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-rw-r--r--sound/core/pcm.c6
-rw-r--r--sound/core/pcm_native.c4
-rw-r--r--sound/oss/sh_dac_audio.c1
-rw-r--r--sound/oss/sound_timer.c2
-rw-r--r--sound/pci/asihpi/hpi6205.c7
-rw-r--r--sound/pci/emu10k1/emu10k1.c4
-rw-r--r--sound/pci/emu10k1/emupcm.c30
-rw-r--r--sound/pci/emu10k1/memory.c4
-rw-r--r--sound/pci/hda/hda_codec.c59
-rw-r--r--sound/pci/hda/hda_codec.h7
-rw-r--r--sound/pci/hda/hda_eld.c49
-rw-r--r--sound/pci/hda/hda_local.h2
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_conexant.c4
-rw-r--r--sound/pci/hda/patch_hdmi.c59
-rw-r--r--sound/pci/hda/patch_intelhdmi.c9
-rw-r--r--sound/pci/hda/patch_nvhdmi.c12
-rw-r--r--sound/pci/hda/patch_realtek.c179
-rw-r--r--sound/pci/hda/patch_sigmatel.c15
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/pci/riptide/riptide.c11
-rw-r--r--sound/soc/codecs/wm8776.c7
-rw-r--r--sound/soc/codecs/wm8994.c23
-rw-r--r--sound/soc/imx/imx-ssi.c3
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/usb/format.c8
26 files changed, 389 insertions, 126 deletions
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index cbe815dfbdc..204af48c5cc 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = {
FORMAT(S18_3BE),
FORMAT(U18_3LE),
FORMAT(U18_3BE),
+ FORMAT(G723_24),
+ FORMAT(G723_24_1B),
+ FORMAT(G723_40),
+ FORMAT(G723_40_1B),
};
const char *snd_pcm_format_name(snd_pcm_format_t format)
{
+ if (format >= ARRAY_SIZE(snd_pcm_format_names))
+ return "Unknown";
return snd_pcm_format_names[format];
}
EXPORT_SYMBOL_GPL(snd_pcm_format_name);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index a3b2a647924..134fc6c2e08 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -978,6 +978,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push)
{
if (substream->runtime->trigger_master != substream)
return 0;
+ /* some drivers might use hw_ptr to recover from the pause -
+ update the hw_ptr now */
+ if (push)
+ snd_pcm_update_hw_ptr(substream);
/* The jiffies check in snd_pcm_update_hw_ptr*() is done by
* a delta betwen the current jiffies, this gives a large enough
* delta, effectively to skip the check once.
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
index fdb58eb83d4..479e3025a8a 100644
--- a/sound/oss/sh_dac_audio.c
+++ b/sound/oss/sh_dac_audio.c
@@ -15,7 +15,6 @@
#include <linux/linkage.h>
#include <linux/slab.h>
#include <linux/fs.h>
-#include <linux/smp_lock.h>
#include <linux/sound.h>
#include <linux/smp_lock.h>
#include <linux/soundcard.h>
diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c
index f0f0c19fbff..48cda6c4c25 100644
--- a/sound/oss/sound_timer.c
+++ b/sound/oss/sound_timer.c
@@ -26,7 +26,7 @@ static unsigned long prev_event_time;
static volatile unsigned long usecs_per_tmr; /* Length of the current interval */
static struct sound_lowlev_timer *tmr;
-static spinlock_t lock;
+static DEFINE_SPINLOCK(lock);
static unsigned long tmr2ticks(int tmr_value)
{
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 3b441344822..22c5fc62553 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao,
}
-static u32 outstream_get_space_available(struct hpi_hostbuffer_status
- *status)
+static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status)
{
return status->size_in_bytes - (status->host_index -
status->dSP_index);
@@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao,
/* write it */
phm->function = HPI_OSTREAM_WRITE;
hw_message(pao, phm, phr);
+
+ if (phr->error)
+ return;
+
/* update status information that the DSP would typically
* update (and will update next time the DSP
* buffer update task reads data from the host BBM buffer)
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 4203782d7cb..aff8387c45c 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
static int enable_ir[SNDRV_CARDS];
static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
+static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
@@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444);
MODULE_PARM_DESC(enable_ir, "Enable IR.");
module_param_array(subsystem, uint, NULL, 0444);
MODULE_PARM_DESC(subsystem, "Force card subsystem model.");
+module_param_array(delay_pcm_irq, uint, NULL, 0444);
+MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0).");
/*
* Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400
*/
@@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
&emu)) < 0)
goto error;
card->private_data = emu;
+ emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f;
if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0)
goto error;
if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0)
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 55b83ef73c6..622bace148e 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
evoice->epcm->ccca_start_addr = start_addr + ccis;
if (extra) {
start_addr += ccis;
- end_addr += ccis;
+ end_addr += ccis + emu->delay_pcm_irq;
}
if (stereo && !extra) {
snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK);
@@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu,
/* Assumption that PT is already 0 so no harm overwriting */
snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]);
snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24));
- snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24));
+ snd_emu10k1_ptr_write(emu, PSST, voice,
+ (start_addr + (extra ? emu->delay_pcm_irq : 0)) |
+ (send_amount[2] << 24));
if (emu->card_capabilities->emu_model)
pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */
else
@@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_
snd_emu10k1_ptr_write(emu, IP, voice, 0);
}
+static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu,
+ struct snd_emu10k1_pcm *epcm,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_runtime *runtime)
+{
+ unsigned int ptr, period_pos;
+
+ /* try to sychronize the current position for the interrupt
+ source voice */
+ period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt;
+ period_pos %= runtime->period_size;
+ ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number);
+ ptr &= ~0x00ffffff;
+ ptr |= epcm->ccca_start_addr + period_pos;
+ snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr);
+}
+
static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
int cmd)
{
@@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
/* follow thru */
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
case SNDRV_PCM_TRIGGER_RESUME:
+ if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE)
+ snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime);
mix = &emu->pcm_mixer[substream->number];
snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix);
snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix);
@@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *
#endif
/*
printk(KERN_DEBUG
- "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n",
- ptr, runtime->buffer_size, runtime->period_size);
+ "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n",
+ (long)ptr, (long)runtime->buffer_size,
+ (long)runtime->period_size);
*/
return ptr;
}
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index ffb1ddb8dc2..957a311514c 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
if (snd_BUG_ON(!hdr))
return NULL;
+ idx = runtime->period_size >= runtime->buffer_size ?
+ (emu->delay_pcm_irq * 2) : 0;
mutex_lock(&hdr->block_mutex);
- blk = search_empty(emu, runtime->dma_bytes);
+ blk = search_empty(emu, runtime->dma_bytes + idx);
if (blk == NULL) {
mutex_unlock(&hdr->block_mutex);
return NULL;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 720a81d711e..3827092cc1d 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -589,6 +589,7 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
bus->ops = temp->ops;
mutex_init(&bus->cmd_mutex);
+ mutex_init(&bus->prepare_mutex);
INIT_LIST_HEAD(&bus->codec_list);
snprintf(bus->workq_name, sizeof(bus->workq_name),
@@ -1068,7 +1069,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
codec->addr = codec_addr;
mutex_init(&codec->spdif_mutex);
mutex_init(&codec->control_mutex);
- mutex_init(&codec->prepare_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);
@@ -1213,6 +1213,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
u32 stream_tag,
int channel_id, int format)
{
+ struct hda_codec *c;
struct hda_cvt_setup *p;
unsigned int oldval, newval;
int i;
@@ -1253,20 +1254,27 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
p->dirty = 0;
/* make other inactive cvts with the same stream-tag dirty */
- for (i = 0; i < codec->cvt_setups.used; i++) {
- p = snd_array_elem(&codec->cvt_setups, i);
- if (!p->active && p->stream_tag == stream_tag)
- p->dirty = 1;
+ list_for_each_entry(c, &codec->bus->codec_list, list) {
+ for (i = 0; i < c->cvt_setups.used; i++) {
+ p = snd_array_elem(&c->cvt_setups, i);
+ if (!p->active && p->stream_tag == stream_tag)
+ p->dirty = 1;
+ }
}
}
EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream);
+static void really_cleanup_stream(struct hda_codec *codec,
+ struct hda_cvt_setup *q);
+
/**
- * snd_hda_codec_cleanup_stream - clean up the codec for closing
+ * __snd_hda_codec_cleanup_stream - clean up the codec for closing
* @codec: the CODEC to clean up
* @nid: the NID to clean up
+ * @do_now: really clean up the stream instead of clearing the active flag
*/
-void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
+void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid,
+ int do_now)
{
struct hda_cvt_setup *p;
@@ -1274,14 +1282,19 @@ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
return;
snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid);
- /* here we just clear the active flag; actual clean-ups will be done
- * in purify_inactive_streams()
- */
p = get_hda_cvt_setup(codec, nid);
- if (p)
- p->active = 0;
+ if (p) {
+ /* here we just clear the active flag when do_now isn't set;
+ * actual clean-ups will be done later in
+ * purify_inactive_streams() called from snd_hda_codec_prpapre()
+ */
+ if (do_now)
+ really_cleanup_stream(codec, p);
+ else
+ p->active = 0;
+ }
}
-EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream);
+EXPORT_SYMBOL_HDA(__snd_hda_codec_cleanup_stream);
static void really_cleanup_stream(struct hda_codec *codec,
struct hda_cvt_setup *q)
@@ -1296,12 +1309,16 @@ static void really_cleanup_stream(struct hda_codec *codec,
/* clean up the all conflicting obsolete streams */
static void purify_inactive_streams(struct hda_codec *codec)
{
+ struct hda_codec *c;
int i;
- for (i = 0; i < codec->cvt_setups.used; i++) {
- struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i);
- if (p->dirty)
- really_cleanup_stream(codec, p);
+ list_for_each_entry(c, &codec->bus->codec_list, list) {
+ for (i = 0; i < c->cvt_setups.used; i++) {
+ struct hda_cvt_setup *p;
+ p = snd_array_elem(&c->cvt_setups, i);
+ if (p->dirty)
+ really_cleanup_stream(c, p);
+ }
}
}
@@ -3492,11 +3509,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
int ret;
- mutex_lock(&codec->prepare_mutex);
+ mutex_lock(&codec->bus->prepare_mutex);
ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream);
if (ret >= 0)
purify_inactive_streams(codec);
- mutex_unlock(&codec->prepare_mutex);
+ mutex_unlock(&codec->bus->prepare_mutex);
return ret;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_prepare);
@@ -3505,9 +3522,9 @@ void snd_hda_codec_cleanup(struct hda_codec *codec,
struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
- mutex_lock(&codec->prepare_mutex);
+ mutex_lock(&codec->bus->prepare_mutex);
hinfo->ops.cleanup(hinfo, codec, substream);
- mutex_unlock(&codec->prepare_mutex);
+ mutex_unlock(&codec->bus->prepare_mutex);
}
EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 3f7a479881e..62c70224010 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -648,6 +648,7 @@ struct hda_bus {
struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1];
struct mutex cmd_mutex;
+ struct mutex prepare_mutex;
/* unsolicited event queue */
struct hda_bus_unsolicited *unsol;
@@ -826,7 +827,6 @@ struct hda_codec {
struct mutex spdif_mutex;
struct mutex control_mutex;
- struct mutex prepare_mutex;
unsigned int spdif_status; /* IEC958 status bits */
unsigned short spdif_ctls; /* SPDIF control bits */
unsigned int spdif_in_enable; /* SPDIF input enable? */
@@ -963,7 +963,10 @@ void snd_hda_codec_cleanup(struct hda_codec *codec,
void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
u32 stream_tag,
int channel_id, int format);
-void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid);
+void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid,
+ int do_now);
+#define snd_hda_codec_cleanup_stream(codec, nid) \
+ __snd_hda_codec_cleanup_stream(codec, nid, 0)
unsigned int snd_hda_calc_stream_format(unsigned int rate,
unsigned int channels,
unsigned int format,
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index d8da18a9e98..26c3ade7358 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -597,3 +597,52 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free);
#endif /* CONFIG_PROC_FS */
+
+/* update PCM info based on ELD */
+void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
+ struct hda_pcm_stream *codec_pars)
+{
+ int i;
+
+ pcm->rates = 0;
+ pcm->formats = 0;
+ pcm->maxbps = 0;
+ pcm->channels_min = -1;
+ pcm->channels_max = 0;
+ for (i = 0; i < eld->sad_count; i++) {
+ struct cea_sad *a = &eld->sad[i];
+ pcm->rates |= a->rates;
+ if (a->channels < pcm->channels_min)
+ pcm->channels_min = a->channels;
+ if (a->channels > pcm->channels_max)
+ pcm->channels_max = a->channels;
+ if (a->format == AUDIO_CODING_TYPE_LPCM) {
+ if (a->sample_bits & AC_SUPPCM_BITS_16) {
+ pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE;
+ if (pcm->maxbps < 16)
+ pcm->maxbps = 16;
+ }
+ if (a->sample_bits & AC_SUPPCM_BITS_20) {
+ pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ if (pcm->maxbps < 20)
+ pcm->maxbps = 20;
+ }
+ if (a->sample_bits & AC_SUPPCM_BITS_24) {
+ pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
+ if (pcm->maxbps < 24)
+ pcm->maxbps = 24;
+ }
+ }
+ }
+
+ if (!codec_pars)
+ return;
+
+ /* restrict the parameters by the values the codec provides */
+ pcm->rates &= codec_pars->rates;
+ pcm->formats &= codec_pars->formats;
+ pcm->channels_min = max(pcm->channels_min, codec_pars->channels_min);
+ pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
+ pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
+}
+EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 7a97f126f6f..28ab4aead48 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -604,6 +604,8 @@ struct hdmi_eld {
int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid);
int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t);
void snd_hdmi_show_eld(struct hdmi_eld *eld);
+void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
+ struct hda_pcm_stream *codec_pars);
#ifdef CONFIG_PROC_FS
int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld,
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 350ee8ac415..4ef5efaaaef 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -656,7 +656,7 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx,
return 0;
if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) {
/* stream is running, let's swap the current ADC */
- snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
spec->cur_adc = spec->adc_nid[idx];
snd_hda_codec_setup_stream(codec, spec->cur_adc,
spec->cur_adc_stream_tag, 0,
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f7e234e5ee9..5cdb80edbd7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1733,7 +1733,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec)
new_adc = spec->adc_nids[spec->cur_adc_idx];
if (spec->cur_adc && spec->cur_adc != new_adc) {
/* stream is running, let's swap the current ADC */
- snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
spec->cur_adc = new_adc;
snd_hda_codec_setup_stream(codec, new_adc,
spec->cur_adc_stream_tag, 0,
@@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
CXT5066_DELL_LAPTOP),
SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
+ SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO),
SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
@@ -3058,6 +3059,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
+ SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
{}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 522e0748ee9..afd6022a96a 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -46,6 +46,7 @@ struct hdmi_spec {
* export one pcm per pipe
*/
struct hda_pcm pcm_rec[MAX_HDMI_CVTS];
+ struct hda_pcm_stream codec_pcm_pars[MAX_HDMI_CVTS];
/*
* nvhdmi specific
@@ -706,8 +707,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
u32 stream_tag, int format)
{
struct hdmi_spec *spec = codec->spec;
- int tag;
- int fmt;
int pinctl;
int new_pinctl = 0;
int i;
@@ -744,24 +743,48 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid,
return -EINVAL;
}
- tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4;
- fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0);
+ snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
+ return 0;
+}
- snd_printdd("hdmi_setup_stream: "
- "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n",
- nid,
- tag == stream_tag ? "" : "new-",
- stream_tag,
- fmt == format ? "" : "new-",
- format);
+/*
+ * HDA PCM callbacks
+ */
+static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct hdmi_spec *spec = codec->spec;
+ struct hdmi_eld *eld;
+ struct hda_pcm_stream *codec_pars;
+ unsigned int idx;
- if (tag != stream_tag)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_CHANNEL_STREAMID,
- stream_tag << 4);
- if (fmt != format)
- snd_hda_codec_write(codec, nid, 0,
- AC_VERB_SET_STREAM_FORMAT, format);
+ for (idx = 0; idx < spec->num_cvts; idx++)
+ if (hinfo->nid == spec->cvt[idx])
+ break;
+ if (snd_BUG_ON(idx >= spec->num_cvts) ||
+ snd_BUG_ON(idx >= spec->num_pins))
+ return -EINVAL;
+
+ /* save the PCM info the codec provides */
+ codec_pars = &spec->codec_pcm_pars[idx];
+ if (!codec_pars->rates)
+ *codec_pars = *hinfo;
+
+ eld = &spec->sink_eld[idx];
+ if (eld->sad_count > 0) {
+ hdmi_eld_update_pcm_info(eld, hinfo, codec_pars);
+ if (hinfo->channels_min > hinfo->channels_max ||
+ !hinfo->rates || !hinfo->formats)
+ return -ENODEV;
+ } else {
+ /* fallback to the codec default */
+ hinfo->channels_min = codec_pars->channels_min;
+ hinfo->channels_max = codec_pars->channels_max;
+ hinfo->rates = codec_pars->rates;
+ hinfo->formats = codec_pars->formats;
+ hinfo->maxbps = codec_pars->maxbps;
+ }
return 0;
}
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 5972d5e7d01..36a9b83a617 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -69,19 +69,12 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format);
}
-static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- return 0;
-}
-
static struct hda_pcm_stream intel_hdmi_pcm_playback = {
.substreams = 1,
.channels_min = 2,
.ops = {
+ .open = hdmi_pcm_open,
.prepare = intel_hdmi_playback_pcm_prepare,
- .cleanup = intel_hdmi_playback_pcm_cleanup,
},
};
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 77e2b4028b9..69b950d527c 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -326,13 +326,6 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo,
return 0;
}
-static int nvhdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
-{
- return 0;
-}
-
static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
@@ -347,12 +340,9 @@ static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo,
static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = {
.substreams = 1,
.channels_min = 2,
- .rates = SUPPORTED_RATES,
- .maxbps = SUPPORTED_MAXBPS,
- .formats = SUPPORTED_FORMATS,
.ops = {
+ .open = hdmi_pcm_open,
.prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89,
- .cleanup = nvhdmi_playback_pcm_cleanup,
},
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 55d6e5b6bb7..627bf996336 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1037,7 +1037,7 @@ static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec)
new_adc = spec->adc_nids[spec->cur_adc_idx];
if (spec->cur_adc && spec->cur_adc != new_adc) {
/* stream is running, let's swap the current ADC */
- snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1);
spec->cur_adc = new_adc;
snd_hda_codec_setup_stream(codec, new_adc,
spec->cur_adc_stream_tag, 0,
@@ -14467,6 +14467,7 @@ static const struct alc_fixup alc269_fixups[] = {
static struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
+ SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
{}
};
@@ -19030,6 +19031,7 @@ static int patch_alc888(struct hda_codec *codec)
/*
* ALC680 support
*/
+#define ALC680_DIGIN_NID ALC880_DIGIN_NID
#define ALC680_DIGOUT_NID ALC880_DIGOUT_NID
#define alc680_modes alc260_modes
@@ -19044,23 +19046,93 @@ static hda_nid_t alc680_adc_nids[3] = {
0x07, 0x08, 0x09
};
+/*
+ * Analog capture ADC cgange
+ */
+static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int pre_mic, pre_line;
+
+ pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
+ pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]);
+
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+
+ if (pre_mic || pre_line) {
+ if (pre_mic)
+ snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0,
+ format);
+ else
+ snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0,
+ format);
+ } else
+ snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format);
+ return 0;
+}
+
+static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ snd_hda_codec_cleanup_stream(codec, 0x07);
+ snd_hda_codec_cleanup_stream(codec, 0x08);
+ snd_hda_codec_cleanup_stream(codec, 0x09);
+ return 0;
+}
+
+static struct hda_pcm_stream alc680_pcm_analog_auto_capture = {
+ .substreams = 1, /* can be overridden */
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+ .ops = {
+ .prepare = alc680_capture_pcm_prepare,
+ .cleanup = alc680_capture_pcm_cleanup
+ },
+};
+
static struct snd_kcontrol_new alc680_base_mixer[] = {
/* output mixer control */
HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT),
{ }
};
-static struct snd_kcontrol_new alc680_capture_mixer[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+static struct hda_bind_ctls alc680_bind_cap_vol = {
+ .ops = &snd_hda_bind_vol,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static struct hda_bind_ctls alc680_bind_cap_switch = {
+ .ops = &snd_hda_bind_sw,
+ .values = {
+ HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT),
+ HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+ 0
+ },
+};
+
+static struct snd_kcontrol_new alc680_master_capture_mixer[] = {
+ HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol),
+ HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch),
{ } /* end */
};
@@ -19068,25 +19140,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = {
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc680_init_verbs[] = {
- /* Unmute DAC0-1 and set vol = 0 */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN},
+
{ }
};
+/* toggle speaker-output according to the hp-jack state */
+static void alc680_base_setup(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x16;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18;
+ spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19;
+}
+
+static void alc680_rec_autoswitch(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int present;
+ hda_nid_t new_adc;
+
+ present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]);
+
+ new_adc = present ? 0x8 : 0x7;
+ __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1);
+ snd_hda_codec_setup_stream(codec, new_adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+
+}
+
+static void alc680_unsol_event(struct hda_codec *codec,
+ unsigned int res)
+{
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc_automute_amp(codec);
+ if ((res >> 26) == ALC880_MIC_EVENT)
+ alc680_rec_autoswitch(codec);
+}
+
+static void alc680_inithook(struct hda_codec *codec)
+{
+ alc_automute_amp(codec);
+ alc680_rec_autoswitch(codec);
+}
+
/* create input playback/capture controls for the given pin */
static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
const char *ctlname, int idx)
@@ -19197,13 +19317,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec)
#define alc680_pcm_analog_capture alc880_pcm_analog_capture
#define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
#define alc680_pcm_digital_playback alc880_pcm_digital_playback
-
-static struct hda_input_mux alc680_capture_source = {
- .num_items = 1,
- .items = {
- { "Mic", 0x0 },
- },
-};
+#define alc680_pcm_digital_capture alc880_pcm_digital_capture
/*
* BIOS auto configuration
@@ -19218,6 +19332,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
alc680_ignore);
if (err < 0)
return err;
+
if (!spec->autocfg.line_outs) {
if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
spec->multiout.max_channels = 2;
@@ -19239,8 +19354,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec)
add_mixer(spec, spec->kctls.list);
add_verb(spec, alc680_init_verbs);
- spec->num_mux_defs = 1;
- spec->input_mux = &alc680_capture_source;
err = alc_auto_add_mic_boost(codec);
if (err < 0)
@@ -19279,17 +19392,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = {
static struct alc_config_preset alc680_presets[] = {
[ALC680_BASE] = {
.mixers = { alc680_base_mixer },
- .cap_mixer = alc680_capture_mixer,
+ .cap_mixer = alc680_master_capture_mixer,
.init_verbs = { alc680_init_verbs },
.num_dacs = ARRAY_SIZE(alc680_dac_nids),
.dac_nids = alc680_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc680_adc_nids),
- .adc_nids = alc680_adc_nids,
- .hp_nid = 0x04,
.dig_out_nid = ALC680_DIGOUT_NID,
.num_channel_mode = ARRAY_SIZE(alc680_modes),
.channel_mode = alc680_modes,
- .input_mux = &alc680_capture_source,
+ .unsol_event = alc680_unsol_event,
+ .setup = alc680_base_setup,
+ .init_hook = alc680_inithook,
+
},
};
@@ -19333,9 +19446,9 @@ static int patch_alc680(struct hda_codec *codec)
setup_preset(codec, &alc680_presets[board_config]);
spec->stream_analog_playback = &alc680_pcm_analog_playback;
- spec->stream_analog_capture = &alc680_pcm_analog_capture;
- spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture;
+ spec->stream_analog_capture = &alc680_pcm_analog_auto_capture;
spec->stream_digital_playback = &alc680_pcm_digital_playback;
+ spec->stream_digital_capture = &alc680_pcm_digital_capture;
if (!spec->adc_nids) {
spec->adc_nids = alc680_adc_nids;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f3f861bd1bf..95148e58026 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -6303,6 +6303,21 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
{ .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
{ .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
{ .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
+ { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
+ { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
{} /* terminator */
};
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 6433e65c950..46774924957 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1776,6 +1776,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.subvendor = 0x1014,
+ .subdevice = 0x0534,
+ .name = "ThinkPad X31",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x1014,
.subdevice = 0x1f00,
.name = "MS-9128",
.type = AC97_TUNE_ALC_JACK
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index f64fb7d988c..ad5202efd7a 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
firmware.firmware.ASIC, firmware.firmware.CODEC,
firmware.firmware.AUXDSP, firmware.firmware.PROG);
+ if (!chip)
+ return 1;
+
for (i = 0; i < FIRMWARE_VERSIONS; i++) {
if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware)))
- break;
- }
- if (i >= FIRMWARE_VERSIONS)
- return 0; /* no match */
+ return 1; /* OK */
- if (!chip)
- return 1; /* OK */
+ }
snd_printdd("Writing Firmware\n");
if (!chip->fw_entry) {
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 4e212ed62ea..f8154e66152 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0001;
break;
- /* FIXME: CHECK A/B */
- case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
- break;
- case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0007;
- break;
default:
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index a87046a96f2..522249d5c2b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -95,6 +95,7 @@ struct wm8994_priv {
struct wm8994_micdet micdet[2];
+ int revision;
struct wm8994_pdata *pdata;
};
@@ -3070,6 +3071,8 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
static int wm8994_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
switch (level) {
case SND_SOC_BIAS_ON:
break;
@@ -3082,11 +3085,16 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_OFF) {
- /* Tweak DC servo configuration for improved
- * performance. */
- snd_soc_write(codec, 0x102, 0x3);
- snd_soc_write(codec, 0x56, 0x3);
- snd_soc_write(codec, 0x102, 0);
+ /* Tweak DC servo and DSP configuration for
+ * improved performance. */
+ if (wm8994->revision < 4) {
+ /* Tweak DC servo and DSP configuration for
+ * improved performance. */
+ snd_soc_write(codec, 0x102, 0x3);
+ snd_soc_write(codec, 0x56, 0x3);
+ snd_soc_write(codec, 0x817, 0);
+ snd_soc_write(codec, 0x102, 0);
+ }
/* Discharge LINEOUT1 & 2 */
snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
@@ -3919,7 +3927,6 @@ static int wm8994_codec_probe(struct platform_device *pdev)
struct wm8994_priv *wm8994;
struct snd_soc_codec *codec;
int i;
- u16 rev;
if (wm8994_codec) {
dev_err(&pdev->dev, "Another WM8994 is registered\n");
@@ -3973,8 +3980,8 @@ static int wm8994_codec_probe(struct platform_device *pdev)
wm8994->reg_cache[i] = 0;
/* Set revision-specific configuration */
- rev = snd_soc_read(codec, WM8994_CHIP_REVISION);
- switch (rev) {
+ wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION);
+ switch (wm8994->revision) {
case 2:
case 3:
wm8994->hubs.dcs_codes = -5;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index a11daa1e905..c81da05a4f1 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -254,6 +254,9 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream,
dma_data = &ssi->dma_params_rx;
}
+ if (ssi->flags & IMX_SSI_SYN)
+ reg = SSI_STCCR;
+
snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 844ae8221a3..acc91daa1c5 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -251,7 +251,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
printk(KERN_WARNING
"ASoC: Failed to create codec register debugfs file\n");
- codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+ codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0644,
codec->debugfs_codec_root,
&codec->pop_time);
if (!codec->debugfs_pop_time)
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 4387f54d73d..3a1375459c0 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -392,6 +392,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
/* fp->channels is already set in this case */
ret = parse_audio_format_rates_v2(chip, fp);
break;
+ default:
+ snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ return -EINVAL;
}
if (fp->channels < 1) {
@@ -452,6 +456,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
ret = parse_audio_format_rates_v2(chip, fp);
break;
}
+ default:
+ snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ return -EINVAL;
}
return ret;
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-21 02:22:09 +0000