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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 cbe815dfbdc8..204af48c5cc1 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 f0f0c19fbff7..48cda6c4c257 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 3b4413448226..22c5fc625533 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 dd8fb86c842b..3827092cc1d2 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 4303353feda9..62c702240108 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 803b298f7411..26c3ade73583 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 31b5d9eeba68..5cdb80edbd7f 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 2bc0f07cf33f..afd6022a96a7 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 d382d3c81c0f..36a9b83a6174 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 f636870dc718..69b950d527c3 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 2cd1ae809e46..627bf9963368 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 f3f861bd1bf8..95148e58026c 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 6433e65c9507..467749249576 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 4e212ed62ea6..f8154e661524 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 a11daa1e905b..c81da05a4f11 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 844ae8221a3a..acc91daa1c55 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 498a2d8fa4bb..4aa4678e0a01 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 b853f8df794f..7754a1034545 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 bb9f938558fd..b0ef9f501896 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 3a1375459c06..69148212aa70 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 c166db0057d3..3ed3901369ce 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 58ed6820a8cf..f49756c1b837 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-pink 1.4.1 (git 2.36.1) at 2024-10-04 22:32:54 +0000