+ * Use snd_hda_get_default_vref() to guess the right value.
+ * Also, this function is only for analog pins, not for HDMI pins.
+ */
+static inline int
+snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, unsigned int val)
+{
+ return _snd_hda_set_pin_ctl(codec, pin, val, false);
+}
+
+/**
+ * snd_hda_set_pin_ctl_cache - Set a pin-control value safely
+ * @codec: the codec instance
+ * @pin: the pin NID to set the control
+ * @val: the pin-control value (AC_PINCTL_* bits)
+ *
+ * Just like snd_hda_set_pin_ctl() but write to cache as well.
+ */
+static inline int
+snd_hda_set_pin_ctl_cache(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int val)
+{
+ return _snd_hda_set_pin_ctl(codec, pin, val, true);
+}
+
/*
* get widget capabilities
*/
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 7143393..d8b2d6d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -28,6 +28,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -1742,9 +1743,7 @@
if (! ad198x_eapd_put(kcontrol, ucontrol))
return 0;
/* change speaker pin appropriately */
- snd_hda_codec_write(codec, 0x05, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->cur_eapd ? PIN_OUT : 0);
+ snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
/* toggle HP mute appropriately */
snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
HDA_AMP_MUTE,
@@ -3103,7 +3102,7 @@
int dac_idx)
{
/* set as output */
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_set_pin_ctl(codec, nid, pin_type);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
switch (nid) {
case 0x11: /* port-A - DAC 03 */
@@ -3157,6 +3156,7 @@
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
int type = cfg->inputs[i].type;
+ int val;
switch (nid) {
case 0x15: /* port-C */
snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
@@ -3165,8 +3165,10 @@
snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_CONNECT_SEL, 0x0);
break;
}
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN);
+ val = PIN_IN;
+ if (type == AUTO_PIN_MIC)
+ val |= snd_hda_get_default_vref(codec, nid);
+ snd_hda_set_pin_ctl(codec, nid, val);
if (nid != AD1988_PIN_CD_NID)
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_MUTE);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 09ccfab..19ae14f 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -26,6 +26,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
/*
*/
@@ -341,8 +342,7 @@
static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_set_pin_ctl(codec, pin, PIN_HP);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
@@ -356,8 +356,8 @@
static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80);
+ snd_hda_set_pin_ctl(codec, pin, PIN_IN |
+ snd_hda_get_default_vref(codec, pin));
if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 21d91d5..d0d3540 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -30,6 +30,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#define WIDGET_CHIP_CTRL 0x15
#define WIDGET_DSP_CTRL 0x16
@@ -239,8 +240,7 @@
static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_set_pin_ctl(codec, pin, PIN_HP);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
@@ -254,9 +254,8 @@
static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
{
if (pin) {
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_VREF80);
+ snd_hda_set_pin_ctl(codec, pin, PIN_IN |
+ snd_hda_get_default_vref(codec, pin));
if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c83ccdb..9647ed4 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -26,6 +26,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_jack.h"
#include <sound/tlv.h>
@@ -933,8 +934,7 @@
pin_ctl = 0;
nid = cfg->speaker_pins[i];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pin_ctl);
+ snd_hda_set_pin_ctl(codec, nid, pin_ctl);
}
if (spec->gpio_eapd_hp) {
unsigned int gpio = hp_present ?
@@ -948,16 +948,14 @@
/* mute HPs if spdif jack (SENSE_B) is present */
for (i = 0; i < cfg->hp_outs; i++) {
nid = cfg->hp_pins[i];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, nid,
(spdif_present && spec->sense_b) ? 0 : PIN_HP);
}
/* SPDIF TX on/off */
if (cfg->dig_outs) {
nid = cfg->dig_out_pins[0];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, nid,
spdif_present ? PIN_OUT : 0);
}
@@ -1024,13 +1022,11 @@
/* set appropriate pin controls */
for (i = 0; i < cfg->line_outs; i++)
- snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, cfg->line_out_pins[i], PIN_OUT);
/* HP */
for (i = 0; i < cfg->hp_outs; i++) {
hda_nid_t nid = cfg->hp_pins[i];
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ snd_hda_set_pin_ctl(codec, nid, PIN_HP);
if (!cfg->speaker_outs)
continue;
if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
@@ -1041,8 +1037,7 @@
/* Speaker */
for (i = 0; i < cfg->speaker_outs; i++)
- snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, cfg->speaker_pins[i], PIN_OUT);
/* SPDIF is enabled on presence detect for CS421x */
if (spec->hp_detect || spec->spdif_detect)
@@ -1063,14 +1058,9 @@
continue;
/* set appropriate pin control and mute first */
ctl = PIN_IN;
- if (cfg->inputs[i].type == AUTO_PIN_MIC) {
- unsigned int caps = snd_hda_query_pin_caps(codec, pin);
- caps >>= AC_PINCAP_VREF_SHIFT;
- if (caps & AC_PINCAP_VREF_80)
- ctl = PIN_VREF80;
- }
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+ if (cfg->inputs[i].type == AUTO_PIN_MIC)
+ ctl |= snd_hda_get_default_vref(codec, pin);
+ snd_hda_set_pin_ctl(codec, pin, ctl);
snd_hda_codec_write(codec, spec->adc_nid[i], 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_MUTE(spec->adc_idx[i]));
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index b6767b4..c8fdaae 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -29,6 +29,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#define NUM_PINS 11
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d906c5b..3acb582 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -30,6 +30,7 @@
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -66,6 +67,7 @@
};
struct conexant_spec {
+ struct hda_gen_spec gen;
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
@@ -141,6 +143,7 @@
unsigned int hp_laptop:1;
unsigned int asus:1;
unsigned int pin_eapd_ctrls:1;
+ unsigned int fixup_stereo_dmic:1;
unsigned int adc_switching:1;
@@ -1601,17 +1604,13 @@
unsigned int pinctl;
/* headphone pin */
pinctl = (spec->hp_present && spec->cur_eapd) ? PIN_HP : 0;
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x16, pinctl);
/* speaker pin */
pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1a, pinctl);
/* on ideapad there is an additional speaker (subwoofer) to mute */
if (spec->ideapad)
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1b, pinctl);
}
/* turn on/off EAPD (+ mute HP) as a master switch */
@@ -1996,8 +1995,7 @@
/* Port A (HP) */
pinctl = (hp_port_a_present(spec) && spec->cur_eapd) ? PIN_HP : 0;
- snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x19, pinctl);
/* Port D (HP/LO) */
pinctl = spec->cur_eapd ? spec->port_d_mode : 0;
@@ -2010,13 +2008,11 @@
if (!hp_port_d_present(spec))
pinctl = 0;
}
- snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1c, pinctl);
/* CLASS_D AMP */
pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinctl);
+ snd_hda_set_pin_ctl(codec, 0x1f, pinctl);
}
/* turn on/off EAPD (+ mute HP) as a master switch */
@@ -2047,8 +2043,7 @@
/* Even though port F is the DC input, the bias is controlled on port B.
* we also leave that port as an active input (but unselected) in DC mode
* just in case that is necessary to make the bias setting take effect. */
- return snd_hda_codec_write_cache(codec, 0x1a, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ return snd_hda_set_pin_ctl_cache(codec, 0x1a,
cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index);
}
@@ -2081,14 +2076,14 @@
}
/* disable DC (port F) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_set_pin_ctl(codec, 0x1e, 0);
/* external mic, port B */
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, 0x1a,
spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0);
/* internal mic, port C */
- snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, 0x1b,
spec->ext_mic_present ? 0 : PIN_VREF80);
}
@@ -3357,9 +3352,7 @@
struct conexant_spec *spec = codec->spec;
int i;
for (i = 0; i < num_pins; i++)
- snd_hda_codec_write(codec, pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- on ? PIN_OUT : 0);
+ snd_hda_set_pin_ctl(codec, pins[i], on ? PIN_OUT : 0);
if (spec->pin_eapd_ctrls)
cx_auto_turn_eapd(codec, num_pins, pins, on);
}
@@ -3976,8 +3969,7 @@
if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
AC_PINCAP_HP_DRV)
val |= AC_PINCTL_HP_EN;
- snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, cfg->hp_pins[i], val);
}
mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
@@ -4030,13 +4022,11 @@
}
for (i = 0; i < cfg->num_inputs; i++) {
- unsigned int type;
+ hda_nid_t pin = cfg->inputs[i].pin;
+ unsigned int type = PIN_IN;
if (cfg->inputs[i].type == AUTO_PIN_MIC)
- type = PIN_VREF80;
- else
- type = PIN_IN;
- snd_hda_codec_write(codec, cfg->inputs[i].pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, type);
+ type |= snd_hda_get_default_vref(codec, pin);
+ snd_hda_set_pin_ctl(codec, pin, type);
}
if (spec->auto_mic) {
@@ -4063,11 +4053,9 @@
struct auto_pin_cfg *cfg = &spec->autocfg;
if (spec->multiout.dig_out_nid)
- snd_hda_codec_write(codec, cfg->dig_out_pins[0], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, cfg->dig_out_pins[0], PIN_OUT);
if (spec->dig_in_nid)
- snd_hda_codec_write(codec, cfg->dig_in_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+ snd_hda_set_pin_ctl(codec, cfg->dig_in_pin, PIN_IN);
}
static int cx_auto_init(struct hda_codec *codec)
@@ -4084,9 +4072,9 @@
static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
const char *dir, int cidx,
- hda_nid_t nid, int hda_dir, int amp_idx)
+ hda_nid_t nid, int hda_dir, int amp_idx, int chs)
{
- static char name[32];
+ static char name[44];
static struct snd_kcontrol_new knew[] = {
HDA_CODEC_VOLUME(name, 0, 0, 0),
HDA_CODEC_MUTE(name, 0, 0, 0),
@@ -4096,7 +4084,7 @@
for (i = 0; i < 2; i++) {
struct snd_kcontrol *kctl;
- knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+ knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, chs, amp_idx,
hda_dir);
knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
knew[i].index = cidx;
@@ -4115,7 +4103,7 @@
}
#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir) \
- cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+ cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0, 3)
#define cx_auto_add_pb_volume(codec, nid, str, idx) \
cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
@@ -4185,6 +4173,36 @@
return 0;
}
+/* Returns zero if this is a normal stereo channel, and non-zero if it should
+ be split in two independent channels.
+ dest_label must be at least 44 characters. */
+static int cx_auto_get_rightch_label(struct hda_codec *codec, const char *label,
+ char *dest_label, int nid)
+{
+ struct conexant_spec *spec = codec->spec;
+ int i;
+
+ if (!spec->fixup_stereo_dmic)
+ return 0;
+
+ for (i = 0; i < AUTO_CFG_MAX_INS; i++) {
+ int def_conf;
+ if (spec->autocfg.inputs[i].pin != nid)
+ continue;
+
+ if (spec->autocfg.inputs[i].type != AUTO_PIN_MIC)
+ return 0;
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT)
+ return 0;
+
+ /* Finally found the inverted internal mic! */
+ snprintf(dest_label, 44, "Inverted %s", label);
+ return 1;
+ }
+ return 0;
+}
+
static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
const char *label, const char *pfx,
int cidx)
@@ -4193,14 +4211,25 @@
int i;
for (i = 0; i < spec->num_adc_nids; i++) {
+ char rightch_label[44];
hda_nid_t adc_nid = spec->adc_nids[i];
int idx = get_input_connection(codec, adc_nid, nid);
if (idx < 0)
continue;
if (codec->single_adc_amp)
idx = 0;
+
+ if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) {
+ /* Make two independent kcontrols for left and right */
+ int err = cx_auto_add_volume_idx(codec, label, pfx,
+ cidx, adc_nid, HDA_INPUT, idx, 1);
+ if (err < 0)
+ return err;
+ return cx_auto_add_volume_idx(codec, rightch_label, pfx,
+ cidx, adc_nid, HDA_INPUT, idx, 2);
+ }
return cx_auto_add_volume_idx(codec, label, pfx,
- cidx, adc_nid, HDA_INPUT, idx);
+ cidx, adc_nid, HDA_INPUT, idx, 3);
}
return 0;
}
@@ -4213,9 +4242,19 @@
int i, con;
nid = spec->imux_info[idx].pin;
- if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
+ if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) {
+ char rightch_label[44];
+ if (cx_auto_get_rightch_label(codec, label, rightch_label, nid)) {
+ int err = cx_auto_add_volume_idx(codec, label, " Boost",
+ cidx, nid, HDA_INPUT, 0, 1);
+ if (err < 0)
+ return err;
+ return cx_auto_add_volume_idx(codec, rightch_label, " Boost",
+ cidx, nid, HDA_INPUT, 0, 2);
+ }
return cx_auto_add_volume(codec, label, " Boost", cidx,
nid, HDA_INPUT);
+ }
con = __select_input_connection(codec, spec->imux_info[idx].adc, nid,
&mux, false, 0);
if (con < 0)
@@ -4370,37 +4409,21 @@
/*
* pin fix-up
*/
-struct cxt_pincfg {
- hda_nid_t nid;
- u32 val;
-};
-
-static void apply_pincfg(struct hda_codec *codec, const struct cxt_pincfg *cfg)
-{
- for (; cfg->nid; cfg++)
- snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-
-}
-
-static void apply_pin_fixup(struct hda_codec *codec,
- const struct snd_pci_quirk *quirk,
- const struct cxt_pincfg **table)
-{
- quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
- if (quirk) {
- snd_printdd(KERN_INFO "hda_codec: applying pincfg for %s\n",
- quirk->name);
- apply_pincfg(codec, table[quirk->value]);
- }
-}
-
enum {
CXT_PINCFG_LENOVO_X200,
CXT_PINCFG_LENOVO_TP410,
+ CXT_FIXUP_STEREO_DMIC,
};
+static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct conexant_spec *spec = codec->spec;
+ spec->fixup_stereo_dmic = 1;
+}
+
/* ThinkPad X200 & co with cxt5051 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
+static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
{ 0x17, 0x21a11000 }, /* dock-mic */
{ 0x19, 0x2121103f }, /* dock-HP */
@@ -4409,16 +4432,26 @@
};
/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */
-static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = {
+static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = {
{ 0x19, 0x042110ff }, /* HP (seq# overridden) */
{ 0x1a, 0x21a190f0 }, /* dock-mic */
{ 0x1c, 0x212140ff }, /* dock-HP */
{}
};
-static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
- [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
- [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410,
+static const struct hda_fixup cxt_fixups[] = {
+ [CXT_PINCFG_LENOVO_X200] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = cxt_pincfg_lenovo_x200,
+ },
+ [CXT_PINCFG_LENOVO_TP410] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = cxt_pincfg_lenovo_tp410,
+ },
+ [CXT_FIXUP_STEREO_DMIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_stereo_dmic,
+ },
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -4432,6 +4465,7 @@
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
+ SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
{}
};
@@ -4471,13 +4505,16 @@
case 0x14f15051:
add_cx5051_fake_mutes(codec);
codec->pin_amp_workaround = 1;
- apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl);
+ snd_hda_pick_fixup(codec, NULL, cxt5051_fixups, cxt_fixups);
break;
default:
codec->pin_amp_workaround = 1;
- apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl);
+ snd_hda_pick_fixup(codec, NULL, cxt5066_fixups, cxt_fixups);
+ break;
}
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
/* Show mute-led control only on HP laptops
* This is a sort of white-list: on HP laptops, EAPD corresponds
* only to the mute-LED without actualy amp function. Meanwhile,
@@ -4556,6 +4593,12 @@
.patch = patch_conexant_auto },
{ .id = 0x14f150b9, .name = "CX20665",
.patch = patch_conexant_auto },
+ { .id = 0x14f1510f, .name = "CX20751/2",
+ .patch = patch_conexant_auto },
+ { .id = 0x14f15110, .name = "CX20751/2",
+ .patch = patch_conexant_auto },
+ { .id = 0x14f15111, .name = "CX20753/4",
+ .patch = patch_conexant_auto },
{} /* terminator */
};
@@ -4576,6 +4619,9 @@
MODULE_ALIAS("snd-hda-codec-id:14f150ac");
MODULE_ALIAS("snd-hda-codec-id:14f150b8");
MODULE_ALIAS("snd-hda-codec-id:14f150b9");
+MODULE_ALIAS("snd-hda-codec-id:14f1510f");
+MODULE_ALIAS("snd-hda-codec-id:14f15110");
+MODULE_ALIAS("snd-hda-codec-id:14f15111");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 83f345f..ad319d4d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1592,10 +1592,10 @@
unsigned int dataDCC2, channel_id;
int i;
struct hdmi_spec *spec = codec->spec;
- struct hda_spdif_out *spdif =
- snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
+ struct hda_spdif_out *spdif;
mutex_lock(&codec->spdif_mutex);
+ spdif = snd_hda_spdif_out_of_nid(codec, spec->cvts[0].cvt_nid);
chs = substream->runtime->channels;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7810913..1ce7fea 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -32,6 +32,7 @@
#include <sound/jack.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -66,8 +67,6 @@
unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */
};
-struct alc_fixup;
-
struct alc_multi_io {
hda_nid_t pin; /* multi-io widget pin NID */
hda_nid_t dac; /* DAC to be connected */
@@ -82,19 +81,33 @@
#define MAX_VOL_NIDS 0x40
+/* make compatible with old code */
+#define alc_apply_pincfgs snd_hda_apply_pincfgs
+#define alc_apply_fixup snd_hda_apply_fixup
+#define alc_pick_fixup snd_hda_pick_fixup
+#define alc_fixup hda_fixup
+#define alc_pincfg hda_pintbl
+#define alc_model_fixup hda_model_fixup
+
+#define ALC_FIXUP_PINS HDA_FIXUP_PINS
+#define ALC_FIXUP_VERBS HDA_FIXUP_VERBS
+#define ALC_FIXUP_FUNC HDA_FIXUP_FUNC
+
+#define ALC_FIXUP_ACT_PRE_PROBE HDA_FIXUP_ACT_PRE_PROBE
+#define ALC_FIXUP_ACT_PROBE HDA_FIXUP_ACT_PROBE
+#define ALC_FIXUP_ACT_INIT HDA_FIXUP_ACT_INIT
+#define ALC_FIXUP_ACT_BUILD HDA_FIXUP_ACT_BUILD
+
+
struct alc_spec {
+ struct hda_gen_spec gen;
+
/* codec parameterization */
const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
unsigned int num_mixers;
const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */
- const struct hda_verb *init_verbs[10]; /* initialization verbs
- * don't forget NULL
- * termination!
- */
- unsigned int num_init_verbs;
-
char stream_name_analog[32]; /* analog PCM stream */
const struct hda_pcm_stream *stream_analog_playback;
const struct hda_pcm_stream *stream_analog_capture;
@@ -210,11 +223,6 @@
unsigned int pll_coef_idx, pll_coef_bit;
unsigned int coef0;
- /* fix-up list */
- int fixup_id;
- const struct alc_fixup *fixup_list;
- const char *fixup_name;
-
/* multi-io */
int multi_ios;
struct alc_multi_io multi_io[4];
@@ -319,13 +327,16 @@
/* for shared I/O, change the pin-control accordingly */
if (spec->shared_mic_hp) {
+ unsigned int val;
+ hda_nid_t pin = spec->autocfg.inputs[1].pin;
/* NOTE: this assumes that there are only two inputs, the
* first is the real internal mic and the second is HP jack.
*/
- snd_hda_codec_write(codec, spec->autocfg.inputs[1].pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->cur_mux[adc_idx] ?
- PIN_VREF80 : PIN_HP);
+ if (spec->cur_mux[adc_idx])
+ val = snd_hda_get_default_vref(codec, pin) | PIN_IN;
+ else
+ val = PIN_HP;
+ snd_hda_set_pin_ctl(codec, pin, val);
spec->automute_speaker = !spec->cur_mux[adc_idx];
call_update_outputs(codec);
}
@@ -338,7 +349,7 @@
nid = get_capsrc(spec, adc_idx);
/* no selection? */
- num_conns = snd_hda_get_conn_list(codec, nid, NULL);
+ num_conns = snd_hda_get_num_conns(codec, nid);
if (num_conns <= 1)
return 1;
@@ -376,25 +387,9 @@
int auto_pin_type)
{
unsigned int val = PIN_IN;
-
- if (auto_pin_type == AUTO_PIN_MIC) {
- unsigned int pincap;
- unsigned int oldval;
- oldval = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- pincap = snd_hda_query_pin_caps(codec, nid);
- pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
- /* if the default pin setup is vref50, we give it priority */
- if ((pincap & AC_PINCAP_VREF_80) && oldval != PIN_VREF50)
- val = PIN_VREF80;
- else if (pincap & AC_PINCAP_VREF_50)
- val = PIN_VREF50;
- else if (pincap & AC_PINCAP_VREF_100)
- val = PIN_VREF100;
- else if (pincap & AC_PINCAP_VREF_GRD)
- val = PIN_VREFGRD;
- }
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ if (auto_pin_type == AUTO_PIN_MIC)
+ val |= snd_hda_get_default_vref(codec, nid);
+ snd_hda_set_pin_ctl(codec, nid, val);
}
/*
@@ -409,13 +404,6 @@
spec->mixers[spec->num_mixers++] = mix;
}
-static void add_verb(struct alc_spec *spec, const struct hda_verb *verb)
-{
- if (snd_BUG_ON(spec->num_init_verbs >= ARRAY_SIZE(spec->init_verbs)))
- return;
- spec->init_verbs[spec->num_init_verbs++] = verb;
-}
-
/*
* GPIO setup tables, used in initialization
*/
@@ -517,9 +505,7 @@
} else
val = 0;
val |= pin_bits;
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- val);
+ snd_hda_set_pin_ctl(codec, nid, val);
break;
case ALC_AUTOMUTE_AMP:
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -1200,6 +1186,16 @@
*/
#define ALC_FIXUP_SKU_IGNORE (2)
+static void alc_fixup_sku_ignore(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->cdefine.fixup = 1;
+ spec->cdefine.sku_cfg = ALC_FIXUP_SKU_IGNORE;
+ }
+}
+
static int alc_auto_parse_customize_define(struct hda_codec *codec)
{
unsigned int ass, tmp, i;
@@ -1403,178 +1399,6 @@
}
/*
- * Fix-up pin default configurations and add default verbs
- */
-
-struct alc_pincfg {
- hda_nid_t nid;
- u32 val;
-};
-
-struct alc_model_fixup {
- const int id;
- const char *name;
-};
-
-struct alc_fixup {
- int type;
- bool chained;
- int chain_id;
- union {
- unsigned int sku;
- const struct alc_pincfg *pins;
- const struct hda_verb *verbs;
- void (*func)(struct hda_codec *codec,
- const struct alc_fixup *fix,
- int action);
- } v;
-};
-
-enum {
- ALC_FIXUP_INVALID,
- ALC_FIXUP_SKU,
- ALC_FIXUP_PINS,
- ALC_FIXUP_VERBS,
- ALC_FIXUP_FUNC,
-};
-
-enum {
- ALC_FIXUP_ACT_PRE_PROBE,
- ALC_FIXUP_ACT_PROBE,
- ALC_FIXUP_ACT_INIT,
- ALC_FIXUP_ACT_BUILD,
-};
-
-static void alc_apply_pincfgs(struct hda_codec *codec,
- const struct alc_pincfg *cfg)
-{
- for (; cfg->nid; cfg++)
- snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-}
-
-static void alc_apply_fixup(struct hda_codec *codec, int action)
-{
- struct alc_spec *spec = codec->spec;
- int id = spec->fixup_id;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- const char *modelname = spec->fixup_name;
-#endif
- int depth = 0;
-
- if (!spec->fixup_list)
- return;
-
- while (id >= 0) {
- const struct alc_fixup *fix = spec->fixup_list + id;
- const struct alc_pincfg *cfg;
-
- switch (fix->type) {
- case ALC_FIXUP_SKU:
- if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply sku override for %s\n",
- codec->chip_name, modelname);
- spec->cdefine.sku_cfg = fix->v.sku;
- spec->cdefine.fixup = 1;
- break;
- case ALC_FIXUP_PINS:
- cfg = fix->v.pins;
- if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply pincfg for %s\n",
- codec->chip_name, modelname);
- alc_apply_pincfgs(codec, cfg);
- break;
- case ALC_FIXUP_VERBS:
- if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply fix-verbs for %s\n",
- codec->chip_name, modelname);
- add_verb(codec->spec, fix->v.verbs);
- break;
- case ALC_FIXUP_FUNC:
- if (!fix->v.func)
- break;
- snd_printdd(KERN_INFO "hda_codec: %s: "
- "Apply fix-func for %s\n",
- codec->chip_name, modelname);
- fix->v.func(codec, fix, action);
- break;
- default:
- snd_printk(KERN_ERR "hda_codec: %s: "
- "Invalid fixup type %d\n",
- codec->chip_name, fix->type);
- break;
- }
- if (!fix->chained)
- break;
- if (++depth > 10)
- break;
- id = fix->chain_id;
- }
-}
-
-static void alc_pick_fixup(struct hda_codec *codec,
- const struct alc_model_fixup *models,
- const struct snd_pci_quirk *quirk,
- const struct alc_fixup *fixlist)
-{
- struct alc_spec *spec = codec->spec;
- const struct snd_pci_quirk *q;
- int id = -1;
- const char *name = NULL;
-
- /* when model=nofixup is given, don't pick up any fixups */
- if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
- spec->fixup_list = NULL;
- spec->fixup_id = -1;
- return;
- }
-
- if (codec->modelname && models) {
- while (models->name) {
- if (!strcmp(codec->modelname, models->name)) {
- id = models->id;
- name = models->name;
- break;
- }
- models++;
- }
- }
- if (id < 0) {
- q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
- if (q) {
- id = q->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- name = q->name;
-#endif
- }
- }
- if (id < 0) {
- for (q = quirk; q->subvendor; q++) {
- unsigned int vendorid =
- q->subdevice | (q->subvendor << 16);
- if (vendorid == codec->subsystem_id) {
- id = q->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- name = q->name;
-#endif
- break;
- }
- }
- }
-
- spec->fixup_id = id;
- if (id >= 0) {
- spec->fixup_list = fixlist;
- spec->fixup_name = name;
- }
-}
-
-/*
* COEF access helper functions
*/
static int alc_read_coef_idx(struct hda_codec *codec,
@@ -1621,8 +1445,7 @@
pin = spec->autocfg.dig_out_pins[i];
if (!pin)
continue;
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ snd_hda_set_pin_ctl(codec, pin, PIN_OUT);
if (!i)
dac = spec->multiout.dig_out_nid;
else
@@ -1635,9 +1458,7 @@
}
pin = spec->autocfg.dig_in_pin;
if (pin)
- snd_hda_codec_write(codec, pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_IN);
+ snd_hda_set_pin_ctl(codec, pin, PIN_IN);
}
/* parse digital I/Os and set up NIDs in BIOS auto-parse mode */
@@ -2068,7 +1889,6 @@
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- unsigned int i;
if (spec->init_hook)
spec->init_hook(codec);
@@ -2076,8 +1896,6 @@
alc_fix_pll(codec);
alc_auto_init_amp(codec, spec->init_amp);
- for (i = 0; i < spec->num_init_verbs; i++)
- snd_hda_sequence_write(codec, spec->init_verbs[i]);
alc_init_special_input_src(codec);
alc_auto_init_std(codec);
@@ -2550,6 +2368,7 @@
{ 0x10ec0269, 0xffff, 0xa023, "ALC259" },
{ 0x10ec0269, 0xffff, 0x6023, "ALC281X" },
{ 0x10ec0269, 0x00f0, 0x0020, "ALC269VC" },
+ { 0x10ec0269, 0x00f0, 0x0030, "ALC269VD" },
{ 0x10ec0887, 0x00f0, 0x0030, "ALC887-VD" },
{ 0x10ec0888, 0x00f0, 0x0030, "ALC888-VD" },
{ 0x10ec0888, 0xf0f0, 0x3020, "ALC886" },
@@ -2725,7 +2544,6 @@
nid = codec->start_nid;
for (i = 0; i < codec->num_nodes; i++, nid++) {
hda_nid_t src;
- const hda_nid_t *list;
unsigned int caps = get_wcaps(codec, nid);
int type = get_wcaps_type(caps);
@@ -2743,13 +2561,14 @@
cap_nids[nums] = src;
break;
}
- n = snd_hda_get_conn_list(codec, src, &list);
+ n = snd_hda_get_num_conns(codec, src);
if (n > 1) {
cap_nids[nums] = src;
break;
} else if (n != 1)
break;
- src = *list;
+ if (snd_hda_get_connections(codec, src, &src, 1) != 1)
+ break;
}
if (++nums >= max_nums)
break;
@@ -2856,8 +2675,7 @@
static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
unsigned int pin_type)
{
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
+ snd_hda_set_pin_ctl(codec, nid, pin_type);
/* unmute pin */
if (nid_has_mute(codec, nid, HDA_OUTPUT))
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
@@ -2891,7 +2709,7 @@
/* mute all loopback inputs */
if (spec->mixer_nid) {
- int nums = snd_hda_get_conn_list(codec, spec->mixer_nid, NULL);
+ int nums = snd_hda_get_num_conns(codec, spec->mixer_nid);
for (i = 0; i < nums; i++)
snd_hda_codec_write(codec, spec->mixer_nid, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
@@ -3521,7 +3339,7 @@
if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) {
type = ALC_CTL_WIDGET_MUTE;
val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT);
- } else if (snd_hda_get_conn_list(codec, nid, NULL) == 1) {
+ } else if (snd_hda_get_num_conns(codec, nid) == 1) {
type = ALC_CTL_WIDGET_MUTE;
val = HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT);
} else {
@@ -3998,9 +3816,7 @@
snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
if (output) {
- snd_hda_codec_update_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
+ snd_hda_set_pin_ctl_cache(codec, nid, PIN_OUT);
if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, 0);
@@ -4009,9 +3825,8 @@
if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_update_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->multi_io[idx].ctl_in);
+ snd_hda_set_pin_ctl_cache(codec, nid,
+ spec->multi_io[idx].ctl_in);
}
return 0;
}
@@ -4084,7 +3899,7 @@
nums = 0;
for (n = 0; n < spec->num_adc_nids; n++) {
hda_nid_t cap = spec->private_capsrc_nids[n];
- int num_conns = snd_hda_get_conn_list(codec, cap, NULL);
+ int num_conns = snd_hda_get_num_conns(codec, cap);
for (i = 0; i < imux->num_items; i++) {
hda_nid_t pin = spec->imux_pins[i];
if (pin) {
@@ -4213,7 +4028,7 @@
if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
HDA_AMP_MUTE, 0);
- } else if (snd_hda_get_conn_list(codec, cap, NULL) > 1) {
+ } else if (snd_hda_get_num_conns(codec, cap) > 1) {
snd_hda_codec_write_cache(codec, cap, 0,
AC_VERB_SET_CONNECT_SEL, idx);
}
@@ -4427,6 +4242,25 @@
return 1;
}
+/* common preparation job for alc_spec */
+static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
+{
+ struct alc_spec *spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ int err;
+
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+ spec->mixer_nid = mixer_nid;
+
+ err = alc_codec_rename_from_preset(codec);
+ if (err < 0) {
+ kfree(spec);
+ return err;
+ }
+ return 0;
+}
+
static int alc880_parse_auto_config(struct hda_codec *codec)
{
static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
@@ -4808,13 +4642,11 @@
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
spec->need_dac_fix = 1;
alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl,
@@ -4890,7 +4722,7 @@
spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
spec->unsol_event = alc_sku_unsol_event;
- add_verb(codec->spec, alc_gpio1_init_verbs);
+ snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs);
}
}
@@ -5001,13 +4833,11 @@
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x07);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x07;
+ spec = codec->spec;
alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -5171,8 +5001,7 @@
val = snd_hda_codec_read(codec, nids[i], 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
val |= AC_PINCTL_VREF_80;
- snd_hda_codec_write(codec, nids[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, nids[i], val);
spec->keep_vref_in_automute = 1;
break;
}
@@ -5193,8 +5022,7 @@
val = snd_hda_codec_read(codec, nids[i], 0,
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
val |= AC_PINCTL_VREF_50;
- snd_hda_codec_write(codec, nids[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, nids[i], val);
}
spec->keep_vref_in_automute = 1;
}
@@ -5225,8 +5053,8 @@
}
},
[ALC882_FIXUP_ACER_ASPIRE_7736] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_sku_ignore,
},
[ALC882_FIXUP_ASUS_W90V] = {
.type = ALC_FIXUP_PINS,
@@ -5476,13 +5304,11 @@
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
switch (codec->vendor_id) {
case 0x10ec0882:
@@ -5494,10 +5320,6 @@
break;
}
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
-
alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
alc882_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -5621,13 +5443,11 @@
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
#if 0
/* pshou 07/11/05 set a zero PCM sample to DAC when FIFO is
@@ -5710,7 +5530,7 @@
if (err > 0) {
if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) {
add_mixer(spec, alc268_beep_mixer);
- add_verb(spec, alc268_beep_init_verbs);
+ snd_hda_gen_add_verbs(&spec->gen, alc268_beep_init_verbs);
}
}
return err;
@@ -5723,13 +5543,12 @@
struct alc_spec *spec;
int i, has_beep, err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
/* ALC268 has no aa-loopback mixer */
+ err = alc_alloc_spec(codec, 0);
+ if (err < 0)
+ return err;
+
+ spec = codec->spec;
/* automatic parse from the BIOS config */
err = alc268_parse_auto_config(codec);
@@ -5796,6 +5615,7 @@
ALC269_TYPE_ALC269VA,
ALC269_TYPE_ALC269VB,
ALC269_TYPE_ALC269VC,
+ ALC269_TYPE_ALC269VD,
};
/*
@@ -5807,8 +5627,21 @@
static const hda_nid_t alc269_ssids[] = { 0, 0x1b, 0x14, 0x21 };
static const hda_nid_t alc269va_ssids[] = { 0x15, 0x1b, 0x14, 0 };
struct alc_spec *spec = codec->spec;
- const hda_nid_t *ssids = spec->codec_variant == ALC269_TYPE_ALC269VA ?
- alc269va_ssids : alc269_ssids;
+ const hda_nid_t *ssids;
+
+ switch (spec->codec_variant) {
+ case ALC269_TYPE_ALC269VA:
+ case ALC269_TYPE_ALC269VC:
+ ssids = alc269va_ssids;
+ break;
+ case ALC269_TYPE_ALC269VB:
+ case ALC269_TYPE_ALC269VD:
+ ssids = alc269_ssids;
+ break;
+ default:
+ ssids = alc269_ssids;
+ break;
+ }
return alc_parse_auto_config(codec, alc269_ignore, ssids);
}
@@ -5825,6 +5658,11 @@
static void alc269_shutup(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->codec_variant != ALC269_TYPE_ALC269VB)
+ return;
+
if ((alc_get_coef0(codec) & 0x00ff) == 0x017)
alc269_toggle_power_output(codec, 0);
if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
@@ -5836,19 +5674,24 @@
#ifdef CONFIG_PM
static int alc269_resume(struct hda_codec *codec)
{
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018) {
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+ (alc_get_coef0(codec) & 0x00ff) == 0x018) {
alc269_toggle_power_output(codec, 0);
msleep(150);
}
codec->patch_ops.init(codec);
- if ((alc_get_coef0(codec) & 0x00ff) == 0x017) {
+ if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+ (alc_get_coef0(codec) & 0x00ff) == 0x017) {
alc269_toggle_power_output(codec, 1);
msleep(200);
}
- if ((alc_get_coef0(codec) & 0x00ff) == 0x018)
+ if (spec->codec_variant == ALC269_TYPE_ALC269VB ||
+ (alc_get_coef0(codec) & 0x00ff) == 0x018)
alc269_toggle_power_output(codec, 1);
snd_hda_codec_resume_amp(codec);
@@ -5946,9 +5789,7 @@
{
struct hda_codec *codec = private_data;
unsigned int pinval = enabled ? 0x20 : 0x24;
- snd_hda_codec_update_cache(codec, 0x19, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinval);
+ snd_hda_set_pin_ctl_cache(codec, 0x19, pinval);
}
static void alc269_fixup_mic2_mute(struct hda_codec *codec,
@@ -6015,8 +5856,8 @@
}
},
[ALC269_FIXUP_SKU_IGNORE] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_sku_ignore,
},
[ALC269_FIXUP_ASUS_G73JW] = {
.type = ALC_FIXUP_PINS,
@@ -6242,19 +6083,13 @@
static int patch_alc269(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err = 0;
+ int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
-
- err = alc_codec_rename_from_preset(codec);
+ err = alc_alloc_spec(codec, 0x0b);
if (err < 0)
- goto error;
+ return err;
+
+ spec = codec->spec;
if (codec->vendor_id == 0x10ec0269) {
spec->codec_variant = ALC269_TYPE_ALC269VA;
@@ -6271,6 +6106,9 @@
err = alc_codec_rename(codec, "ALC3202");
spec->codec_variant = ALC269_TYPE_ALC269VC;
break;
+ case 0x0030:
+ spec->codec_variant = ALC269_TYPE_ALC269VD;
+ break;
default:
alc_fix_pll_init(codec, 0x20, 0x04, 15);
}
@@ -6346,8 +6184,7 @@
if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)))
val |= AC_PINCTL_IN_EN;
val |= AC_PINCTL_VREF_50;
- snd_hda_codec_write(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ snd_hda_set_pin_ctl(codec, 0x0f, val);
spec->keep_vref_in_automute = 1;
}
@@ -6401,13 +6238,11 @@
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x15);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x15;
+ spec = codec->spec;
alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -6504,13 +6339,11 @@
struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
@@ -6522,7 +6355,7 @@
if (codec->vendor_id == 0x10ec0660) {
/* always turn on EAPD */
- add_verb(spec, alc660vd_eapd_verbs);
+ snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs);
}
if (!spec->no_analog) {
@@ -6635,8 +6468,8 @@
}
},
[ALC662_FIXUP_SKU_IGNORE] = {
- .type = ALC_FIXUP_SKU,
- .v.sku = ALC_FIXUP_SKU_IGNORE,
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_sku_ignore,
},
[ALC662_FIXUP_HP_RP5800] = {
.type = ALC_FIXUP_PINS,
@@ -6849,25 +6682,19 @@
static int patch_alc662(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err = 0;
+ int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (!spec)
- return -ENOMEM;
+ err = alc_alloc_spec(codec, 0x0b);
+ if (err < 0)
+ return err;
- codec->spec = spec;
-
- spec->mixer_nid = 0x0b;
+ spec = codec->spec;
/* handle multiple HPs as is */
spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
alc_fix_pll_init(codec, 0x20, 0x04, 15);
- err = alc_codec_rename_from_preset(codec);
- if (err < 0)
- goto error;
-
if ((alc_get_coef0(codec) & (1 << 14)) &&
codec->bus->pci->subsystem_vendor == 0x1025 &&
spec->cdefine.platform_type == 1) {
@@ -6930,16 +6757,12 @@
*/
static int patch_alc680(struct hda_codec *codec)
{
- struct alc_spec *spec;
int err;
- spec = kzalloc(sizeof(*spec), GFP_KERNEL);
- if (spec == NULL)
- return -ENOMEM;
-
- codec->spec = spec;
-
/* ALC680 has no aa-loopback mixer */
+ err = alc_alloc_spec(codec, 0);
+ if (err < 0)
+ return err;
/* automatic parse from the BIOS config */
err = alc680_parse_auto_config(codec);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4742cac..7db8228 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -36,6 +36,7 @@
#include <sound/tlv.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_beep.h"
#include "hda_jack.h"
@@ -221,6 +222,7 @@
unsigned char aloopback_shift;
/* power management */
+ unsigned int power_map_bits;
unsigned int num_pwrs;
const hda_nid_t *pwr_nids;
const hda_nid_t *dac_list;
@@ -314,6 +316,9 @@
struct hda_vmaster_mute_hook vmaster_mute;
};
+#define AC_VERB_IDT_SET_POWER_MAP 0x7ec
+#define AC_VERB_IDT_GET_POWER_MAP 0xfec
+
static const hda_nid_t stac9200_adc_nids[1] = {
0x03,
};
@@ -681,8 +686,7 @@
pinctl &= ~AC_PINCTL_VREFEN;
pinctl |= (new_vref & AC_PINCTL_VREFEN);
- error = snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+ error = snd_hda_set_pin_ctl_cache(codec, nid, pinctl);
if (error < 0)
return error;
@@ -706,8 +710,7 @@
else
pincfg |= AC_PINCTL_IN_EN;
- error = snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg);
+ error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg);
if (error < 0)
return error;
else
@@ -2505,27 +2508,10 @@
return 0;
}
-static unsigned int stac92xx_get_default_vref(struct hda_codec *codec,
- hda_nid_t nid)
-{
- unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
- pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
- if (pincap & AC_PINCAP_VREF_100)
- return AC_PINCTL_VREF_100;
- if (pincap & AC_PINCAP_VREF_80)
- return AC_PINCTL_VREF_80;
- if (pincap & AC_PINCAP_VREF_50)
- return AC_PINCTL_VREF_50;
- if (pincap & AC_PINCAP_VREF_GRD)
- return AC_PINCTL_VREF_GRD;
- return 0;
-}
-
static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
{
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_set_pin_ctl_cache(codec, nid, pin_type);
}
#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info
@@ -2594,7 +2580,7 @@
hda_nid_t nid = kcontrol->private_value;
unsigned int vref = stac92xx_vref_get(codec, nid);
- if (vref == stac92xx_get_default_vref(codec, nid))
+ if (vref == snd_hda_get_default_vref(codec, nid))
ucontrol->value.enumerated.item[0] = 0;
else if (vref == AC_PINCTL_VREF_GRD)
ucontrol->value.enumerated.item[0] = 1;
@@ -2613,7 +2599,7 @@
hda_nid_t nid = kcontrol->private_value;
if (ucontrol->value.enumerated.item[0] == 0)
- new_vref = stac92xx_get_default_vref(codec, nid);
+ new_vref = snd_hda_get_default_vref(codec, nid);
else if (ucontrol->value.enumerated.item[0] == 1)
new_vref = AC_PINCTL_VREF_GRD;
else if (ucontrol->value.enumerated.item[0] == 2)
@@ -2679,7 +2665,7 @@
else {
unsigned int pinctl = AC_PINCTL_IN_EN;
if (io_idx) /* set VREF for mic */
- pinctl |= stac92xx_get_default_vref(codec, nid);
+ pinctl |= snd_hda_get_default_vref(codec, nid);
stac92xx_auto_set_pinctl(codec, nid, pinctl);
}
@@ -2847,7 +2833,7 @@
char name[22];
if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) {
- if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
+ if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
&& nid == spec->line_switch)
control = STAC_CTL_WIDGET_IO_SWITCH;
else if (snd_hda_query_pin_caps(codec, nid)
@@ -4250,13 +4236,6 @@
val = snd_hda_get_bool_hint(codec, "eapd_switch");
if (val >= 0)
spec->eapd_switch = val;
- get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
- if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
- spec->gpio_mask |= spec->gpio_led;
- spec->gpio_dir |= spec->gpio_led;
- if (spec->gpio_led_polarity)
- spec->gpio_data |= spec->gpio_led;
- }
}
static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins,
@@ -4354,7 +4333,7 @@
unsigned int pinctl, conf;
if (type == AUTO_PIN_MIC) {
/* for mic pins, force to initialize */
- pinctl = stac92xx_get_default_vref(codec, nid);
+ pinctl = snd_hda_get_default_vref(codec, nid);
pinctl |= AC_PINCTL_IN_EN;
stac92xx_auto_set_pinctl(codec, nid, pinctl);
} else {
@@ -4390,10 +4369,18 @@
hda_nid_t nid = spec->pwr_nids[i];
int pinctl, def_conf;
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ def_conf = get_defcfg_connect(def_conf);
+ if (def_conf == AC_JACK_PORT_NONE) {
+ /* power off unused ports */
+ stac_toggle_power_map(codec, nid, 0);
+ continue;
+ }
/* power on when no jack detection is available */
/* or when the VREF is used for controlling LED */
if (!spec->hp_detect ||
- spec->vref_mute_led_nid == nid) {
+ spec->vref_mute_led_nid == nid ||
+ !is_jack_detectable(codec, nid)) {
stac_toggle_power_map(codec, nid, 1);
continue;
}
@@ -4411,15 +4398,6 @@
stac_toggle_power_map(codec, nid, 1);
continue;
}
- def_conf = snd_hda_codec_get_pincfg(codec, nid);
- def_conf = get_defcfg_connect(def_conf);
- /* skip any ports that don't have jacks since presence
- * detection is useless */
- if (def_conf != AC_JACK_PORT_COMPLEX) {
- if (def_conf != AC_JACK_PORT_NONE)
- stac_toggle_power_map(codec, nid, 1);
- continue;
- }
if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
stac_issue_unsol_event(codec, nid);
continue;
@@ -4432,6 +4410,12 @@
/* sync mute LED */
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+
+ /* sync the power-map */
+ if (spec->num_pwrs)
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_IDT_SET_POWER_MAP,
+ spec->power_map_bits);
if (spec->dac_list)
stac92xx_power_down(codec);
return 0;
@@ -4460,8 +4444,7 @@
struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
def_conf = snd_hda_codec_get_pincfg(codec, pin->nid);
if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)
- snd_hda_codec_write(codec, pin->nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+ snd_hda_set_pin_ctl(codec, pin->nid, 0);
}
}
@@ -4517,9 +4500,7 @@
pin_ctl |= flag;
if (old_ctl != pin_ctl)
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_ctl);
+ snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl);
}
static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -4528,9 +4509,7 @@
unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
if (pin_ctl & flag)
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_ctl & ~flag);
+ snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag);
}
static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
@@ -4682,14 +4661,18 @@
idx = 1 << idx;
- val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0xff;
+ val = spec->power_map_bits;
if (enable)
val &= ~idx;
else
val |= idx;
/* power down unused output ports */
- snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val);
+ if (val != spec->power_map_bits) {
+ spec->power_map_bits = val;
+ snd_hda_codec_write(codec, codec->afg, 0,
+ AC_VERB_IDT_SET_POWER_MAP, val);
+ }
}
static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid)
@@ -4866,6 +4849,11 @@
struct sigmatel_spec *spec = codec->spec;
const struct dmi_device *dev = NULL;
+ if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
+ get_int_hint(codec, "gpio_led_polarity",
+ &spec->gpio_led_polarity);
+ return 1;
+ }
if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) {
while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
NULL, dev))) {
@@ -4952,7 +4940,8 @@
{
if (nid == codec->afg)
snd_iprintf(buffer, "Power-Map: 0x%02x\n",
- snd_hda_codec_read(codec, nid, 0, 0x0fec, 0x0));
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_IDT_GET_POWER_MAP, 0));
}
static void analog_loop_proc_hook(struct snd_info_buffer *buffer,
@@ -5009,20 +4998,6 @@
return 0;
}
-static int stac92xx_pre_resume(struct hda_codec *codec)
-{
- struct sigmatel_spec *spec = codec->spec;
-
- /* sync mute LED */
- if (spec->vref_mute_led_nid)
- stac_vrefout_set(codec, spec->vref_mute_led_nid,
- spec->vref_led);
- else if (spec->gpio_led)
- stac_gpio_set(codec, spec->gpio_mask,
- spec->gpio_dir, spec->gpio_data);
- return 0;
-}
-
static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
unsigned int power_state)
{
@@ -5046,7 +5021,6 @@
#else
#define stac92xx_suspend NULL
#define stac92xx_resume NULL
-#define stac92xx_pre_resume NULL
#define stac92xx_set_power_state NULL
#endif /* CONFIG_PM */
@@ -5592,9 +5566,6 @@
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
-#ifdef CONFIG_PM
- codec->patch_ops.pre_resume = stac92xx_pre_resume;
-#endif
}
err = stac92xx_parse_auto_config(codec);
@@ -5901,9 +5872,6 @@
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
-#ifdef CONFIG_PM
- codec->patch_ops.pre_resume = stac92xx_pre_resume;
-#endif
}
spec->multiout.dac_nids = spec->dac_nids;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 06214fd..82b3680 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -54,6 +54,7 @@
#include <sound/asoundef.h>
#include "hda_codec.h"
#include "hda_local.h"
+#include "hda_auto_parser.h"
#include "hda_jack.h"
/* Pin Widget NID */
@@ -484,7 +485,7 @@
if (!path)
return;
- num = snd_hda_get_conn_list(codec, mix_nid, NULL);
+ num = snd_hda_get_num_conns(codec, mix_nid);
for (i = 0; i < num; i++) {
if (i == idx)
val = AMP_IN_UNMUTE(i);
@@ -532,8 +533,7 @@
{
if (!pin)
return;
- snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_type);
+ snd_hda_set_pin_ctl(codec, pin, pin_type);
if (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_EAPD)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
@@ -662,12 +662,12 @@
hda_nid_t nid = cfg->inputs[i].pin;
if (spec->smart51_enabled && is_smart51_pins(codec, nid))
ctl = PIN_OUT;
- else if (cfg->inputs[i].type == AUTO_PIN_MIC)
- ctl = PIN_VREF50;
- else
+ else {
ctl = PIN_IN;
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+ if (cfg->inputs[i].type == AUTO_PIN_MIC)
+ ctl |= snd_hda_get_default_vref(codec, nid);
+ }
+ snd_hda_set_pin_ctl(codec, nid, ctl);
}
/* init input-src */
@@ -1006,9 +1006,7 @@
AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
parm |= out_in;
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- parm);
+ snd_hda_set_pin_ctl(codec, nid, parm);
if (out_in == AC_PINCTL_OUT_EN) {
mute_aa_path(codec, 1);
notify_aa_path_ctls(codec);
@@ -1647,8 +1645,7 @@
parm &= ~AC_PINCTL_OUT_EN;
else
parm |= AC_PINCTL_OUT_EN;
- snd_hda_codec_write(codec, pins[i], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, parm);
+ snd_hda_set_pin_ctl(codec, pins[i], parm);
}
}
@@ -1709,8 +1706,7 @@
if (gpio_data == 0x02) {
/* unmute line out */
- snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
+ snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0],
PIN_OUT);
if (vol_counter & 0x20) {
/* decrease volume */
@@ -1728,9 +1724,7 @@
}
} else if (!(gpio_data & 0x02)) {
/* mute line out */
- snd_hda_codec_write(codec, spec->autocfg.line_out_pins[0], 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- 0);
+ snd_hda_set_pin_ctl(codec, spec->autocfg.line_out_pins[0], 0);
}
}
@@ -2757,8 +2751,7 @@
struct via_spec *spec = codec->spec;
if (!spec->dig_in_nid)
return;
- snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
+ snd_hda_set_pin_ctl(codec, spec->autocfg.dig_in_pin, PIN_IN);
}
/* initialize the unsolicited events */
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 132a86e..5be2e12 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2803,22 +2803,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ice1712_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ice1712_ids,
.probe = snd_ice1712_probe,
.remove = __devexit_p(snd_ice1712_remove),
};
-static int __init alsa_card_ice1712_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ice1712_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ice1712_init)
-module_exit(alsa_card_ice1712_exit)
+module_pci_driver(ice1712_driver);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 812d10e..a01a00d 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2873,7 +2873,7 @@
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver vt1724_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_vt1724_ids,
.probe = snd_vt1724_probe,
@@ -2884,15 +2884,4 @@
#endif
};
-static int __init alsa_card_ice1724_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ice1724_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ice1724_init)
-module_exit(alsa_card_ice1724_exit)
+module_pci_driver(vt1724_driver);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index e0a4263..f4e2dd4 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3338,7 +3338,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver intel8x0_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_intel8x0_ids,
.probe = snd_intel8x0_probe,
@@ -3349,16 +3349,4 @@
#endif
};
-
-static int __init alsa_card_intel8x0_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_intel8x0_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_intel8x0_init)
-module_exit(alsa_card_intel8x0_exit)
+module_pci_driver(intel8x0_driver);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index d689913..fc27a6a 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1324,7 +1324,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver intel8x0m_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_intel8x0m_ids,
.probe = snd_intel8x0m_probe,
@@ -1335,16 +1335,4 @@
#endif
};
-
-static int __init alsa_card_intel8x0m_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_intel8x0m_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_intel8x0m_init)
-module_exit(alsa_card_intel8x0m_exit)
+module_pci_driver(intel8x0m_driver);
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 8fea45a..e69ce5f 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2476,22 +2476,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver korg1212_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_korg1212_ids,
.probe = snd_korg1212_probe,
.remove = __devexit_p(snd_korg1212_remove),
};
-static int __init alsa_card_korg1212_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_korg1212_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_korg1212_init)
-module_exit(alsa_card_korg1212_exit)
+module_pci_driver(korg1212_driver);
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 3759827..ac15166 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -770,22 +770,11 @@
MODULE_DEVICE_TABLE(pci, lola_ids);
/* pci_driver definition */
-static struct pci_driver driver = {
+static struct pci_driver lola_driver = {
.name = KBUILD_MODNAME,
.id_table = lola_ids,
.probe = lola_probe,
.remove = __devexit_p(lola_remove),
};
-static int __init alsa_card_lola_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_lola_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_lola_init)
-module_exit(alsa_card_lola_exit)
+module_pci_driver(lola_driver);
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d94c0c2..d1ab4370 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -1141,24 +1141,11 @@
}
-static struct pci_driver driver = {
+static struct pci_driver lx6464es_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_lx6464es_ids,
.probe = snd_lx6464es_probe,
.remove = __devexit_p(snd_lx6464es_remove),
};
-
-/* module initialization */
-static int __init mod_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit mod_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(mod_init);
-module_exit(mod_exit);
+module_pci_driver(lx6464es_driver);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 78229b0..deef213 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2837,7 +2837,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver m3_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_m3_ids,
.probe = snd_m3_probe,
@@ -2848,15 +2848,4 @@
#endif
};
-static int __init alsa_card_m3_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_m3_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_m3_init)
-module_exit(alsa_card_m3_exit)
+module_pci_driver(m3_driver);
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 487837c..0762610 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1380,22 +1380,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver mixart_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_mixart_ids,
.probe = snd_mixart_probe,
.remove = __devexit_p(snd_mixart_remove),
};
-static int __init alsa_card_mixart_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_mixart_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_mixart_init)
-module_exit(alsa_card_mixart_exit)
+module_pci_driver(mixart_driver);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index ade2c64..8159b05 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1742,7 +1742,7 @@
}
-static struct pci_driver driver = {
+static struct pci_driver nm256_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_nm256_ids,
.probe = snd_nm256_probe,
@@ -1753,16 +1753,4 @@
#endif
};
-
-static int __init alsa_card_nm256_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_nm256_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_nm256_init)
-module_exit(alsa_card_nm256_exit)
+module_pci_driver(nm256_driver);
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index eab663e..610275b 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -94,6 +94,7 @@
MODEL_2CH_OUTPUT,
MODEL_HG2PCI,
MODEL_XONAR_DG,
+ MODEL_XONAR_DGX,
};
static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = {
@@ -109,6 +110,8 @@
{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
/* Asus Xonar DG */
{ OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG },
+ /* Asus Xonar DGX */
+ { OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX },
/* PCI 2.0 HD Audio */
{ OXYGEN_PCI_SUBID(0x13f6, 0x8782), .driver_data = MODEL_2CH_OUTPUT },
/* Kuroutoshikou CMI8787-HG2PCI */
@@ -827,6 +830,11 @@
break;
case MODEL_XONAR_DG:
chip->model = model_xonar_dg;
+ chip->model.shortname = "Xonar DG";
+ break;
+ case MODEL_XONAR_DGX:
+ chip->model = model_xonar_dg;
+ chip->model.shortname = "Xonar DGX";
break;
}
if (id->driver_data == MODEL_MERIDIAN ||
@@ -870,15 +878,4 @@
#endif
};
-static int __init alsa_card_oxygen_init(void)
-{
- return pci_register_driver(&oxygen_driver);
-}
-
-static void __exit alsa_card_oxygen_exit(void)
-{
- pci_unregister_driver(&oxygen_driver);
-}
-
-module_init(alsa_card_oxygen_init)
-module_exit(alsa_card_oxygen_exit)
+module_pci_driver(oxygen_driver);
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 3fdee49..19962c6 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -100,15 +100,4 @@
.shutdown = oxygen_pci_shutdown,
};
-static int __init alsa_card_xonar_init(void)
-{
- return pci_register_driver(&xonar_driver);
-}
-
-static void __exit alsa_card_xonar_exit(void)
-{
- pci_unregister_driver(&xonar_driver);
-}
-
-module_init(alsa_card_xonar_init)
-module_exit(alsa_card_xonar_exit)
+module_pci_driver(xonar_driver);
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index 793bdf0..77acd79 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -1,5 +1,5 @@
/*
- * card driver for the Xonar DG
+ * card driver for the Xonar DG/DGX
*
* Copyright (c) Clemens Ladisch <clemens@ladisch.de>
*
@@ -17,8 +17,8 @@
*/
/*
- * Xonar DG
- * --------
+ * Xonar DG/DGX
+ * ------------
*
* CMI8788:
*
@@ -581,7 +581,6 @@
}
struct oxygen_model model_xonar_dg = {
- .shortname = "Xonar DG",
.longname = "C-Media Oxygen HD Audio",
.chip = "CMI8786",
.init = dg_init,
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index fd1809a..0435f45 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1607,22 +1607,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver pcxhr_driver = {
.name = KBUILD_MODNAME,
.id_table = pcxhr_ids,
.probe = pcxhr_probe,
.remove = __devexit_p(pcxhr_remove),
};
-static int __init pcxhr_module_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit pcxhr_module_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(pcxhr_module_init)
-module_exit(pcxhr_module_exit)
+module_pci_driver(pcxhr_driver);
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 0481d94..cbeb3f7 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1837,8 +1837,7 @@
}
if (chip->irq >= 0)
free_irq(chip->irq, chip);
- if (chip->fw_entry)
- release_firmware(chip->fw_entry);
+ release_firmware(chip->fw_entry);
release_and_free_resource(chip->res_port);
kfree(chip);
return 0;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index b4819d5..46b3629 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1984,22 +1984,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver rme32_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_rme32_ids,
.probe = snd_rme32_probe,
.remove = __devexit_p(snd_rme32_remove),
};
-static int __init alsa_card_rme32_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_rme32_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_rme32_init)
-module_exit(alsa_card_rme32_exit)
+module_pci_driver(rme32_driver);
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index ba89415..9b98dc4 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -2395,22 +2395,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver rme96_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_rme96_ids,
.probe = snd_rme96_probe,
.remove = __devexit_p(snd_rme96_remove),
};
-static int __init alsa_card_rme96_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_rme96_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_rme96_init)
-module_exit(alsa_card_rme96_exit)
+module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 0b2aea2..0d6930c 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5636,22 +5636,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver hdsp_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_hdsp_ids,
.probe = snd_hdsp_probe,
.remove = __devexit_p(snd_hdsp_remove),
};
-static int __init alsa_card_hdsp_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hdsp_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hdsp_init)
-module_exit(alsa_card_hdsp_exit)
+module_pci_driver(hdsp_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bc030a2..b8ac871 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1988,6 +1988,13 @@
period = hdspm_read(hdspm, HDSPM_RD_PLL_FREQ);
rate = hdspm_calc_dds_value(hdspm, period);
+ if (rate > 207000) {
+ /* Unreasonable high sample rate as seen on PCI MADI cards.
+ * Use the cached value instead.
+ */
+ rate = hdspm->system_sample_rate;
+ }
+
return rate;
}
@@ -6918,23 +6925,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver hdspm_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_hdspm_ids,
.probe = snd_hdspm_probe,
.remove = __devexit_p(snd_hdspm_remove),
};
-
-static int __init alsa_card_hdspm_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hdspm_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hdspm_init)
-module_exit(alsa_card_hdspm_exit)
+module_pci_driver(hdspm_driver);
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index b737d16..a15fc10 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -2631,22 +2631,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver rme9652_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_rme9652_ids,
.probe = snd_rme9652_probe,
.remove = __devexit_p(snd_rme9652_remove),
};
-static int __init alsa_card_hammerfall_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_hammerfall_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_hammerfall_init)
-module_exit(alsa_card_hammerfall_exit)
+module_pci_driver(rme9652_driver);
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index ff500a8..1552642 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1488,15 +1488,4 @@
#endif
};
-static int __init sis7019_init(void)
-{
- return pci_register_driver(&sis7019_driver);
-}
-
-static void __exit sis7019_exit(void)
-{
- pci_unregister_driver(&sis7019_driver);
-}
-
-module_init(sis7019_init);
-module_exit(sis7019_exit);
+module_pci_driver(sis7019_driver);
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 54cc802..baa9946b 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1530,22 +1530,11 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver sonicvibes_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_sonic_ids,
.probe = snd_sonic_probe,
.remove = __devexit_p(snd_sonic_remove),
};
-static int __init alsa_card_sonicvibes_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_sonicvibes_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_sonicvibes_init)
-module_exit(alsa_card_sonicvibes_exit)
+module_pci_driver(sonicvibes_driver);
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 5f1def7..611983e 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -172,7 +172,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver trident_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_trident_ids,
.probe = snd_trident_probe,
@@ -183,15 +183,4 @@
#endif
};
-static int __init alsa_card_trident_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_trident_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_trident_init)
-module_exit(alsa_card_trident_exit)
+module_pci_driver(trident_driver);
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 7563040..b5afab4 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2619,7 +2619,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver via82xx_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_via82xx_ids,
.probe = snd_via82xx_probe,
@@ -2630,15 +2630,4 @@
#endif
};
-static int __init alsa_card_via82xx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_via82xx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_via82xx_init)
-module_exit(alsa_card_via82xx_exit)
+module_pci_driver(via82xx_driver);
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5efcbca..59fd47e 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1223,7 +1223,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver via82xx_modem_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_via82xx_modem_ids,
.probe = snd_via82xx_probe,
@@ -1234,15 +1234,4 @@
#endif
};
-static int __init alsa_card_via82xx_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_via82xx_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_via82xx_init)
-module_exit(alsa_card_via82xx_exit)
+module_pci_driver(via82xx_modem_driver);
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 6a534bf..1ea1f65 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -289,7 +289,7 @@
}
#endif
-static struct pci_driver driver = {
+static struct pci_driver vx222_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_vx222_ids,
.probe = snd_vx222_probe,
@@ -300,15 +300,4 @@
#endif
};
-static int __init alsa_card_vx222_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_vx222_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_vx222_init)
-module_exit(alsa_card_vx222_exit)
+module_pci_driver(vx222_driver);
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 94ab728..9a1d01d 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -350,7 +350,7 @@
pci_set_drvdata(pci, NULL);
}
-static struct pci_driver driver = {
+static struct pci_driver ymfpci_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_ymfpci_ids,
.probe = snd_card_ymfpci_probe,
@@ -361,15 +361,4 @@
#endif
};
-static int __init alsa_card_ymfpci_init(void)
-{
- return pci_register_driver(&driver);
-}
-
-static void __exit alsa_card_ymfpci_exit(void)
-{
- pci_unregister_driver(&driver);
-}
-
-module_init(alsa_card_ymfpci_init)
-module_exit(alsa_card_ymfpci_exit)
+module_pci_driver(ymfpci_driver);
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index b11f82b..f8b01c7 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -433,7 +433,7 @@
/*
* "driver" definition
*/
-static struct platform_driver driver = {
+static struct platform_driver sh_dac_driver = {
.probe = snd_sh_dac_probe,
.remove = snd_sh_dac_remove,
.driver = {
@@ -441,4 +441,4 @@
},
};
-module_platform_driver(driver);
+module_platform_driver(sh_dac_driver);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 5a537b7..e0d45fd 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -565,22 +565,22 @@
attn_tlv),
SOC_SINGLE_TLV("SPK-IP Mono Volume",
- CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("SPK-XSP Mono Volume",
- CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("SPK-ASP Mono Volume",
- CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("SPK-VSP Mono Volume",
- CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-IP Mono Volume",
- CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-XSP Mono Volume",
- CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-ASP Mono Volume",
- CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv),
SOC_SINGLE_TLV("ESL-VSP Mono Volume",
- CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv),
+ CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv),
SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum),
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 2f9870a..993639d 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1127,7 +1127,7 @@
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
WM8994_AIF2DACL_ENA |
WM8994_AIF2DACR_ENA, 0);
- snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
WM8994_AIF2ADCL_ENA |
WM8994_AIF2ADCR_ENA, 0);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 9ccfa5e..57a2fa7 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -109,11 +109,12 @@
- PandaBoard (4430)
- PandaBoardES (4460)
-config SND_OMAP_SOC_OMAP4_HDMI
- tristate "SoC Audio support for Texas Instruments OMAP4 HDMI"
- depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4
+config SND_OMAP_SOC_OMAP_HDMI
+ tristate "SoC Audio support for Texas Instruments OMAP HDMI"
+ depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS
select SND_OMAP_SOC_HDMI
select SND_SOC_OMAP_HDMI_CODEC
+ select OMAP4_DSS_HDMI_AUDIO
help
Say Y if you want to add support for SoC HDMI audio on Texas Instruments
OMAP4 chips
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 1d656bc..0e14dd3 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@
snd-soc-omap3beagle-objs := omap3beagle.o
snd-soc-zoom2-objs := zoom2.o
snd-soc-igep0020-objs := igep0020.o
-snd-soc-omap4-hdmi-objs := omap4-hdmi-card.o
+snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o
obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
@@ -41,4 +41,4 @@
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP4_HDMI) += snd-soc-omap4-hdmi.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index e5f4444..34835e8 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -109,6 +109,47 @@
dev_dbg(mcbsp->dev, "***********************\n");
}
+static irqreturn_t omap_mcbsp_irq_handler(int irq, void *dev_id)
+{
+ struct omap_mcbsp *mcbsp = dev_id;
+ u16 irqst;
+
+ irqst = MCBSP_READ(mcbsp, IRQST);
+ dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst);
+
+ if (irqst & RSYNCERREN)
+ dev_err(mcbsp->dev, "RX Frame Sync Error!\n");
+ if (irqst & RFSREN)
+ dev_dbg(mcbsp->dev, "RX Frame Sync\n");
+ if (irqst & REOFEN)
+ dev_dbg(mcbsp->dev, "RX End Of Frame\n");
+ if (irqst & RRDYEN)
+ dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n");
+ if (irqst & RUNDFLEN)
+ dev_err(mcbsp->dev, "RX Buffer Underflow!\n");
+ if (irqst & ROVFLEN)
+ dev_err(mcbsp->dev, "RX Buffer Overflow!\n");
+
+ if (irqst & XSYNCERREN)
+ dev_err(mcbsp->dev, "TX Frame Sync Error!\n");
+ if (irqst & XFSXEN)
+ dev_dbg(mcbsp->dev, "TX Frame Sync\n");
+ if (irqst & XEOFEN)
+ dev_dbg(mcbsp->dev, "TX End Of Frame\n");
+ if (irqst & XRDYEN)
+ dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n");
+ if (irqst & XUNDFLEN)
+ dev_err(mcbsp->dev, "TX Buffer Underflow!\n");
+ if (irqst & XOVFLEN)
+ dev_err(mcbsp->dev, "TX Buffer Overflow!\n");
+ if (irqst & XEMPTYEOFEN)
+ dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n");
+
+ MCBSP_WRITE(mcbsp, IRQST, irqst);
+
+ return IRQ_HANDLED;
+}
+
static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id)
{
struct omap_mcbsp *mcbsp_tx = dev_id;
@@ -176,6 +217,10 @@
/* Enable wakeup behavior */
if (mcbsp->pdata->has_wakeup)
MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN);
+
+ /* Enable TX/RX sync error interrupts by default */
+ if (mcbsp->irq)
+ MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN);
}
/**
@@ -489,23 +534,25 @@
MCBSP_WRITE(mcbsp, SPCR1, 0);
MCBSP_WRITE(mcbsp, SPCR2, 0);
- err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler,
- 0, "McBSP", (void *)mcbsp);
- if (err != 0) {
- dev_err(mcbsp->dev, "Unable to request TX IRQ %d "
- "for McBSP%d\n", mcbsp->tx_irq,
- mcbsp->id);
- goto err_clk_disable;
- }
-
- if (mcbsp->rx_irq) {
- err = request_irq(mcbsp->rx_irq,
- omap_mcbsp_rx_irq_handler,
- 0, "McBSP", (void *)mcbsp);
+ if (mcbsp->irq) {
+ err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0,
+ "McBSP", (void *)mcbsp);
if (err != 0) {
- dev_err(mcbsp->dev, "Unable to request RX IRQ %d "
- "for McBSP%d\n", mcbsp->rx_irq,
- mcbsp->id);
+ dev_err(mcbsp->dev, "Unable to request IRQ\n");
+ goto err_clk_disable;
+ }
+ } else {
+ err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0,
+ "McBSP TX", (void *)mcbsp);
+ if (err != 0) {
+ dev_err(mcbsp->dev, "Unable to request TX IRQ\n");
+ goto err_clk_disable;
+ }
+
+ err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0,
+ "McBSP RX", (void *)mcbsp);
+ if (err != 0) {
+ dev_err(mcbsp->dev, "Unable to request RX IRQ\n");
goto err_free_irq;
}
}
@@ -542,9 +589,16 @@
if (mcbsp->pdata->has_wakeup)
MCBSP_WRITE(mcbsp, WAKEUPEN, 0);
- if (mcbsp->rx_irq)
+ /* Disable interrupt requests */
+ if (mcbsp->irq)
+ MCBSP_WRITE(mcbsp, IRQEN, 0);
+
+ if (mcbsp->irq) {
+ free_irq(mcbsp->irq, (void *)mcbsp);
+ } else {
free_irq(mcbsp->rx_irq, (void *)mcbsp);
- free_irq(mcbsp->tx_irq, (void *)mcbsp);
+ free_irq(mcbsp->tx_irq, (void *)mcbsp);
+ }
reg_cache = mcbsp->reg_cache;
@@ -754,7 +808,7 @@
THRESHOLD_PROP_BUILDER(max_rx_thres);
static const char *dma_op_modes[] = {
- "element", "threshold", "frame",
+ "element", "threshold",
};
static ssize_t dma_op_mode_show(struct device *dev,
@@ -949,13 +1003,24 @@
else
mcbsp->phys_dma_base = res->start;
- mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx");
- mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx");
+ /*
+ * OMAP1, 2 uses two interrupt lines: TX, RX
+ * OMAP2430, OMAP3 SoC have combined IRQ line as well.
+ * OMAP4 and newer SoC only have the combined IRQ line.
+ * Use the combined IRQ if available since it gives better debugging
+ * possibilities.
+ */
+ mcbsp->irq = platform_get_irq_byname(pdev, "common");
+ if (mcbsp->irq == -ENXIO) {
+ mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx");
- /* From OMAP4 there will be a single irq line */
- if (mcbsp->tx_irq == -ENXIO) {
- mcbsp->tx_irq = platform_get_irq(pdev, 0);
- mcbsp->rx_irq = 0;
+ if (mcbsp->tx_irq == -ENXIO) {
+ mcbsp->irq = platform_get_irq(pdev, 0);
+ mcbsp->tx_irq = 0;
+ } else {
+ mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx");
+ mcbsp->irq = 0;
+ }
}
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx");
diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h
index a944fcc..262a615 100644
--- a/sound/soc/omap/mcbsp.h
+++ b/sound/soc/omap/mcbsp.h
@@ -217,17 +217,20 @@
/********************** McBSP DMA operating modes **************************/
#define MCBSP_DMA_MODE_ELEMENT 0
#define MCBSP_DMA_MODE_THRESHOLD 1
-#define MCBSP_DMA_MODE_FRAME 2
-/********************** McBSP WAKEUPEN bit definitions *********************/
+/********************** McBSP WAKEUPEN/IRQST/IRQEN bit definitions *********/
#define RSYNCERREN BIT(0)
#define RFSREN BIT(1)
#define REOFEN BIT(2)
#define RRDYEN BIT(3)
+#define RUNDFLEN BIT(4)
+#define ROVFLEN BIT(5)
#define XSYNCERREN BIT(7)
#define XFSXEN BIT(8)
#define XEOFEN BIT(9)
#define XRDYEN BIT(10)
+#define XUNDFLEN BIT(11)
+#define XOVFLEN BIT(12)
#define XEMPTYEOFEN BIT(14)
/* Clock signal muxing options */
@@ -295,6 +298,7 @@
int configured;
u8 free;
+ int irq;
int rx_irq;
int tx_irq;
diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
index 93bb8ee..9d93793 100644
--- a/sound/soc/omap/omap-abe-twl6040.c
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -40,6 +40,11 @@
#include "omap-pcm.h"
#include "../codecs/twl6040.h"
+struct abe_twl6040 {
+ int jack_detection; /* board can detect jack events */
+ int mclk_freq; /* MCLK frequency speed for twl6040 */
+};
+
static int omap_abe_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -47,13 +52,13 @@
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = codec->card;
- struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+ struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int clk_id, freq;
int ret;
clk_id = twl6040_get_clk_id(rtd->codec);
if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
- freq = pdata->mclk_freq;
+ freq = priv->mclk_freq;
else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
freq = 32768;
else
@@ -128,6 +133,9 @@
SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
+
+ /* Digital microphones */
+ SND_SOC_DAPM_MIC("Digital Mic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
@@ -173,6 +181,7 @@
struct snd_soc_card *card = codec->card;
struct snd_soc_dapm_context *dapm = &codec->dapm;
struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+ struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
int hs_trim;
int ret = 0;
@@ -196,7 +205,7 @@
TWL6040_HSF_TRIM_RIGHT(hs_trim));
/* Headset jack detection only if it is supported */
- if (pdata->jack_detection) {
+ if (priv->jack_detection) {
ret = snd_soc_jack_new(codec, "Headset Jack",
SND_JACK_HEADSET, &hs_jack);
if (ret)
@@ -210,10 +219,6 @@
return ret;
}
-static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Digital Mic", NULL),
-};
-
static const struct snd_soc_dapm_route dmic_audio_map[] = {
{"DMic", NULL, "Digital Mic"},
{"Digital Mic", NULL, "Digital Mic1 Bias"},
@@ -223,19 +228,13 @@
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
- ARRAY_SIZE(dmic_dapm_widgets));
- if (ret)
- return ret;
return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
ARRAY_SIZE(dmic_audio_map));
}
/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link twl6040_dmic_dai[] = {
+static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
{
.name = "TWL6040",
.stream_name = "TWL6040",
@@ -258,19 +257,6 @@
},
};
-static struct snd_soc_dai_link twl6040_only_dai[] = {
- {
- .name = "TWL6040",
- .stream_name = "TWL6040",
- .cpu_dai_name = "omap-mcpdm",
- .codec_dai_name = "twl6040-legacy",
- .platform_name = "omap-pcm-audio",
- .codec_name = "twl6040-codec",
- .init = omap_abe_twl6040_init,
- .ops = &omap_abe_ops,
- },
-};
-
/* Audio machine driver */
static struct snd_soc_card omap_abe_card = {
.owner = THIS_MODULE,
@@ -285,6 +271,8 @@
{
struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
struct snd_soc_card *card = &omap_abe_card;
+ struct abe_twl6040 *priv;
+ int num_links = 0;
int ret;
card->dev = &pdev->dev;
@@ -294,6 +282,10 @@
return -ENODEV;
}
+ priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
+ if (priv == NULL)
+ return -ENOMEM;
+
if (pdata->card_name) {
card->name = pdata->card_name;
} else {
@@ -301,18 +293,24 @@
return -ENODEV;
}
- if (!pdata->mclk_freq) {
+ priv->jack_detection = pdata->jack_detection;
+ priv->mclk_freq = pdata->mclk_freq;
+
+
+ if (!priv->mclk_freq) {
dev_err(&pdev->dev, "MCLK frequency missing\n");
return -ENODEV;
}
- if (pdata->has_dmic) {
- card->dai_link = twl6040_dmic_dai;
- card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
- } else {
- card->dai_link = twl6040_only_dai;
- card->num_links = ARRAY_SIZE(twl6040_only_dai);
- }
+ if (pdata->has_dmic)
+ num_links = 2;
+ else
+ num_links = 1;
+
+ card->dai_link = abe_twl6040_dai_links;
+ card->num_links = num_links;
+
+ snd_soc_card_set_drvdata(card, priv);
ret = snd_soc_register_card(card);
if (ret)
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 4dcb5a7..75f5dca 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -32,6 +32,7 @@
#include <linux/io.h>
#include <linux/slab.h>
#include <linux/pm_runtime.h>
+#include <linux/of_device.h>
#include <plat/dma.h>
#include <sound/core.h>
@@ -528,10 +529,17 @@
return 0;
}
+static const struct of_device_id omap_dmic_of_match[] = {
+ { .compatible = "ti,omap4-dmic", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, omap_dmic_of_match);
+
static struct platform_driver asoc_dmic_driver = {
.driver = {
.name = "omap-dmic",
.owner = THIS_MODULE,
+ .of_match_table = omap_dmic_of_match,
},
.probe = asoc_dmic_probe,
.remove = __devexit_p(asoc_dmic_remove),
diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c
new file mode 100644
index 0000000..eaa2ea0
--- /dev/null
+++ b/sound/soc/omap/omap-hdmi-card.c
@@ -0,0 +1,87 @@
+/*
+ * omap-hdmi-card.c
+ *
+ * OMAP ALSA SoC machine driver for TI OMAP HDMI
+ * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@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
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <video/omapdss.h>
+
+#define DRV_NAME "omap-hdmi-audio"
+
+static struct snd_soc_dai_link omap_hdmi_dai = {
+ .name = "HDMI",
+ .stream_name = "HDMI",
+ .cpu_dai_name = "omap-hdmi-audio-dai",
+ .platform_name = "omap-pcm-audio",
+ .codec_name = "hdmi-audio-codec",
+ .codec_dai_name = "omap-hdmi-hifi",
+};
+
+static struct snd_soc_card snd_soc_omap_hdmi = {
+ .name = "OMAPHDMI",
+ .owner = THIS_MODULE,
+ .dai_link = &omap_hdmi_dai,
+ .num_links = 1,
+};
+
+static __devinit int omap_hdmi_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &snd_soc_omap_hdmi;
+ int ret;
+
+ card->dev = &pdev->dev;
+
+ ret = snd_soc_register_card(card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ card->dev = NULL;
+ return ret;
+ }
+ return 0;
+}
+
+static int __devexit omap_hdmi_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+ card->dev = NULL;
+ return 0;
+}
+
+static struct platform_driver omap_hdmi_driver = {
+ .driver = {
+ .name = DRV_NAME,
+ .owner = THIS_MODULE,
+ },
+ .probe = omap_hdmi_probe,
+ .remove = __devexit_p(omap_hdmi_remove),
+};
+
+module_platform_driver(omap_hdmi_driver);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("OMAP HDMI machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c
index 38e0def..a08245d 100644
--- a/sound/soc/omap/omap-hdmi.c
+++ b/sound/soc/omap/omap-hdmi.c
@@ -30,21 +30,28 @@
#include <sound/pcm_params.h>
#include <sound/initval.h>
#include <sound/soc.h>
+#include <sound/asound.h>
+#include <sound/asoundef.h>
+#include <video/omapdss.h>
#include <plat/dma.h>
#include "omap-pcm.h"
#include "omap-hdmi.h"
-#define DRV_NAME "hdmi-audio-dai"
+#define DRV_NAME "omap-hdmi-audio-dai"
-static struct omap_pcm_dma_data omap_hdmi_dai_dma_params = {
- .name = "HDMI playback",
- .sync_mode = OMAP_DMA_SYNC_PACKET,
+struct hdmi_priv {
+ struct omap_pcm_dma_data dma_params;
+ struct omap_dss_audio dss_audio;
+ struct snd_aes_iec958 iec;
+ struct snd_cea_861_aud_if cea;
+ struct omap_dss_device *dssdev;
};
static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
int err;
/*
* Make sure that the period bytes are multiple of the DMA packet size.
@@ -52,46 +59,201 @@
*/
err = snd_pcm_hw_constraint_step(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128);
- if (err < 0)
+ if (err < 0) {
+ dev_err(dai->dev, "could not apply constraint\n");
return err;
+ }
+ if (!priv->dssdev->driver->audio_supported(priv->dssdev)) {
+ dev_err(dai->dev, "audio not supported\n");
+ return -ENODEV;
+ }
return 0;
}
+static int omap_hdmi_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ return priv->dssdev->driver->audio_enable(priv->dssdev);
+}
+
static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+ struct snd_aes_iec958 *iec = &priv->iec;
+ struct snd_cea_861_aud_if *cea = &priv->cea;
int err = 0;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- omap_hdmi_dai_dma_params.packet_size = 16;
+ priv->dma_params.packet_size = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- omap_hdmi_dai_dma_params.packet_size = 32;
+ priv->dma_params.packet_size = 32;
+ break;
+ default:
+ dev_err(dai->dev, "format not supported!\n");
+ return -EINVAL;
+ }
+
+ priv->dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
+
+ snd_soc_dai_set_dma_data(dai, substream,
+ &priv->dma_params);
+
+ /*
+ * fill the IEC-60958 channel status word
+ */
+
+ /* specify IEC-60958-3 (commercial use) */
+ iec->status[0] &= ~IEC958_AES0_PROFESSIONAL;
+
+ /* specify that the audio is LPCM*/
+ iec->status[0] &= ~IEC958_AES0_NONAUDIO;
+
+ iec->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
+
+ iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE;
+
+ iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID;
+
+ iec->status[1] = IEC958_AES1_CON_GENERAL;
+
+ iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC;
+
+ iec->status[2] |= IEC958_AES2_CON_CHANNEL_UNSPEC;
+
+ switch (params_rate(params)) {
+ case 32000:
+ iec->status[3] |= IEC958_AES3_CON_FS_32000;
+ break;
+ case 44100:
+ iec->status[3] |= IEC958_AES3_CON_FS_44100;
+ break;
+ case 48000:
+ iec->status[3] |= IEC958_AES3_CON_FS_48000;
+ break;
+ case 88200:
+ iec->status[3] |= IEC958_AES3_CON_FS_88200;
+ break;
+ case 96000:
+ iec->status[3] |= IEC958_AES3_CON_FS_96000;
+ break;
+ case 176400:
+ iec->status[3] |= IEC958_AES3_CON_FS_176400;
+ break;
+ case 192000:
+ iec->status[3] |= IEC958_AES3_CON_FS_192000;
+ break;
+ default:
+ dev_err(dai->dev, "rate not supported!\n");
+ return -EINVAL;
+ }
+
+ /* specify the clock accuracy */
+ iec->status[3] |= IEC958_AES3_CON_CLOCK_1000PPM;
+
+ /*
+ * specify the word length. The same word length value can mean
+ * two different lengths. Hence, we need to specify the maximum
+ * word length as well.
+ */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ iec->status[4] |= IEC958_AES4_CON_WORDLEN_20_16;
+ iec->status[4] &= ~IEC958_AES4_CON_MAX_WORDLEN_24;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iec->status[4] |= IEC958_AES4_CON_WORDLEN_24_20;
+ iec->status[4] |= IEC958_AES4_CON_MAX_WORDLEN_24;
+ break;
+ default:
+ dev_err(dai->dev, "format not supported!\n");
+ return -EINVAL;
+ }
+
+ /*
+ * Fill the CEA-861 audio infoframe (see spec for details)
+ */
+
+ cea->db1_ct_cc = (params_channels(params) - 1)
+ & CEA861_AUDIO_INFOFRAME_DB1CC;
+ cea->db1_ct_cc |= CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM;
+
+ cea->db2_sf_ss = CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM;
+ cea->db2_sf_ss |= CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM;
+
+ cea->db3 = 0; /* not used, all zeros */
+
+ /*
+ * The OMAP HDMI IP requires to use the 8-channel channel code when
+ * transmitting more than two channels.
+ */
+ if (params_channels(params) == 2)
+ cea->db4_ca = 0x0;
+ else
+ cea->db4_ca = 0x13;
+
+ cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED;
+ /* the expression is trivial but makes clear what we are doing */
+ cea->db5_dminh_lsv |= (0 & CEA861_AUDIO_INFOFRAME_DB5_LSV);
+
+ priv->dss_audio.iec = iec;
+ priv->dss_audio.cea = cea;
+
+ err = priv->dssdev->driver->audio_config(priv->dssdev,
+ &priv->dss_audio);
+
+ return err;
+}
+
+static int omap_hdmi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ err = priv->dssdev->driver->audio_start(priv->dssdev);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ priv->dssdev->driver->audio_stop(priv->dssdev);
break;
default:
err = -EINVAL;
}
-
- omap_hdmi_dai_dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
-
- snd_soc_dai_set_dma_data(dai, substream,
- &omap_hdmi_dai_dma_params);
-
return err;
}
+static void omap_hdmi_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ priv->dssdev->driver->audio_disable(priv->dssdev);
+}
+
static const struct snd_soc_dai_ops omap_hdmi_dai_ops = {
.startup = omap_hdmi_dai_startup,
.hw_params = omap_hdmi_dai_hw_params,
+ .prepare = omap_hdmi_dai_prepare,
+ .trigger = omap_hdmi_dai_trigger,
+ .shutdown = omap_hdmi_dai_shutdown,
};
static struct snd_soc_dai_driver omap_hdmi_dai = {
.playback = {
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 8,
.rates = OMAP_HDMI_RATES,
.formats = OMAP_HDMI_FORMATS,
},
@@ -102,31 +264,77 @@
{
int ret;
struct resource *hdmi_rsrc;
+ struct hdmi_priv *hdmi_data;
+ bool hdmi_dev_found = false;
+
+ hdmi_data = devm_kzalloc(&pdev->dev, sizeof(*hdmi_data), GFP_KERNEL);
+ if (hdmi_data == NULL) {
+ dev_err(&pdev->dev, "Cannot allocate memory for HDMI data\n");
+ return -ENOMEM;
+ }
hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!hdmi_rsrc) {
dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n");
- return -EINVAL;
+ return -ENODEV;
}
- omap_hdmi_dai_dma_params.port_addr = hdmi_rsrc->start
+ hdmi_data->dma_params.port_addr = hdmi_rsrc->start
+ OMAP_HDMI_AUDIO_DMA_PORT;
hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!hdmi_rsrc) {
dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n");
- return -EINVAL;
+ return -ENODEV;
}
- omap_hdmi_dai_dma_params.dma_req = hdmi_rsrc->start;
+ hdmi_data->dma_params.dma_req = hdmi_rsrc->start;
+ hdmi_data->dma_params.name = "HDMI playback";
+ hdmi_data->dma_params.sync_mode = OMAP_DMA_SYNC_PACKET;
+ /*
+ * TODO: We assume that there is only one DSS HDMI device. Future
+ * OMAP implementations may support more than one HDMI devices and
+ * we should provided separate audio support for all of them.
+ */
+ /* Find an HDMI device. */
+ for_each_dss_dev(hdmi_data->dssdev) {
+ omap_dss_get_device(hdmi_data->dssdev);
+
+ if (!hdmi_data->dssdev->driver) {
+ omap_dss_put_device(hdmi_data->dssdev);
+ continue;
+ }
+
+ if (hdmi_data->dssdev->type == OMAP_DISPLAY_TYPE_HDMI) {
+ hdmi_dev_found = true;
+ break;
+ }
+ }
+
+ if (!hdmi_dev_found) {
+ dev_err(&pdev->dev, "no driver for HDMI display found\n");
+ return -ENODEV;
+ }
+
+ dev_set_drvdata(&pdev->dev, hdmi_data);
ret = snd_soc_register_dai(&pdev->dev, &omap_hdmi_dai);
+
return ret;
}
static int __devexit omap_hdmi_remove(struct platform_device *pdev)
{
+ struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev);
+
snd_soc_unregister_dai(&pdev->dev);
+
+ if (hdmi_data == NULL) {
+ dev_err(&pdev->dev, "cannot obtain HDMi data\n");
+ return -ENODEV;
+ }
+
+ omap_dss_put_device(hdmi_data->dssdev);
return 0;
}
diff --git a/sound/soc/omap/omap-hdmi.h b/sound/soc/omap/omap-hdmi.h
index 34c298d..6ad2bf4 100644
--- a/sound/soc/omap/omap-hdmi.h
+++ b/sound/soc/omap/omap-hdmi.h
@@ -28,7 +28,9 @@
#define OMAP_HDMI_AUDIO_DMA_PORT 0x8c
#define OMAP_HDMI_RATES (SNDRV_PCM_RATE_32000 | \
- SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
#define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE)
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6912ac7..1046083 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -71,18 +71,17 @@
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
- if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
- /*
- * Configure McBSP threshold based on either:
- * packet_size, when the sDMA is in packet mode, or
- * based on the period size.
- */
- if (dma_data->packet_size)
- words = dma_data->packet_size;
- else
- words = snd_pcm_lib_period_bytes(substream) /
- (mcbsp->wlen / 8);
+ /*
+ * Configure McBSP threshold based on either:
+ * packet_size, when the sDMA is in packet mode, or based on the
+ * period size in THRESHOLD mode, otherwise use McBSP threshold = 1
+ * for mono streams.
+ */
+ if (dma_data->packet_size)
+ words = dma_data->packet_size;
+ else if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+ words = snd_pcm_lib_period_bytes(substream) /
+ (mcbsp->wlen / 8);
else
words = 1;
@@ -139,13 +138,15 @@
if (mcbsp->pdata->buffer_size) {
/*
* Rule for the buffer size. We should not allow
- * smaller buffer than the FIFO size to avoid underruns
+ * smaller buffer than the FIFO size to avoid underruns.
+ * This applies only for the playback stream.
*/
- snd_pcm_hw_rule_add(substream->runtime, 0,
- SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- omap_mcbsp_hwrule_min_buffersize,
- mcbsp,
- SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ omap_mcbsp_hwrule_min_buffersize,
+ mcbsp,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
/* Make sure, that the period size is always even */
snd_pcm_hw_constraint_step(substream->runtime, 0,
@@ -230,6 +231,7 @@
unsigned int format, div, framesize, master;
dma_data = &mcbsp->dma_data[substream->stream];
+ channels = params_channels(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
@@ -245,7 +247,6 @@
}
if (mcbsp->pdata->buffer_size) {
dma_data->set_threshold = omap_mcbsp_set_threshold;
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
int period_words, max_thrsh;
@@ -283,6 +284,10 @@
} else {
sync_mode = OMAP_DMA_SYNC_FRAME;
}
+ } else if (channels > 1) {
+ /* Use packet mode for non mono streams */
+ pkt_size = channels;
+ sync_mode = OMAP_DMA_SYNC_PACKET;
}
}
@@ -301,7 +306,7 @@
regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7));
regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7));
format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
- wpf = channels = params_channels(params);
+ wpf = channels;
if (channels == 2 && (format == SND_SOC_DAIFMT_I2S ||
format == SND_SOC_DAIFMT_LEFT_J)) {
/* Use dual-phase frames */
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 3970556..59d47ab5 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -33,6 +33,7 @@
#include <linux/irq.h>
#include <linux/slab.h>
#include <linux/pm_runtime.h>
+#include <linux/of_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -507,10 +508,17 @@
return 0;
}
+static const struct of_device_id omap_mcpdm_of_match[] = {
+ { .compatible = "ti,omap4-mcpdm", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, omap_mcpdm_of_match);
+
static struct platform_driver asoc_mcpdm_driver = {
.driver = {
.name = "omap-mcpdm",
.owner = THIS_MODULE,
+ .of_match_table = omap_mcpdm_of_match,
},
.probe = asoc_mcpdm_probe,
diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c
deleted file mode 100644
index 28d689b..0000000
--- a/sound/soc/omap/omap4-hdmi-card.c
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * omap4-hdmi-card.c
- *
- * OMAP ALSA SoC machine driver for TI OMAP4 HDMI
- * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@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
- *
- */
-
-#include <linux/module.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <asm/mach-types.h>
-#include <video/omapdss.h>
-
-#define DRV_NAME "omap4-hdmi-audio"
-
-static int omap4_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- int i;
- struct omap_overlay_manager *mgr = NULL;
- struct device *dev = substream->pcm->card->dev;
-
- /* Find DSS HDMI device */
- for (i = 0; i < omap_dss_get_num_overlay_managers(); i++) {
- mgr = omap_dss_get_overlay_manager(i);
- if (mgr && mgr->device
- && mgr->device->type == OMAP_DISPLAY_TYPE_HDMI)
- break;
- }
-
- if (i == omap_dss_get_num_overlay_managers()) {
- dev_err(dev, "HDMI display device not found!\n");
- return -ENODEV;
- }
-
- /* Make sure HDMI is power-on to avoid L3 interconnect errors */
- if (mgr->device->state != OMAP_DSS_DISPLAY_ACTIVE) {
- dev_err(dev, "HDMI display is not active!\n");
- return -EIO;
- }
-
- return 0;
-}
-
-static struct snd_soc_ops omap4_hdmi_dai_ops = {
- .hw_params = omap4_hdmi_dai_hw_params,
-};
-
-static struct snd_soc_dai_link omap4_hdmi_dai = {
- .name = "HDMI",
- .stream_name = "HDMI",
- .cpu_dai_name = "hdmi-audio-dai",
- .platform_name = "omap-pcm-audio",
- .codec_name = "omapdss_hdmi",
- .codec_dai_name = "hdmi-audio-codec",
- .ops = &omap4_hdmi_dai_ops,
-};
-
-static struct snd_soc_card snd_soc_omap4_hdmi = {
- .name = "OMAP4HDMI",
- .owner = THIS_MODULE,
- .dai_link = &omap4_hdmi_dai,
- .num_links = 1,
-};
-
-static __devinit int omap4_hdmi_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &snd_soc_omap4_hdmi;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = snd_soc_register_card(card);
- if (ret) {
- dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
- card->dev = NULL;
- return ret;
- }
- return 0;
-}
-
-static int __devexit omap4_hdmi_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- card->dev = NULL;
- return 0;
-}
-
-static struct platform_driver omap4_hdmi_driver = {
- .driver = {
- .name = "omap4-hdmi-audio",
- .owner = THIS_MODULE,
- },
- .probe = omap4_hdmi_probe,
- .remove = __devexit_p(omap4_hdmi_remove),
-};
-
-module_platform_driver(omap4_hdmi_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("OMAP4 HDMI machine ASoC driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/sound_core.c b/sound/sound_core.c
index c6e81fb..fb9255c 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -361,7 +361,7 @@
struct device *dev)
{
const int chain = unit % SOUND_STEP;
- int max_unit = 128 + chain;
+ int max_unit = 256;
const char *name;
char _name[16];
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 4a7be7b..d5b5c33 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -131,8 +131,9 @@
subs = &as->substream[idx];
if (!subs->num_formats)
continue;
- snd_usb_release_substream_urbs(subs, 1);
subs->interface = -1;
+ subs->data_endpoint = NULL;
+ subs->sync_endpoint = NULL;
}
}
@@ -276,6 +277,7 @@
static int snd_usb_audio_free(struct snd_usb_audio *chip)
{
+ mutex_destroy(&chip->mutex);
kfree(chip);
return 0;
}
@@ -336,6 +338,7 @@
return -ENOMEM;
}
+ mutex_init(&chip->mutex);
mutex_init(&chip->shutdown_mutex);
chip->index = idx;
chip->dev = dev;
@@ -348,6 +351,7 @@
chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
le16_to_cpu(dev->descriptor.idProduct));
INIT_LIST_HEAD(&chip->pcm_list);
+ INIT_LIST_HEAD(&chip->ep_list);
INIT_LIST_HEAD(&chip->midi_list);
INIT_LIST_HEAD(&chip->mixer_list);
@@ -565,6 +569,10 @@
list_for_each(p, &chip->pcm_list) {
snd_usb_stream_disconnect(p);
}
+ /* release the endpoint resources */
+ list_for_each(p, &chip->ep_list) {
+ snd_usb_endpoint_free(p);
+ }
/* release the midi resources */
list_for_each(p, &chip->midi_list) {
snd_usbmidi_disconnect(p);
diff --git a/sound/usb/card.h b/sound/usb/card.h
index da5fa1a..0d37238 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -30,20 +30,71 @@
};
struct snd_usb_substream;
+struct snd_usb_endpoint;
struct snd_urb_ctx {
struct urb *urb;
unsigned int buffer_size; /* size of data buffer, if data URB */
struct snd_usb_substream *subs;
+ struct snd_usb_endpoint *ep;
int index; /* index for urb array */
int packets; /* number of packets per urb */
+ int packet_size[MAX_PACKS_HS]; /* size of packets for next submission */
+ struct list_head ready_list;
};
-struct snd_urb_ops {
- int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
- int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
- int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
- int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u);
+struct snd_usb_endpoint {
+ struct snd_usb_audio *chip;
+
+ int use_count;
+ int ep_num; /* the referenced endpoint number */
+ int type; /* SND_USB_ENDPOINT_TYPE_* */
+ unsigned long flags;
+
+ void (*prepare_data_urb) (struct snd_usb_substream *subs,
+ struct urb *urb);
+ void (*retire_data_urb) (struct snd_usb_substream *subs,
+ struct urb *urb);
+
+ struct snd_usb_substream *data_subs;
+ struct snd_usb_endpoint *sync_master;
+ struct snd_usb_endpoint *sync_slave;
+
+ struct snd_urb_ctx urb[MAX_URBS];
+
+ struct snd_usb_packet_info {
+ uint32_t packet_size[MAX_PACKS_HS];
+ int packets;
+ } next_packet[MAX_URBS];
+ int next_packet_read_pos, next_packet_write_pos;
+ struct list_head ready_playback_urbs;
+
+ unsigned int nurbs; /* # urbs */
+ unsigned long active_mask; /* bitmask of active urbs */
+ unsigned long unlink_mask; /* bitmask of unlinked urbs */
+ char *syncbuf; /* sync buffer for all sync URBs */
+ dma_addr_t sync_dma; /* DMA address of syncbuf */
+
+ unsigned int pipe; /* the data i/o pipe */
+ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */
+ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */
+ int freqshift; /* how much to shift the feedback value to get Q16.16 */
+ unsigned int freqmax; /* maximum sampling rate, used for buffer management */
+ unsigned int phase; /* phase accumulator */
+ unsigned int maxpacksize; /* max packet size in bytes */
+ unsigned int maxframesize; /* max packet size in frames */
+ unsigned int curpacksize; /* current packet size in bytes (for capture) */
+ unsigned int curframesize; /* current packet size in frames (for capture) */
+ unsigned int syncmaxsize; /* sync endpoint packet size */
+ unsigned int fill_max:1; /* fill max packet size always */
+ unsigned int datainterval; /* log_2 of data packet interval */
+ unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */
+ unsigned char silence_value;
+ unsigned int stride;
+ int iface, alt_idx;
+
+ spinlock_t lock;
+ struct list_head list;
};
struct snd_usb_substream {
@@ -57,21 +108,6 @@
unsigned int cur_rate; /* current rate (for hw_params callback) */
unsigned int period_bytes; /* current period bytes (for hw_params callback) */
unsigned int altset_idx; /* USB data format: index of alternate setting */
- unsigned int datapipe; /* the data i/o pipe */
- unsigned int syncpipe; /* 1 - async out or adaptive in */
- unsigned int datainterval; /* log_2 of data packet interval */
- unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */
- unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */
- unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */
- int freqshift; /* how much to shift the feedback value to get Q16.16 */
- unsigned int freqmax; /* maximum sampling rate, used for buffer management */
- unsigned int phase; /* phase accumulator */
- unsigned int maxpacksize; /* max packet size in bytes */
- unsigned int maxframesize; /* max packet size in frames */
- unsigned int curpacksize; /* current packet size in bytes (for capture) */
- unsigned int curframesize; /* current packet size in frames (for capture) */
- unsigned int syncmaxsize; /* sync endpoint packet size */
- unsigned int fill_max: 1; /* fill max packet size always */
unsigned int txfr_quirk:1; /* allow sub-frame alignment */
unsigned int fmt_type; /* USB audio format type (1-3) */
@@ -82,11 +118,10 @@
unsigned long active_mask; /* bitmask of active urbs */
unsigned long unlink_mask; /* bitmask of unlinked urbs */
- unsigned int nurbs; /* # urbs */
- struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */
- struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */
- char *syncbuf; /* sync buffer for all sync URBs */
- dma_addr_t sync_dma; /* DMA address of syncbuf */
+ /* data and sync endpoints for this stream */
+ struct snd_usb_endpoint *data_endpoint;
+ struct snd_usb_endpoint *sync_endpoint;
+ unsigned long flags;
u64 formats; /* format bitmasks (all or'ed) */
unsigned int num_formats; /* number of supported audio formats (list) */
@@ -94,7 +129,6 @@
struct snd_pcm_hw_constraint_list rate_list; /* limited rates */
spinlock_t lock;
- struct snd_urb_ops ops; /* callbacks (must be filled at init) */
int last_frame_number; /* stored frame number */
int last_delay; /* stored delay */
};
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 08dcce5..e690690 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -20,9 +20,11 @@
#include <linux/ratelimit.h>
#include <linux/usb.h>
#include <linux/usb/audio.h>
+#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include "usbaudio.h"
#include "helper.h"
@@ -30,6 +32,36 @@
#include "endpoint.h"
#include "pcm.h"
+#define EP_FLAG_ACTIVATED 0
+#define EP_FLAG_RUNNING 1
+
+/*
+ * snd_usb_endpoint is a model that abstracts everything related to an
+ * USB endpoint and its streaming.
+ *
+ * There are functions to activate and deactivate the streaming URBs and
+ * optional callbacks to let the pcm logic handle the actual content of the
+ * packets for playback and record. Thus, the bus streaming and the audio
+ * handlers are fully decoupled.
+ *
+ * There are two different types of endpoints in audio applications.
+ *
+ * SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both
+ * inbound and outbound traffic.
+ *
+ * SND_USB_ENDPOINT_TYPE_SYNC endpoints are for inbound traffic only and
+ * expect the payload to carry Q10.14 / Q16.16 formatted sync information
+ * (3 or 4 bytes).
+ *
+ * Each endpoint has to be configured prior to being used by calling
+ * snd_usb_endpoint_set_params().
+ *
+ * The model incorporates a reference counting, so that multiple users
+ * can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and
+ * only the first user will effectively start the URBs, and only the last
+ * one to stop it will tear the URBs down again.
+ */
+
/*
* convert a sampling rate into our full speed format (fs/1000 in Q16.16)
* this will overflow at approx 524 kHz
@@ -49,727 +81,16 @@
}
/*
- * unlink active urbs.
- */
-static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep)
-{
- struct snd_usb_audio *chip = subs->stream->chip;
- unsigned int i;
- int async;
-
- subs->running = 0;
-
- if (!force && subs->stream->chip->shutdown) /* to be sure... */
- return -EBADFD;
-
- async = !can_sleep && chip->async_unlink;
-
- if (!async && in_interrupt())
- return 0;
-
- for (i = 0; i < subs->nurbs; i++) {
- if (test_bit(i, &subs->active_mask)) {
- if (!test_and_set_bit(i, &subs->unlink_mask)) {
- struct urb *u = subs->dataurb[i].urb;
- if (async)
- usb_unlink_urb(u);
- else
- usb_kill_urb(u);
- }
- }
- }
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- if (test_bit(i+16, &subs->active_mask)) {
- if (!test_and_set_bit(i+16, &subs->unlink_mask)) {
- struct urb *u = subs->syncurb[i].urb;
- if (async)
- usb_unlink_urb(u);
- else
- usb_kill_urb(u);
- }
- }
- }
- }
- return 0;
-}
-
-
-/*
* release a urb data
*/
static void release_urb_ctx(struct snd_urb_ctx *u)
{
- if (u->urb) {
- if (u->buffer_size)
- usb_free_coherent(u->subs->dev, u->buffer_size,
- u->urb->transfer_buffer,
- u->urb->transfer_dma);
- usb_free_urb(u->urb);
- u->urb = NULL;
- }
-}
-
-/*
- * wait until all urbs are processed.
- */
-static int wait_clear_urbs(struct snd_usb_substream *subs)
-{
- unsigned long end_time = jiffies + msecs_to_jiffies(1000);
- unsigned int i;
- int alive;
-
- do {
- alive = 0;
- for (i = 0; i < subs->nurbs; i++) {
- if (test_bit(i, &subs->active_mask))
- alive++;
- }
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- if (test_bit(i + 16, &subs->active_mask))
- alive++;
- }
- }
- if (! alive)
- break;
- schedule_timeout_uninterruptible(1);
- } while (time_before(jiffies, end_time));
- if (alive)
- snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive);
- return 0;
-}
-
-/*
- * release a substream
- */
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force)
-{
- int i;
-
- /* stop urbs (to be sure) */
- deactivate_urbs(subs, force, 1);
- wait_clear_urbs(subs);
-
- for (i = 0; i < MAX_URBS; i++)
- release_urb_ctx(&subs->dataurb[i]);
- for (i = 0; i < SYNC_URBS; i++)
- release_urb_ctx(&subs->syncurb[i]);
- usb_free_coherent(subs->dev, SYNC_URBS * 4,
- subs->syncbuf, subs->sync_dma);
- subs->syncbuf = NULL;
- subs->nurbs = 0;
-}
-
-/*
- * complete callback from data urb
- */
-static void snd_complete_urb(struct urb *urb)
-{
- struct snd_urb_ctx *ctx = urb->context;
- struct snd_usb_substream *subs = ctx->subs;
- struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
- int err = 0;
-
- if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) ||
- !subs->running || /* can be stopped during retire callback */
- (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 ||
- (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
- clear_bit(ctx->index, &subs->active_mask);
- if (err < 0) {
- snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err);
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- }
- }
-}
-
-
-/*
- * complete callback from sync urb
- */
-static void snd_complete_sync_urb(struct urb *urb)
-{
- struct snd_urb_ctx *ctx = urb->context;
- struct snd_usb_substream *subs = ctx->subs;
- struct snd_pcm_substream *substream = ctx->subs->pcm_substream;
- int err = 0;
-
- if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) ||
- !subs->running || /* can be stopped during retire callback */
- (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 ||
- (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) {
- clear_bit(ctx->index + 16, &subs->active_mask);
- if (err < 0) {
- snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err);
- snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- }
- }
-}
-
-
-/*
- * initialize a substream for plaback/capture
- */
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
- unsigned int period_bytes,
- unsigned int rate,
- unsigned int frame_bits)
-{
- unsigned int maxsize, i;
- int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
- unsigned int urb_packs, total_packs, packs_per_ms;
- struct snd_usb_audio *chip = subs->stream->chip;
-
- /* calculate the frequency in 16.16 format */
- if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
- subs->freqn = get_usb_full_speed_rate(rate);
- else
- subs->freqn = get_usb_high_speed_rate(rate);
- subs->freqm = subs->freqn;
- subs->freqshift = INT_MIN;
- /* calculate max. frequency */
- if (subs->maxpacksize) {
- /* whatever fits into a max. size packet */
- maxsize = subs->maxpacksize;
- subs->freqmax = (maxsize / (frame_bits >> 3))
- << (16 - subs->datainterval);
- } else {
- /* no max. packet size: just take 25% higher than nominal */
- subs->freqmax = subs->freqn + (subs->freqn >> 2);
- maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3))
- >> (16 - subs->datainterval);
- }
- subs->phase = 0;
-
- if (subs->fill_max)
- subs->curpacksize = subs->maxpacksize;
- else
- subs->curpacksize = maxsize;
-
- if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL)
- packs_per_ms = 8 >> subs->datainterval;
- else
- packs_per_ms = 1;
-
- if (is_playback) {
- urb_packs = max(chip->nrpacks, 1);
- urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
- } else
- urb_packs = 1;
- urb_packs *= packs_per_ms;
- if (subs->syncpipe)
- urb_packs = min(urb_packs, 1U << subs->syncinterval);
-
- /* decide how many packets to be used */
- if (is_playback) {
- unsigned int minsize, maxpacks;
- /* determine how small a packet can be */
- minsize = (subs->freqn >> (16 - subs->datainterval))
- * (frame_bits >> 3);
- /* with sync from device, assume it can be 12% lower */
- if (subs->syncpipe)
- minsize -= minsize >> 3;
- minsize = max(minsize, 1u);
- total_packs = (period_bytes + minsize - 1) / minsize;
- /* we need at least two URBs for queueing */
- if (total_packs < 2) {
- total_packs = 2;
- } else {
- /* and we don't want too long a queue either */
- maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
- total_packs = min(total_packs, maxpacks);
- }
- } else {
- while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
- urb_packs >>= 1;
- total_packs = MAX_URBS * urb_packs;
- }
- subs->nurbs = (total_packs + urb_packs - 1) / urb_packs;
- if (subs->nurbs > MAX_URBS) {
- /* too much... */
- subs->nurbs = MAX_URBS;
- total_packs = MAX_URBS * urb_packs;
- } else if (subs->nurbs < 2) {
- /* too little - we need at least two packets
- * to ensure contiguous playback/capture
- */
- subs->nurbs = 2;
- }
-
- /* allocate and initialize data urbs */
- for (i = 0; i < subs->nurbs; i++) {
- struct snd_urb_ctx *u = &subs->dataurb[i];
- u->index = i;
- u->subs = subs;
- u->packets = (i + 1) * total_packs / subs->nurbs
- - i * total_packs / subs->nurbs;
- u->buffer_size = maxsize * u->packets;
- if (subs->fmt_type == UAC_FORMAT_TYPE_II)
- u->packets++; /* for transfer delimiter */
- u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
- if (!u->urb)
- goto out_of_memory;
- u->urb->transfer_buffer =
- usb_alloc_coherent(subs->dev, u->buffer_size,
- GFP_KERNEL, &u->urb->transfer_dma);
- if (!u->urb->transfer_buffer)
- goto out_of_memory;
- u->urb->pipe = subs->datapipe;
- u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
- u->urb->interval = 1 << subs->datainterval;
- u->urb->context = u;
- u->urb->complete = snd_complete_urb;
- }
-
- if (subs->syncpipe) {
- /* allocate and initialize sync urbs */
- subs->syncbuf = usb_alloc_coherent(subs->dev, SYNC_URBS * 4,
- GFP_KERNEL, &subs->sync_dma);
- if (!subs->syncbuf)
- goto out_of_memory;
- for (i = 0; i < SYNC_URBS; i++) {
- struct snd_urb_ctx *u = &subs->syncurb[i];
- u->index = i;
- u->subs = subs;
- u->packets = 1;
- u->urb = usb_alloc_urb(1, GFP_KERNEL);
- if (!u->urb)
- goto out_of_memory;
- u->urb->transfer_buffer = subs->syncbuf + i * 4;
- u->urb->transfer_dma = subs->sync_dma + i * 4;
- u->urb->transfer_buffer_length = 4;
- u->urb->pipe = subs->syncpipe;
- u->urb->transfer_flags = URB_ISO_ASAP |
- URB_NO_TRANSFER_DMA_MAP;
- u->urb->number_of_packets = 1;
- u->urb->interval = 1 << subs->syncinterval;
- u->urb->context = u;
- u->urb->complete = snd_complete_sync_urb;
- }
- }
- return 0;
-
-out_of_memory:
- snd_usb_release_substream_urbs(subs, 0);
- return -ENOMEM;
-}
-
-/*
- * prepare urb for full speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 10.14 frequency is passed through the pipe.
- */
-static int prepare_capture_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned char *cp = urb->transfer_buffer;
- struct snd_urb_ctx *ctx = urb->context;
-
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->iso_frame_desc[0].length = 3;
- urb->iso_frame_desc[0].offset = 0;
- cp[0] = subs->freqn >> 2;
- cp[1] = subs->freqn >> 10;
- cp[2] = subs->freqn >> 18;
- return 0;
-}
-
-/*
- * prepare urb for high speed capture sync pipe
- *
- * fill the length and offset of each urb descriptor.
- * the fixed 12.13 frequency is passed as 16.16 through the pipe.
- */
-static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned char *cp = urb->transfer_buffer;
- struct snd_urb_ctx *ctx = urb->context;
-
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->iso_frame_desc[0].length = 4;
- urb->iso_frame_desc[0].offset = 0;
- cp[0] = subs->freqn;
- cp[1] = subs->freqn >> 8;
- cp[2] = subs->freqn >> 16;
- cp[3] = subs->freqn >> 24;
- return 0;
-}
-
-/*
- * process after capture sync complete
- * - nothing to do
- */
-static int retire_capture_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- return 0;
-}
-
-/*
- * prepare urb for capture data pipe
- *
- * fill the offset and length of each descriptor.
- *
- * we use a temporary buffer to write the captured data.
- * since the length of written data is determined by host, we cannot
- * write onto the pcm buffer directly... the data is thus copied
- * later at complete callback to the global buffer.
- */
-static int prepare_capture_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- int i, offs;
- struct snd_urb_ctx *ctx = urb->context;
-
- offs = 0;
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- for (i = 0; i < ctx->packets; i++) {
- urb->iso_frame_desc[i].offset = offs;
- urb->iso_frame_desc[i].length = subs->curpacksize;
- offs += subs->curpacksize;
- }
- urb->transfer_buffer_length = offs;
- urb->number_of_packets = ctx->packets;
- return 0;
-}
-
-/*
- * process after capture complete
- *
- * copy the data from each desctiptor to the pcm buffer, and
- * update the current position.
- */
-static int retire_capture_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned long flags;
- unsigned char *cp;
- int i;
- unsigned int stride, frames, bytes, oldptr;
- int period_elapsed = 0;
-
- stride = runtime->frame_bits >> 3;
-
- for (i = 0; i < urb->number_of_packets; i++) {
- cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
- if (urb->iso_frame_desc[i].status && printk_ratelimit()) {
- snd_printdd("frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
- // continue;
- }
- bytes = urb->iso_frame_desc[i].actual_length;
- frames = bytes / stride;
- if (!subs->txfr_quirk)
- bytes = frames * stride;
- if (bytes % (runtime->sample_bits >> 3) != 0) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
- int oldbytes = bytes;
-#endif
- bytes = frames * stride;
- snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
- oldbytes, bytes);
- }
- /* update the current pointer */
- spin_lock_irqsave(&subs->lock, flags);
- oldptr = subs->hwptr_done;
- subs->hwptr_done += bytes;
- if (subs->hwptr_done >= runtime->buffer_size * stride)
- subs->hwptr_done -= runtime->buffer_size * stride;
- frames = (bytes + (oldptr % stride)) / stride;
- subs->transfer_done += frames;
- if (subs->transfer_done >= runtime->period_size) {
- subs->transfer_done -= runtime->period_size;
- period_elapsed = 1;
- }
- spin_unlock_irqrestore(&subs->lock, flags);
- /* copy a data chunk */
- if (oldptr + bytes > runtime->buffer_size * stride) {
- unsigned int bytes1 =
- runtime->buffer_size * stride - oldptr;
- memcpy(runtime->dma_area + oldptr, cp, bytes1);
- memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
- } else {
- memcpy(runtime->dma_area + oldptr, cp, bytes);
- }
- }
- if (period_elapsed)
- snd_pcm_period_elapsed(subs->pcm_substream);
- return 0;
-}
-
-/*
- * Process after capture complete when paused. Nothing to do.
- */
-static int retire_paused_capture_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- return 0;
-}
-
-
-/*
- * prepare urb for playback sync pipe
- *
- * set up the offset and length to receive the current frequency.
- */
-static int prepare_playback_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- struct snd_urb_ctx *ctx = urb->context;
-
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize);
- urb->iso_frame_desc[0].offset = 0;
- return 0;
-}
-
-/*
- * process after playback sync complete
- *
- * Full speed devices report feedback values in 10.14 format as samples per
- * frame, high speed devices in 16.16 format as samples per microframe.
- * Because the Audio Class 1 spec was written before USB 2.0, many high speed
- * devices use a wrong interpretation, some others use an entirely different
- * format. Therefore, we cannot predict what format any particular device uses
- * and must detect it automatically.
- */
-static int retire_playback_sync_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned int f;
- int shift;
- unsigned long flags;
-
- if (urb->iso_frame_desc[0].status != 0 ||
- urb->iso_frame_desc[0].actual_length < 3)
- return 0;
-
- f = le32_to_cpup(urb->transfer_buffer);
- if (urb->iso_frame_desc[0].actual_length == 3)
- f &= 0x00ffffff;
- else
- f &= 0x0fffffff;
- if (f == 0)
- return 0;
-
- if (unlikely(subs->freqshift == INT_MIN)) {
- /*
- * The first time we see a feedback value, determine its format
- * by shifting it left or right until it matches the nominal
- * frequency value. This assumes that the feedback does not
- * differ from the nominal value more than +50% or -25%.
- */
- shift = 0;
- while (f < subs->freqn - subs->freqn / 4) {
- f <<= 1;
- shift++;
- }
- while (f > subs->freqn + subs->freqn / 2) {
- f >>= 1;
- shift--;
- }
- subs->freqshift = shift;
- }
- else if (subs->freqshift >= 0)
- f <<= subs->freqshift;
- else
- f >>= -subs->freqshift;
-
- if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) {
- /*
- * If the frequency looks valid, set it.
- * This value is referred to in prepare_playback_urb().
- */
- spin_lock_irqsave(&subs->lock, flags);
- subs->freqm = f;
- spin_unlock_irqrestore(&subs->lock, flags);
- } else {
- /*
- * Out of range; maybe the shift value is wrong.
- * Reset it so that we autodetect again the next time.
- */
- subs->freqshift = INT_MIN;
- }
-
- return 0;
-}
-
-/* determine the number of frames in the next packet */
-static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
-{
- if (subs->fill_max)
- return subs->maxframesize;
- else {
- subs->phase = (subs->phase & 0xffff)
- + (subs->freqm << subs->datainterval);
- return min(subs->phase >> 16, subs->maxframesize);
- }
-}
-
-/*
- * Prepare urb for streaming before playback starts or when paused.
- *
- * We don't have any data, so we send silence.
- */
-static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned int i, offs, counts;
- struct snd_urb_ctx *ctx = urb->context;
- int stride = runtime->frame_bits >> 3;
-
- offs = 0;
- urb->dev = ctx->subs->dev;
- for (i = 0; i < ctx->packets; ++i) {
- counts = snd_usb_audio_next_packet_size(subs);
- urb->iso_frame_desc[i].offset = offs * stride;
- urb->iso_frame_desc[i].length = counts * stride;
- offs += counts;
- }
- urb->number_of_packets = ctx->packets;
- urb->transfer_buffer_length = offs * stride;
- memset(urb->transfer_buffer,
- runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
- offs * stride);
- return 0;
-}
-
-/*
- * prepare urb for playback data pipe
- *
- * Since a URB can handle only a single linear buffer, we must use double
- * buffering when the data to be transferred overflows the buffer boundary.
- * To avoid inconsistencies when updating hwptr_done, we use double buffering
- * for all URBs.
- */
-static int prepare_playback_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- int i, stride;
- unsigned int counts, frames, bytes;
- unsigned long flags;
- int period_elapsed = 0;
- struct snd_urb_ctx *ctx = urb->context;
-
- stride = runtime->frame_bits >> 3;
-
- frames = 0;
- urb->dev = ctx->subs->dev; /* we need to set this at each time */
- urb->number_of_packets = 0;
- spin_lock_irqsave(&subs->lock, flags);
- for (i = 0; i < ctx->packets; i++) {
- counts = snd_usb_audio_next_packet_size(subs);
- /* set up descriptor */
- urb->iso_frame_desc[i].offset = frames * stride;
- urb->iso_frame_desc[i].length = counts * stride;
- frames += counts;
- urb->number_of_packets++;
- subs->transfer_done += counts;
- if (subs->transfer_done >= runtime->period_size) {
- subs->transfer_done -= runtime->period_size;
- period_elapsed = 1;
- if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
- if (subs->transfer_done > 0) {
- /* FIXME: fill-max mode is not
- * supported yet */
- frames -= subs->transfer_done;
- counts -= subs->transfer_done;
- urb->iso_frame_desc[i].length =
- counts * stride;
- subs->transfer_done = 0;
- }
- i++;
- if (i < ctx->packets) {
- /* add a transfer delimiter */
- urb->iso_frame_desc[i].offset =
- frames * stride;
- urb->iso_frame_desc[i].length = 0;
- urb->number_of_packets++;
- }
- break;
- }
- }
- if (period_elapsed) /* finish at the period boundary */
- break;
- }
- bytes = frames * stride;
- if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
- /* err, the transferred area goes over buffer boundary. */
- unsigned int bytes1 =
- runtime->buffer_size * stride - subs->hwptr_done;
- memcpy(urb->transfer_buffer,
- runtime->dma_area + subs->hwptr_done, bytes1);
- memcpy(urb->transfer_buffer + bytes1,
- runtime->dma_area, bytes - bytes1);
- } else {
- memcpy(urb->transfer_buffer,
- runtime->dma_area + subs->hwptr_done, bytes);
- }
- subs->hwptr_done += bytes;
- if (subs->hwptr_done >= runtime->buffer_size * stride)
- subs->hwptr_done -= runtime->buffer_size * stride;
-
- /* update delay with exact number of samples queued */
- runtime->delay = subs->last_delay;
- runtime->delay += frames;
- subs->last_delay = runtime->delay;
-
- /* realign last_frame_number */
- subs->last_frame_number = usb_get_current_frame_number(subs->dev);
- subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
-
- spin_unlock_irqrestore(&subs->lock, flags);
- urb->transfer_buffer_length = bytes;
- if (period_elapsed)
- snd_pcm_period_elapsed(subs->pcm_substream);
- return 0;
-}
-
-/*
- * process after playback data complete
- * - decrease the delay count again
- */
-static int retire_playback_urb(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime,
- struct urb *urb)
-{
- unsigned long flags;
- int stride = runtime->frame_bits >> 3;
- int processed = urb->transfer_buffer_length / stride;
- int est_delay;
-
- spin_lock_irqsave(&subs->lock, flags);
-
- est_delay = snd_usb_pcm_delay(subs, runtime->rate);
- /* update delay with exact number of samples played */
- if (processed > subs->last_delay)
- subs->last_delay = 0;
- else
- subs->last_delay -= processed;
- runtime->delay = subs->last_delay;
-
- /*
- * Report when delay estimate is off by more than 2ms.
- * The error should be lower than 2ms since the estimate relies
- * on two reads of a counter updated every ms.
- */
- if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
- snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
- est_delay, subs->last_delay);
-
- spin_unlock_irqrestore(&subs->lock, flags);
- return 0;
+ if (u->buffer_size)
+ usb_free_coherent(u->ep->chip->dev, u->buffer_size,
+ u->urb->transfer_buffer,
+ u->urb->transfer_dma);
+ usb_free_urb(u->urb);
+ u->urb = NULL;
}
static const char *usb_error_string(int err)
@@ -797,170 +118,1044 @@
}
}
-/*
- * set up and start data/sync urbs
+/**
+ * snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type
+ *
+ * @ep: The snd_usb_endpoint
+ *
+ * Determine whether an endpoint is driven by an implicit feedback
+ * data endpoint source.
*/
-static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime)
+int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep)
{
- unsigned int i;
+ return ep->sync_master &&
+ ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA &&
+ ep->type == SND_USB_ENDPOINT_TYPE_DATA &&
+ usb_pipeout(ep->pipe);
+}
+
+/*
+ * For streaming based on information derived from sync endpoints,
+ * prepare_outbound_urb_sizes() will call next_packet_size() to
+ * determine the number of samples to be sent in the next packet.
+ *
+ * For implicit feedback, next_packet_size() is unused.
+ */
+static int next_packet_size(struct snd_usb_endpoint *ep)
+{
+ unsigned long flags;
+ int ret;
+
+ if (ep->fill_max)
+ return ep->maxframesize;
+
+ spin_lock_irqsave(&ep->lock, flags);
+ ep->phase = (ep->phase & 0xffff)
+ + (ep->freqm << ep->datainterval);
+ ret = min(ep->phase >> 16, ep->maxframesize);
+ spin_unlock_irqrestore(&ep->lock, flags);
+
+ return ret;
+}
+
+static void retire_outbound_urb(struct snd_usb_endpoint *ep,
+ struct snd_urb_ctx *urb_ctx)
+{
+ if (ep->retire_data_urb)
+ ep->retire_data_urb(ep->data_subs, urb_ctx->urb);
+}
+
+static void retire_inbound_urb(struct snd_usb_endpoint *ep,
+ struct snd_urb_ctx *urb_ctx)
+{
+ struct urb *urb = urb_ctx->urb;
+
+ if (ep->sync_slave)
+ snd_usb_handle_sync_urb(ep->sync_slave, ep, urb);
+
+ if (ep->retire_data_urb)
+ ep->retire_data_urb(ep->data_subs, urb);
+}
+
+static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep,
+ struct snd_urb_ctx *ctx)
+{
+ int i;
+
+ for (i = 0; i < ctx->packets; ++i)
+ ctx->packet_size[i] = next_packet_size(ep);
+}
+
+/*
+ * Prepare a PLAYBACK urb for submission to the bus.
+ */
+static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
+ struct snd_urb_ctx *ctx)
+{
+ int i;
+ struct urb *urb = ctx->urb;
+ unsigned char *cp = urb->transfer_buffer;
+
+ urb->dev = ep->chip->dev; /* we need to set this at each time */
+
+ switch (ep->type) {
+ case SND_USB_ENDPOINT_TYPE_DATA:
+ if (ep->prepare_data_urb) {
+ ep->prepare_data_urb(ep->data_subs, urb);
+ } else {
+ /* no data provider, so send silence */
+ unsigned int offs = 0;
+ for (i = 0; i < ctx->packets; ++i) {
+ int counts = ctx->packet_size[i];
+ urb->iso_frame_desc[i].offset = offs * ep->stride;
+ urb->iso_frame_desc[i].length = counts * ep->stride;
+ offs += counts;
+ }
+
+ urb->number_of_packets = ctx->packets;
+ urb->transfer_buffer_length = offs * ep->stride;
+ memset(urb->transfer_buffer, ep->silence_value,
+ offs * ep->stride);
+ }
+ break;
+
+ case SND_USB_ENDPOINT_TYPE_SYNC:
+ if (snd_usb_get_speed(ep->chip->dev) >= USB_SPEED_HIGH) {
+ /*
+ * fill the length and offset of each urb descriptor.
+ * the fixed 12.13 frequency is passed as 16.16 through the pipe.
+ */
+ urb->iso_frame_desc[0].length = 4;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = ep->freqn;
+ cp[1] = ep->freqn >> 8;
+ cp[2] = ep->freqn >> 16;
+ cp[3] = ep->freqn >> 24;
+ } else {
+ /*
+ * fill the length and offset of each urb descriptor.
+ * the fixed 10.14 frequency is passed through the pipe.
+ */
+ urb->iso_frame_desc[0].length = 3;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = ep->freqn >> 2;
+ cp[1] = ep->freqn >> 10;
+ cp[2] = ep->freqn >> 18;
+ }
+
+ break;
+ }
+}
+
+/*
+ * Prepare a CAPTURE or SYNC urb for submission to the bus.
+ */
+static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep,
+ struct snd_urb_ctx *urb_ctx)
+{
+ int i, offs;
+ struct urb *urb = urb_ctx->urb;
+
+ urb->dev = ep->chip->dev; /* we need to set this at each time */
+
+ switch (ep->type) {
+ case SND_USB_ENDPOINT_TYPE_DATA:
+ offs = 0;
+ for (i = 0; i < urb_ctx->packets; i++) {
+ urb->iso_frame_desc[i].offset = offs;
+ urb->iso_frame_desc[i].length = ep->curpacksize;
+ offs += ep->curpacksize;
+ }
+
+ urb->transfer_buffer_length = offs;
+ urb->number_of_packets = urb_ctx->packets;
+ break;
+
+ case SND_USB_ENDPOINT_TYPE_SYNC:
+ urb->iso_frame_desc[0].length = min(4u, ep->syncmaxsize);
+ urb->iso_frame_desc[0].offset = 0;
+ break;
+ }
+}
+
+/*
+ * Send output urbs that have been prepared previously. URBs are dequeued
+ * from ep->ready_playback_urbs and in case there there aren't any available
+ * or there are no packets that have been prepared, this function does
+ * nothing.
+ *
+ * The reason why the functionality of sending and preparing URBs is separated
+ * is that host controllers don't guarantee the order in which they return
+ * inbound and outbound packets to their submitters.
+ *
+ * This function is only used for implicit feedback endpoints. For endpoints
+ * driven by dedicated sync endpoints, URBs are immediately re-submitted
+ * from their completion handler.
+ */
+static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
+{
+ while (test_bit(EP_FLAG_RUNNING, &ep->flags)) {
+
+ unsigned long flags;
+ struct snd_usb_packet_info *uninitialized_var(packet);
+ struct snd_urb_ctx *ctx = NULL;
+ struct urb *urb;
+ int err, i;
+
+ spin_lock_irqsave(&ep->lock, flags);
+ if (ep->next_packet_read_pos != ep->next_packet_write_pos) {
+ packet = ep->next_packet + ep->next_packet_read_pos;
+ ep->next_packet_read_pos++;
+ ep->next_packet_read_pos %= MAX_URBS;
+
+ /* take URB out of FIFO */
+ if (!list_empty(&ep->ready_playback_urbs))
+ ctx = list_first_entry(&ep->ready_playback_urbs,
+ struct snd_urb_ctx, ready_list);
+ }
+ spin_unlock_irqrestore(&ep->lock, flags);
+
+ if (ctx == NULL)
+ return;
+
+ list_del_init(&ctx->ready_list);
+ urb = ctx->urb;
+
+ /* copy over the length information */
+ for (i = 0; i < packet->packets; i++)
+ ctx->packet_size[i] = packet->packet_size[i];
+
+ /* call the data handler to fill in playback data */
+ prepare_outbound_urb(ep, ctx);
+
+ err = usb_submit_urb(ctx->urb, GFP_ATOMIC);
+ if (err < 0)
+ snd_printk(KERN_ERR "Unable to submit urb #%d: %d (urb %p)\n",
+ ctx->index, err, ctx->urb);
+ else
+ set_bit(ctx->index, &ep->active_mask);
+ }
+}
+
+/*
+ * complete callback for urbs
+ */
+static void snd_complete_urb(struct urb *urb)
+{
+ struct snd_urb_ctx *ctx = urb->context;
+ struct snd_usb_endpoint *ep = ctx->ep;
int err;
- if (subs->stream->chip->shutdown)
- return -EBADFD;
+ if (unlikely(urb->status == -ENOENT || /* unlinked */
+ urb->status == -ENODEV || /* device removed */
+ urb->status == -ECONNRESET || /* unlinked */
+ urb->status == -ESHUTDOWN || /* device disabled */
+ ep->chip->shutdown)) /* device disconnected */
+ goto exit_clear;
- for (i = 0; i < subs->nurbs; i++) {
- if (snd_BUG_ON(!subs->dataurb[i].urb))
- return -EINVAL;
- if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) {
- snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i);
- goto __error;
+ if (usb_pipeout(ep->pipe)) {
+ retire_outbound_urb(ep, ctx);
+ /* can be stopped during retire callback */
+ if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
+ goto exit_clear;
+
+ if (snd_usb_endpoint_implict_feedback_sink(ep)) {
+ unsigned long flags;
+
+ spin_lock_irqsave(&ep->lock, flags);
+ list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
+ spin_unlock_irqrestore(&ep->lock, flags);
+ queue_pending_output_urbs(ep);
+
+ goto exit_clear;
+ }
+
+ prepare_outbound_urb_sizes(ep, ctx);
+ prepare_outbound_urb(ep, ctx);
+ } else {
+ retire_inbound_urb(ep, ctx);
+ /* can be stopped during retire callback */
+ if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
+ goto exit_clear;
+
+ prepare_inbound_urb(ep, ctx);
+ }
+
+ err = usb_submit_urb(urb, GFP_ATOMIC);
+ if (err == 0)
+ return;
+
+ snd_printk(KERN_ERR "cannot submit urb (err = %d)\n", err);
+ //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+
+exit_clear:
+ clear_bit(ctx->index, &ep->active_mask);
+}
+
+/**
+ * snd_usb_add_endpoint: Add an endpoint to an USB audio chip
+ *
+ * @chip: The chip
+ * @alts: The USB host interface
+ * @ep_num: The number of the endpoint to use
+ * @direction: SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE
+ * @type: SND_USB_ENDPOINT_TYPE_DATA or SND_USB_ENDPOINT_TYPE_SYNC
+ *
+ * If the requested endpoint has not been added to the given chip before,
+ * a new instance is created. Otherwise, a pointer to the previoulsy
+ * created instance is returned. In case of any error, NULL is returned.
+ *
+ * New endpoints will be added to chip->ep_list and must be freed by
+ * calling snd_usb_endpoint_free().
+ */
+struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts,
+ int ep_num, int direction, int type)
+{
+ struct list_head *p;
+ struct snd_usb_endpoint *ep;
+ int ret, is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
+
+ mutex_lock(&chip->mutex);
+
+ list_for_each(p, &chip->ep_list) {
+ ep = list_entry(p, struct snd_usb_endpoint, list);
+ if (ep->ep_num == ep_num &&
+ ep->iface == alts->desc.bInterfaceNumber &&
+ ep->alt_idx == alts->desc.bAlternateSetting) {
+ snd_printdd(KERN_DEBUG "Re-using EP %x in iface %d,%d @%p\n",
+ ep_num, ep->iface, ep->alt_idx, ep);
+ goto __exit_unlock;
}
}
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- if (snd_BUG_ON(!subs->syncurb[i].urb))
- return -EINVAL;
- if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) {
- snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i);
- goto __error;
+
+ snd_printdd(KERN_DEBUG "Creating new %s %s endpoint #%x\n",
+ is_playback ? "playback" : "capture",
+ type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync",
+ ep_num);
+
+ /* select the alt setting once so the endpoints become valid */
+ ret = usb_set_interface(chip->dev, alts->desc.bInterfaceNumber,
+ alts->desc.bAlternateSetting);
+ if (ret < 0) {
+ snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n",
+ __func__, ret);
+ ep = NULL;
+ goto __exit_unlock;
+ }
+
+ ep = kzalloc(sizeof(*ep), GFP_KERNEL);
+ if (!ep)
+ goto __exit_unlock;
+
+ ep->chip = chip;
+ spin_lock_init(&ep->lock);
+ ep->type = type;
+ ep->ep_num = ep_num;
+ ep->iface = alts->desc.bInterfaceNumber;
+ ep->alt_idx = alts->desc.bAlternateSetting;
+ INIT_LIST_HEAD(&ep->ready_playback_urbs);
+ ep_num &= USB_ENDPOINT_NUMBER_MASK;
+
+ if (is_playback)
+ ep->pipe = usb_sndisocpipe(chip->dev, ep_num);
+ else
+ ep->pipe = usb_rcvisocpipe(chip->dev, ep_num);
+
+ if (type == SND_USB_ENDPOINT_TYPE_SYNC) {
+ if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ get_endpoint(alts, 1)->bRefresh >= 1 &&
+ get_endpoint(alts, 1)->bRefresh <= 9)
+ ep->syncinterval = get_endpoint(alts, 1)->bRefresh;
+ else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL)
+ ep->syncinterval = 1;
+ else if (get_endpoint(alts, 1)->bInterval >= 1 &&
+ get_endpoint(alts, 1)->bInterval <= 16)
+ ep->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
+ else
+ ep->syncinterval = 3;
+
+ ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
+ }
+
+ list_add_tail(&ep->list, &chip->ep_list);
+
+__exit_unlock:
+ mutex_unlock(&chip->mutex);
+
+ return ep;
+}
+
+/*
+ * wait until all urbs are processed.
+ */
+static int wait_clear_urbs(struct snd_usb_endpoint *ep)
+{
+ unsigned long end_time = jiffies + msecs_to_jiffies(1000);
+ unsigned int i;
+ int alive;
+
+ do {
+ alive = 0;
+ for (i = 0; i < ep->nurbs; i++)
+ if (test_bit(i, &ep->active_mask))
+ alive++;
+
+ if (!alive)
+ break;
+
+ schedule_timeout_uninterruptible(1);
+ } while (time_before(jiffies, end_time));
+
+ if (alive)
+ snd_printk(KERN_ERR "timeout: still %d active urbs on EP #%x\n",
+ alive, ep->ep_num);
+
+ return 0;
+}
+
+/*
+ * unlink active urbs.
+ */
+static int deactivate_urbs(struct snd_usb_endpoint *ep, int force, int can_sleep)
+{
+ unsigned int i;
+ int async;
+
+ if (!force && ep->chip->shutdown) /* to be sure... */
+ return -EBADFD;
+
+ async = !can_sleep && ep->chip->async_unlink;
+
+ clear_bit(EP_FLAG_RUNNING, &ep->flags);
+
+ INIT_LIST_HEAD(&ep->ready_playback_urbs);
+ ep->next_packet_read_pos = 0;
+ ep->next_packet_write_pos = 0;
+
+ if (!async && in_interrupt())
+ return 0;
+
+ for (i = 0; i < ep->nurbs; i++) {
+ if (test_bit(i, &ep->active_mask)) {
+ if (!test_and_set_bit(i, &ep->unlink_mask)) {
+ struct urb *u = ep->urb[i].urb;
+ if (async)
+ usb_unlink_urb(u);
+ else
+ usb_kill_urb(u);
}
}
}
- subs->active_mask = 0;
- subs->unlink_mask = 0;
- subs->running = 1;
- for (i = 0; i < subs->nurbs; i++) {
- err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC);
+ return 0;
+}
+
+/*
+ * release an endpoint's urbs
+ */
+static void release_urbs(struct snd_usb_endpoint *ep, int force)
+{
+ int i;
+
+ /* route incoming urbs to nirvana */
+ ep->retire_data_urb = NULL;
+ ep->prepare_data_urb = NULL;
+
+ /* stop urbs */
+ deactivate_urbs(ep, force, 1);
+ wait_clear_urbs(ep);
+
+ for (i = 0; i < ep->nurbs; i++)
+ release_urb_ctx(&ep->urb[i]);
+
+ if (ep->syncbuf)
+ usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
+ ep->syncbuf, ep->sync_dma);
+
+ ep->syncbuf = NULL;
+ ep->nurbs = 0;
+}
+
+/*
+ * configure a data endpoint
+ */
+static int data_ep_set_params(struct snd_usb_endpoint *ep,
+ struct snd_pcm_hw_params *hw_params,
+ struct audioformat *fmt,
+ struct snd_usb_endpoint *sync_ep)
+{
+ unsigned int maxsize, i, urb_packs, total_packs, packs_per_ms;
+ int period_bytes = params_period_bytes(hw_params);
+ int format = params_format(hw_params);
+ int is_playback = usb_pipeout(ep->pipe);
+ int frame_bits = snd_pcm_format_physical_width(params_format(hw_params)) *
+ params_channels(hw_params);
+
+ ep->datainterval = fmt->datainterval;
+ ep->stride = frame_bits >> 3;
+ ep->silence_value = format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0;
+
+ /* calculate max. frequency */
+ if (ep->maxpacksize) {
+ /* whatever fits into a max. size packet */
+ maxsize = ep->maxpacksize;
+ ep->freqmax = (maxsize / (frame_bits >> 3))
+ << (16 - ep->datainterval);
+ } else {
+ /* no max. packet size: just take 25% higher than nominal */
+ ep->freqmax = ep->freqn + (ep->freqn >> 2);
+ maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
+ >> (16 - ep->datainterval);
+ }
+
+ if (ep->fill_max)
+ ep->curpacksize = ep->maxpacksize;
+ else
+ ep->curpacksize = maxsize;
+
+ if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL)
+ packs_per_ms = 8 >> ep->datainterval;
+ else
+ packs_per_ms = 1;
+
+ if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) {
+ urb_packs = max(ep->chip->nrpacks, 1);
+ urb_packs = min(urb_packs, (unsigned int) MAX_PACKS);
+ } else {
+ urb_packs = 1;
+ }
+
+ urb_packs *= packs_per_ms;
+
+ if (sync_ep && !snd_usb_endpoint_implict_feedback_sink(ep))
+ urb_packs = min(urb_packs, 1U << sync_ep->syncinterval);
+
+ /* decide how many packets to be used */
+ if (is_playback && !snd_usb_endpoint_implict_feedback_sink(ep)) {
+ unsigned int minsize, maxpacks;
+ /* determine how small a packet can be */
+ minsize = (ep->freqn >> (16 - ep->datainterval))
+ * (frame_bits >> 3);
+ /* with sync from device, assume it can be 12% lower */
+ if (sync_ep)
+ minsize -= minsize >> 3;
+ minsize = max(minsize, 1u);
+ total_packs = (period_bytes + minsize - 1) / minsize;
+ /* we need at least two URBs for queueing */
+ if (total_packs < 2) {
+ total_packs = 2;
+ } else {
+ /* and we don't want too long a queue either */
+ maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
+ total_packs = min(total_packs, maxpacks);
+ }
+ } else {
+ while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+ urb_packs >>= 1;
+ total_packs = MAX_URBS * urb_packs;
+ }
+
+ ep->nurbs = (total_packs + urb_packs - 1) / urb_packs;
+ if (ep->nurbs > MAX_URBS) {
+ /* too much... */
+ ep->nurbs = MAX_URBS;
+ total_packs = MAX_URBS * urb_packs;
+ } else if (ep->nurbs < 2) {
+ /* too little - we need at least two packets
+ * to ensure contiguous playback/capture
+ */
+ ep->nurbs = 2;
+ }
+
+ /* allocate and initialize data urbs */
+ for (i = 0; i < ep->nurbs; i++) {
+ struct snd_urb_ctx *u = &ep->urb[i];
+ u->index = i;
+ u->ep = ep;
+ u->packets = (i + 1) * total_packs / ep->nurbs
+ - i * total_packs / ep->nurbs;
+ u->buffer_size = maxsize * u->packets;
+
+ if (fmt->fmt_type == UAC_FORMAT_TYPE_II)
+ u->packets++; /* for transfer delimiter */
+ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
+ if (!u->urb)
+ goto out_of_memory;
+
+ u->urb->transfer_buffer =
+ usb_alloc_coherent(ep->chip->dev, u->buffer_size,
+ GFP_KERNEL, &u->urb->transfer_dma);
+ if (!u->urb->transfer_buffer)
+ goto out_of_memory;
+ u->urb->pipe = ep->pipe;
+ u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
+ u->urb->interval = 1 << ep->datainterval;
+ u->urb->context = u;
+ u->urb->complete = snd_complete_urb;
+ INIT_LIST_HEAD(&u->ready_list);
+ }
+
+ return 0;
+
+out_of_memory:
+ release_urbs(ep, 0);
+ return -ENOMEM;
+}
+
+/*
+ * configure a sync endpoint
+ */
+static int sync_ep_set_params(struct snd_usb_endpoint *ep,
+ struct snd_pcm_hw_params *hw_params,
+ struct audioformat *fmt)
+{
+ int i;
+
+ ep->syncbuf = usb_alloc_coherent(ep->chip->dev, SYNC_URBS * 4,
+ GFP_KERNEL, &ep->sync_dma);
+ if (!ep->syncbuf)
+ return -ENOMEM;
+
+ for (i = 0; i < SYNC_URBS; i++) {
+ struct snd_urb_ctx *u = &ep->urb[i];
+ u->index = i;
+ u->ep = ep;
+ u->packets = 1;
+ u->urb = usb_alloc_urb(1, GFP_KERNEL);
+ if (!u->urb)
+ goto out_of_memory;
+ u->urb->transfer_buffer = ep->syncbuf + i * 4;
+ u->urb->transfer_dma = ep->sync_dma + i * 4;
+ u->urb->transfer_buffer_length = 4;
+ u->urb->pipe = ep->pipe;
+ u->urb->transfer_flags = URB_ISO_ASAP |
+ URB_NO_TRANSFER_DMA_MAP;
+ u->urb->number_of_packets = 1;
+ u->urb->interval = 1 << ep->syncinterval;
+ u->urb->context = u;
+ u->urb->complete = snd_complete_urb;
+ }
+
+ ep->nurbs = SYNC_URBS;
+
+ return 0;
+
+out_of_memory:
+ release_urbs(ep, 0);
+ return -ENOMEM;
+}
+
+/**
+ * snd_usb_endpoint_set_params: configure an snd_usb_endpoint
+ *
+ * @ep: the snd_usb_endpoint to configure
+ * @hw_params: the hardware parameters
+ * @fmt: the USB audio format information
+ * @sync_ep: the sync endpoint to use, if any
+ *
+ * Determine the number of URBs to be used on this endpoint.
+ * An endpoint must be configured before it can be started.
+ * An endpoint that is already running can not be reconfigured.
+ */
+int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
+ struct snd_pcm_hw_params *hw_params,
+ struct audioformat *fmt,
+ struct snd_usb_endpoint *sync_ep)
+{
+ int err;
+
+ if (ep->use_count != 0) {
+ snd_printk(KERN_WARNING "Unable to change format on ep #%x: already in use\n",
+ ep->ep_num);
+ return -EBUSY;
+ }
+
+ /* release old buffers, if any */
+ release_urbs(ep, 0);
+
+ ep->datainterval = fmt->datainterval;
+ ep->maxpacksize = fmt->maxpacksize;
+ ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX);
+
+ if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL)
+ ep->freqn = get_usb_full_speed_rate(params_rate(hw_params));
+ else
+ ep->freqn = get_usb_high_speed_rate(params_rate(hw_params));
+
+ /* calculate the frequency in 16.16 format */
+ ep->freqm = ep->freqn;
+ ep->freqshift = INT_MIN;
+
+ ep->phase = 0;
+
+ switch (ep->type) {
+ case SND_USB_ENDPOINT_TYPE_DATA:
+ err = data_ep_set_params(ep, hw_params, fmt, sync_ep);
+ break;
+ case SND_USB_ENDPOINT_TYPE_SYNC:
+ err = sync_ep_set_params(ep, hw_params, fmt);
+ break;
+ default:
+ err = -EINVAL;
+ }
+
+ snd_printdd(KERN_DEBUG "Setting params for ep #%x (type %d, %d urbs), ret=%d\n",
+ ep->ep_num, ep->type, ep->nurbs, err);
+
+ return err;
+}
+
+/**
+ * snd_usb_endpoint_start: start an snd_usb_endpoint
+ *
+ * @ep: the endpoint to start
+ *
+ * A call to this function will increment the use count of the endpoint.
+ * In case it is not already running, the URBs for this endpoint will be
+ * submitted. Otherwise, this function does nothing.
+ *
+ * Must be balanced to calls of snd_usb_endpoint_stop().
+ *
+ * Returns an error if the URB submission failed, 0 in all other cases.
+ */
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
+{
+ int err;
+ unsigned int i;
+
+ if (ep->chip->shutdown)
+ return -EBADFD;
+
+ /* already running? */
+ if (++ep->use_count != 1)
+ return 0;
+
+ if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags)))
+ return -EINVAL;
+
+ /* just to be sure */
+ deactivate_urbs(ep, 0, 1);
+ wait_clear_urbs(ep);
+
+ ep->active_mask = 0;
+ ep->unlink_mask = 0;
+ ep->phase = 0;
+
+ /*
+ * If this endpoint has a data endpoint as implicit feedback source,
+ * don't start the urbs here. Instead, mark them all as available,
+ * wait for the record urbs to return and queue the playback urbs
+ * from that context.
+ */
+
+ set_bit(EP_FLAG_RUNNING, &ep->flags);
+
+ if (snd_usb_endpoint_implict_feedback_sink(ep)) {
+ for (i = 0; i < ep->nurbs; i++) {
+ struct snd_urb_ctx *ctx = ep->urb + i;
+ list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
+ }
+
+ return 0;
+ }
+
+ for (i = 0; i < ep->nurbs; i++) {
+ struct urb *urb = ep->urb[i].urb;
+
+ if (snd_BUG_ON(!urb))
+ goto __error;
+
+ if (usb_pipeout(ep->pipe)) {
+ prepare_outbound_urb_sizes(ep, urb->context);
+ prepare_outbound_urb(ep, urb->context);
+ } else {
+ prepare_inbound_urb(ep, urb->context);
+ }
+
+ err = usb_submit_urb(urb, GFP_ATOMIC);
if (err < 0) {
- snd_printk(KERN_ERR "cannot submit datapipe "
- "for urb %d, error %d: %s\n",
+ snd_printk(KERN_ERR "cannot submit urb %d, error %d: %s\n",
i, err, usb_error_string(err));
goto __error;
}
- set_bit(i, &subs->active_mask);
+ set_bit(i, &ep->active_mask);
}
- if (subs->syncpipe) {
- for (i = 0; i < SYNC_URBS; i++) {
- err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC);
- if (err < 0) {
- snd_printk(KERN_ERR "cannot submit syncpipe "
- "for urb %d, error %d: %s\n",
- i, err, usb_error_string(err));
- goto __error;
- }
- set_bit(i + 16, &subs->active_mask);
- }
- }
+
return 0;
- __error:
- // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN);
- deactivate_urbs(subs, 0, 0);
+__error:
+ clear_bit(EP_FLAG_RUNNING, &ep->flags);
+ ep->use_count--;
+ deactivate_urbs(ep, 0, 0);
return -EPIPE;
}
-
-/*
+/**
+ * snd_usb_endpoint_stop: stop an snd_usb_endpoint
+ *
+ * @ep: the endpoint to stop (may be NULL)
+ *
+ * A call to this function will decrement the use count of the endpoint.
+ * In case the last user has requested the endpoint stop, the URBs will
+ * actually be deactivated.
+ *
+ * Must be balanced to calls of snd_usb_endpoint_start().
*/
-static struct snd_urb_ops audio_urb_ops[2] = {
- {
- .prepare = prepare_nodata_playback_urb,
- .retire = retire_playback_urb,
- .prepare_sync = prepare_playback_sync_urb,
- .retire_sync = retire_playback_sync_urb,
- },
- {
- .prepare = prepare_capture_urb,
- .retire = retire_capture_urb,
- .prepare_sync = prepare_capture_sync_urb,
- .retire_sync = retire_capture_sync_urb,
- },
-};
+void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
+ int force, int can_sleep, int wait)
+{
+ if (!ep)
+ return;
-/*
- * initialize the substream instance.
+ if (snd_BUG_ON(ep->use_count == 0))
+ return;
+
+ if (snd_BUG_ON(!test_bit(EP_FLAG_ACTIVATED, &ep->flags)))
+ return;
+
+ if (--ep->use_count == 0) {
+ deactivate_urbs(ep, force, can_sleep);
+ ep->data_subs = NULL;
+ ep->sync_slave = NULL;
+ ep->retire_data_urb = NULL;
+ ep->prepare_data_urb = NULL;
+
+ if (wait)
+ wait_clear_urbs(ep);
+ }
+}
+
+/**
+ * snd_usb_endpoint_activate: activate an snd_usb_endpoint
+ *
+ * @ep: the endpoint to activate
+ *
+ * If the endpoint is not currently in use, this functions will select the
+ * correct alternate interface setting for the interface of this endpoint.
+ *
+ * In case of any active users, this functions does nothing.
+ *
+ * Returns an error if usb_set_interface() failed, 0 in all other
+ * cases.
*/
-
-void snd_usb_init_substream(struct snd_usb_stream *as,
- int stream, struct audioformat *fp)
+int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep)
{
- struct snd_usb_substream *subs = &as->substream[stream];
-
- INIT_LIST_HEAD(&subs->fmt_list);
- spin_lock_init(&subs->lock);
-
- subs->stream = as;
- subs->direction = stream;
- subs->dev = as->chip->dev;
- subs->txfr_quirk = as->chip->txfr_quirk;
- subs->ops = audio_urb_ops[stream];
- if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH)
- subs->ops.prepare_sync = prepare_capture_sync_urb_hs;
-
- snd_usb_set_pcm_ops(as->pcm, stream);
-
- list_add_tail(&fp->list, &subs->fmt_list);
- subs->formats |= fp->formats;
- subs->endpoint = fp->endpoint;
- subs->num_formats++;
- subs->fmt_type = fp->fmt_type;
-}
-
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd)
-{
- struct snd_usb_substream *subs = substream->runtime->private_data;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- subs->ops.prepare = prepare_playback_urb;
+ if (ep->use_count != 0)
return 0;
- case SNDRV_PCM_TRIGGER_STOP:
- return deactivate_urbs(subs, 0, 0);
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- subs->ops.prepare = prepare_nodata_playback_urb;
+
+ if (!ep->chip->shutdown &&
+ !test_and_set_bit(EP_FLAG_ACTIVATED, &ep->flags)) {
+ int ret;
+
+ ret = usb_set_interface(ep->chip->dev, ep->iface, ep->alt_idx);
+ if (ret < 0) {
+ snd_printk(KERN_ERR "%s() usb_set_interface() failed, ret = %d\n",
+ __func__, ret);
+ clear_bit(EP_FLAG_ACTIVATED, &ep->flags);
+ return ret;
+ }
+
return 0;
}
- return -EINVAL;
+ return -EBUSY;
}
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+/**
+ * snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint
+ *
+ * @ep: the endpoint to deactivate
+ *
+ * If the endpoint is not currently in use, this functions will select the
+ * alternate interface setting 0 for the interface of this endpoint.
+ *
+ * In case of any active users, this functions does nothing.
+ *
+ * Returns an error if usb_set_interface() failed, 0 in all other
+ * cases.
+ */
+int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep)
{
- struct snd_usb_substream *subs = substream->runtime->private_data;
+ if (!ep)
+ return -EINVAL;
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- subs->ops.retire = retire_capture_urb;
- return start_urbs(subs, substream->runtime);
- case SNDRV_PCM_TRIGGER_STOP:
- return deactivate_urbs(subs, 0, 0);
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- subs->ops.retire = retire_paused_capture_urb;
+ if (ep->use_count != 0)
return 0;
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- subs->ops.retire = retire_capture_urb;
+
+ if (!ep->chip->shutdown &&
+ test_and_clear_bit(EP_FLAG_ACTIVATED, &ep->flags)) {
+ int ret;
+
+ ret = usb_set_interface(ep->chip->dev, ep->iface, 0);
+ if (ret < 0) {
+ snd_printk(KERN_ERR "%s(): usb_set_interface() failed, ret = %d\n",
+ __func__, ret);
+ return ret;
+ }
+
return 0;
}
- return -EINVAL;
+ return -EBUSY;
}
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime)
+/**
+ * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint
+ *
+ * @ep: the list header of the endpoint to free
+ *
+ * This function does not care for the endpoint's use count but will tear
+ * down all the streaming URBs immediately and free all resources.
+ */
+void snd_usb_endpoint_free(struct list_head *head)
{
- /* clear urbs (to be sure) */
- deactivate_urbs(subs, 0, 1);
- wait_clear_urbs(subs);
+ struct snd_usb_endpoint *ep;
- /* for playback, submit the URBs now; otherwise, the first hwptr_done
- * updates for all URBs would happen at the same time when starting */
- if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) {
- subs->ops.prepare = prepare_nodata_playback_urb;
- return start_urbs(subs, runtime);
+ ep = list_entry(head, struct snd_usb_endpoint, list);
+ release_urbs(ep, 1);
+ kfree(ep);
+}
+
+/**
+ * snd_usb_handle_sync_urb: parse an USB sync packet
+ *
+ * @ep: the endpoint to handle the packet
+ * @sender: the sending endpoint
+ * @urb: the received packet
+ *
+ * This function is called from the context of an endpoint that received
+ * the packet and is used to let another endpoint object handle the payload.
+ */
+void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
+ struct snd_usb_endpoint *sender,
+ const struct urb *urb)
+{
+ int shift;
+ unsigned int f;
+ unsigned long flags;
+
+ snd_BUG_ON(ep == sender);
+
+ /*
+ * In case the endpoint is operating in implicit feedback mode, prepare
+ * a new outbound URB that has the same layout as the received packet
+ * and add it to the list of pending urbs. queue_pending_output_urbs()
+ * will take care of them later.
+ */
+ if (snd_usb_endpoint_implict_feedback_sink(ep) &&
+ ep->use_count != 0) {
+
+ /* implicit feedback case */
+ int i, bytes = 0;
+ struct snd_urb_ctx *in_ctx;
+ struct snd_usb_packet_info *out_packet;
+
+ in_ctx = urb->context;
+
+ /* Count overall packet size */
+ for (i = 0; i < in_ctx->packets; i++)
+ if (urb->iso_frame_desc[i].status == 0)
+ bytes += urb->iso_frame_desc[i].actual_length;
+
+ /*
+ * skip empty packets. At least M-Audio's Fast Track Ultra stops
+ * streaming once it received a 0-byte OUT URB
+ */
+ if (bytes == 0)
+ return;
+
+ spin_lock_irqsave(&ep->lock, flags);
+ out_packet = ep->next_packet + ep->next_packet_write_pos;
+
+ /*
+ * Iterate through the inbound packet and prepare the lengths
+ * for the output packet. The OUT packet we are about to send
+ * will have the same amount of payload bytes than the IN
+ * packet we just received.
+ */
+
+ out_packet->packets = in_ctx->packets;
+ for (i = 0; i < in_ctx->packets; i++) {
+ if (urb->iso_frame_desc[i].status == 0)
+ out_packet->packet_size[i] =
+ urb->iso_frame_desc[i].actual_length / ep->stride;
+ else
+ out_packet->packet_size[i] = 0;
+ }
+
+ ep->next_packet_write_pos++;
+ ep->next_packet_write_pos %= MAX_URBS;
+ spin_unlock_irqrestore(&ep->lock, flags);
+ queue_pending_output_urbs(ep);
+
+ return;
}
- return 0;
+ /*
+ * process after playback sync complete
+ *
+ * Full speed devices report feedback values in 10.14 format as samples
+ * per frame, high speed devices in 16.16 format as samples per
+ * microframe.
+ *
+ * Because the Audio Class 1 spec was written before USB 2.0, many high
+ * speed devices use a wrong interpretation, some others use an
+ * entirely different format.
+ *
+ * Therefore, we cannot predict what format any particular device uses
+ * and must detect it automatically.
+ */
+
+ if (urb->iso_frame_desc[0].status != 0 ||
+ urb->iso_frame_desc[0].actual_length < 3)
+ return;
+
+ f = le32_to_cpup(urb->transfer_buffer);
+ if (urb->iso_frame_desc[0].actual_length == 3)
+ f &= 0x00ffffff;
+ else
+ f &= 0x0fffffff;
+
+ if (f == 0)
+ return;
+
+ if (unlikely(ep->freqshift == INT_MIN)) {
+ /*
+ * The first time we see a feedback value, determine its format
+ * by shifting it left or right until it matches the nominal
+ * frequency value. This assumes that the feedback does not
+ * differ from the nominal value more than +50% or -25%.
+ */
+ shift = 0;
+ while (f < ep->freqn - ep->freqn / 4) {
+ f <<= 1;
+ shift++;
+ }
+ while (f > ep->freqn + ep->freqn / 2) {
+ f >>= 1;
+ shift--;
+ }
+ ep->freqshift = shift;
+ } else if (ep->freqshift >= 0)
+ f <<= ep->freqshift;
+ else
+ f >>= -ep->freqshift;
+
+ if (likely(f >= ep->freqn - ep->freqn / 8 && f <= ep->freqmax)) {
+ /*
+ * If the frequency looks valid, set it.
+ * This value is referred to in prepare_playback_urb().
+ */
+ spin_lock_irqsave(&ep->lock, flags);
+ ep->freqm = f;
+ spin_unlock_irqrestore(&ep->lock, flags);
+ } else {
+ /*
+ * Out of range; maybe the shift value is wrong.
+ * Reset it so that we autodetect again the next time.
+ */
+ ep->freqshift = INT_MIN;
+ }
}
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 88eb63a..ee2723f 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -1,21 +1,29 @@
#ifndef __USBAUDIO_ENDPOINT_H
#define __USBAUDIO_ENDPOINT_H
-void snd_usb_init_substream(struct snd_usb_stream *as,
- int stream,
- struct audioformat *fp);
+#define SND_USB_ENDPOINT_TYPE_DATA 0
+#define SND_USB_ENDPOINT_TYPE_SYNC 1
-int snd_usb_init_substream_urbs(struct snd_usb_substream *subs,
- unsigned int period_bytes,
- unsigned int rate,
- unsigned int frame_bits);
+struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts,
+ int ep_num, int direction, int type);
-void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force);
+int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
+ struct snd_pcm_hw_params *hw_params,
+ struct audioformat *fmt,
+ struct snd_usb_endpoint *sync_ep);
-int snd_usb_substream_prepare(struct snd_usb_substream *subs,
- struct snd_pcm_runtime *runtime);
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
+void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
+ int force, int can_sleep, int wait);
+int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep);
+void snd_usb_endpoint_free(struct list_head *head);
-int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd);
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd);
+int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep);
+
+void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
+ struct snd_usb_endpoint *sender,
+ const struct urb *urb);
#endif /* __USBAUDIO_ENDPOINT_H */
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ab23869..4f40ba8 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -486,7 +486,7 @@
/*
* TLV callback for mixer volume controls
*/
-static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *_tlv)
{
struct usb_mixer_elem_info *cval = kcontrol->private_data;
@@ -770,6 +770,26 @@
struct snd_kcontrol *kctl)
{
switch (cval->mixer->chip->usb_id) {
+ case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */
+ case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
+ if (strcmp(kctl->id.name, "Effect Duration") == 0) {
+ snd_printk(KERN_INFO
+ "usb-audio: set quirk for FTU Effect Duration\n");
+ cval->min = 0x0000;
+ cval->max = 0x7f00;
+ cval->res = 0x0100;
+ break;
+ }
+ if (strcmp(kctl->id.name, "Effect Volume") == 0 ||
+ strcmp(kctl->id.name, "Effect Feedback Volume") == 0) {
+ snd_printk(KERN_INFO
+ "usb-audio: set quirks for FTU Effect Feedback/Volume\n");
+ cval->min = 0x00;
+ cval->max = 0x7f;
+ break;
+ }
+ break;
+
case USB_ID(0x0471, 0x0101):
case USB_ID(0x0471, 0x0104):
case USB_ID(0x0471, 0x0105):
@@ -1121,9 +1141,6 @@
len = snd_usb_copy_string_desc(state, nameid,
kctl->id.name, sizeof(kctl->id.name));
- /* get min/max values */
- get_min_max_with_quirks(cval, 0, kctl);
-
switch (control) {
case UAC_FU_MUTE:
case UAC_FU_VOLUME:
@@ -1155,17 +1172,7 @@
}
append_ctl_name(kctl, control == UAC_FU_MUTE ?
" Switch" : " Volume");
- if (control == UAC_FU_VOLUME) {
- check_mapped_dB(map, cval);
- if (cval->dBmin < cval->dBmax || !cval->initialized) {
- kctl->tlv.c = mixer_vol_tlv;
- kctl->vd[0].access |=
- SNDRV_CTL_ELEM_ACCESS_TLV_READ |
- SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
- }
- }
break;
-
default:
if (! len)
strlcpy(kctl->id.name, audio_feature_info[control-1].name,
@@ -1173,6 +1180,19 @@
break;
}
+ /* get min/max values */
+ get_min_max_with_quirks(cval, 0, kctl);
+
+ if (control == UAC_FU_VOLUME) {
+ check_mapped_dB(map, cval);
+ if (cval->dBmin < cval->dBmax || !cval->initialized) {
+ kctl->tlv.c = snd_usb_mixer_vol_tlv;
+ kctl->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+ }
+ }
+
range = (cval->max - cval->min) / cval->res;
/* Are there devices with volume range more than 255? I use a bit more
* to be sure. 384 is a resolution magic number found on Logitech
@@ -1388,7 +1408,7 @@
for (pin = 0; pin < input_pins; pin++) {
err = parse_audio_unit(state, desc->baSourceID[pin]);
if (err < 0)
- return err;
+ continue;
err = check_input_term(state, desc->baSourceID[pin], &iterm);
if (err < 0)
return err;
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 81b2d8a..a7f3d45 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -68,4 +68,7 @@
int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer,
struct snd_kcontrol *kctl);
+int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+ unsigned int size, unsigned int __user *_tlv);
+
#endif /* __USBMIXER_H */
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index f1324c4..41daaa2 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -288,6 +288,15 @@
{ 0 } /* terminator */
};
+static struct usbmix_name_map ebox44_map[] = {
+ { 4, NULL }, /* FU */
+ { 6, NULL }, /* MU */
+ { 7, NULL }, /* FU */
+ { 10, NULL }, /* FU */
+ { 11, NULL }, /* MU */
+ { 0 }
+};
+
/* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+"
* most importand difference is SU[8], it should be set to "Capture Source"
* to make alsamixer and PA working properly.
@@ -371,6 +380,10 @@
.map = scratch_live_map,
.ignore_ctl_error = 1,
},
+ {
+ .id = USB_ID(0x200c, 0x1018),
+ .map = ebox44_map,
+ },
{ 0 } /* terminator */
};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index ab125ee..41f4b69 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -42,6 +42,77 @@
extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
+/* private_free callback */
+static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
+{
+ kfree(kctl->private_data);
+ kctl->private_data = NULL;
+}
+
+/* This function allows for the creation of standard UAC controls.
+ * See the quirks for M-Audio FTUs or Ebox-44.
+ * If you don't want to set a TLV callback pass NULL.
+ *
+ * Since there doesn't seem to be a devices that needs a multichannel
+ * version, we keep it mono for simplicity.
+ */
+static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer,
+ unsigned int unitid,
+ unsigned int control,
+ unsigned int cmask,
+ int val_type,
+ const char *name,
+ snd_kcontrol_tlv_rw_t *tlv_callback)
+{
+ int err;
+ struct usb_mixer_elem_info *cval;
+ struct snd_kcontrol *kctl;
+
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (!cval)
+ return -ENOMEM;
+
+ cval->id = unitid;
+ cval->mixer = mixer;
+ cval->val_type = val_type;
+ cval->channels = 1;
+ cval->control = control;
+ cval->cmask = cmask;
+
+ /* get_min_max() is called only for integer volumes later,
+ * so provide a short-cut for booleans */
+ cval->min = 0;
+ cval->max = 1;
+ cval->res = 0;
+ cval->dBmin = 0;
+ cval->dBmax = 0;
+
+ /* Create control */
+ kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval);
+ if (!kctl) {
+ kfree(cval);
+ return -ENOMEM;
+ }
+
+ /* Set name */
+ snprintf(kctl->id.name, sizeof(kctl->id.name), name);
+ kctl->private_free = usb_mixer_elem_free;
+
+ /* set TLV */
+ if (tlv_callback) {
+ kctl->tlv.c = tlv_callback;
+ kctl->vd[0].access |=
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK;
+ }
+ /* Add control to mixer */
+ err = snd_usb_mixer_add_control(mixer, kctl);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
/*
* Sound Blaster remote control configuration
*
@@ -495,60 +566,218 @@
}
/* M-Audio FastTrack Ultra quirks */
+/* FTU Effect switch */
+struct snd_ftu_eff_switch_priv_val {
+ struct usb_mixer_interface *mixer;
+ int cached_value;
+ int is_cached;
+};
-/* private_free callback */
-static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
+static int snd_ftu_eff_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
{
- kfree(kctl->private_data);
- kctl->private_data = NULL;
+ static const char *texts[8] = {"Room 1",
+ "Room 2",
+ "Room 3",
+ "Hall 1",
+ "Hall 2",
+ "Plate",
+ "Delay",
+ "Echo"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item > 7)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
}
-static int snd_maudio_ftu_create_ctl(struct usb_mixer_interface *mixer,
- int in, int out, const char *name)
+static int snd_ftu_eff_switch_get(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
{
- struct usb_mixer_elem_info *cval;
- struct snd_kcontrol *kctl;
+ struct snd_usb_audio *chip;
+ struct usb_mixer_interface *mixer;
+ struct snd_ftu_eff_switch_priv_val *pval;
+ int err;
+ unsigned char value[2];
- cval = kzalloc(sizeof(*cval), GFP_KERNEL);
- if (!cval)
+ const int id = 6;
+ const int validx = 1;
+ const int val_len = 2;
+
+ value[0] = 0x00;
+ value[1] = 0x00;
+
+ pval = (struct snd_ftu_eff_switch_priv_val *)
+ kctl->private_value;
+
+ if (pval->is_cached) {
+ ucontrol->value.enumerated.item[0] = pval->cached_value;
+ return 0;
+ }
+
+ mixer = (struct usb_mixer_interface *) pval->mixer;
+ if (snd_BUG_ON(!mixer))
+ return -EINVAL;
+
+ chip = (struct snd_usb_audio *) mixer->chip;
+ if (snd_BUG_ON(!chip))
+ return -EINVAL;
+
+
+ err = snd_usb_ctl_msg(chip->dev,
+ usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+ validx << 8, snd_usb_ctrl_intf(chip) | (id << 8),
+ value, val_len);
+ if (err < 0)
+ return err;
+
+ ucontrol->value.enumerated.item[0] = value[0];
+ pval->cached_value = value[0];
+ pval->is_cached = 1;
+
+ return 0;
+}
+
+static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_usb_audio *chip;
+ struct snd_ftu_eff_switch_priv_val *pval;
+
+ struct usb_mixer_interface *mixer;
+ int changed, cur_val, err, new_val;
+ unsigned char value[2];
+
+
+ const int id = 6;
+ const int validx = 1;
+ const int val_len = 2;
+
+ changed = 0;
+
+ pval = (struct snd_ftu_eff_switch_priv_val *)
+ kctl->private_value;
+ cur_val = pval->cached_value;
+ new_val = ucontrol->value.enumerated.item[0];
+
+ mixer = (struct usb_mixer_interface *) pval->mixer;
+ if (snd_BUG_ON(!mixer))
+ return -EINVAL;
+
+ chip = (struct snd_usb_audio *) mixer->chip;
+ if (snd_BUG_ON(!chip))
+ return -EINVAL;
+
+ if (!pval->is_cached) {
+ /* Read current value */
+ err = snd_usb_ctl_msg(chip->dev,
+ usb_rcvctrlpipe(chip->dev, 0), UAC_GET_CUR,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+ validx << 8, snd_usb_ctrl_intf(chip) | (id << 8),
+ value, val_len);
+ if (err < 0)
+ return err;
+
+ cur_val = value[0];
+ pval->cached_value = cur_val;
+ pval->is_cached = 1;
+ }
+ /* update value if needed */
+ if (cur_val != new_val) {
+ value[0] = new_val;
+ value[1] = 0;
+ err = snd_usb_ctl_msg(chip->dev,
+ usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+ validx << 8, snd_usb_ctrl_intf(chip) | (id << 8),
+ value, val_len);
+ if (err < 0)
+ return err;
+
+ pval->cached_value = new_val;
+ pval->is_cached = 1;
+ changed = 1;
+ }
+
+ return changed;
+}
+
+static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer)
+{
+ static struct snd_kcontrol_new template = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Effect Program Switch",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_ftu_eff_switch_info,
+ .get = snd_ftu_eff_switch_get,
+ .put = snd_ftu_eff_switch_put
+ };
+
+ int err;
+ struct snd_kcontrol *kctl;
+ struct snd_ftu_eff_switch_priv_val *pval;
+
+ pval = kzalloc(sizeof(*pval), GFP_KERNEL);
+ if (!pval)
return -ENOMEM;
- cval->id = 5;
- cval->mixer = mixer;
- cval->val_type = USB_MIXER_S16;
- cval->channels = 1;
- cval->control = out + 1;
- cval->cmask = 1 << in;
+ pval->cached_value = 0;
+ pval->is_cached = 0;
+ pval->mixer = mixer;
- kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval);
+ template.private_value = (unsigned long) pval;
+ kctl = snd_ctl_new1(&template, mixer->chip);
if (!kctl) {
- kfree(cval);
+ kfree(pval);
return -ENOMEM;
}
- snprintf(kctl->id.name, sizeof(kctl->id.name), name);
- kctl->private_free = usb_mixer_elem_free;
- return snd_usb_mixer_add_control(mixer, kctl);
+ err = snd_ctl_add(mixer->chip->card, kctl);
+ if (err < 0)
+ return err;
+
+ return 0;
}
-static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer)
+/* Create volume controls for FTU devices*/
+static int snd_ftu_create_volume_ctls(struct usb_mixer_interface *mixer)
{
char name[64];
+ unsigned int control, cmask;
int in, out, err;
+ const unsigned int id = 5;
+ const int val_type = USB_MIXER_S16;
+
for (out = 0; out < 8; out++) {
+ control = out + 1;
for (in = 0; in < 8; in++) {
+ cmask = 1 << in;
snprintf(name, sizeof(name),
- "AIn%d - Out%d Capture Volume", in + 1, out + 1);
- err = snd_maudio_ftu_create_ctl(mixer, in, out, name);
+ "AIn%d - Out%d Capture Volume",
+ in + 1, out + 1);
+ err = snd_create_std_mono_ctl(mixer, id, control,
+ cmask, val_type, name,
+ &snd_usb_mixer_vol_tlv);
if (err < 0)
return err;
}
-
for (in = 8; in < 16; in++) {
+ cmask = 1 << in;
snprintf(name, sizeof(name),
- "DIn%d - Out%d Playback Volume", in - 7, out + 1);
- err = snd_maudio_ftu_create_ctl(mixer, in, out, name);
+ "DIn%d - Out%d Playback Volume",
+ in - 7, out + 1);
+ err = snd_create_std_mono_ctl(mixer, id, control,
+ cmask, val_type, name,
+ &snd_usb_mixer_vol_tlv);
if (err < 0)
return err;
}
@@ -557,6 +786,191 @@
return 0;
}
+/* This control needs a volume quirk, see mixer.c */
+static int snd_ftu_create_effect_volume_ctl(struct usb_mixer_interface *mixer)
+{
+ static const char name[] = "Effect Volume";
+ const unsigned int id = 6;
+ const int val_type = USB_MIXER_U8;
+ const unsigned int control = 2;
+ const unsigned int cmask = 0;
+
+ return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type,
+ name, snd_usb_mixer_vol_tlv);
+}
+
+/* This control needs a volume quirk, see mixer.c */
+static int snd_ftu_create_effect_duration_ctl(struct usb_mixer_interface *mixer)
+{
+ static const char name[] = "Effect Duration";
+ const unsigned int id = 6;
+ const int val_type = USB_MIXER_S16;
+ const unsigned int control = 3;
+ const unsigned int cmask = 0;
+
+ return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type,
+ name, snd_usb_mixer_vol_tlv);
+}
+
+/* This control needs a volume quirk, see mixer.c */
+static int snd_ftu_create_effect_feedback_ctl(struct usb_mixer_interface *mixer)
+{
+ static const char name[] = "Effect Feedback Volume";
+ const unsigned int id = 6;
+ const int val_type = USB_MIXER_U8;
+ const unsigned int control = 4;
+ const unsigned int cmask = 0;
+
+ return snd_create_std_mono_ctl(mixer, id, control, cmask, val_type,
+ name, NULL);
+}
+
+static int snd_ftu_create_effect_return_ctls(struct usb_mixer_interface *mixer)
+{
+ unsigned int cmask;
+ int err, ch;
+ char name[48];
+
+ const unsigned int id = 7;
+ const int val_type = USB_MIXER_S16;
+ const unsigned int control = 7;
+
+ for (ch = 0; ch < 4; ++ch) {
+ cmask = 1 << ch;
+ snprintf(name, sizeof(name),
+ "Effect Return %d Volume", ch + 1);
+ err = snd_create_std_mono_ctl(mixer, id, control,
+ cmask, val_type, name,
+ snd_usb_mixer_vol_tlv);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+static int snd_ftu_create_effect_send_ctls(struct usb_mixer_interface *mixer)
+{
+ unsigned int cmask;
+ int err, ch;
+ char name[48];
+
+ const unsigned int id = 5;
+ const int val_type = USB_MIXER_S16;
+ const unsigned int control = 9;
+
+ for (ch = 0; ch < 8; ++ch) {
+ cmask = 1 << ch;
+ snprintf(name, sizeof(name),
+ "Effect Send AIn%d Volume", ch + 1);
+ err = snd_create_std_mono_ctl(mixer, id, control, cmask,
+ val_type, name,
+ snd_usb_mixer_vol_tlv);
+ if (err < 0)
+ return err;
+ }
+ for (ch = 8; ch < 16; ++ch) {
+ cmask = 1 << ch;
+ snprintf(name, sizeof(name),
+ "Effect Send DIn%d Volume", ch - 7);
+ err = snd_create_std_mono_ctl(mixer, id, control, cmask,
+ val_type, name,
+ snd_usb_mixer_vol_tlv);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+static int snd_ftu_create_mixer(struct usb_mixer_interface *mixer)
+{
+ int err;
+
+ err = snd_ftu_create_volume_ctls(mixer);
+ if (err < 0)
+ return err;
+
+ err = snd_ftu_create_effect_switch(mixer);
+ if (err < 0)
+ return err;
+ err = snd_ftu_create_effect_volume_ctl(mixer);
+ if (err < 0)
+ return err;
+
+ err = snd_ftu_create_effect_duration_ctl(mixer);
+ if (err < 0)
+ return err;
+
+ err = snd_ftu_create_effect_feedback_ctl(mixer);
+ if (err < 0)
+ return err;
+
+ err = snd_ftu_create_effect_return_ctls(mixer);
+ if (err < 0)
+ return err;
+
+ err = snd_ftu_create_effect_send_ctls(mixer);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+
+/*
+ * Create mixer for Electrix Ebox-44
+ *
+ * The mixer units from this device are corrupt, and even where they
+ * are valid they presents mono controls as L and R channels of
+ * stereo. So we create a good mixer in code.
+ */
+
+static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer)
+{
+ int err;
+
+ err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN,
+ "Headphone Playback Switch", NULL);
+ if (err < 0)
+ return err;
+ err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16,
+ "Headphone A Mix Playback Volume", NULL);
+ if (err < 0)
+ return err;
+ err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16,
+ "Headphone B Mix Playback Volume", NULL);
+ if (err < 0)
+ return err;
+
+ err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN,
+ "Output Playback Switch", NULL);
+ if (err < 0)
+ return err;
+ err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16,
+ "Output A Playback Volume", NULL);
+ if (err < 0)
+ return err;
+ err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16,
+ "Output B Playback Volume", NULL);
+ if (err < 0)
+ return err;
+
+ err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN,
+ "Input Capture Switch", NULL);
+ if (err < 0)
+ return err;
+ err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16,
+ "Input A Capture Volume", NULL);
+ if (err < 0)
+ return err;
+ err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16,
+ "Input B Capture Volume", NULL);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
unsigned char samplerate_id)
{
@@ -600,7 +1014,7 @@
case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */
- err = snd_maudio_ftu_create_mixer(mixer);
+ err = snd_ftu_create_mixer(mixer);
break;
case USB_ID(0x0b05, 0x1739):
@@ -619,6 +1033,10 @@
snd_nativeinstruments_ta10_mixers,
ARRAY_SIZE(snd_nativeinstruments_ta10_mixers));
break;
+
+ case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */
+ err = snd_ebox44_create_mixer(mixer);
+ break;
}
return err;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 0eed611..cdf8b76 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -16,6 +16,7 @@
#include <linux/init.h>
#include <linux/slab.h>
+#include <linux/ratelimit.h>
#include <linux/usb.h>
#include <linux/usb/audio.h>
#include <linux/usb/audio-v2.h>
@@ -34,6 +35,9 @@
#include "clock.h"
#include "power.h"
+#define SUBSTREAM_FLAG_DATA_EP_STARTED 0
+#define SUBSTREAM_FLAG_SYNC_EP_STARTED 1
+
/* return the estimated delay based on USB frame counters */
snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
unsigned int rate)
@@ -208,6 +212,84 @@
}
}
+static int start_endpoints(struct snd_usb_substream *subs)
+{
+ int err;
+
+ if (!subs->data_endpoint)
+ return -EINVAL;
+
+ if (!test_and_set_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) {
+ struct snd_usb_endpoint *ep = subs->data_endpoint;
+
+ snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep);
+
+ ep->data_subs = subs;
+ err = snd_usb_endpoint_start(ep);
+ if (err < 0) {
+ clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
+ return err;
+ }
+ }
+
+ if (subs->sync_endpoint &&
+ !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
+ struct snd_usb_endpoint *ep = subs->sync_endpoint;
+
+ snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep);
+
+ ep->sync_slave = subs->data_endpoint;
+ err = snd_usb_endpoint_start(ep);
+ if (err < 0) {
+ clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static void stop_endpoints(struct snd_usb_substream *subs,
+ int force, int can_sleep, int wait)
+{
+ if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags))
+ snd_usb_endpoint_stop(subs->sync_endpoint,
+ force, can_sleep, wait);
+
+ if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags))
+ snd_usb_endpoint_stop(subs->data_endpoint,
+ force, can_sleep, wait);
+}
+
+static int activate_endpoints(struct snd_usb_substream *subs)
+{
+ if (subs->sync_endpoint) {
+ int ret;
+
+ ret = snd_usb_endpoint_activate(subs->sync_endpoint);
+ if (ret < 0)
+ return ret;
+ }
+
+ return snd_usb_endpoint_activate(subs->data_endpoint);
+}
+
+static int deactivate_endpoints(struct snd_usb_substream *subs)
+{
+ int reta, retb;
+
+ reta = snd_usb_endpoint_deactivate(subs->sync_endpoint);
+ retb = snd_usb_endpoint_deactivate(subs->data_endpoint);
+
+ if (reta < 0)
+ return reta;
+
+ if (retb < 0)
+ return retb;
+
+ return 0;
+}
+
/*
* find a matching format and set up the interface
*/
@@ -219,7 +301,7 @@
struct usb_interface *iface;
unsigned int ep, attr;
int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
- int err;
+ int err, implicit_fb = 0;
iface = usb_ifnum_to_if(dev, fmt->iface);
if (WARN_ON(!iface))
@@ -232,40 +314,11 @@
if (fmt == subs->cur_audiofmt)
return 0;
- /* close the old interface */
- if (subs->interface >= 0 && subs->interface != fmt->iface) {
- if (usb_set_interface(subs->dev, subs->interface, 0) < 0) {
- snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n",
- dev->devnum, fmt->iface, fmt->altsetting);
- return -EIO;
- }
- subs->interface = -1;
- subs->altset_idx = 0;
- }
-
- /* set interface */
- if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) {
- if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) {
- snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n",
- dev->devnum, fmt->iface, fmt->altsetting);
- return -EIO;
- }
- snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting);
- subs->interface = fmt->iface;
- subs->altset_idx = fmt->altset_idx;
- }
-
- /* create a data pipe */
- ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK;
- if (is_playback)
- subs->datapipe = usb_sndisocpipe(dev, ep);
- else
- subs->datapipe = usb_rcvisocpipe(dev, ep);
- subs->datainterval = fmt->datainterval;
- subs->syncpipe = subs->syncinterval = 0;
- subs->maxpacksize = fmt->maxpacksize;
- subs->syncmaxsize = 0;
- subs->fill_max = 0;
+ subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+ alts, fmt->endpoint, subs->direction,
+ SND_USB_ENDPOINT_TYPE_DATA);
+ if (!subs->data_endpoint)
+ return -EINVAL;
/* we need a sync pipe in async OUT or adaptive IN mode */
/* check the number of EP, since some devices have broken
@@ -273,8 +326,25 @@
* assume it as adaptive-out or sync-in.
*/
attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+ switch (subs->stream->chip->usb_id) {
+ case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+ case USB_ID(0x0763, 0x2081):
+ if (is_playback) {
+ implicit_fb = 1;
+ ep = 0x81;
+ iface = usb_ifnum_to_if(dev, 2);
+
+ if (!iface || iface->num_altsetting == 0)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+ goto add_sync_ep;
+ }
+ }
+
if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
- (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
+ (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
altsd->bNumEndpoints >= 2) {
/* check sync-pipe endpoint */
/* ... and check descriptor size before accessing bSynchAddress
@@ -282,7 +352,8 @@
the audio fields in the endpoint descriptors */
if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 ||
(get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
- get_endpoint(alts, 1)->bSynchAddress != 0)) {
+ get_endpoint(alts, 1)->bSynchAddress != 0 &&
+ !implicit_fb)) {
snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
dev->devnum, fmt->iface, fmt->altsetting);
return -EINVAL;
@@ -290,33 +361,27 @@
ep = get_endpoint(alts, 1)->bEndpointAddress;
if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
(( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
- (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+ (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) ||
+ ( is_playback && !implicit_fb))) {
snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
dev->devnum, fmt->iface, fmt->altsetting);
return -EINVAL;
}
- ep &= USB_ENDPOINT_NUMBER_MASK;
- if (is_playback)
- subs->syncpipe = usb_rcvisocpipe(dev, ep);
- else
- subs->syncpipe = usb_sndisocpipe(dev, ep);
- if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
- get_endpoint(alts, 1)->bRefresh >= 1 &&
- get_endpoint(alts, 1)->bRefresh <= 9)
- subs->syncinterval = get_endpoint(alts, 1)->bRefresh;
- else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
- subs->syncinterval = 1;
- else if (get_endpoint(alts, 1)->bInterval >= 1 &&
- get_endpoint(alts, 1)->bInterval <= 16)
- subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
- else
- subs->syncinterval = 3;
- subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
- }
- /* always fill max packet size */
- if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)
- subs->fill_max = 1;
+ implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+ == USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+add_sync_ep:
+ subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+ alts, ep, !subs->direction,
+ implicit_fb ?
+ SND_USB_ENDPOINT_TYPE_DATA :
+ SND_USB_ENDPOINT_TYPE_SYNC);
+ if (!subs->sync_endpoint)
+ return -EINVAL;
+
+ subs->data_endpoint->sync_master = subs->sync_endpoint;
+ }
if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0)
return err;
@@ -390,15 +455,30 @@
if (changed) {
mutex_lock(&subs->stream->chip->shutdown_mutex);
/* format changed */
- snd_usb_release_substream_urbs(subs, 0);
- /* influenced: period_bytes, channels, rate, format, */
- ret = snd_usb_init_substream_urbs(subs, params_period_bytes(hw_params),
- params_rate(hw_params),
- snd_pcm_format_physical_width(params_format(hw_params)) *
- params_channels(hw_params));
+ stop_endpoints(subs, 0, 0, 0);
+ deactivate_endpoints(subs);
+
+ ret = activate_endpoints(subs);
+ if (ret < 0)
+ goto unlock;
+
+ ret = snd_usb_endpoint_set_params(subs->data_endpoint, hw_params, fmt,
+ subs->sync_endpoint);
+ if (ret < 0)
+ goto unlock;
+
+ if (subs->sync_endpoint)
+ ret = snd_usb_endpoint_set_params(subs->sync_endpoint,
+ hw_params, fmt, NULL);
+unlock:
mutex_unlock(&subs->stream->chip->shutdown_mutex);
}
+ if (ret == 0) {
+ subs->interface = fmt->iface;
+ subs->altset_idx = fmt->altset_idx;
+ }
+
return ret;
}
@@ -415,7 +495,7 @@
subs->cur_rate = 0;
subs->period_bytes = 0;
mutex_lock(&subs->stream->chip->shutdown_mutex);
- snd_usb_release_substream_urbs(subs, 0);
+ stop_endpoints(subs, 0, 1, 1);
mutex_unlock(&subs->stream->chip->shutdown_mutex);
return snd_pcm_lib_free_vmalloc_buffer(substream);
}
@@ -435,19 +515,28 @@
return -ENXIO;
}
+ if (snd_BUG_ON(!subs->data_endpoint))
+ return -EIO;
+
/* some unit conversions in runtime */
- subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize);
- subs->curframesize = bytes_to_frames(runtime, subs->curpacksize);
+ subs->data_endpoint->maxframesize =
+ bytes_to_frames(runtime, subs->data_endpoint->maxpacksize);
+ subs->data_endpoint->curframesize =
+ bytes_to_frames(runtime, subs->data_endpoint->curpacksize);
/* reset the pointer */
subs->hwptr_done = 0;
subs->transfer_done = 0;
- subs->phase = 0;
subs->last_delay = 0;
subs->last_frame_number = 0;
runtime->delay = 0;
- return snd_usb_substream_prepare(subs, runtime);
+ /* for playback, submit the URBs now; otherwise, the first hwptr_done
+ * updates for all URBs would happen at the same time when starting */
+ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
+ return start_endpoints(subs);
+
+ return 0;
}
static struct snd_pcm_hardware snd_usb_hardware =
@@ -699,6 +788,9 @@
int count = 0, needs_knot = 0;
int err;
+ kfree(subs->rate_list.list);
+ subs->rate_list.list = NULL;
+
list_for_each_entry(fp, &subs->fmt_list, list) {
if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)
return 0;
@@ -842,16 +934,171 @@
static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
{
+ int ret;
struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
struct snd_usb_substream *subs = &as->substream[direction];
- if (!as->chip->shutdown && subs->interface >= 0) {
- usb_set_interface(subs->dev, subs->interface, 0);
- subs->interface = -1;
- }
+ stop_endpoints(subs, 0, 0, 0);
+ ret = deactivate_endpoints(subs);
subs->pcm_substream = NULL;
snd_usb_autosuspend(subs->stream->chip);
- return 0;
+
+ return ret;
+}
+
+/* Since a URB can handle only a single linear buffer, we must use double
+ * buffering when the data to be transferred overflows the buffer boundary.
+ * To avoid inconsistencies when updating hwptr_done, we use double buffering
+ * for all URBs.
+ */
+static void retire_capture_urb(struct snd_usb_substream *subs,
+ struct urb *urb)
+{
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ unsigned int stride, frames, bytes, oldptr;
+ int i, period_elapsed = 0;
+ unsigned long flags;
+ unsigned char *cp;
+
+ stride = runtime->frame_bits >> 3;
+
+ for (i = 0; i < urb->number_of_packets; i++) {
+ cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset;
+ if (urb->iso_frame_desc[i].status && printk_ratelimit()) {
+ snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status);
+ // continue;
+ }
+ bytes = urb->iso_frame_desc[i].actual_length;
+ frames = bytes / stride;
+ if (!subs->txfr_quirk)
+ bytes = frames * stride;
+ if (bytes % (runtime->sample_bits >> 3) != 0) {
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+ int oldbytes = bytes;
+#endif
+ bytes = frames * stride;
+ snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n",
+ oldbytes, bytes);
+ }
+ /* update the current pointer */
+ spin_lock_irqsave(&subs->lock, flags);
+ oldptr = subs->hwptr_done;
+ subs->hwptr_done += bytes;
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
+ frames = (bytes + (oldptr % stride)) / stride;
+ subs->transfer_done += frames;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ period_elapsed = 1;
+ }
+ spin_unlock_irqrestore(&subs->lock, flags);
+ /* copy a data chunk */
+ if (oldptr + bytes > runtime->buffer_size * stride) {
+ unsigned int bytes1 =
+ runtime->buffer_size * stride - oldptr;
+ memcpy(runtime->dma_area + oldptr, cp, bytes1);
+ memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
+ } else {
+ memcpy(runtime->dma_area + oldptr, cp, bytes);
+ }
+ }
+
+ if (period_elapsed)
+ snd_pcm_period_elapsed(subs->pcm_substream);
+}
+
+static void prepare_playback_urb(struct snd_usb_substream *subs,
+ struct urb *urb)
+{
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ struct snd_urb_ctx *ctx = urb->context;
+ unsigned int counts, frames, bytes;
+ int i, stride, period_elapsed = 0;
+ unsigned long flags;
+
+ stride = runtime->frame_bits >> 3;
+
+ frames = 0;
+ urb->number_of_packets = 0;
+ spin_lock_irqsave(&subs->lock, flags);
+ for (i = 0; i < ctx->packets; i++) {
+ counts = ctx->packet_size[i];
+ /* set up descriptor */
+ urb->iso_frame_desc[i].offset = frames * stride;
+ urb->iso_frame_desc[i].length = counts * stride;
+ frames += counts;
+ urb->number_of_packets++;
+ subs->transfer_done += counts;
+ if (subs->transfer_done >= runtime->period_size) {
+ subs->transfer_done -= runtime->period_size;
+ period_elapsed = 1;
+ if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
+ if (subs->transfer_done > 0) {
+ /* FIXME: fill-max mode is not
+ * supported yet */
+ frames -= subs->transfer_done;
+ counts -= subs->transfer_done;
+ urb->iso_frame_desc[i].length =
+ counts * stride;
+ subs->transfer_done = 0;
+ }
+ i++;
+ if (i < ctx->packets) {
+ /* add a transfer delimiter */
+ urb->iso_frame_desc[i].offset =
+ frames * stride;
+ urb->iso_frame_desc[i].length = 0;
+ urb->number_of_packets++;
+ }
+ break;
+ }
+ }
+ if (period_elapsed &&
+ !snd_usb_endpoint_implict_feedback_sink(subs->data_endpoint)) /* finish at the period boundary */
+ break;
+ }
+ bytes = frames * stride;
+ if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
+ /* err, the transferred area goes over buffer boundary. */
+ unsigned int bytes1 =
+ runtime->buffer_size * stride - subs->hwptr_done;
+ memcpy(urb->transfer_buffer,
+ runtime->dma_area + subs->hwptr_done, bytes1);
+ memcpy(urb->transfer_buffer + bytes1,
+ runtime->dma_area, bytes - bytes1);
+ } else {
+ memcpy(urb->transfer_buffer,
+ runtime->dma_area + subs->hwptr_done, bytes);
+ }
+ subs->hwptr_done += bytes;
+ if (subs->hwptr_done >= runtime->buffer_size * stride)
+ subs->hwptr_done -= runtime->buffer_size * stride;
+ runtime->delay += frames;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ urb->transfer_buffer_length = bytes;
+ if (period_elapsed)
+ snd_pcm_period_elapsed(subs->pcm_substream);
+}
+
+/*
+ * process after playback data complete
+ * - decrease the delay count again
+ */
+static void retire_playback_urb(struct snd_usb_substream *subs,
+ struct urb *urb)
+{
+ unsigned long flags;
+ struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+ int stride = runtime->frame_bits >> 3;
+ int processed = urb->transfer_buffer_length / stride;
+
+ spin_lock_irqsave(&subs->lock, flags);
+ if (processed > runtime->delay)
+ runtime->delay = 0;
+ else
+ runtime->delay -= processed;
+ spin_unlock_irqrestore(&subs->lock, flags);
}
static int snd_usb_playback_open(struct snd_pcm_substream *substream)
@@ -874,6 +1121,63 @@
return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE);
}
+static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream,
+ int cmd)
+{
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ subs->data_endpoint->prepare_data_urb = prepare_playback_urb;
+ subs->data_endpoint->retire_data_urb = retire_playback_urb;
+ subs->running = 1;
+ return 0;
+ case SNDRV_PCM_TRIGGER_STOP:
+ stop_endpoints(subs, 0, 0, 0);
+ subs->running = 0;
+ return 0;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ subs->data_endpoint->prepare_data_urb = NULL;
+ subs->data_endpoint->retire_data_urb = NULL;
+ subs->running = 0;
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
+int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int err;
+ struct snd_usb_substream *subs = substream->runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ err = start_endpoints(subs);
+ if (err < 0)
+ return err;
+
+ subs->data_endpoint->retire_data_urb = retire_capture_urb;
+ subs->running = 1;
+ return 0;
+ case SNDRV_PCM_TRIGGER_STOP:
+ stop_endpoints(subs, 0, 0, 0);
+ subs->running = 0;
+ return 0;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ subs->data_endpoint->retire_data_urb = NULL;
+ subs->running = 0;
+ return 0;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ subs->data_endpoint->retire_data_urb = retire_capture_urb;
+ subs->running = 1;
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
static struct snd_pcm_ops snd_usb_playback_ops = {
.open = snd_usb_playback_open,
.close = snd_usb_playback_close,
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index 961c9a2..ebc1a5b 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -25,6 +25,7 @@
#include "usbaudio.h"
#include "helper.h"
#include "card.h"
+#include "endpoint.h"
#include "proc.h"
/* convert our full speed USB rate into sampling rate in Hz */
@@ -115,28 +116,33 @@
}
}
+static void proc_dump_ep_status(struct snd_usb_substream *subs,
+ struct snd_usb_endpoint *ep,
+ struct snd_info_buffer *buffer)
+{
+ if (!ep)
+ return;
+ snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize);
+ snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n",
+ snd_usb_get_speed(subs->dev) == USB_SPEED_FULL
+ ? get_full_speed_hz(ep->freqm)
+ : get_high_speed_hz(ep->freqm),
+ ep->freqm >> 16, ep->freqm & 0xffff);
+ if (ep->freqshift != INT_MIN) {
+ int res = 16 - ep->freqshift;
+ snd_iprintf(buffer, " Feedback Format = %d.%d\n",
+ (ep->syncmaxsize > 3 ? 32 : 24) - res, res);
+ }
+}
+
static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer)
{
if (subs->running) {
- unsigned int i;
snd_iprintf(buffer, " Status: Running\n");
snd_iprintf(buffer, " Interface = %d\n", subs->interface);
snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx);
- snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs);
- for (i = 0; i < subs->nurbs; i++)
- snd_iprintf(buffer, "%d ", subs->dataurb[i].packets);
- snd_iprintf(buffer, "]\n");
- snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize);
- snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n",
- snd_usb_get_speed(subs->dev) == USB_SPEED_FULL
- ? get_full_speed_hz(subs->freqm)
- : get_high_speed_hz(subs->freqm),
- subs->freqm >> 16, subs->freqm & 0xffff);
- if (subs->freqshift != INT_MIN)
- snd_iprintf(buffer, " Feedback Format = %d.%d\n",
- (subs->syncmaxsize > 3 ? 32 : 24)
- - (16 - subs->freqshift),
- 16 - subs->freqshift);
+ proc_dump_ep_status(subs, subs->data_endpoint, buffer);
+ proc_dump_ep_status(subs, subs->sync_endpoint, buffer);
} else {
snd_iprintf(buffer, " Status: Stop\n");
}
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 5ff8010..6b7d7a2 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -73,6 +73,31 @@
}
}
+/*
+ * initialize the substream instance.
+ */
+
+static void snd_usb_init_substream(struct snd_usb_stream *as,
+ int stream,
+ struct audioformat *fp)
+{
+ struct snd_usb_substream *subs = &as->substream[stream];
+
+ INIT_LIST_HEAD(&subs->fmt_list);
+ spin_lock_init(&subs->lock);
+
+ subs->stream = as;
+ subs->direction = stream;
+ subs->dev = as->chip->dev;
+ subs->txfr_quirk = as->chip->txfr_quirk;
+
+ snd_usb_set_pcm_ops(as->pcm, stream);
+
+ list_add_tail(&fp->list, &subs->fmt_list);
+ subs->formats |= fp->formats;
+ subs->num_formats++;
+ subs->fmt_type = fp->fmt_type;
+}
/*
* add this endpoint to the chip instance.
@@ -94,9 +119,9 @@
if (as->fmt_type != fp->fmt_type)
continue;
subs = &as->substream[stream];
- if (!subs->endpoint)
+ if (!subs->data_endpoint)
continue;
- if (subs->endpoint == fp->endpoint) {
+ if (subs->data_endpoint->ep_num == fp->endpoint) {
list_add_tail(&fp->list, &subs->fmt_list);
subs->num_formats++;
subs->formats |= fp->formats;
@@ -109,7 +134,7 @@
if (as->fmt_type != fp->fmt_type)
continue;
subs = &as->substream[stream];
- if (subs->endpoint)
+ if (subs->data_endpoint)
continue;
err = snd_pcm_new_stream(as->pcm, stream, 1);
if (err < 0)
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 3e2b035..b8233eb 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@
struct snd_card *card;
struct usb_interface *pm_intf;
u32 usb_id;
+ struct mutex mutex;
struct mutex shutdown_mutex;
unsigned int shutdown:1;
unsigned int probing:1;
@@ -46,6 +47,7 @@
int num_suspended_intf;
struct list_head pcm_list; /* list of pcm streams */
+ struct list_head ep_list; /* list of audio-related endpoints */
int pcm_devs;
struct list_head midi_list; /* list of midi interfaces */