From f67d8176ba9a3dbc33454cd67057184b2ef5ee31 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 4 Feb 2009 23:30:19 +0100
Subject: ALSA: hda - Add quirk for FSC Amilo Xi2550

Added model=fujisu-pi2515 for FSC Amilo Xi2550 with ALC883 codec.

Refernece: Novell bnc#450979
	https://bugzilla.novell.com/show_bug.cgi?id=450979

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 0040101f6150..a3baa33aedfd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8515,6 +8515,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
+	SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550",
+		      ALC883_FUJITSU_PI2515),
 	SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515),
 	SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530",
 		ALC888_FUJITSU_XA3530),
-- 
cgit v1.2.3


From e8c0ee5d77ec0f144c753a622c67dd96fa195d50 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 5 Feb 2009 07:34:28 +0100
Subject: ALSA: hda - Fix misc workqueue issues

Some fixes regarding snd-hda-intel workqueue:
- Use create_singlethread_workqueue() instead of create_workqueue()
  as per-CPU work isn't required.
- Allocate workq name string properly
- Renamed the workq name to "hd-audio*" to be more obvious.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 9 +++++----
 sound/pci/hda/hda_codec.h | 1 +
 2 files changed, 6 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b7bba7dc7cf1..0b708134d12f 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
 {
 	struct hda_bus *bus;
 	int err;
-	char qname[8];
 	static struct snd_device_ops dev_ops = {
 		.dev_register = snd_hda_bus_dev_register,
 		.dev_free = snd_hda_bus_dev_free,
@@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card,
 	mutex_init(&bus->cmd_mutex);
 	INIT_LIST_HEAD(&bus->codec_list);
 
-	snprintf(qname, sizeof(qname), "hda%d", card->number);
-	bus->workq = create_workqueue(qname);
+	snprintf(bus->workq_name, sizeof(bus->workq_name),
+		 "hd-audio%d", card->number);
+	bus->workq = create_singlethread_workqueue(bus->workq_name);
 	if (!bus->workq) {
-		snd_printk(KERN_ERR "cannot create workqueue %s\n", qname);
+		snd_printk(KERN_ERR "cannot create workqueue %s\n",
+			   bus->workq_name);
 		kfree(bus);
 		return -ENOMEM;
 	}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 5810ef588402..09a332ada0c6 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -614,6 +614,7 @@ struct hda_bus {
 
 	/* unsolicited event queue */
 	struct hda_bus_unsolicited *unsol;
+	char workq_name[16];
 	struct workqueue_struct *workq;	/* common workqueue for codecs */
 
 	/* assigned PCMs */
-- 
cgit v1.2.3


From c6e8f2daadc6d61a32b7486a1058c8f1f9baa499 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 12:45:52 +0100
Subject: ALSA: hda - Add missing initialization for ALC272

ALC272 needs EAPD for speaker outputs as well as other similar ALC
codecs.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a3baa33aedfd..ac1a6e728430 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1037,6 +1037,7 @@ do_sku:
 		case 0x10ec0267:
 		case 0x10ec0268:
 		case 0x10ec0269:
+		case 0x10ec0272:
 		case 0x10ec0660:
 		case 0x10ec0662:
 		case 0x10ec0663:
-- 
cgit v1.2.3


From 4a5a4c56b443a213fa9c2ad27984a8681a3d7087 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Feb 2009 12:46:59 +0100
Subject: ALSA: hda - Add missing COEF initialization for ALC887

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ac1a6e728430..ae5c8a0d1479 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1066,6 +1066,7 @@ do_sku:
 		case 0x10ec0882:
 		case 0x10ec0883:
 		case 0x10ec0885:
+		case 0x10ec0887:
 		case 0x10ec0889:
 			snd_hda_codec_write(codec, 0x20, 0,
 					    AC_VERB_SET_COEF_INDEX, 7);
-- 
cgit v1.2.3


From 44a678d04babaa751c0ee98e006ede9576fa9e00 Mon Sep 17 00:00:00 2001
From: Mackenzie Morgan <macoafi@gmail.com>
Date: Tue, 10 Feb 2009 17:13:43 +0100
Subject: ALSA: hda - Add quirk for Asus z37e (1043:8284)

Added a quirk for Asus Z37E for fixing suspend/hibernation problem.

Reference:
	https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/25896
	http://launchpadlibrarian.net/17053575/0001-Add-quirk-for-ASUS-Z37E-to-make-sound-audible-afte.patch
	https://bugtrack.alsa-project.org/alsa-bug/bug_view_page.php?bug_id=4282

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ae5c8a0d1479..ed8fcbd60003 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8478,6 +8478,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
-- 
cgit v1.2.3


From a1667e4eea0a7085815d1532d7630bb4611271d0 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Wed, 11 Feb 2009 15:22:28 +0800
Subject: ALSA: hda - allow multi-channel HDMI audio playback when ELD is not
 present

The YAMAHA AV-X1800 requires audio infoframe to include speaker-channel
mapping to play >2 channel HDMI audio. In theory that mapping should be
derived from its speaker configurations contained in its ELD. However we
currently cannot get ELD in console before the KMS functionalities are ready.
This is a more or less general issue at least in the near future. As a
workaround, we propose to allow playback of mult-channel audio when ELD
is not available.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_intelhdmi.c | 12 ++++++++----
 1 file changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 3564f4e4b74c..a8643509e2af 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -419,13 +419,17 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
 	/*
 	 * CA defaults to 0 for basic stereo audio
 	 */
-	if (!eld->eld_ver)
-		return 0;
-	if (!eld->spk_alloc)
-		return 0;
 	if (channels <= 2)
 		return 0;
 
+	/*
+	 * HDMI sink's ELD info cannot always be retrieved for now, e.g.
+	 * in console or for audio devices. Assume the highest speakers
+	 * configuration, to _not_ prohibit multi-channel audio playback.
+	 */
+	if (!eld->spk_alloc)
+		eld->spk_alloc = 0xffff;
+
 	/*
 	 * expand ELD's speaker allocation mask
 	 *
-- 
cgit v1.2.3


From 606c0cee695bbd0c2bf32132999e35cff5a6dd9e Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Wed, 11 Feb 2009 15:22:29 +0800
Subject: ALSA: hda - enable HDMI audio pin out at module loading time

We found that enabling/disabling HDMI audio pin out at stream start/stop
time will kill the leading 500ms or so sound samples. Avoid this by enabling
pin out once and for ever at module loading time.

The leading ~500ms audio samples will still be lost when switching from
X-channel playback to Y-channel playback where X != Y. However there's no
much we can do about it: the audio infoframe has to change and it looks like
either G45 or YAMAHA requires some time to switch the configuration.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_intelhdmi.c | 42 +++++++++++++++++++----------------------
 1 file changed, 19 insertions(+), 23 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index a8643509e2af..f2610d67e187 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = {
 	{} /* terminator */
 };
 
-static struct hda_verb pinout_disable_verb[] = {
-	{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00},
-	{}
-};
-
 static struct hda_verb unsolicited_response_verb[] = {
 	{PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
 						  INTEL_HDMI_EVENT_TAG},
@@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
 
 static void hdmi_enable_output(struct hda_codec *codec)
 {
-	/* Enable Audio InfoFrame Transmission */
-	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
-	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
-						AC_DIPXMIT_BEST);
 	/* Unmute */
 	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
 		snd_hda_codec_write(codec, PIN_NID, 0,
@@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec)
 	snd_hda_sequence_write(codec, pinout_enable_verb);
 }
 
-static void hdmi_disable_output(struct hda_codec *codec)
+/*
+ * Enable Audio InfoFrame Transmission
+ */
+static void hdmi_start_infoframe_trans(struct hda_codec *codec)
 {
-	snd_hda_sequence_write(codec, pinout_disable_verb);
-	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
-		snd_hda_codec_write(codec, PIN_NID, 0,
-				AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+						AC_DIPXMIT_BEST);
+}
 
-	/*
-	 * FIXME: noises may arise when playing music after reloading the
-	 * kernel module, until the next X restart or monitor repower.
-	 */
+/*
+ * Disable Audio InfoFrame Transmission
+ */
+static void hdmi_stop_infoframe_trans(struct hda_codec *codec)
+{
+	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+						AC_DIPXMIT_DISABLE);
 }
 
 static int hdmi_get_channel_count(struct hda_codec *codec)
@@ -489,6 +487,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 	hdmi_setup_channel_mapping(codec, &ai);
 
 	hdmi_fill_audio_infoframe(codec, &ai);
+	hdmi_start_infoframe_trans(codec);
 }
 
 
@@ -566,7 +565,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo,
 {
 	struct intel_hdmi_spec *spec = codec->spec;
 
-	hdmi_disable_output(codec);
+	hdmi_stop_infoframe_trans(codec);
 
 	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
@@ -586,8 +585,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 
 	hdmi_setup_audio_infoframe(codec, substream);
 
-	hdmi_enable_output(codec);
-
 	return 0;
 }
 
@@ -632,8 +629,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec)
 
 static int intel_hdmi_init(struct hda_codec *codec)
 {
-	/* disable audio output as early as possible */
-	hdmi_disable_output(codec);
+	hdmi_enable_output(codec);
 
 	snd_hda_sequence_write(codec, unsolicited_response_verb);
 
-- 
cgit v1.2.3


From 9a957a24e3b4008d84e204cdf25849ae4d5592a2 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Wed, 11 Feb 2009 15:22:30 +0800
Subject: ALSA: hda - compute checksum in HDMI audio infoframe

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_intelhdmi.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index f2610d67e187..90b11374a0a8 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -366,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
 					struct hdmi_audio_infoframe *ai)
 {
 	u8 *params = (u8 *)ai;
+	u8 sum = 0;
 	int i;
 
 	hdmi_debug_dip_size(codec);
 	hdmi_clear_dip_buffers(codec); /* be paranoid */
 
+	for (i = 0; i < sizeof(ai); i++)
+		sum += params[i];
+	ai->checksum = - sum;
+
 	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
 	for (i = 0; i < sizeof(ai); i++)
 		hdmi_write_dip_byte(codec, PIN_NID, params[i]);
-- 
cgit v1.2.3


From a57c0eb65576c810c408f0a086afac179242a21c Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Wed, 11 Feb 2009 15:22:31 +0800
Subject: ALSA: hda - add id for Intel IbexPeak integrated HDMI codec

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_intelhdmi.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 90b11374a0a8..fcc77fec4487 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -684,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
 	{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
 	{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
 	{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
+	{ .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
 	{ .id = 0x10951392, .name = "SiI1392 HDMI",     .patch = patch_intel_hdmi },
 	{} /* terminator */
 };
@@ -692,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb");
 MODULE_ALIAS("snd-hda-codec-id:80862801");
 MODULE_ALIAS("snd-hda-codec-id:80862802");
 MODULE_ALIAS("snd-hda-codec-id:80862803");
+MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:10951392");
 
 MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From 26a74f1f61c5bba1c0b46e67e91e921e941f76d7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Feb 2009 00:13:19 +0100
Subject: ALSA: hda - Register (new) devices at reconfig

The devices that have been newly added during reconfig must be
registered.  Otherwise they won't be visible to user-space.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 300ab407cf42..482fb0304ca9 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -175,7 +175,7 @@ static int reconfig_codec(struct hda_codec *codec)
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
 		return err;
-	return 0;
+	return snd_card_register(codec->bus->card);
 }
 
 /*
-- 
cgit v1.2.3


From 92258a3ed2f583c8720ef570f5c62b28e6c58d71 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Thu, 12 Feb 2009 17:27:27 -0200
Subject: ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5
 model

Change HP dv7 quirk: although reported to work with hp-m4 model
(https://bugzilla.novell.com/show_bug.cgi?id=445321), the original
report doesn't contain info about testing of internal microphone.

Recently I received a report about internal mic not working
(https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be
related with the forced line in on pin 0x0e done with hp-m4 model. Thus
change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported
to be fixed on a provided kernel with this change
(https://qa.mandriva.com/show_bug.cgi?id=44855#c196).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 38428e22428f..aa814a3c2d8c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1799,7 +1799,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
 		      "HP dv5", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
-		      "HP dv7", STAC_HP_M4),
+		      "HP dv7", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
 		      "HP dv4", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
-- 
cgit v1.2.3


From 3a08e30de2facffe8e1a25bf4fa62cbc920fbaf6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 11:37:08 +0100
Subject: ALSA: hda - Add missing terminator in slave dig-out array

Added the missing terminator for ad1989b_slave_dig_outs[].

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2e7371ec2e23..7006d62ca6c2 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1885,8 +1885,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 #define AD1988_SPDIF_OUT_HDMI	0x0b
 #define AD1988_SPDIF_IN		0x07
 
-static hda_nid_t ad1989b_slave_dig_outs[2] = {
-	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI
+static hda_nid_t ad1989b_slave_dig_outs[] = {
+	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
 };
 
 static struct hda_input_mux ad1988_6stack_capture_source = {
-- 
cgit v1.2.3


From 9411e21cd0cc4fd046b4f448417b0e103e80951c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 13 Feb 2009 11:32:28 +0100
Subject: ALSA: hda - Add snd_hda_multi_out_dig_cleanup()

Added the helper function snd_hda_multi_out_dig_cleanup() to clean up
the digital outputs with multi setup.  This call is needed in cases
the codec supports multiple digital outputs as slaves.  Otherwise the
slave widgets aren't properly cleaned up.

For a single digital output (e.g. in patch_conexant.c), this call isn't
needed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 10 ++++++++++
 sound/pci/hda/hda_local.h      |  2 ++
 sound/pci/hda/patch_analog.c   | 11 ++++++++++-
 sound/pci/hda/patch_sigmatel.c | 11 ++++++++++-
 4 files changed, 32 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0b708134d12f..d03f99298be9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3088,6 +3088,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare);
 
+int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
+				  struct hda_multi_out *mout)
+{
+	mutex_lock(&codec->spdif_mutex);
+	cleanup_dig_out_stream(codec, mout->dig_out_nid);
+	mutex_unlock(&codec->spdif_mutex);
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup);
+
 /*
  * release the digital out
  */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 1dd8716c387f..44f189cb97ae 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -251,6 +251,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec,
 				  unsigned int stream_tag,
 				  unsigned int format,
 				  struct snd_pcm_substream *substream);
+int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec,
+				  struct hda_multi_out *mout);
 int snd_hda_multi_out_analog_open(struct hda_codec *codec,
 				  struct hda_multi_out *mout,
 				  struct snd_pcm_substream *substream,
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 7006d62ca6c2..e48612323aa0 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -275,6 +275,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     format, substream);
 }
 
+static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					   struct hda_codec *codec,
+					   struct snd_pcm_substream *substream)
+{
+	struct ad198x_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 /*
  * Analog capture
  */
@@ -333,7 +341,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = {
 	.ops = {
 		.open = ad198x_dig_playback_pcm_open,
 		.close = ad198x_dig_playback_pcm_close,
-		.prepare = ad198x_dig_playback_pcm_prepare
+		.prepare = ad198x_dig_playback_pcm_prepare,
+		.cleanup = ad198x_dig_playback_pcm_cleanup
 	},
 };
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index aa814a3c2d8c..8027edf3c8f2 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2442,6 +2442,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     stream_tag, format, substream);
 }
 
+static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 
 /*
  * Analog capture callbacks
@@ -2486,7 +2494,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
 	.ops = {
 		.open = stac92xx_dig_playback_pcm_open,
 		.close = stac92xx_dig_playback_pcm_close,
-		.prepare = stac92xx_dig_playback_pcm_prepare
+		.prepare = stac92xx_dig_playback_pcm_prepare,
+		.cleanup = stac92xx_dig_playback_pcm_cleanup
 	},
 };
 
-- 
cgit v1.2.3


From d14a7e0bfc4aed6452a436c9836903fbd1a5d6ad Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 Feb 2009 10:13:03 +0100
Subject: Revert "Sound: hda - Restore PCI configuration space with interrupts
 off"

This reverts commit 32e176c14d7a425b681ef003c9061001ddb7fc7b.

That commit caused a regression with suspend on Thinkpad SL300.

Reference: kernel bug#12711
	http://bugzilla.kernel.org/show_bug.cgi?id=12711

Tested-by:  Alexandre Rostovtsev <tetromino@gmail.com>
Acked-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 8 ++------
 1 file changed, 2 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 11e791b965f6..c8d9178f47e5 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1947,16 +1947,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 	return 0;
 }
 
-static int azx_resume_early(struct pci_dev *pci)
-{
-	return pci_restore_state(pci);
-}
-
 static int azx_resume(struct pci_dev *pci)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
 	struct azx *chip = card->private_data;
 
+	pci_set_power_state(pci, PCI_D0);
+	pci_restore_state(pci);
 	if (pci_enable_device(pci) < 0) {
 		printk(KERN_ERR "hda-intel: pci_enable_device failed, "
 		       "disabling device\n");
@@ -2468,7 +2465,6 @@ static struct pci_driver driver = {
 	.remove = __devexit_p(azx_remove),
 #ifdef CONFIG_PM
 	.suspend = azx_suspend,
-	.resume_early = azx_resume_early,
 	.resume = azx_resume,
 #endif
 };
-- 
cgit v1.2.3


From 6ce6c473a7fd742fdb0db95841e2c4c6b37337c5 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Tue, 17 Feb 2009 09:50:30 +0100
Subject: sound: virtuoso: revert "do not overwrite EEPROM on Xonar D2/D2X"

This reverts commit 7e86c0e6850504ec9516b953f316a47277825e33 ("do not
overwrite EEPROM on Xonar D2/D2X") because it did not actually help with
the problem.

More user reports show that the overwriting of the EEPROM is not
triggered by using this driver but by installing Linux, and that the
installation of any other operating system (even one without any CMI8788
driver) has the same effect.  In other words, the presence of this
driver does not have any effect on the occurrence of the error.  (So
far, the available evidence seems to point to a BIOS bug.)

Furthermore, it turns out that the EEPROM chip is protected against
stray write commands by the command format and by requiring a separate
write-enable command, so the error scenario in the previous commit (that
SPI writes can be misinterpreted as an EEPROM write command) is not even
theoretically possible.

The mixer control that was removed as a consequence of the previous
commit can only be partially emulated in userspace, which also means it
cannot be seen be the in-kernel OSS API emulation, so it is better to
revert that change.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/virtuoso.c | 17 ++++-------------
 1 file changed, 4 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 18c7c91786bc..6c870c12a177 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -26,7 +26,7 @@
  * SPI 0 -> 1st PCM1796 (front)
  * SPI 1 -> 2nd PCM1796 (surround)
  * SPI 2 -> 3rd PCM1796 (center/LFE)
- * SPI 4 -> 4th PCM1796 (back) and EEPROM self-destruct (do not use!)
+ * SPI 4 -> 4th PCM1796 (back)
  *
  * GPIO 2 -> M0 of CS5381
  * GPIO 3 -> M1 of CS5381
@@ -207,12 +207,6 @@ static void xonar_gpio_changed(struct oxygen *chip);
 static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
 				     u8 reg, u8 value)
 {
-	/*
-	 * We don't want to do writes on SPI 4 because the EEPROM, which shares
-	 * the same pin, might get confused and broken.  We'd better take care
-	 * that the driver works with the default register values ...
-	 */
-#if 0
 	/* maps ALSA channel pair number to SPI output */
 	static const u8 codec_map[4] = {
 		0, 1, 2, 4
@@ -223,7 +217,6 @@ static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec,
 			 (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) |
 			 OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
 			 (reg << 8) | value);
-#endif
 }
 
 static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec,
@@ -757,9 +750,6 @@ static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
 
 static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 {
-	if (!strncmp(template->name, "Master Playback ", 16))
-		/* disable volume/mute because they would require SPI writes */
-		return 1;
 	if (!strncmp(template->name, "CD Capture ", 11))
 		/* CD in is actually connected to the video in pin */
 		template->private_value ^= AC97_CD ^ AC97_VIDEO;
@@ -850,8 +840,9 @@ static const struct oxygen_model model_xonar_d2 = {
 	.dac_volume_min = 0x0f,
 	.dac_volume_max = 0xff,
 	.misc_flags = OXYGEN_MISC_MIDI,
-	.function_flags = OXYGEN_FUNCTION_SPI,
-	.dac_i2s_format = OXYGEN_I2S_FORMAT_I2S,
+	.function_flags = OXYGEN_FUNCTION_SPI |
+			  OXYGEN_FUNCTION_ENABLE_SPI_4_5,
+	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
-- 
cgit v1.2.3


From e32740d9786b8a6c54f6e3d670567d9ef57b3b8c Mon Sep 17 00:00:00 2001
From: Harvey Harrison <harvey.harrison@gmail.com>
Date: Thu, 19 Feb 2009 11:58:37 -0800
Subject: ALSA: pcxhr.h replace signed one-bit bitfields

The usage and comments make it clear values of 1/0 were intended
rather than -1/0

Noticed by sparse:
sound/pci/pcxhr/pcxhr.h:100:20: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:101:22: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:102:24: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:103:21: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:104:25: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:105:20: error: dubious one-bit signed bitfield

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/pcxhr/pcxhr.h | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c92..69d87dee6995 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -97,12 +97,12 @@ struct pcxhr_mgr {
 	int capture_chips;
 	int fw_file_set;
 	int firmware_num;
-	int is_hr_stereo:1;
-	int board_has_aes1:1;	/* if 1 board has AES1 plug and SRC */
-	int board_has_analog:1;	/* if 0 the board is digital only */
-	int board_has_mic:1;	/* if 1 the board has microphone input */
-	int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
-	int mono_capture:1;	/* if 1 the board does mono capture */
+	unsigned int is_hr_stereo:1;
+	unsigned int board_has_aes1:1;	/* if 1 board has AES1 plug and SRC */
+	unsigned int board_has_analog:1; /* if 0 the board is digital only */
+	unsigned int board_has_mic:1; /* if 1 the board has microphone input */
+	unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
+	unsigned int mono_capture:1; /* if 1 the board does mono capture */
 
 	struct snd_dma_buffer hostport;
 
-- 
cgit v1.2.3


From 55290e1932102f57ea17e7cff895914c2dbdb4c4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 20 Feb 2009 15:59:01 +0100
Subject: ALSA: hda - Fix parse of init_verbs sysfs entry

Fixed the parse of init_verbs hwdep sysfs entry.
Simplieied using sscanf.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c | 15 ++++++++-------
 1 file changed, 8 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 482fb0304ca9..4ae51dcb81af 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -277,18 +277,19 @@ static ssize_t init_verbs_store(struct device *dev,
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
-	char *p;
-	struct hda_verb verb, *v;
+	struct hda_verb *v;
+	int nid, verb, param;
 
-	verb.nid = simple_strtoul(buf, &p, 0);
-	verb.verb = simple_strtoul(p, &p, 0);
-	verb.param = simple_strtoul(p, &p, 0);
-	if (!verb.nid || !verb.verb || !verb.param)
+	if (sscanf(buf, "%i %i %i", &nid, &verb, &param) != 3)
+		return -EINVAL;
+	if (!nid || !verb)
 		return -EINVAL;
 	v = snd_array_new(&codec->init_verbs);
 	if (!v)
 		return -ENOMEM;
-	*v = verb;
+	v->nid = nid;
+	v->verb = verb;
+	v->param = param;
 	return count;
 }
 
-- 
cgit v1.2.3


From e8bf069c419c1dc0657e02636441fe1179a9db14 Mon Sep 17 00:00:00 2001
From: Anssi Hannula <anssi.hannula@gmail.com>
Date: Sun, 22 Feb 2009 14:42:54 +0200
Subject: ALSA: aw2: do not grab every saa7146 based device

Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not
check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver
grabs them as well.

According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the
subsystem ids set to 0, the saa7146 default.

Fix conflicts with DVB devices by checking for subsystem ids = 0
specifically.

Signed-off-by: Anssi Hannula <anssi.hannula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/aw2/aw2-alsa.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f8..c7c54e7748e9 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
 
 static struct pci_device_id snd_aw2_ids[] = {
-	{PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID,
+	{PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
 	 0, 0, 0},
 	{0}
 };
-- 
cgit v1.2.3


From 2d4663816064fabb68935f920bbd7ccdc7f9392d Mon Sep 17 00:00:00 2001
From: Luke Yelavich <themuso@ubuntu.com>
Date: Mon, 23 Feb 2009 13:00:33 +1100
Subject: ALSA: hda - add another MacBook Pro 3,1 SSID

Reference: Ubuntu bug #33245
    https://bugs.launchpad.net/bugs/332456

Signed-off-by: Luke Yelavich <themuso@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ed8fcbd60003..f6571224b34e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7017,6 +7017,7 @@ static int patch_alc882(struct hda_codec *codec)
 		case 0x106b3e00: /* iMac 24 Aluminium */
 			board_config = ALC885_IMAC24;
 			break;
+		case 0x106b00a0: /* MacBookPro3,1 - Another revision */
 		case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
 		case 0x106b00a4: /* MacbookPro4,1 */
 		case 0x106b2c00: /* Macbook Pro rev3 */
-- 
cgit v1.2.3


From cc374c477c9bf95f409fed16426856d86a97394f Mon Sep 17 00:00:00 2001
From: Juan Jesus Garcia de Soria <skandalfo@gmail.com>
Date: Mon, 23 Feb 2009 08:11:59 +0100
Subject: ALSA: hda - Quirk for Acer Aspire 6530G

The Acer Aspire 6530G needs the 4930G "model" for the front mic to
work properly.

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 f6571224b34e..a680be0d4534 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8470,6 +8470,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
+		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
-- 
cgit v1.2.3


From 1f9da5544073d38e05139f8ce9da24e78653c73e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 24 Feb 2009 15:31:02 +0100
Subject: ALSA: emu10k1 - Fix digital/analog switch on audigy2 ZS

Fix the inverted logic of shared spdif switch.

Reference: Novell bnc#478496
	https://bugzilla.novell.com/show_bug.cgi?id=478496

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/emu10k1_main.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d66..101a1c13a20d 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1528,6 +1528,7 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0151_chip = 1,
 	 .spk71 = 1,
 	 .spdif_bug = 1,
+	 .invert_shared_spdif = 1,	/* digital/analog switch swapped */
 	 .ac97_chip = 1} ,
 	{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
 	 .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]",
-- 
cgit v1.2.3


From ea18aa464452c3e6550320d247c0306aaa2d156f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 27 Feb 2009 17:36:33 +0100
Subject: ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3

Fix num_dmuxes initialization for dell-m4-1 and dell-m4-3 models
of IDT 92HD71bxx codec, which was wrongly set to zero.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8027edf3c8f2..3bc427645da8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4989,7 +4989,7 @@ again:
 	case STAC_DELL_M4_3:
 		spec->num_dmics = 1;
 		spec->num_smuxes = 0;
-		spec->num_dmuxes = 0;
+		spec->num_dmuxes = 1;
 		break;
 	default:
 		spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
-- 
cgit v1.2.3


From bb543c969467f33c3a1a0ccfcfcd9a508cd81c54 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 27 Feb 2009 17:44:07 +0100
Subject: ALSA: hda - Add quirk for new HP xw series

Added model=hp-bpc for new HP xw series (103c:170b).

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a680be0d4534..6c26afcb8262 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10557,6 +10557,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
 	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
 	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
-- 
cgit v1.2.3


From 38f1df27e3191d76e983cb9c6b4392582fd32fda Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 1 Mar 2009 10:55:44 +0100
Subject: ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268

Some Toshiba laptops have another ALC268 codec on slot#3 that conflicts
with the primary codec.  The codec#3 is for the digital I/O, and should
be fixed by the driver, but it'd need a bunch of changes.

So, let's fix the probe problem temporarily by setting the default
probe_mask value.

Reference: kernel bugzilla #12735
	http://bugzilla.kernel.org/show_bug.cgi?id=12735

Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8d9178f47e5..5e909e0da04b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2095,6 +2095,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
 	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
 	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
+	/* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */
+	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03),
 	{}
 };
 
-- 
cgit v1.2.3


From 14b97595e0e1f47b6f809e180e5bcd8dcd995690 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 09:42:07 +0100
Subject: ALSA: hda - Fix typos in slave controls in patch_sigmatel.c

"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3bc427645da8..995b413078f5 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1207,7 +1207,7 @@ static const char *slave_vols[] = {
 	"LFE Playback Volume",
 	"Side Playback Volume",
 	"Headphone Playback Volume",
-	"Headphone Playback Volume",
+	"Headphone2 Playback Volume",
 	"Speaker Playback Volume",
 	"External Speaker Playback Volume",
 	"Speaker2 Playback Volume",
@@ -1221,7 +1221,7 @@ static const char *slave_sws[] = {
 	"LFE Playback Switch",
 	"Side Playback Switch",
 	"Headphone Playback Switch",
-	"Headphone Playback Switch",
+	"Headphone2 Playback Switch",
 	"Speaker Playback Switch",
 	"External Speaker Playback Switch",
 	"Speaker2 Playback Switch",
-- 
cgit v1.2.3


From c50ff7c04225c945b13d410d50fde6ff6c59d7ee Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 6 Mar 2009 09:43:58 +0100
Subject: ALSA: hda - Fix headphone-detect regression with multiple HP jacks

The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output.  Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.

Reference: Novell bnc#482052
	https://bugzilla.novell.com/show_bug.cgi?id=482052

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 995b413078f5..6094344fb223 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3516,6 +3516,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (! spec->autocfg.line_outs)
 		return 0; /* can't find valid pin config */
 
+#if 0 /* FIXME: temporarily disabled */
 	/* If we have no real line-out pin and multiple hp-outs, HPs should
 	 * be set up as multi-channel outputs.
 	 */
@@ -3535,6 +3536,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
 		spec->autocfg.hp_outs = 0;
 	}
+#endif /* FIXME: temporarily disabled */
 	if (spec->autocfg.mono_out_pin) {
 		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
 			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
-- 
cgit v1.2.3


From 09240cf429505891d6123ce14a29f58f2a60121e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Mar 2009 07:47:18 +0100
Subject: ALSA: hda - Fix DMA mask for ATI controllers

ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA.  As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 5e909e0da04b..643f0e499292 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2210,9 +2210,17 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	gcap = azx_readw(chip, GCAP);
 	snd_printdd("chipset global capabilities = 0x%x\n", gcap);
 
+	/* ATI chips seems buggy about 64bit DMA addresses */
+	if (chip->driver_type == AZX_DRIVER_ATI)
+		gcap &= ~0x01;
+
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK))
 		pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK);
+	else {
+		pci_set_dma_mask(pci, DMA_32BIT_MASK);
+		pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK);
+	}
 
 	/* read number of streams from GCAP register instead of using
 	 * hardcoded value
-- 
cgit v1.2.3


From c673ba1c23941173c16ff24c7cb34199e826c8b5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 17 Mar 2009 07:49:14 +0100
Subject: ALSA: hda - Workaround for buggy DMA position on ATI controllers

The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 31 ++++++++++++++++++-------------
 1 file changed, 18 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 643f0e499292..f3b5723c2859 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2059,26 +2059,31 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
 {
 	const struct snd_pci_quirk *q;
 
-	/* Check VIA HD Audio Controller exist */
-	if (chip->pci->vendor == PCI_VENDOR_ID_VIA &&
-	    chip->pci->device == VIA_HDAC_DEVICE_ID) {
+	switch (fix) {
+	case POS_FIX_LPIB:
+	case POS_FIX_POSBUF:
+		return fix;
+	}
+
+	/* Check VIA/ATI HD Audio Controller exist */
+	switch (chip->driver_type) {
+	case AZX_DRIVER_VIA:
+	case AZX_DRIVER_ATI:
 		chip->via_dmapos_patch = 1;
 		/* Use link position directly, avoid any transfer problem. */
 		return POS_FIX_LPIB;
 	}
 	chip->via_dmapos_patch = 0;
 
-	if (fix == POS_FIX_AUTO) {
-		q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
-		if (q) {
-			printk(KERN_INFO
-				    "hda_intel: position_fix set to %d "
-				    "for device %04x:%04x\n",
-				    q->value, q->subvendor, q->subdevice);
-			return q->value;
-		}
+	q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
+	if (q) {
+		printk(KERN_INFO
+		       "hda_intel: position_fix set to %d "
+		       "for device %04x:%04x\n",
+		       q->value, q->subvendor, q->subdevice);
+		return q->value;
 	}
-	return fix;
+	return POS_FIX_AUTO;
 }
 
 /*
-- 
cgit v1.2.3


From 82f5d57163abed2e5ff271d03217b6f90c616eb8 Mon Sep 17 00:00:00 2001
From: Jiri Slaby <jirislaby@gmail.com>
Date: Wed, 11 Mar 2009 20:11:41 +0100
Subject: ALSA: mixart, fix lock imbalance

There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.

Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/mixart/mixart.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index f23a73577c22..bb162507fe6c 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs,
 	/* set the format to the board */
 	err = mixart_set_format(stream, format);
 	if(err < 0) {
+		mutex_unlock(&mgr->setup_mutex);
 		return err;
 	}
 
-- 
cgit v1.2.3