Merge master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa

* master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: (28 commits)
  [ALSA] Kconfig SND_SEQUENCER_OSS help text fix
  [ALSA] Add Aux input switch control for Aureon Universe
  [ALSA] pcxhr - Fix the crash with REV01 board
  [ALSA] sound/pci/hda: use create_singlethread_workqueue()
  [ALSA] hda-intel - Add support of ATI SB600
  [ALSA] cs4281 - Fix the check of timeout in probe
  [ALSA] cs4281 - Fix the check of right channel
  [ALSA] Test volume resolution of usb audio at initialization
  [ALSA] maestro3.c: fix BUG, optimization
  [ALSA] HDA/Realtek: multiple input mux definitions and pin mode additions
  [ALSA] AdLib FM card driver
  [ALSA] Fix / clean up PCM-OSS setup hooks
  [ALSA] Clean up PCM codes (take 2)
  [ALSA] Tiny clean up of PCM codes
  [ALSA] ISA drivers bailing on first !enable[i]
  [ALSA] Remove obsolete kfree_nocheck call
  [ALSA] Remove obsolete kfree_nocheck call
  [ALSA] Add snd-als300 driver for Avance Logic ALS300/ALS300+ soundcards
  [ALSA] Add snd-riptide driver for Conexant Riptide chip
  [ALSA] hda-codec - Fix noisy output wtih AD1986A 3stack model
  ...

5 files changed, 288 insertions, 76 deletions

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b42dff7ceed0..5bee3b536478 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -295,7 +295,7 @@ static int init_unsol_queue(struct hda_bus *bus)
 		snd_printk(KERN_ERR "hda_codec: can't allocate unsolicited queue\n");
 		return -ENOMEM;
 	}
-	unsol->workq = create_workqueue("hda_codec");
+	unsol->workq = create_singlethread_workqueue("hda_codec");
 	if (! unsol->workq) {
 		snd_printk(KERN_ERR "hda_codec: can't create workqueue\n");
 		kfree(unsol);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c096606970ff..0ad60ae29011 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -81,6 +81,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, ESB2},"
 			 "{Intel, ICH8},"
 			 "{ATI, SB450},"
+			 "{ATI, SB600},"
 			 "{VIA, VT8251},"
 			 "{VIA, VT8237A},"
 			 "{SiS, SIS966},"
@@ -1619,6 +1620,7 @@ static struct pci_device_id azx_ids[] = {
 	{ 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ESB2 */
 	{ 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */
 	{ 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */
+	{ 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */
 	{ 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */
 	{ 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */
 	{ 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 32401bd8c229..2bfe37e8543c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -44,6 +44,7 @@ struct ad198x_spec {
 					 * dig_out_nid and hp_nid are optional
 					 */
 	unsigned int cur_eapd;
+	unsigned int need_dac_fix;
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -836,10 +837,14 @@ static int patch_ad1986a(struct hda_codec *codec)
 	case AD1986A_3STACK:
 		spec->num_mixers = 2;
 		spec->mixers[1] = ad1986a_3st_mixers;
-		spec->num_init_verbs = 2;
+		spec->num_init_verbs = 3;
 		spec->init_verbs[1] = ad1986a_3st_init_verbs;
+		spec->init_verbs[2] = ad1986a_ch2_init;
 		spec->channel_mode = ad1986a_modes;
 		spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
+		spec->need_dac_fix = 1;
+		spec->multiout.max_channels = 2;
+		spec->multiout.num_dacs = 1;
 		break;
 	case AD1986A_LAPTOP:
 		spec->mixers[0] = ad1986a_laptop_mixers;
@@ -1555,6 +1560,8 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct ad198x_spec *spec = codec->spec;
+	if (spec->need_dac_fix)
+		spec->multiout.num_dacs = spec->multiout.max_channels / 2;
 	return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
 				   spec->num_channel_mode, &spec->multiout.max_channels);
 }
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4c6c9ec8ea5b..66bbdb60f50b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -52,6 +52,7 @@ enum {
 	ALC880_CLEVO,
 	ALC880_TCL_S700,
 	ALC880_LG,
+	ALC880_LG_LW,
 #ifdef CONFIG_SND_DEBUG
 	ALC880_TEST,
 #endif
@@ -131,6 +132,7 @@ struct alc_spec {
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 
 	/* capture source */
+	unsigned int num_mux_defs;
 	const struct hda_input_mux *input_mux;
 	unsigned int cur_mux[3];
 
@@ -172,6 +174,7 @@ struct alc_config_preset {
 	hda_nid_t dig_in_nid;
 	unsigned int num_channel_mode;
 	const struct hda_channel_mode *channel_mode;
+	unsigned int num_mux_defs;
 	const struct hda_input_mux *input_mux;
 	void (*unsol_event)(struct hda_codec *, unsigned int);
 	void (*init_hook)(struct hda_codec *);
@@ -185,7 +188,10 @@ static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	return snd_hda_input_mux_info(spec->input_mux, uinfo);
+	unsigned int mux_idx = snd_ctl_get_ioffidx(kcontrol, &uinfo->id);
+	if (mux_idx >= spec->num_mux_defs)
+		mux_idx = 0;
+	return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo);
 }
 
 static int alc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -203,7 +209,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+	unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
+	return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
 				     spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
 }
 
@@ -245,7 +252,8 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va
  * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
  * are requested.  Therefore order this list so that this behaviour will not
  * cause problems when mixer clients move through the enum sequentially.
- * NIDs 0x0f and 0x10 have been observed to have this behaviour.
+ * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
+ * March 2006.
  */
 static char *alc_pin_mode_names[] = {
 	"Mic 50pc bias", "Mic 80pc bias",
@@ -255,19 +263,27 @@ static unsigned char alc_pin_mode_values[] = {
 	PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
 };
 /* The control can present all 5 options, or it can limit the options based
- * in the pin being assumed to be exclusively an input or an output pin.
+ * in the pin being assumed to be exclusively an input or an output pin.  In
+ * addition, "input" pins may or may not process the mic bias option
+ * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
+ * accept requests for bias as of chip versions up to March 2006) and/or
+ * wiring in the computer.
  */
-#define ALC_PIN_DIR_IN    0x00
-#define ALC_PIN_DIR_OUT   0x01
-#define ALC_PIN_DIR_INOUT 0x02
+#define ALC_PIN_DIR_IN              0x00
+#define ALC_PIN_DIR_OUT             0x01
+#define ALC_PIN_DIR_INOUT           0x02
+#define ALC_PIN_DIR_IN_NOMICBIAS    0x03
+#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
 
-/* Info about the pin modes supported by the three different pin directions. 
+/* Info about the pin modes supported by the different pin direction modes. 
  * For each direction the minimum and maximum values are given.
  */
-static signed char alc_pin_mode_dir_info[3][2] = {
+static signed char alc_pin_mode_dir_info[5][2] = {
 	{ 0, 2 },    /* ALC_PIN_DIR_IN */
 	{ 3, 4 },    /* ALC_PIN_DIR_OUT */
 	{ 0, 4 },    /* ALC_PIN_DIR_INOUT */
+	{ 2, 2 },    /* ALC_PIN_DIR_IN_NOMICBIAS */
+	{ 2, 4 },    /* ALC_PIN_DIR_INOUT_NOMICBIAS */
 };
 #define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
 #define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
@@ -329,9 +345,10 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 		 * input modes.
 		 *
 		 * Dynamically switching the input/output buffers probably
-		 * reduces noise slightly, particularly on input.  However,
-		 * havingboth input and output buffers enabled
-		 * simultaneously doesn't seem to be problematic.
+		 * reduces noise slightly (particularly on input) so we'll
+		 * do it.  However, having both input and output buffers
+		 * enabled simultaneously doesn't seem to be problematic if
+		 * this turns out to be necessary in the future.
 		 */
 		if (val <= 2) {
 			snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
@@ -483,6 +500,9 @@ static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *
 	spec->multiout.dig_out_nid = preset->dig_out_nid;
 	spec->multiout.hp_nid = preset->hp_nid;
 	
+	spec->num_mux_defs = preset->num_mux_defs;
+	if (! spec->num_mux_defs)
+		spec->num_mux_defs = 1;
 	spec->input_mux = preset->input_mux;
 
 	spec->num_adc_nids = preset->num_adc_nids;
@@ -1427,6 +1447,82 @@ static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
 }
 
 /*
+ * LG LW20
+ *
+ * Pin assignment:
+ *   Speaker-out: 0x14
+ *   Mic-In: 0x18
+ *   Built-in Mic-In: 0x19 (?)
+ *   HP-Out: 0x1b
+ *   SPDIF-Out: 0x1e
+ */
+
+/* seems analog CD is not working */
+static struct hda_input_mux alc880_lg_lw_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Internal Mic", 0x1 },
+	},
+};
+
+static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_verb alc880_lg_lw_init_verbs[] = {
+	/* set capture source to mic-in */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+	/* speaker-out */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* HP-out */
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* mic-in to input */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* built-in mic */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* jack sense */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
+	{ }
+};
+
+/* toggle speaker-output according to the hp-jack state */
+static void alc880_lg_lw_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
+				 0x80, present ? 0x80 : 0);
+	snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
+				 0x80, present ? 0x80 : 0);
+}
+
+static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	/* Looks like the unsol event is incompatible with the standard
+	 * definition.  4bit tag is placed at 28 bit!
+	 */
+	if ((res >> 28) == 0x01)
+		alc880_lg_lw_automute(codec);
+}
+
+/*
  * Common callbacks
  */
 
@@ -2078,6 +2174,9 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 	{ .modelname = "lg", .config = ALC880_LG },
 	{ .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG },
 
+	{ .modelname = "lg-lw", .config = ALC880_LG_LW },
+	{ .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW },
+
 #ifdef CONFIG_SND_DEBUG
 	{ .modelname = "test", .config = ALC880_TEST },
 #endif
@@ -2268,6 +2367,19 @@ static struct alc_config_preset alc880_presets[] = {
 		.unsol_event = alc880_lg_unsol_event,
 		.init_hook = alc880_lg_automute,
 	},
+	[ALC880_LG_LW] = {
+		.mixers = { alc880_lg_lw_mixer },
+		.init_verbs = { alc880_volume_init_verbs,
+				alc880_lg_lw_init_verbs },
+		.num_dacs = 1, 
+		.dac_nids = alc880_dac_nids,
+		.dig_out_nid = ALC880_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+		.channel_mode = alc880_2_jack_modes,
+		.input_mux = &alc880_lg_lw_capture_source,
+		.unsol_event = alc880_lg_lw_unsol_event,
+		.init_hook = alc880_lg_lw_automute,
+	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST] = {
 		.mixers = { alc880_test_mixer },
@@ -2593,6 +2705,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 
 	spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs;
 
+	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
 	return 1;
@@ -2722,30 +2835,56 @@ static struct hda_input_mux alc260_capture_source = {
 };
 
 /* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
- * headphone jack and the internal CD lines.
+ * headphone jack and the internal CD lines since these are the only pins at
+ * which audio can appear.  For flexibility, also allow the option of
+ * recording the mixer output on the second ADC (ADC0 doesn't have a
+ * connection to the mixer output).
  */
-static struct hda_input_mux alc260_fujitsu_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic/Line", 0x0 },
-		{ "CD", 0x4 },
-		{ "Headphone", 0x2 },
+static struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
+	{
+		.num_items = 3,
+		.items = {
+			{ "Mic/Line", 0x0 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x2 },
+		},
 	},
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic/Line", 0x0 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x2 },
+			{ "Mixer", 0x5 },
+		},
+	},
+
 };
 
-/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configutation to
- * the Fujitsu S702x, but jacks are marked differently. We won't allow
- * retasking the Headphone jack, so it won't be available here.
+/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
+ * the Fujitsu S702x, but jacks are marked differently.
  */
-static struct hda_input_mux alc260_acer_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
+static struct hda_input_mux alc260_acer_capture_sources[2] = {
+	{
+		.num_items = 4,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x5 },
+		},
+	},
+	{
+		.num_items = 5,
+		.items = {
+			{ "Mic", 0x0 },
+			{ "Line", 0x2 },
+			{ "CD", 0x4 },
+			{ "Headphone", 0x6 },
+			{ "Mixer", 0x5 },
+		},
 	},
 };
-
 /*
  * This is just place-holder, so there's something for alc_build_pcms to look
  * at when it calculates the maximum number of channels. ALC260 has no mixer
@@ -2806,6 +2945,9 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = {
 	{ } /* end */
 };
 
+/* Fujitsu S702x series laptops.  ALC260 pin usage: Mic/Line jack = 0x12, 
+ * HP jack = 0x14, CD audio =  0x16, internal speaker = 0x10.
+ */
 static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
@@ -2822,9 +2964,28 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
 	{ } /* end */
 };
 
+/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks.  Note that current
+ * versions of the ALC260 don't act on requests to enable mic bias from NID
+ * 0x0f (used to drive the headphone jack in these laptops).  The ALC260
+ * datasheet doesn't mention this restriction.  At this stage it's not clear
+ * whether this behaviour is intentional or is a hardware bug in chip
+ * revisions available in early 2006.  Therefore for now allow the
+ * "Headphone Jack Mode" control to span all choices, but if it turns out
+ * that the lack of mic bias for this NID is intentional we could change the
+ * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ *
+ * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
+ * don't appear to make the mic bias available from the "line" jack, even
+ * though the NID used for this jack (0x14) can supply it.  The theory is
+ * that perhaps Acer have included blocking capacitors between the ALC260
+ * and the output jack.  If this turns out to be the case for all such
+ * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
+ * to ALC_PIN_DIR_INOUT_NOMICBIAS.
+ */
 static struct snd_kcontrol_new alc260_acer_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
+	ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
@@ -3038,7 +3199,8 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = {
 };
 
 /* Initialisation sequence for ALC260 as configured in Fujitsu S702x
- * laptops.
+ * laptops.  ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
+ * audio = 0x16, internal speaker = 0x10.
  */
 static struct hda_verb alc260_fujitsu_init_verbs[] = {
 	/* Disable all GPIOs */
@@ -3185,10 +3347,10 @@ static struct hda_verb alc260_acer_init_verbs[] = {
 	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	/* Do similar with the second ADC: mute capture input amp and
-	 * set ADC connection to line (on line1 pin)
+	 * set ADC connection to mic to match ALSA's default state.
 	 */
 	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	/* Mute all inputs to mixer widget (even unconnected ones) */
 	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
@@ -3213,26 +3375,35 @@ static hda_nid_t alc260_test_dac_nids[1] = {
 static hda_nid_t alc260_test_adc_nids[2] = {
 	0x04, 0x05,
 };
-/* This is a bit messy since the two input muxes in the ALC260 have slight
- * variations in their signal assignments.  The ideal way to deal with this
- * is to extend alc_spec.input_mux to allow a different input MUX for each
- * ADC.  For the purposes of the test model it's sufficient to just list
- * both options for affected signal indices.  The separate input mux
- * functionality only needs to be considered if a model comes along which
- * actually uses signals 0x5, 0x6 and 0x7 for something which makes sense to
- * record.
+/* For testing the ALC260, each input MUX needs its own definition since
+ * the signal assignments are different.  This assumes that the first ADC 
+ * is NID 0x04.
  */
-static struct hda_input_mux alc260_test_capture_source = {
-	.num_items = 8,
-	.items = {
-		{ "MIC1 pin", 0x0 },
-		{ "MIC2 pin", 0x1 },
-		{ "LINE1 pin", 0x2 },
-		{ "LINE2 pin", 0x3 },
-		{ "CD pin", 0x4 },
-		{ "LINE-OUT pin (cap1), Mixer (cap2)", 0x5 },
-		{ "HP-OUT pin (cap1), LINE-OUT pin (cap2)", 0x6 },
-		{ "HP-OUT pin (cap2 only)", 0x7 },
+static struct hda_input_mux alc260_test_capture_sources[2] = {
+	{
+		.num_items = 7,
+		.items = {
+			{ "MIC1 pin", 0x0 },
+			{ "MIC2 pin", 0x1 },
+			{ "LINE1 pin", 0x2 },
+			{ "LINE2 pin", 0x3 },
+			{ "CD pin", 0x4 },
+			{ "LINE-OUT pin", 0x5 },
+			{ "HP-OUT pin", 0x6 },
+		},
+        },
+	{
+		.num_items = 8,
+		.items = {
+			{ "MIC1 pin", 0x0 },
+			{ "MIC2 pin", 0x1 },
+			{ "LINE1 pin", 0x2 },
+			{ "LINE2 pin", 0x3 },
+			{ "CD pin", 0x4 },
+			{ "Mixer", 0x5 },
+			{ "LINE-OUT pin", 0x6 },
+			{ "HP-OUT pin", 0x7 },
+		},
         },
 };
 static struct snd_kcontrol_new alc260_test_mixer[] = {
@@ -3244,7 +3415,17 @@ static struct snd_kcontrol_new alc260_test_mixer[] = {
 	HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
 
-	/* Modes for retasking pin widgets */
+	/* Modes for retasking pin widgets
+	 * Note: the ALC260 doesn't seem to act on requests to enable mic
+         * bias from NIDs 0x0f and 0x10.  The ALC260 datasheet doesn't
+         * mention this restriction.  At this stage it's not clear whether
+         * this behaviour is intentional or is a hardware bug in chip
+         * revisions available at least up until early 2006.  Therefore for
+         * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
+         * choices, but if it turns out that the lack of mic bias for these
+         * NIDs is intentional we could change their modes from
+         * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
+	 */
 	ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
 	ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
 	ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
@@ -3606,6 +3787,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 
 	spec->init_verbs[spec->num_init_verbs++] = alc260_volume_init_verbs;
 
+	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
 	/* check whether NID 0x04 is valid */
@@ -3711,7 +3893,8 @@ static struct alc_config_preset alc260_presets[] = {
 		.adc_nids = alc260_dual_adc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
-		.input_mux = &alc260_fujitsu_capture_source,
+		.num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
+		.input_mux = alc260_fujitsu_capture_sources,
 	},
 	[ALC260_ACER] = {
 		.mixers = { alc260_acer_mixer,
@@ -3723,7 +3906,8 @@ static struct alc_config_preset alc260_presets[] = {
 		.adc_nids = alc260_dual_adc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
-		.input_mux = &alc260_acer_capture_source,
+		.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
+		.input_mux = alc260_acer_capture_sources,
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC260_TEST] = {
@@ -3736,7 +3920,8 @@ static struct alc_config_preset alc260_presets[] = {
 		.adc_nids = alc260_test_adc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc260_modes),
 		.channel_mode = alc260_modes,
-		.input_mux = &alc260_test_capture_source,
+		.num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
+		.input_mux = alc260_test_capture_sources,
 	},
 #endif
 };
@@ -3828,7 +4013,6 @@ static struct hda_input_mux alc882_capture_source = {
 		{ "CD", 0x4 },
 	},
 };
-
 #define alc882_mux_enum_info alc_mux_enum_info
 #define alc882_mux_enum_get alc_mux_enum_get
 
@@ -4730,6 +4914,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
 
 	spec->init_verbs[spec->num_init_verbs++] = alc262_volume_init_verbs;
+	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
 	return 1;
@@ -5406,6 +5591,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 
 	spec->init_verbs[spec->num_init_verbs++] = alc861_auto_init_verbs;
 
+	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux;
 
 	spec->adc_nids = alc861_adc_nids;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b56ca4019392..abe9493f0a2c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -534,6 +534,22 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int pincap = snd_hda_param_read(codec, nid,
+						 AC_PAR_PIN_CAP);
+	pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+	if (pincap & AC_PINCAP_VREF_100)
+		return AC_PINCTL_VREF_100;
+	if (pincap & AC_PINCAP_VREF_80)
+		return AC_PINCTL_VREF_80;
+	if (pincap & AC_PINCAP_VREF_50)
+		return AC_PINCTL_VREF_50;
+	if (pincap & AC_PINCAP_VREF_GRD)
+		return AC_PINCTL_VREF_GRD;
+	return 0;
+}
+
 static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type)
 
 {
@@ -571,9 +587,12 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 
 	if (val)
 		stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
-	else
-		stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
-
+	else {
+		unsigned int pinctl = AC_PINCTL_IN_EN;
+		if (io_idx) /* set VREF for mic */
+			pinctl |= stac92xx_get_vref(codec, nid);
+		stac92xx_auto_set_pinctl(codec, nid, pinctl);
+	}
         return 1;
 }
 
@@ -767,13 +786,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin
 		return 0;
 
 	wid_caps = get_wcaps(codec, pin);
-	if (wid_caps & AC_WCAP_UNSOL_CAP) {
-		/* Enable unsolicited responses on the HP widget */
-		snd_hda_codec_write(codec, pin, 0,
-				AC_VERB_SET_UNSOLICITED_ENABLE,
-				STAC_UNSOL_ENABLE);
+	if (wid_caps & AC_WCAP_UNSOL_CAP)
 		spec->hp_detect = 1;
-	}
 
 	nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
 	for (i = 0; i < cfg->line_outs; i++) {
@@ -896,13 +910,8 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec,
 		return 0;
 
 	wid_caps = get_wcaps(codec, pin);
-	if (wid_caps & AC_WCAP_UNSOL_CAP) {
-		/* Enable unsolicited responses on the HP widget */
-		snd_hda_codec_write(codec, pin, 0,
-				AC_VERB_SET_UNSOLICITED_ENABLE,
-				STAC_UNSOL_ENABLE);
+	if (wid_caps & AC_WCAP_UNSOL_CAP)
 		spec->hp_detect = 1;
-	}
 
 	return 0;
 }
@@ -944,6 +953,10 @@ static int stac92xx_init(struct hda_codec *codec)
 
 	/* set up pins */
 	if (spec->hp_detect) {
+		/* Enable unsolicited responses on the HP widget */
+		snd_hda_codec_write(codec, cfg->hp_pin, 0,
+				AC_VERB_SET_UNSOLICITED_ENABLE,
+				STAC_UNSOL_ENABLE);
 		/* fake event to set up pins */
 		codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
 	} else {
@@ -951,9 +964,13 @@ static int stac92xx_init(struct hda_codec *codec)
 		stac92xx_auto_init_hp_out(codec);
 	}
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (cfg->input_pins[i])
-			stac92xx_auto_set_pinctl(codec, cfg->input_pins[i],
-						 AC_PINCTL_IN_EN);
+		hda_nid_t nid = cfg->input_pins[i];
+		if (nid) {
+			unsigned int pinctl = AC_PINCTL_IN_EN;
+			if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC)
+				pinctl |= stac92xx_get_vref(codec, nid);
+			stac92xx_auto_set_pinctl(codec, nid, pinctl);
+		}
 	}
 	if (cfg->dig_out_pin)
 		stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin,