From 302e9c5af4fb3ea258917ee6a32e9e45f578b231 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@suse.cz>
Date: Wed, 5 Jul 2006 17:39:49 +0200
Subject: [ALSA] HDA codec & CA0106 - add/fix TLV support

Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/hda/patch_analog.c | 17 +++++++++++++++++
 1 file changed, 17 insertions(+)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 6823f2bc10b..54506d4e57d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -452,6 +452,19 @@ static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl
 	return change;
 }
 
+static int ad1986a_pcm_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
+			       unsigned int size, unsigned int __user *_tlv)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad198x_spec *ad = codec->spec;
+
+	mutex_lock(&ad->amp_mutex);
+	snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, _tlv);
+	mutex_unlock(&ad->amp_mutex);
+	return 0;
+}
+
+
 #define ad1986a_pcm_amp_sw_info		snd_hda_mixer_amp_switch_info
 
 static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -488,9 +501,13 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "PCM Playback Volume",
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			  SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+			  SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,
 		.info = ad1986a_pcm_amp_vol_info,
 		.get = ad1986a_pcm_amp_vol_get,
 		.put = ad1986a_pcm_amp_vol_put,
+		.tlv.c = ad1986a_pcm_amp_tlv,
 		.private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
 	},
 	{
-- 
cgit v1.2.3-70-g09d2


From 0a197f005a27766f5c9e0d960e7650748ec1ee4f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Jul 2006 14:51:14 +0200
Subject: [ALSA] Add model entry for Samsung X10 laptop

Added the proper model entry (laptop-eapd) for Samsung X10-T2300
Culesa laptop with AD1986A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/hda/patch_analog.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 54506d4e57d..e547442e6fe 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -820,6 +820,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
 	  .config = AD1986A_LAPTOP_EAPD }, /* Samsung X60 Chane */
 	{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc024,
 	  .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */
+	{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc026,
+	  .config = AD1986A_LAPTOP_EAPD }, /* Samsung X10-T2300 Culesa */
 	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1153,
 	  .config = AD1986A_LAPTOP_EAPD }, /* ASUS M9 */
 	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x1213,
-- 
cgit v1.2.3-70-g09d2


From 827a56ea3d9c3d5f80c5520ba9d487f9b7069238 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Jul 2006 14:51:16 +0200
Subject: [ALSA] Added model for ASUS M2NPV-VM mobo

Added the proper model (3stack) for ASUS M2NPV-VM mobo with AD1986A
codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/hda/patch_analog.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e547442e6fe..8955397cca6 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -808,6 +808,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
 	  .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */
 	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3,
 	  .config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */
+	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x81cb,
+	  .config = AD1986A_3STACK }, /* ASUS M2NPV-VM */
 	{ .modelname = "laptop",	.config = AD1986A_LAPTOP },
 	{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e,
 	  .config = AD1986A_LAPTOP }, /* FSC V2060 */
-- 
cgit v1.2.3-70-g09d2


From 4e195a7b78618c89b06547f3140e67a69ec23272 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 28 Jul 2006 14:47:34 +0200
Subject: [ALSA] Fix noisy output with shared channel mode with hd-audio

- Fix the wrong initialization of num_dacs when changing the channel
  mode between 2 and multi-channel modes.  It must be evaluated
  after calling snd_hda_ch_mode_put()
- Added the similar check of num_dacs fix in Realtek code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/hda/patch_analog.c  |  8 +++++---
 sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++---
 2 files changed, 29 insertions(+), 6 deletions(-)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 8955397cca6..077f1ce01ee 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1647,10 +1647,12 @@ 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)
+	int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+				      spec->num_channel_mode,
+				      &spec->multiout.max_channels);
+	if (! err && 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);
+	return err;
 }
 
 /* 6-stack mode */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 378e5f111e3..991f1079116 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -155,6 +155,7 @@ struct alc_spec {
 	/* channel model */
 	const struct hda_channel_mode *channel_mode;
 	int num_channel_mode;
+	int need_dac_fix;
 
 	/* PCM information */
 	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */
@@ -192,6 +193,7 @@ struct alc_config_preset {
 	hda_nid_t dig_in_nid;
 	unsigned int num_channel_mode;
 	const struct hda_channel_mode *channel_mode;
+	int need_dac_fix;
 	unsigned int num_mux_defs;
 	const struct hda_input_mux *input_mux;
 	void (*unsol_event)(struct hda_codec *, unsigned int);
@@ -264,9 +266,12 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
-				   spec->num_channel_mode,
-				   &spec->multiout.max_channels);
+	int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
+				      spec->num_channel_mode,
+				      &spec->multiout.max_channels);
+	if (! err && spec->need_dac_fix)
+		spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+	return err;
 }
 
 /*
@@ -546,6 +551,7 @@ static void setup_preset(struct alc_spec *spec,
 	
 	spec->channel_mode = preset->channel_mode;
 	spec->num_channel_mode = preset->num_channel_mode;
+	spec->need_dac_fix = preset->need_dac_fix;
 
 	spec->multiout.max_channels = spec->channel_mode[0].channels;
 
@@ -2278,6 +2284,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dac_nids = alc880_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
 		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_3ST_DIG] = {
@@ -2288,6 +2295,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dig_out_nid = ALC880_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
 		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_TCL_S700] = {
@@ -2380,6 +2388,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dac_nids = alc880_asus_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
 		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_ASUS_DIG] = {
@@ -2391,6 +2400,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dig_out_nid = ALC880_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
 		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_ASUS_DIG2] = {
@@ -2402,6 +2412,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dig_out_nid = ALC880_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
 		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_ASUS_W1V] = {
@@ -2413,6 +2424,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dig_out_nid = ALC880_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
 		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_UNIWILL_DIG] = {
@@ -2423,6 +2435,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dig_out_nid = ALC880_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
 		.channel_mode = alc880_asus_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_CLEVO] = {
@@ -2434,6 +2447,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
 		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_LG] = {
@@ -2445,6 +2459,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.dig_out_nid = ALC880_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
 		.channel_mode = alc880_lg_ch_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc880_lg_capture_source,
 		.unsol_event = alc880_lg_unsol_event,
 		.init_hook = alc880_lg_automute,
@@ -4437,6 +4452,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.dig_in_nid = ALC882_DIGIN_NID,
 		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
 		.channel_mode = alc882_ch_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc882_capture_source,
 	},
 	[ALC882_6ST_DIG] = {
@@ -5075,6 +5091,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.dig_in_nid = ALC883_DIGIN_NID,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
 		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
 	},	
 	[ALC883_3ST_6ch] = {
@@ -5086,6 +5103,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.adc_nids = alc883_adc_nids,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
 		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
 	},	
 	[ALC883_6ST_DIG] = {
@@ -6554,6 +6572,7 @@ static struct alc_config_preset alc861_presets[] = {
 		.dac_nids = alc861_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
 		.channel_mode = alc861_threestack_modes,
+		.need_dac_fix = 1,
 		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
 		.adc_nids = alc861_adc_nids,
 		.input_mux = &alc861_capture_source,
@@ -6566,6 +6585,7 @@ static struct alc_config_preset alc861_presets[] = {
 		.dig_out_nid = ALC861_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
 		.channel_mode = alc861_threestack_modes,
+		.need_dac_fix = 1,
 		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
 		.adc_nids = alc861_adc_nids,
 		.input_mux = &alc861_capture_source,
@@ -6589,6 +6609,7 @@ static struct alc_config_preset alc861_presets[] = {
 		.dac_nids = alc660_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
 		.channel_mode = alc861_threestack_modes,
+		.need_dac_fix = 1,
 		.num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
 		.adc_nids = alc861_adc_nids,
 		.input_mux = &alc861_capture_source,
-- 
cgit v1.2.3-70-g09d2


From 7cf0a95310f21f3c986288a483801b1d5694dee1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 17 Aug 2006 16:23:07 +0200
Subject: [ALSA] Fix compile errors with older gcc

Fixed compile errors with older gcc for initialization of a union.
sound/pci/ca0106/ca0106_mixer.c: At top level:
sound/pci/ca0106/ca0106_mixer.c:499: unknown field 'p' specified in initializer
sound/pci/ca0106/ca0106_mixer.c:499: warning: missing braces around initializer
sound/pci/ca0106/ca0106_mixer.c:499: warning: (near initialization for 'snd_ca0106_volume_ctls[0].tlv')

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/ca0106/ca0106_mixer.c | 4 ++--
 sound/pci/emu10k1/p16v.c        | 2 +-
 sound/pci/hda/hda_local.h       | 2 +-
 sound/pci/hda/patch_analog.c    | 2 +-
 4 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 6d64438cecc..9855f528ea7 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -478,7 +478,7 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
 	.info =	 snd_ca0106_volume_info,			\
 	.get =   snd_ca0106_volume_get,				\
 	.put =   snd_ca0106_volume_put,				\
-	.tlv.p = snd_ca0106_db_scale1,				\
+	.tlv = { .p = snd_ca0106_db_scale1 },			\
 	.private_value = ((chid) << 8) | (reg)			\
 }
 
@@ -490,7 +490,7 @@ static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
 	.info =  snd_ca0106_i2c_volume_info,			\
 	.get =   snd_ca0106_i2c_volume_get,			\
 	.put =   snd_ca0106_i2c_volume_put,			\
-	.tlv.p = snd_ca0106_db_scale2,				\
+	.tlv = { .p = snd_ca0106_db_scale2 },			\
 	.private_value = chid					\
 }
 
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 1e44714b862..4e0f95438f4 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -794,7 +794,7 @@ static DECLARE_TLV_DB_SCALE(snd_p16v_db_scale1, -5175, 25, 1);
 	.info = snd_p16v_volume_info, \
 	.get = snd_p16v_volume_get, \
 	.put = snd_p16v_volume_put, \
-	.tlv.p = snd_p16v_db_scale1, \
+	.tlv = { .p = snd_p16v_db_scale1 },	\
 	.private_value = ((xreg) | ((xhl) << 8)) \
 }
 
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 0f0ae685a9c..ff24266fe35 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -36,7 +36,7 @@
 	  .info = snd_hda_mixer_amp_volume_info, \
 	  .get = snd_hda_mixer_amp_volume_get, \
 	  .put = snd_hda_mixer_amp_volume_put, \
-	  .tlv.c = snd_hda_mixer_amp_tlv, \
+	  .tlv = { .c = snd_hda_mixer_amp_tlv },		\
 	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
 /* stereo volume with index */
 #define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 077f1ce01ee..043256c67d1 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -507,7 +507,7 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 		.info = ad1986a_pcm_amp_vol_info,
 		.get = ad1986a_pcm_amp_vol_get,
 		.put = ad1986a_pcm_amp_vol_put,
-		.tlv.c = ad1986a_pcm_amp_tlv,
+		.tlv = { .c = ad1986a_pcm_amp_tlv },
 		.private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
 	},
 	{
-- 
cgit v1.2.3-70-g09d2


From c256652466127872f1b2e510431dc25524ba40ba Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 17 Aug 2006 18:21:36 +0200
Subject: [ALSA] Add missing TLV callbacks for HD-audio codecs

Added missing TLV callbacks for HD-audio codec supports.
Also cleaned up the tlv callback for ad1986a (no mutex is needed there).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/hda/patch_analog.c   | 16 ++--------------
 sound/pci/hda/patch_realtek.c  |  1 +
 sound/pci/hda/patch_sigmatel.c |  1 +
 3 files changed, 4 insertions(+), 14 deletions(-)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 043256c67d1..71abc2aa61a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -452,19 +452,6 @@ static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl
 	return change;
 }
 
-static int ad1986a_pcm_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
-			       unsigned int size, unsigned int __user *_tlv)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *ad = codec->spec;
-
-	mutex_lock(&ad->amp_mutex);
-	snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, _tlv);
-	mutex_unlock(&ad->amp_mutex);
-	return 0;
-}
-
-
 #define ad1986a_pcm_amp_sw_info		snd_hda_mixer_amp_switch_info
 
 static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
@@ -507,7 +494,7 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 		.info = ad1986a_pcm_amp_vol_info,
 		.get = ad1986a_pcm_amp_vol_get,
 		.put = ad1986a_pcm_amp_vol_put,
-		.tlv = { .c = ad1986a_pcm_amp_tlv },
+		.tlv = { .c = snd_hda_mixer_amp_tlv },
 		.private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
 	},
 	{
@@ -654,6 +641,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
 		.info = snd_hda_mixer_amp_volume_info,
 		.get = snd_hda_mixer_amp_volume_get,
 		.put = ad1986a_laptop_master_vol_put,
+		.tlv = { .c = snd_hda_mixer_amp_tlv },
 		.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
 	},
 	{
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f857e963ff4..79d361260b2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5540,6 +5540,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
 		.info = snd_hda_mixer_amp_volume_info,
 		.get = snd_hda_mixer_amp_volume_get,
 		.put = alc262_fujitsu_master_vol_put,
+		.tlv = { .c = snd_hda_mixer_amp_tlv },
 		.private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
 	},
 	{
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7eaf755b014..887b52e96ec 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1528,6 +1528,7 @@ static struct snd_kcontrol_new vaio_mixer[] = {
 		.info = snd_hda_mixer_amp_volume_info,
 		.get = snd_hda_mixer_amp_volume_get,
 		.put = vaio_master_vol_put,
+		.tlv = { .c = snd_hda_mixer_amp_tlv },
 		.private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
 	},
 	{
-- 
cgit v1.2.3-70-g09d2


From eb06ed8f4c2440558ebf465e8baeac6367d90201 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 20 Sep 2006 17:10:27 +0200
Subject: [ALSA] hda-codec - Support multiple headphone pins

Some machines have multiple headpohne pins (usually on the lpatop
and on the docking station) while the current hda-codec driver
assumes a single headphone pin.  Now it supports multiple hp pins
(at least for detection).
The sigmatel 92xx code supports this new multiple hp pins.
It detects all hp pins for auto-muting, too.
Also, the driver checks speaker pins in addition.  In some cases,
all line-out, speaker and hp-pins coexist.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
---
 sound/pci/hda/hda_codec.c      |  23 +++--
 sound/pci/hda/hda_local.h      |   3 +-
 sound/pci/hda/patch_analog.c   |   4 +-
 sound/pci/hda/patch_realtek.c  |  18 ++--
 sound/pci/hda/patch_sigmatel.c | 202 +++++++++++++++++++++++++++--------------
 5 files changed, 164 insertions(+), 86 deletions(-)

(limited to 'sound/pci/hda/patch_analog.c')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8b2c080c85a..07360996caa 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2012,7 +2012,7 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)
  * in the order of front, rear, CLFE, side, ...
  *
  * If more extra outputs (speaker and headphone) are found, the pins are
- * assisnged to hp_pin and speaker_pins[], respectively.  If no line-out jack
+ * assisnged to hp_pins[] and speaker_pins[], respectively.  If no line-out jack
  * is detected, one of speaker of HP pins is assigned as the primary
  * output, i.e. to line_out_pins[0].  So, line_outs is always positive
  * if any analog output exists.
@@ -2074,7 +2074,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c
 			cfg->speaker_outs++;
 			break;
 		case AC_JACK_HP_OUT:
-			cfg->hp_pin = nid;
+			if (cfg->hp_outs >= ARRAY_SIZE(cfg->hp_pins))
+				continue;
+			cfg->hp_pins[cfg->hp_outs] = nid;
+			cfg->hp_outs++;
 			break;
 		case AC_JACK_MIC_IN:
 			if (loc == AC_JACK_LOC_FRONT)
@@ -2147,8 +2150,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c
 		   cfg->speaker_outs, cfg->speaker_pins[0],
 		   cfg->speaker_pins[1], cfg->speaker_pins[2],
 		   cfg->speaker_pins[3], cfg->speaker_pins[4]);
-	snd_printd("   hp=0x%x, dig_out=0x%x, din_in=0x%x\n",
-		   cfg->hp_pin, cfg->dig_out_pin, cfg->dig_in_pin);
+	snd_printd("   hp_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
+		   cfg->hp_outs, cfg->hp_pins[0],
+		   cfg->hp_pins[1], cfg->hp_pins[2],
+		   cfg->hp_pins[3], cfg->hp_pins[4]);
 	snd_printd("   inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x,"
 		   " cd=0x%x, aux=0x%x\n",
 		   cfg->input_pins[AUTO_PIN_MIC],
@@ -2169,10 +2174,12 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c
 			       sizeof(cfg->speaker_pins));
 			cfg->speaker_outs = 0;
 			memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
-		} else if (cfg->hp_pin) {
-			cfg->line_outs = 1;
-			cfg->line_out_pins[0] = cfg->hp_pin;
-			cfg->hp_pin = 0;
+		} else if (cfg->hp_outs) {
+			cfg->line_outs = cfg->hp_outs;
+			memcpy(cfg->line_out_pins, cfg->hp_pins,
+			       sizeof(cfg->hp_pins));
+			cfg->hp_outs = 0;
+			memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
 		}
 	}
 
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index ff24266fe35..f9416c36396 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -229,7 +229,8 @@ struct auto_pin_cfg {
 	hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */
 	int speaker_outs;
 	hda_nid_t speaker_pins[5];
-	hda_nid_t hp_pin;
+	int hp_outs;
+	hda_nid_t hp_pins[5];
 	hda_nid_t input_pins[AUTO_PIN_LAST];
 	hda_nid_t dig_out_pin;
 	hda_nid_t dig_in_pin;
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 71abc2aa61a..511df07fa2a 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2471,7 +2471,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec)
 	pin = spec->autocfg.speaker_pins[0];
 	if (pin) /* connect to front */
 		ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
-	pin = spec->autocfg.hp_pin;
+	pin = spec->autocfg.hp_pins[0];
 	if (pin) /* connect to front */
 		ad1988_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
 }
@@ -2523,7 +2523,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	    (err = ad1988_auto_create_extra_out(codec,
 						spec->autocfg.speaker_pins[0],
 						"Speaker")) < 0 ||
-	    (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin,
+	    (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
 						"Headphone")) < 0 ||
 	    (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
 		return err;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ba9e050e201..d08d2e399c8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2753,7 +2753,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec)
 	pin = spec->autocfg.speaker_pins[0];
 	if (pin) /* connect to front */
 		alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
-	pin = spec->autocfg.hp_pin;
+	pin = spec->autocfg.hp_pins[0];
 	if (pin) /* connect to front */
 		alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
 }
@@ -2794,7 +2794,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	    (err = alc880_auto_create_extra_out(spec,
 						spec->autocfg.speaker_pins[0],
 						"Speaker")) < 0 ||
-	    (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin,
+	    (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
 						"Headphone")) < 0 ||
 	    (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
 		return err;
@@ -3736,7 +3736,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 			return err;
 	}
 
-	nid = cfg->hp_pin;
+	nid = cfg->hp_pins[0];
 	if (nid) {
 		err = alc260_add_playback_controls(spec, nid, "Headphone");
 		if (err < 0)
@@ -3806,7 +3806,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec)
 	if (nid)
 		alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
 
-	nid = spec->autocfg.hp_pin;
+	nid = spec->autocfg.hp_pins[0];
 	if (nid)
 		alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0);
 }	
@@ -4526,7 +4526,7 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t pin;
 
-	pin = spec->autocfg.hp_pin;
+	pin = spec->autocfg.hp_pins[0];
 	if (pin) /* connect to front */
 		alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); /* use dac 0 */
 }
@@ -5207,7 +5207,7 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t pin;
 
-	pin = spec->autocfg.hp_pin;
+	pin = spec->autocfg.hp_pins[0];
 	if (pin) /* connect to front */
 		/* use dac 0 */
 		alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
@@ -5630,7 +5630,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct
 				return err;
 		}
 	}
-	nid = cfg->hp_pin;
+	nid = cfg->hp_pins[0];
 	if (nid) {
 		/* spec->multiout.hp_nid = 2; */
 		if (nid == 0x16) {
@@ -6630,7 +6630,7 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t pin;
 
-	pin = spec->autocfg.hp_pin;
+	pin = spec->autocfg.hp_pins[0];
 	if (pin) /* connect to front */
 		alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, spec->multiout.dac_nids[0]);
 }
@@ -6665,7 +6665,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 
 	if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 ||
 	    (err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
-	    (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 ||
+	    (err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0])) < 0 ||
 	    (err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
 		return err;
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index bcbbe111ab9..7cc06426520 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1011,11 +1011,29 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
 	return 0;
 }
 
+/* create volume control/switch for the given prefx type */
+static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs)
+{
+	char name[32];
+	int err;
+
+	sprintf(name, "%s Playback Volume", pfx);
+	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
+				   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+	sprintf(name, "%s Playback Switch", pfx);
+	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
+				   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+	if (err < 0)
+		return err;
+	return 0;
+}
+
 /* add playback controls from the parsed DAC table */
 static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
 					       const struct auto_pin_cfg *cfg)
 {
-	char name[32];
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
@@ -1030,26 +1048,15 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
 
 		if (i == 2) {
 			/* Center/LFE */
-			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Center Playback Volume",
-					       HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
+			err = create_controls(spec, "Center", nid, 1);
+			if (err < 0)
 				return err;
-			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "LFE Playback Volume",
-					       HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
-				return err;
-			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Center Playback Switch",
-					       HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
-				return err;
-			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "LFE Playback Switch",
-					       HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
+			err = create_controls(spec, "LFE", nid, 2);
+			if (err < 0)
 				return err;
 		} else {
-			sprintf(name, "%s Playback Volume", chname[i]);
-			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
-					       HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
-				return err;
-			sprintf(name, "%s Playback Switch", chname[i]);
-			if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
-					       HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+			err = create_controls(spec, chname[i], nid, 3);
+			if (err < 0)
 				return err;
 		}
 	}
@@ -1065,39 +1072,85 @@ static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
 	return 0;
 }
 
-/* add playback controls for HP output */
-static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg)
+static int check_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid)
 {
-	struct sigmatel_spec *spec = codec->spec;
-	hda_nid_t pin = cfg->hp_pin;
-	hda_nid_t nid;
-	int i, err;
-	unsigned int wid_caps;
+	int i;
 
-	if (! pin)
-		return 0;
+	for (i = 0; i < spec->multiout.num_dacs; i++) {
+		if (spec->multiout.dac_nids[i] == nid)
+			return 1;
+	}
+	if (spec->multiout.hp_nid == nid)
+		return 1;
+	return 0;
+}
 
-	wid_caps = get_wcaps(codec, pin);
-	if (wid_caps & AC_WCAP_UNSOL_CAP)
-		spec->hp_detect = 1;
+static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
+{
+	if (!spec->multiout.hp_nid)
+		spec->multiout.hp_nid = nid;
+	else if (spec->multiout.num_dacs > 4) {
+		printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid);
+		return 1;
+	} else {
+		spec->multiout.dac_nids[spec->multiout.num_dacs] = nid;
+		spec->multiout.num_dacs++;
+	}
+	return 0;
+}
 
-	nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
-	for (i = 0; i < cfg->line_outs; i++) {
-		if (! spec->multiout.dac_nids[i])
+/* add playback controls for Speaker and HP outputs */
+static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
+					struct auto_pin_cfg *cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	hda_nid_t nid;
+	int i, old_num_dacs, err;
+
+	old_num_dacs = spec->multiout.num_dacs;
+	for (i = 0; i < cfg->hp_outs; i++) {
+		unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]);
+		if (wid_caps & AC_WCAP_UNSOL_CAP)
+			spec->hp_detect = 1;
+		nid = snd_hda_codec_read(codec, cfg->hp_pins[i], 0,
+					 AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+		if (check_in_dac_nids(spec, nid))
+			nid = 0;
+		if (! nid)
 			continue;
-		if (spec->multiout.dac_nids[i] == nid)
-			return 0;
+		add_spec_dacs(spec, nid);
+	}
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		nid = snd_hda_codec_read(codec, cfg->speaker_pins[0], 0,
+					 AC_VERB_GET_CONNECT_LIST, 0) & 0xff;
+		if (check_in_dac_nids(spec, nid))
+			nid = 0;
+		if (check_in_dac_nids(spec, nid))
+			nid = 0;
+		if (! nid)
+			continue;
+		add_spec_dacs(spec, nid);
 	}
 
-	spec->multiout.hp_nid = nid;
-
-	/* control HP volume/switch on the output mixer amp */
-	if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, "Headphone Playback Volume",
-					HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
-		return err;
-	if ((err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
-					HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
-		return err;
+	for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) {
+		static const char *pfxs[] = {
+			"Speaker", "External Speaker", "Speaker2",
+		};
+		err = create_controls(spec, pfxs[i - old_num_dacs],
+				      spec->multiout.dac_nids[i], 3);
+		if (err < 0)
+			return err;
+	}
+	if (spec->multiout.hp_nid) {
+		const char *pfx;
+		if (old_num_dacs == spec->multiout.num_dacs)
+			pfx = "Master";
+		else
+			pfx = "Headphone";
+		err = create_controls(spec, pfx, spec->multiout.hp_nid, 3);
+		if (err < 0)
+			return err;
+	}
 
 	return 0;
 }
@@ -1160,11 +1213,20 @@ static void stac92xx_auto_init_multi_out(struct hda_codec *codec)
 static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	hda_nid_t pin;
+	int i;
 
-	pin = spec->autocfg.hp_pin;
-	if (pin) /* connect to front */
-		stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+	for (i = 0; i < spec->autocfg.hp_outs; i++) {
+		hda_nid_t pin;
+		pin = spec->autocfg.hp_pins[i];
+		if (pin) /* connect to front */
+			stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+	}
+	for (i = 0; i < spec->autocfg.speaker_outs; i++) {
+		hda_nid_t pin;
+		pin = spec->autocfg.speaker_pins[i];
+		if (pin) /* connect to front */
+			stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN);
+	}
 }
 
 static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
@@ -1210,7 +1272,7 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec,
 					struct auto_pin_cfg *cfg)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	hda_nid_t pin = cfg->hp_pin;
+	hda_nid_t pin = cfg->hp_pins[0];
 	unsigned int wid_caps;
 
 	if (! pin)
@@ -1266,16 +1328,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
 	}
 
 	if (lfe_pin) {
-		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL,
-					   "LFE Playback Volume",
-					   HDA_COMPOSE_AMP_VAL(lfe_pin, 1, 0,
-							       HDA_OUTPUT));
-		if (err < 0)
-			return err;
-		err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE,
-					   "LFE Playback Switch",
-					   HDA_COMPOSE_AMP_VAL(lfe_pin, 1, 0,
-							       HDA_OUTPUT));
+		err = create_controls(spec, "LFE", lfe_pin, 1);
 		if (err < 0)
 			return err;
 	}
@@ -1363,9 +1416,11 @@ 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);
+		for (i = 0; i < cfg->hp_outs; i++)
+			if (get_wcaps(codec, cfg->hp_pins[i]) & AC_WCAP_UNSOL_CAP)
+				snd_hda_codec_write(codec, cfg->hp_pins[i], 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);
 		/* enable the headphones by default.  If/when unsol_event detection works, this will be ignored */
@@ -1447,21 +1502,36 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	if ((res >> 26) != STAC_HP_EVENT)
 		return;
 
-	presence = snd_hda_codec_read(codec, cfg->hp_pin, 0,
-			AC_VERB_GET_PIN_SENSE, 0x00) >> 31;
+	presence = 0;
+	for (i = 0; i < cfg->hp_outs; i++) {
+		int p = snd_hda_codec_read(codec, cfg->hp_pins[i], 0,
+					   AC_VERB_GET_PIN_SENSE, 0x00);
+		if (p & (1 << 31))
+			presence++;
+	}
 
 	if (presence) {
 		/* disable lineouts, enable hp */
 		for (i = 0; i < cfg->line_outs; i++)
 			stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
 						AC_PINCTL_OUT_EN);
-		stac92xx_set_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN);
+		for (i = 0; i < cfg->speaker_outs; i++)
+			stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
+						AC_PINCTL_OUT_EN);
+		for (i = 0; i < cfg->hp_outs; i++)
+			stac92xx_set_pinctl(codec, cfg->hp_pins[i],
+					    AC_PINCTL_OUT_EN);
 	} else {
 		/* enable lineouts, disable hp */
 		for (i = 0; i < cfg->line_outs; i++)
 			stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
 						AC_PINCTL_OUT_EN);
-		stac92xx_reset_pinctl(codec, cfg->hp_pin, AC_PINCTL_OUT_EN);
+		for (i = 0; i < cfg->speaker_outs; i++)
+			stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
+						AC_PINCTL_OUT_EN);
+		for (i = 0; i < cfg->hp_outs; i++)
+			stac92xx_reset_pinctl(codec, cfg->hp_pins[i],
+					      AC_PINCTL_OUT_EN);
 	}
 } 
 
-- 
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