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-rw-r--r--sound/core/pcm.c6
-rw-r--r--sound/oss/sound_timer.c2
-rw-r--r--sound/pci/asihpi/hpi6205.c7
-rw-r--r--sound/pci/hda/hda_codec.c33
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_eld.c4
-rw-r--r--sound/pci/hda/patch_conexant.c2
-rw-r--r--sound/pci/hda/patch_hdmi.c21
-rw-r--r--sound/pci/hda/patch_intelhdmi.c8
-rw-r--r--sound/pci/hda/patch_nvhdmi.c8
-rw-r--r--sound/pci/hda/patch_realtek.c177
-rw-r--r--sound/pci/hda/patch_sigmatel.c15
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/soc/codecs/wm8776.c7
-rw-r--r--sound/soc/imx/imx-ssi.c3
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/usb/card.c17
-rw-r--r--sound/usb/clock.c3
-rw-r--r--sound/usb/endpoint.c11
-rw-r--r--sound/usb/format.c22
-rw-r--r--sound/usb/mixer.c10
-rw-r--r--sound/usb/pcm.c3
22 files changed, 247 insertions, 122 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/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/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index dd8fb86c842..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;
@@ -1253,10 +1254,12 @@ 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);
@@ -1306,12 +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);
+ }
}
}
@@ -3502,11 +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);
@@ -3515,9 +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 4303353feda..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? */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 803b298f741..26c3ade7358 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -596,6 +596,8 @@ 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)
@@ -644,5 +646,3 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
}
EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info);
-
-#endif /* CONFIG_PROC_FS */
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 31b5d9eeba6..5cdb80edbd7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -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 2bc0f07cf33..afd6022a96a 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -707,8 +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;
@@ -745,24 +743,7 @@ 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_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);
-
- 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);
+ snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
return 0;
}
diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index d382d3c81c0..36a9b83a617 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -69,20 +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 f636870dc71..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,
@@ -350,7 +343,6 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = {
.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 2cd1ae809e4..627bf996336 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -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/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/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/card.c b/sound/usb/card.c
index 498a2d8fa4b..4aa4678e0a0 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
}
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n",
+ protocol);
+ /* fall through */
+
case UAC_VERSION_1: {
struct uac1_ac_header_descriptor *h1 = control_header;
@@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
break;
}
-
- default:
- snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol);
- return -EINVAL;
}
return 0;
@@ -480,7 +481,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
goto __error;
}
- chip->ctrl_intf = alts;
+ /*
+ * For devices with more than one control interface, we assume the
+ * first contains the audio controls. We might need a more specific
+ * check here in the future.
+ */
+ if (!chip->ctrl_intf)
+ chip->ctrl_intf = alts;
if (err > 0) {
/* create normal USB audio interfaces */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index b853f8df794..7754a103454 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
switch (altsd->bInterfaceProtocol) {
case UAC_VERSION_1:
+ default:
return set_sample_rate_v1(chip, iface, alts, fmt, rate);
case UAC_VERSION_2:
return set_sample_rate_v2(chip, iface, alts, fmt, rate);
}
-
- return -EINVAL;
}
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index bb9f938558f..b0ef9f50189 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
/* get audio formats */
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n",
+ dev->devnum, iface_no, altno, protocol);
+ protocol = UAC_VERSION_1;
+ /* fall through */
+
case UAC_VERSION_1: {
struct uac1_as_header_descriptor *as =
snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
@@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
dev->devnum, iface_no, altno, as->bTerminalLink);
continue;
}
-
- default:
- snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n",
- dev->devnum, iface_no, altno, protocol);
- continue;
}
/* get format type */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 3a1375459c0..69148212aa7 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
u64 pcm_formats;
switch (protocol) {
- case UAC_VERSION_1: {
+ case UAC_VERSION_1:
+ default: {
struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
sample_width = fmt->bBitResolution;
sample_bytes = fmt->bSubframeSize;
@@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
format <<= 1;
break;
}
-
- default:
- return -EINVAL;
}
pcm_formats = 0;
@@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
* audio class v2 uses class specific EP0 range requests for that.
*/
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ /* fall through */
case UAC_VERSION_1:
fp->channels = fmt->bNrChannels;
ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7);
@@ -392,10 +394,6 @@ 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) {
@@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
fp->channels = 1;
switch (protocol) {
+ default:
+ snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n",
+ chip->dev->devnum, fp->iface, fp->altsetting, protocol);
+ /* fall through */
case UAC_VERSION_1: {
struct uac_format_type_ii_discrete_descriptor *fmt = _fmt;
brate = le16_to_cpu(fmt->wMaxBitRate);
@@ -456,10 +458,6 @@ 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;
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index c166db0057d..3ed3901369c 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
}
host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
- mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol;
+ switch (get_iface_desc(host_iface)->bInterfaceProtocol) {
+ case UAC_VERSION_1:
+ default:
+ mixer->protocol = UAC_VERSION_1;
+ break;
+ case UAC_VERSION_2:
+ mixer->protocol = UAC_VERSION_2;
+ break;
+ }
if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
(err = snd_usb_mixer_status_create(mixer)) < 0)
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 58ed6820a8c..f49756c1b83 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
switch (altsd->bInterfaceProtocol) {
case UAC_VERSION_1:
+ default:
return init_pitch_v1(chip, iface, alts, fmt);
case UAC_VERSION_2:
return init_pitch_v2(chip, iface, alts, fmt);
}
-
- return -EINVAL;
}
/*
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-23 21:40:21 +0000