From adabb3ec8b0bcbd2ca81973d33c3da726b939c7c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 3 Aug 2011 07:48:37 +0200
Subject: ALSA: hda - Fix digital-mic mono recording on ASUS Eee PC

The digital-mic unit on ASUS Eee PC gives PDM signals instead of the
normal stereo PCM, thus you can't record a mono stream from the stereo
stream as is; the summed stereo signal results in almost zero level, and
you'll hear only soft noise.

As a workaround, use ALC269-specific COEF to manipulate the dmic route
for mono, like used for ALC271x.  This is implemented as a fix-up, thus
it works only with model=auto or without REALTEK_QUIRKS Kconfig.

Reported-and-tested-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++
 1 file changed, 26 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e125c60..9a1aa09 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4484,6 +4484,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec,
 	spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
 }
 
+static void alc269_fixup_stereo_dmic(struct hda_codec *codec,
+				     const struct alc_fixup *fix, int action)
+{
+	int coef;
+
+	if (action != ALC_FIXUP_ACT_INIT)
+		return;
+	/* The digital-mic unit sends PDM (differential signal) instead of
+	 * the standard PCM, thus you can't record a valid mono stream as is.
+	 * Below is a workaround specific to ALC269 to control the dmic
+	 * signal source as mono.
+	 */
+	coef = alc_read_coef_idx(codec, 0x07);
+	alc_write_coef_idx(codec, 0x07, coef | 0x80);
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -4494,6 +4510,7 @@ enum {
 	ALC275_FIXUP_SONY_HWEQ,
 	ALC271_FIXUP_DMIC,
 	ALC269_FIXUP_PCM_44K,
+	ALC269_FIXUP_STEREO_DMIC,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -4556,10 +4573,19 @@ static const struct alc_fixup alc269_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
 	},
+	[ALC269_FIXUP_STEREO_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_stereo_dmic,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
-- 
cgit v1.1


From c3540b81ee707bc8a7a83e850adf5feb3b84c04d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 4 Aug 2011 15:19:26 +0200
Subject: ALSA: hda - Use auto-parser for ASUS UX50, Eee PC P901, S101 and
 P1005

It works fine with auto-parser and now the digital mic workaround was
implemented in auto-parser fixup, let's drop the static model quirks for
these models.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/alc269_quirks.c | 7 -------
 1 file changed, 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/alc269_quirks.c b/sound/pci/hda/alc269_quirks.c
index 14fdcf2..5ac0e21 100644
--- a/sound/pci/hda/alc269_quirks.c
+++ b/sound/pci/hda/alc269_quirks.c
@@ -531,17 +531,10 @@ static const struct snd_pci_quirk alc269_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1653, "ASUS U50", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1693, "ASUS F50N", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS F5Q", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1723, "ASUS P80", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1743, "ASUS U80", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1773, "ASUS U20A", ALC269_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1883, "ASUS F81Se", ALC269_AMIC),
-	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
-		      ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
-		      ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC),
-	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC),
 	SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK),
 	SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC),
-- 
cgit v1.1


From 3d56c8e6b039f2bc1d7c62d3b19e696c3129c8c0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 5 Aug 2011 12:30:12 +0200
Subject: ALSA: hdspm - Fix uninitialized compile warnings

Put the exception checks for io_type switch() for possible mistakes in
future.  Also this shuts up annoying compile warnings.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdspm.c | 19 +++++++++++++------
 1 file changed, 13 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 6edc67c..493e394 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -1339,6 +1339,10 @@ static u64 hdspm_calc_dds_value(struct hdspm *hdspm, u64 period)
 		break;
 	case MADIface:
 		freq_const = 131072000000000ULL;
+		break;
+	default:
+		snd_BUG();
+		return 0;
 	}
 
 	return div_u64(freq_const, period);
@@ -1356,16 +1360,19 @@ static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
 
 	switch (hdspm->io_type) {
 	case MADIface:
-	  n = 131072000000000ULL;  /* 125 MHz */
-	  break;
+		n = 131072000000000ULL;  /* 125 MHz */
+		break;
 	case MADI:
 	case AES32:
-	  n = 110069313433624ULL;  /* 105 MHz */
-	  break;
+		n = 110069313433624ULL;  /* 105 MHz */
+		break;
 	case RayDAT:
 	case AIO:
-	  n = 104857600000000ULL;  /* 100 MHz */
-	  break;
+		n = 104857600000000ULL;  /* 100 MHz */
+		break;
+	default:
+		snd_BUG();
+		return;
 	}
 
 	n = div_u64(n, rate);
-- 
cgit v1.1


From 81c0a78b644f0e265a01d5a5f5ab397b791bad08 Mon Sep 17 00:00:00 2001
From: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Date: Fri, 5 Aug 2011 18:51:29 +0800
Subject: ALSA: hda - Fix a complile warning in patch_via.c

  sound/pci/hda/patch_via.c:2087: warning: 'dac' may be used uninitialized in this function

Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 84d8798..4ebfbd8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2084,7 +2084,7 @@ static int via_auto_create_speaker_ctls(struct hda_codec *codec)
 	struct via_spec *spec = codec->spec;
 	struct nid_path *path;
 	bool check_dac;
-	hda_nid_t pin, dac;
+	hda_nid_t pin, dac = 0;
 	int err;
 
 	pin = spec->autocfg.speaker_pins[0];
-- 
cgit v1.1


From 67ada8367c323ce13d0268c87cf09bf8af956e92 Mon Sep 17 00:00:00 2001
From: Thomas Meyer <thomas@m3y3r.de>
Date: Sat, 6 Aug 2011 13:26:20 +0200
Subject: ALSA: asihpi - use kzalloc()

 Use kzalloc rather than kmalloc followed by memset with 0

 This considers some simple cases that are common and easy to validate
 Note in particular that there are no ...s in the rule, so all of the
 matched code has to be contiguous

 The semantic patch that makes this output is available
 in scripts/coccinelle/api/alloc/kzalloc-simple.cocci.

 More information about semantic patching is available at
 http://coccinelle.lip6.fr/

Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpicmn.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 65b7ca1..bd47521 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -631,13 +631,12 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count,
 	if (!p_cache)
 		return NULL;
 
-	p_cache->p_info =
-		kmalloc(sizeof(*p_cache->p_info) * control_count, GFP_KERNEL);
+	p_cache->p_info = kzalloc(sizeof(*p_cache->p_info) * control_count,
+				  GFP_KERNEL);
 	if (!p_cache->p_info) {
 		kfree(p_cache);
 		return NULL;
 	}
-	memset(p_cache->p_info, 0, sizeof(*p_cache->p_info) * control_count);
 	p_cache->cache_size_in_bytes = size_in_bytes;
 	p_cache->control_count = control_count;
 	p_cache->p_cache = p_dsp_control_buffer;
-- 
cgit v1.1


From df944f66784e6d4f2f50739263a4947885d8b6ae Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 7 Aug 2011 17:34:07 +0200
Subject: ALSA: Fix dependency of CONFIG_SND_TEA575X

CONFIG_SND_TEA575X is enabled by RADIO_SF16FMR2, but the latter one is
no PCI device.  Since tea575x-tuner itself is independent from the board
bus type, the config should be moved out of SND_PCI dependency.

Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Acked-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/Kconfig | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 50abf5b..8816804 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -1,5 +1,10 @@
 # ALSA PCI drivers
 
+config SND_TEA575X
+	tristate
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
+	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+
 menuconfig SND_PCI
 	bool "PCI sound devices"
 	depends on PCI
@@ -563,11 +568,6 @@ config SND_FM801_TEA575X_BOOL
 	  FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and
 	  SF64-PCR) into the snd-fm801 driver.
 
-config SND_TEA575X
-	tristate
-	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
-
 source "sound/pci/hda/Kconfig"
 
 config SND_HDSP
-- 
cgit v1.1


From 96b635977984a88ecdb9cc76b8a54db7297f36e0 Mon Sep 17 00:00:00 2001
From: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Date: Wed, 10 Aug 2011 16:01:04 +0800
Subject: ALSA: hda - Add CONFIG_SND_HDA_POWER_SAVE to stac_vrefout_set()

In commit 45eebda7, it add new function stac_vrefout_set, but it
is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so
add the macro to avoid such warning:

  sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used

Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa376b5..5145b66 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -673,6 +673,7 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
 static int stac_vrefout_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
 {
@@ -696,6 +697,7 @@ static int stac_vrefout_set(struct hda_codec *codec,
 
 	return 1;
 }
+#endif
 
 static unsigned int stac92xx_vref_set(struct hda_codec *codec,
 					hda_nid_t nid, unsigned int new_vref)
-- 
cgit v1.1


From a5a3973da8b52944bc5909852714e55771c31ce7 Mon Sep 17 00:00:00 2001
From: Julia Lawall <julia@diku.dk>
Date: Wed, 10 Aug 2011 11:49:04 +0200
Subject: ALSA: azt3328 - adjust error handling code to include debugging code

snd_azf3328_dbgcallenter is called at the very beginning of the function,
so it could be useful to call snd_azf3328_dbgcallleave at all exit points.

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/azt3328.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index e4d76a2..579fc0dce 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2625,16 +2625,19 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 	int err;
 
 	snd_azf3328_dbgcallenter();
-	if (dev >= SNDRV_CARDS)
-		return -ENODEV;
+	if (dev >= SNDRV_CARDS) {
+		err = -ENODEV;
+		goto out;
+	}
 	if (!enable[dev]) {
 		dev++;
-		return -ENOENT;
+		err = -ENOENT;
+		goto out;
 	}
 
 	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 	if (err < 0)
-		return err;
+		goto out;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
-- 
cgit v1.1


From eade7b281c9fc18401b989c77d5e5e660b25a3b7 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sun, 14 Aug 2011 22:43:01 -0400
Subject: ALSA: ac97: Add HP Compaq dc5100 SFF(PT003AW) to Headphone Jack Sense
 whitelist

BugLink: https://bugs.launchpad.net/bugs/826081

The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.

Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_patch.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 200c9a1..a872d0a 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1909,6 +1909,7 @@ static unsigned int ad1981_jacks_whitelist[] = {
 	0x103c0944, /* HP nc6220 */
 	0x103c0934, /* HP nc8220 */
 	0x103c006d, /* HP nx9105 */
+	0x103c300d, /* HP Compaq dc5100 SFF(PT003AW) */
 	0x17340088, /* FSC Scenic-W */
 	0 /* end */
 };
-- 
cgit v1.1


From c503ad466da44ca23c658986629bf7a2e2eabbb7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 16 Aug 2011 14:23:20 +0200
Subject: ALSA: hda - Fix duplicated capture-volume creation for ALC268 models

Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations.  The capture mixers must be put to cap_mixer field
instead of mixers array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/alc268_quirks.c | 36 ++++++++++++++++++------------------
 1 file changed, 18 insertions(+), 18 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/alc268_quirks.c b/sound/pci/hda/alc268_quirks.c
index be58bf2..2e5876c 100644
--- a/sound/pci/hda/alc268_quirks.c
+++ b/sound/pci/hda/alc268_quirks.c
@@ -476,8 +476,8 @@ static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = {
 
 static const struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -492,8 +492,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_3ST] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_base_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
 		.dac_nids = alc268_dac_nids,
@@ -507,8 +507,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.input_mux = &alc268_capture_source,
 	},
 	[ALC268_TOSHIBA] = {
-		.mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_toshiba_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -525,8 +525,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_acer_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -543,8 +543,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_DMIC] = {
-		.mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_acer_dmic_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -561,9 +561,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
-		.mixers = { alc268_acer_aspire_one_mixer,
-			    alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer},
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -579,8 +578,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_DELL] = {
-		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
-			    alc268_capture_nosrc_mixer },
+		.mixers = { alc268_dell_mixer, alc268_beep_mixer},
+		.cap_mixer = alc268_capture_nosrc_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_dell_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -596,8 +595,8 @@ static const struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc_inithook,
 	},
 	[ALC268_ZEPTO] = {
-		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
-			    alc268_beep_mixer },
+		.mixers = { alc268_base_mixer, alc268_beep_mixer },
+		.cap_mixer = alc268_capture_alt_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_toshiba_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -616,7 +615,8 @@ static const struct alc_config_preset alc268_presets[] = {
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC268_TEST] = {
-		.mixers = { alc268_test_mixer, alc268_capture_mixer },
+		.mixers = { alc268_test_mixer },
+		.cap_mixer = alc268_capture_mixer,
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_volume_init_verbs,
 				alc268_beep_init_verbs },
-- 
cgit v1.1


From 3fe45aeaf2033c9eaa5028ed5ba68b466008876f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 18 Aug 2011 15:13:17 +0200
Subject: ALSA: hda - Add "PCM" volume to vmaster slave list

The new parser may use "PCM" volume, but it was missing the vmaster
slave list, thus "Master" volume didn't control it.

Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a1aa09..fcb11af 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1784,6 +1784,7 @@ static const char * const alc_slave_vols[] = {
 	"Speaker Playback Volume",
 	"Mono Playback Volume",
 	"Line-Out Playback Volume",
+	"PCM Playback Volume",
 	NULL,
 };
 
@@ -1798,6 +1799,7 @@ static const char * const alc_slave_sws[] = {
 	"Mono Playback Switch",
 	"IEC958 Playback Switch",
 	"Line-Out Playback Switch",
+	"PCM Playback Switch",
 	NULL,
 };
 
-- 
cgit v1.1


From b6acf013bdc6f6ff9643030add85832d44034a28 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 20 Aug 2011 09:14:45 +0200
Subject: ALSA: hda - Don't spew too many ELD errors

Currently HD-audio driver shows the all error ELD byte as an error
in the kernel message.  This is annoying when the video driver doesn't
set the correct ELD from the beginning. e.g. radeon sends a zero-byte
data, but we still check ELD with the fixed 128 byte as a workaround
for some broken devices, it spews 128-times errors.

For avoiding this, the driver aborts reading when the first byte is
invalid.  In such a case, the whole data is certainly invalid.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_eld.c | 31 +++++++++++++++++++------------
 1 file changed, 19 insertions(+), 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 28ce17d..c34f730 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = {
 	SNDRV_PCM_RATE_192000,	/* 7: 192000Hz */
 };
 
-static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid,
+static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid,
 					int byte_index)
 {
 	unsigned int val;
 
 	val = snd_hda_codec_read(codec, nid, 0,
 					AC_VERB_GET_HDMI_ELDD, byte_index);
-
 #ifdef BE_PARANOID
 	printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val);
 #endif
-
-	if ((val & AC_ELDD_ELD_VALID) == 0) {
-		snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n",
-								byte_index);
-		val = 0;
-	}
-
-	return val & AC_ELDD_ELD_DATA;
+	return val;
 }
 
 #define GRAB_BITS(buf, byte, lowbit, bits) 		\
@@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld,
 	if (!buf)
 		return -ENOMEM;
 
-	for (i = 0; i < size; i++)
-		buf[i] = hdmi_get_eld_byte(codec, nid, i);
+	for (i = 0; i < size; i++) {
+		unsigned int val = hdmi_get_eld_data(codec, nid, i);
+		if (!(val & AC_ELDD_ELD_VALID)) {
+			if (!i) {
+				snd_printd(KERN_INFO
+					   "HDMI: invalid ELD data\n");
+				ret = -EINVAL;
+				goto error;
+			}
+			snd_printd(KERN_INFO
+				  "HDMI: invalid ELD data byte %d\n", i);
+			val = 0;
+		} else
+			val &= AC_ELDD_ELD_DATA;
+		buf[i] = val;
+	}
 
 	ret = hdmi_update_eld(eld, buf, size);
 
+error:
 	kfree(buf);
 	return ret;
 }
-- 
cgit v1.1


From 1b004d03d8670bdd871e0f297ed20bc510e404de Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 20 Aug 2011 09:19:59 +0200
Subject: ALSA: hda - Fix error check from snd_hda_get_conn_index() in
 patch_cirrus.c

snd_hda_get_conn_index() returns a negative value while the current code
stores it in an unsigned int.  It must be stored in a signed integer.

Reported-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 47d6ffc..d6c93d9 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
 static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 			 unsigned int *idxp)
 {
-	int i;
+	int i, idx;
 	hda_nid_t nid;
 
 	nid = codec->start_nid;
@@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 		type = get_wcaps_type(get_wcaps(codec, nid));
 		if (type != AC_WID_AUD_IN)
 			continue;
-		*idxp = snd_hda_get_conn_index(codec, nid, pin, false);
-		if (*idxp >= 0)
+		idx = snd_hda_get_conn_index(codec, nid, pin, false);
+		if (idx >= 0) {
+			*idxp = idx;
 			return nid;
+		}
 	}
 	return 0;
 }
-- 
cgit v1.1


From 675c1aa3c4a7290e537e854d0af7cdf9692bd396 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 23 Aug 2011 12:36:28 +0200
Subject: ALSA: hda - Fix output-path initialization for Realtek auto-parser

When the headphone or speaker output has no own DAC, initialize the path
using the primary DAC.  Otherwise the path won't be set properly and
can result in the silence.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 20 +++++++++++++-------
 1 file changed, 13 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fcb11af..0fefc10 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
 static void alc_auto_init_extra_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
+	hda_nid_t pin, dac;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
-						  spec->multiout.hp_nid);
+	if (pin) {
+		dac = spec->multiout.hp_nid;
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+	}
 	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
-					spec->multiout.extra_out_nid[0]);
+	if (pin) {
+		dac = spec->multiout.extra_out_nid[0];
+		if (!dac)
+			dac = spec->multiout.dac_nids[0];
+		alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+	}
 }
 
 /*
-- 
cgit v1.1


From 3c715a98844f72cec0fa3ef2b68232b8f751468b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 23 Aug 2011 12:41:09 +0200
Subject: ALSA: hda - Update jack-sense info even when no automute is set

The internal states, jack_present and line_jack_present should be
updated upon unsolicited events even if no automute is set.
Otherwise the wrong state is referred when the automute behavior is
changed by the mixer control.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0fefc10..7cabd73 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute)
-		return;
 	spec->jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
 			     spec->autocfg.hp_pins);
+	if (!spec->automute)
+		return;
 	update_speakers(codec);
 }
 
@@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (!spec->automute || !spec->detect_line)
-		return;
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
+	if (!spec->automute || !spec->detect_line)
+		return;
 	update_speakers(codec);
 }
 
-- 
cgit v1.1


From 1f015f5fdc4003f3f2a7c66efdb1acf7a2d230bf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 23 Aug 2011 14:57:08 +0200
Subject: ALSA: hda - Fix double-headphone/speaker paths for Cxt auto-parser

When multiple headphones or speakers are assigned but no individual
DACs are available, the driver should take the first HP/SPK DAC instead
of another primary output.  The patch adds a bit-flag to dac field of
struct pin_dac_pair indicating that it's a slave DAC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 13 ++++++++++---
 1 file changed, 10 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..4c462c3 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
 
 #define MAX_AUTO_DACS	5
 
+#define DAC_SLAVE_FLAG	0x8000	/* filled dac is a slave */
+
 /* fill analog DAC list from the widget tree */
 static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 {
@@ -3379,6 +3381,8 @@ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
+		if (!filled[nums].dac && i > 0 && filled[0].dac)
+			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
 		nums++;
 	}
 	return nums;
@@ -3407,7 +3411,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac)
+		if (!dac || (dac & DAC_SLAVE_FLAG))
 			continue;
 		switch (spec->dac_info[i].type) {
 		case AUTO_PIN_LINE_OUT:
@@ -4035,6 +4039,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
 		nid = spec->dac_info[i].dac;
 		if (!nid)
 			nid = spec->multiout.dac_nids[0];
+		else if (nid & DAC_SLAVE_FLAG)
+			nid &= ~DAC_SLAVE_FLAG;
 		select_connection(codec, spec->dac_info[i].pin, nid);
 	}
 	if (spec->auto_mute) {
@@ -4191,7 +4197,8 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	for (i = 0; i < spec->dac_info_filled; i++) {
 		const char *label;
 		int idx, type;
-		if (!spec->dac_info[i].dac)
+		hda_nid_t dac = spec->dac_info[i].dac;
+		if (!dac || (dac & DAC_SLAVE_FLAG))
 			continue;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
@@ -4211,7 +4218,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 			idx = num_spk++;
 			break;
 		}
-		err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+		err = try_add_pb_volume(codec, dac,
 					spec->dac_info[i].pin,
 					label, idx);
 		if (err < 0)
-- 
cgit v1.1


From 7675535958175b85b8117bcee245d9ecbc4d3d74 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 24 Aug 2011 10:53:10 +0200
Subject: ALSA: hda/conexant - Enable ADC-switching for auto-mic mode, too

The ADC-switching can work also in the auto-mic mode, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4c462c3..5616444 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3866,7 +3866,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
 	}
 	if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
 		cx_auto_check_auto_mic(codec);
-	if (imux->num_items > 1 && !spec->auto_mic) {
+	if (imux->num_items > 1) {
 		for (i = 1; i < imux->num_items; i++) {
 			if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
 				spec->adc_switching = 1;
-- 
cgit v1.1


From 468c5458856236cde6df1b0654d32bf6625349a5 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 25 Aug 2011 13:16:02 +0200
Subject: ALSA: hda: Conexant: Allow different output types to share DAC

Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.

BugLink: http://bugs.launchpad.net/bugs/817943
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 46 +++++++++++++++++++++++++-----------------
 1 file changed, 27 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 5616444..7696d05 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3372,18 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
 /* fill pin_dac_pair list from the pin and dac list */
 static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
 			      int num_pins, hda_nid_t *dacs, int *rest,
-			      struct pin_dac_pair *filled, int type)
+			      struct pin_dac_pair *filled, int nums, 
+			      int type)
 {
-	int i, nums;
+	int i, start = nums;
 
-	nums = 0;
-	for (i = 0; i < num_pins; i++) {
+	for (i = 0; i < num_pins; i++, nums++) {
 		filled[nums].pin = pins[i];
 		filled[nums].type = type;
 		filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
-		if (!filled[nums].dac && i > 0 && filled[0].dac)
+		if (filled[nums].dac) 
+			continue;
+		if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+			filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+			continue;
+		}
+		if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
 			filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
-		nums++;
+			continue;
+		}
+		snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
 	}
 	return nums;
 }
@@ -3399,14 +3407,14 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 	rest = fill_cx_auto_dacs(codec, dacs);
 	/* parse all analog output pins */
 	nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
-				  dacs, &rest, spec->dac_info,
-				  AUTO_PIN_LINE_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_HP_OUT);
-	nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
-				  dacs, &rest, spec->dac_info + nums,
-				  AUTO_PIN_SPEAKER_OUT);
+			  dacs, &rest, spec->dac_info, 0,
+			  AUTO_PIN_LINE_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_HP_OUT);
+	nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+			  dacs, &rest, spec->dac_info, nums,
+			  AUTO_PIN_SPEAKER_OUT);
 	spec->dac_info_filled = nums;
 	/* fill multiout struct */
 	for (i = 0; i < nums; i++) {
@@ -4173,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
 			     hda_nid_t pin, const char *name, int idx)
 {
 	unsigned int caps;
-	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
-	if (caps & AC_AMPCAP_NUM_STEPS)
-		return cx_auto_add_pb_volume(codec, dac, name, idx);
+	if (dac && !(dac & DAC_SLAVE_FLAG)) {
+		caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+		if (caps & AC_AMPCAP_NUM_STEPS)
+			return cx_auto_add_pb_volume(codec, dac, name, idx);
+	}
 	caps = query_amp_caps(codec, pin, HDA_OUTPUT);
 	if (caps & AC_AMPCAP_NUM_STEPS)
 		return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4198,8 +4208,6 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 		const char *label;
 		int idx, type;
 		hda_nid_t dac = spec->dac_info[i].dac;
-		if (!dac || (dac & DAC_SLAVE_FLAG))
-			continue;
 		type = spec->dac_info[i].type;
 		if (type == AUTO_PIN_LINE_OUT)
 			type = spec->autocfg.line_out_type;
-- 
cgit v1.1


From 99e14c9d4140e9ed2c8810322a377e2a51b356eb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Sep 2011 10:33:16 +0200
Subject: ALSA: hda - Terminate the recursive connection search properly

The recursive search of widget connections in snd_hda_get_conn_index()
must be terminated at the pin and the audio-out widgets.  Otherwise
you'll get "too deep connection" warnings unnecessarily.

Reported-by: Francis Moreau <francis.moro@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3e7850c..f3aefef 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 		return -1;
 	}
 	recursive++;
-	for (i = 0; i < nums; i++)
+	for (i = 0; i < nums; i++) {
+		unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i]));
+		if (type == AC_WID_PIN || type == AC_WID_AUD_OUT)
+			continue;
 		if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0)
 			return i;
+	}
 	return -1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
-- 
cgit v1.1


From 2e1210bc3d065a6e26ff5fef228a9a7e08921d2c Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 14 Sep 2011 13:22:54 +0200
Subject: ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name

This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452a10 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index d6c93d9..c45f3e6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -535,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name,
 		      int index, unsigned int pval, int dir,
 		      struct snd_kcontrol **kctlp)
 {
-	char tmp[32];
+	char tmp[44];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
-- 
cgit v1.1


From 8974bd51a77824d91010176f9a5da28513c2e1f5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 19 Sep 2011 11:31:34 +0200
Subject: ALSA: hda/realtek - Fix auto-mute with HP+LO configuration

When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work.  It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.

The patch fixes the problem and add a comment to indicate the
relationship briefly.

BugLink: http://bugs.launchpad.net/bugs/851697

Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Cc: stable@kernel.org (3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7cabd73..0503c99 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -168,7 +168,7 @@ struct alc_spec {
 	unsigned int auto_mic_valid_imux:1;	/* valid imux for auto-mic */
 	unsigned int automute:1;	/* HP automute enabled */
 	unsigned int detect_line:1;	/* Line-out detection enabled */
-	unsigned int automute_lines:1;	/* automute line-out as well */
+	unsigned int automute_lines:1;	/* automute line-out as well; NOP when automute_hp_lo isn't set */
 	unsigned int automute_hp_lo:1;	/* both HP and LO available */
 
 	/* other flags */
@@ -551,7 +551,7 @@ static void update_speakers(struct hda_codec *codec)
 	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] ||
 	    spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0])
 		return;
-	if (!spec->automute_lines || !spec->automute)
+	if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines))
 		on = 0;
 	else
 		on = spec->jack_present;
@@ -803,7 +803,7 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol,
 	unsigned int val;
 	if (!spec->automute)
 		val = 0;
-	else if (!spec->automute_lines)
+	else if (!spec->automute_hp_lo || !spec->automute_lines)
 		val = 1;
 	else
 		val = 2;
@@ -824,7 +824,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol,
 		spec->automute = 0;
 		break;
 	case 1:
-		if (spec->automute && !spec->automute_lines)
+		if (spec->automute &&
+		    (!spec->automute_hp_lo || !spec->automute_lines))
 			return 0;
 		spec->automute = 1;
 		spec->automute_lines = 0;
-- 
cgit v1.1


From 46724c2e023cb7ba5cd5000dee6481f0a15ebed9 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 20 Sep 2011 09:02:22 +0200
Subject: ALSA: HDA: Add support for IDT 92HD93

Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 5145b66..1b7c114 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -6573,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
+	{ .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
-- 
cgit v1.1


From 2ba34e43ba0469086d1ff81c13a9aa9070c1a0e1 Mon Sep 17 00:00:00 2001
From: Ben Hutchings <ben@decadent.org.uk>
Date: Thu, 22 Sep 2011 14:38:58 +0100
Subject: ALSA: fm801: Fix double free in case of error in tuner detection

Commit 9676001559fce06e37c7dc230ab275f605556176
("ALSA: fm801: add error handling if auto-detect fails") added
incorrect error handling.

Once we have successfully called snd_device_new(), the cleanup
function fm801_free() will automatically be called by snd_card_free()
and we must *not* also call fm801_free() directly.

Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/fm801.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f9123f0..a96456d 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1236,7 +1236,6 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
 		if (snd_tea575x_init(&chip->tea)) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
 			return -ENODEV;
 		}
 	} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1251,7 +1250,6 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		}
 		if (tea575x_tuner == 4) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			snd_fm801_free(chip);
 			return -ENODEV;
 		}
 	}
-- 
cgit v1.1


From c37279b92aba2893578f61076cd2eef5c5fa0e99 Mon Sep 17 00:00:00 2001
From: Ben Hutchings <ben@decadent.org.uk>
Date: Thu, 22 Sep 2011 14:39:52 +0100
Subject: ALSA: fm801: Gracefully handle failure of tuner auto-detect

Commit 9676001559fce06e37c7dc230ab275f605556176
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.

As a bonus, this should fix init and cleanup for the case where the
tuner is explicitly disabled.

Reported-and-tested-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/fm801.c | 13 ++++++++++---
 1 file changed, 10 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index a96456d..32b02d9 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 module_param_array(tea575x_tuner, int, NULL, 0444);
 MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
 
+#define TUNER_DISABLED		(1<<3)
 #define TUNER_ONLY		(1<<4)
 #define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
 
@@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip)
 
       __end_hw:
 #ifdef CONFIG_SND_FM801_TEA575X_BOOL
-	snd_tea575x_exit(&chip->tea);
+	if (!(chip->tea575x_tuner & TUNER_DISABLED))
+		snd_tea575x_exit(&chip->tea);
 #endif
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
@@ -1250,10 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		}
 		if (tea575x_tuner == 4) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
-			return -ENODEV;
+			chip->tea575x_tuner = TUNER_DISABLED;
 		}
 	}
-	strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card));
+	if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
+		strlcpy(chip->tea.card,
+			snd_fm801_tea575x_gpios[(tea575x_tuner &
+						 TUNER_TYPE_MASK) - 1].name,
+			sizeof(chip->tea.card));
+	}
 #endif
 
 	*rchip = chip;
-- 
cgit v1.1


From 6656b15d675c9c6a049db48d50994b3cd4e76bd6 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Sat, 24 Sep 2011 08:30:44 +0200
Subject: ALSA: HDA: No power nids on 92HD93

This patch is necessary to make internal speakers work on this chip.

Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Tested-by: Alex Wolfson <alex.wolfson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 1b7c114..987e3cf 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5630,6 +5630,7 @@ again:
 	switch (codec->vendor_id) {
 	case 0x111d76d1:
 	case 0x111d76d9:
+	case 0x111d76df:
 	case 0x111d76e5:
 	case 0x111d7666:
 	case 0x111d7667:
-- 
cgit v1.1


From 5fe6e0151dbd969f5fbcd94d05c968b76d76952b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 26 Sep 2011 10:41:21 +0200
Subject: ALSA: hda/realtek - Avoid bogus HP-pin assignment

When the headphone pin is assigned as primary output to line_out_pins[],
the automatic HP-pin assignment by ASSID must be suppressed.  Otherwise
a wrong pin might be assigned to the headphone and breaks the auto-mute.

Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
 sound/pci/hda/patch_realtek.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0503c99..91a5381 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1321,7 +1321,9 @@ do_sku:
 	 * 15   : 1 --> enable the function "Mute internal speaker
 	 *	        when the external headphone out jack is plugged"
 	 */
-	if (!spec->autocfg.hp_pins[0]) {
+	if (!spec->autocfg.hp_pins[0] &&
+	    !(spec->autocfg.line_out_pins[0] &&
+	      spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)) {
 		hda_nid_t nid;
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
-- 
cgit v1.1


From e0d32e335fc0e354db93a807736243f5035fbc67 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 26 Sep 2011 15:19:55 +0200
Subject: ALSA: hda/realtek - Don't detect LO jack when identical with HP

The spec->autocfg.line_out_pins[] may contain the same pins as hp_pins[]
depending on the configuration.  When they are identical, detecting the
line_jack_present flag screws up the auto-mute because alc_line_automute()
is called unconditionally at initialization while it won't be triggered
by unsol events, thus the old line_jack_present flag is kept for the
whole run.

For fixing this buggy behavior, the driver needs to check whether the
line-outs are really individual, and skip if same as headphone jacks.

Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 91a5381..7a73621 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -578,6 +578,10 @@ static void alc_line_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
+	/* check LO jack only when it's different from HP */
+	if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0])
+		return;
+
 	spec->line_jack_present =
 		detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
 			     spec->autocfg.line_out_pins);
-- 
cgit v1.1


From 798cb7e897210dbf827f69a302c0967e0380ebac Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 30 Sep 2011 08:52:26 +0200
Subject: ALSA: hda - Fix a regression of the position-buffer check

The commit a810364a0424c297242c6c66071a42f7675a5568
    ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.

This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().

Reported-and-tested-by: Rocko Requin <rockorequin@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index be69822..e9a2a87 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1924,7 +1924,8 @@ static unsigned int azx_via_get_position(struct azx *chip,
 }
 
 static unsigned int azx_get_position(struct azx *chip,
-				     struct azx_dev *azx_dev)
+				     struct azx_dev *azx_dev,
+				     bool with_check)
 {
 	unsigned int pos;
 	int stream = azx_dev->substream->stream;
@@ -1940,7 +1941,7 @@ static unsigned int azx_get_position(struct azx *chip,
 	default:
 		/* use the position buffer */
 		pos = le32_to_cpu(*azx_dev->posbuf);
-		if (chip->position_fix[stream] == POS_FIX_AUTO) {
+		if (with_check && chip->position_fix[stream] == POS_FIX_AUTO) {
 			if (!pos || pos == (u32)-1) {
 				printk(KERN_WARNING
 				       "hda-intel: Invalid position buffer, "
@@ -1964,7 +1965,7 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream)
 	struct azx *chip = apcm->chip;
 	struct azx_dev *azx_dev = get_azx_dev(substream);
 	return bytes_to_frames(substream->runtime,
-			       azx_get_position(chip, azx_dev));
+			       azx_get_position(chip, azx_dev, false));
 }
 
 /*
@@ -1987,7 +1988,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 		return -1;	/* bogus (too early) interrupt */
 
 	stream = azx_dev->substream->stream;
-	pos = azx_get_position(chip, azx_dev);
+	pos = azx_get_position(chip, azx_dev, true);
 
 	if (WARN_ONCE(!azx_dev->period_bytes,
 		      "hda-intel: zero azx_dev->period_bytes"))
-- 
cgit v1.1