From 4453dba54de7e517b0cd6f5e4a3f4af3b34f9e79 Mon Sep 17 00:00:00 2001
From: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Date: Fri, 6 Feb 2009 12:01:04 +0200
Subject: ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback

Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit
4eaa9819dc08d7bfd1065ce530e31b18a119dcaf.

This was causing arbitrary register writes when touching the controls
defined with SOC_DAPM_SINGLE_AIC3X.

Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic3x.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index b47a749c5ea..aea0cb72d80 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -165,10 +165,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value & 0xff;
-	int shift = (kcontrol->private_value >> 8) & 0x0f;
-	int mask = (kcontrol->private_value >> 16) & 0xff;
-	int invert = (kcontrol->private_value >> 24) & 0x01;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
 	unsigned short val, val_mask;
 	int ret;
 	struct snd_soc_dapm_path *path;
-- 
cgit v1.2.3


From 397d5aeeb5a2c9ca6108899a04b35a51cd904503 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jarkko.nikula@nokia.com>
Date: Fri, 6 Feb 2009 12:01:05 +0200
Subject: ASoC: WM8990: Fix kcontrol's private value use in put callback

Function wm899x_outpga_put_volsw_vu misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit
4eaa9819dc08d7bfd1065ce530e31b18a119dcaf.

This is very similar fix than fix to TLV320AIC3X codec made by
Eero Nurkkala <ext-eero.nurkkala@nokia.com>. This fix is compile tested
only.

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/wm8990.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 1cbb7b9b51c..a5731faa150 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -176,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value & 0xff;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int reg = mc->reg;
 	int ret;
 	u16 val;
 
-- 
cgit v1.2.3


From f6f35bbe7c6494e66590cf519e21da2dd8d59e01 Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Sun, 8 Feb 2009 15:22:25 +0100
Subject: [ARM] AACI: timeout will reach -1

With a postfix decrement the timeout will reach -1 rather than 0,
so the warning will not be issued.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
---
 sound/arm/aaci.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 89096e811a4..772901e41ec 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	 */
 	do {
 		v = readl(aaci->base + AACI_SLFR);
-	} while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--);
+	} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
 
 	if (!timeout)
 		dev_err(&aaci->dev->dev,
@@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	 */
 	do {
 		v = readl(aaci->base + AACI_SLFR);
-	} while ((v & SLFR_1TXB) && timeout--);
+	} while ((v & SLFR_1TXB) && --timeout);
 
 	if (!timeout) {
 		dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
@@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	do {
 		cond_resched();
 		v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
-	} while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--);
+	} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
 
 	if (!timeout) {
 		dev_err(&aaci->dev->dev, "timeout on RX valid\n");
-- 
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')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ae5c8a0d147..ed8fcbd6000 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 272edb00493af32c609f43bdf1d75141756fd999 Mon Sep 17 00:00:00 2001
From: "Lopez Cruz, Misael" <x0052729@ti.com>
Date: Tue, 10 Feb 2009 02:05:07 -0600
Subject: ASoC: Update SDP3430 machine driver for snd_soc_card

This patch replaces "snd_soc_machine" structure by "snd_soc_card" in
SP3430 driver. This change is needed in SDP3430 driver to reflect
changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch
(875065491fba8eb13219f16c36e79a6fb4e15c68).

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/sdp3430.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index ad97836818b..e226fa75669 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -91,7 +91,7 @@ static struct snd_soc_dai_link sdp3430_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+static struct snd_soc_card snd_soc_sdp3430 = {
 	.name = "SDP3430",
 	.platform = &omap_soc_platform,
 	.dai_link = &sdp3430_dai,
@@ -100,7 +100,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = {
 
 /* Audio subsystem */
 static struct snd_soc_device sdp3430_snd_devdata = {
-	.machine = &snd_soc_machine_sdp3430,
+	.card = &snd_soc_sdp3430,
 	.codec_dev = &soc_codec_dev_twl4030,
 };
 
-- 
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')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 3564f4e4b74..a8643509e2a 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')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index a8643509e2a..f2610d67e18 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')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index f2610d67e18..90b11374a0a 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')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 90b11374a0a..fcc77fec448 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 32cf9a16f4af01573ddec1eb073111fc20a9d7d4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Feb 2009 00:06:42 +0100
Subject: ALSA: mtpav - Fix initial value for input hwport

Fix the initial value for input hwport.  The old value (-1) may cause
Oops when an realtime MIDI byte is received before the input port is
explicitly given.
Instead, now it's set to the broadcasting as default.

Tested-by: Holger Dehnhardt <dehnhardt@ahdehnhardt.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/mtpav.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5b89c0883d6..48b64e6b267 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -706,7 +706,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
 	mtp_card->card = card;
 	mtp_card->irq = -1;
 	mtp_card->share_irq = 0;
-	mtp_card->inmidiport = 0xffffffff;
 	mtp_card->inmidistate = 0;
 	mtp_card->outmidihwport = 0xffffffff;
 	init_timer(&mtp_card->timer);
@@ -719,6 +718,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
 	if (err < 0)
 		goto __error;
 
+	mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST;
+
 	err = snd_mtpav_get_ISA(mtp_card);
 	if (err < 0)
 		goto __error;
-- 
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')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 300ab407cf4..482fb0304ca 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')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 38428e22428..aa814a3c2d8 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')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 2e7371ec2e2..7006d62ca6c 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')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 0b708134d12..d03f99298be 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 1dd8716c387..44f189cb97a 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 7006d62ca6c..e48612323aa 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 aa814a3c2d8..8027edf3c8f 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 14fa43f53ff3a9c3d8b9662574b7369812a31a97 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 12 Feb 2009 19:33:19 +0000
Subject: ASoC: Only register AC97 bus if it's not done already

ASoC supports both explicit codec drivers for AC97 devices and a simple
driver which uses the standard ALSA AC97 framework for codec support.
When used with the generic AC97 codec support that will provide the
ad hoc AC97 device for drivers like touchscreens to attach to so the
core shouldn't do so.

Reported-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 55fdb4abb17..ec3f8bb4b51 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1385,7 +1385,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 
 	mutex_lock(&codec->mutex);
 #ifdef CONFIG_SND_SOC_AC97_BUS
-	if (ac97) {
+	/* Only instantiate AC97 if not already done by the adaptor
+	 * for the generic AC97 subsystem.
+	 */
+	if (ac97 && strcmp(codec->name, "AC97") != 0) {
 		ret = soc_ac97_dev_register(codec);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: AC97 device register failed\n");
-- 
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')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 11e791b965f..c8d9178f47e 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 e156ac4c571e3be741bc411e58820b74a9295c72 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 16 Feb 2009 15:22:39 +0100
Subject: sound: usb-audio: fix uninitialized variable with M-Audio MIDI
 interfaces

Fix the snd_usbmidi_create_endpoints_midiman() function, which forgot to
set the out_interval member of the endpoint info structure for Midiman/
M-Audio devices.  Since kernel 2.6.24, any non-zero value makes the
driver use interrupt transfers instead of bulk transfers.  With EHCI
controllers, these random interval values result in unbearably large
latencies for output MIDI transfers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: David <devurandom@foobox.com>
Tested-by: David <devurandom@foobox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/usbmidi.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 320641ab5be..26bad373fe6 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi,
 	}
 
 	ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+	ep_info.out_interval = 0;
 	ep_info.out_cables = endpoint->out_cables & 0x5555;
 	err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
 	if (err < 0)
-- 
cgit v1.2.3


From 0412558c873f716efe902b397af0653a550f7341 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 16 Feb 2009 22:48:12 +0100
Subject: ALSA: usb-audio - Fix non-continuous rate detection

The detection of non-continuous rates (given via rate tables) isn't
processed properly (e.g. for type II).

This patch fixes and simplifies the detection code.

Tested-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/usbaudio.c | 17 +++++++++--------
 1 file changed, 9 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 2ab83129d9b..80863093d2c 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
 		 * build the rate table and bitmap flags
 		 */
 		int r, idx;
-		unsigned int nonzero_rates = 0;
 
 		fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
 		if (fp->rate_table == NULL) {
@@ -2532,24 +2531,26 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
 			return -1;
 		}
 
-		fp->nr_rates = nr_rates;
-		fp->rate_min = fp->rate_max = combine_triple(&fmt[8]);
+		fp->nr_rates = 0;
+		fp->rate_min = fp->rate_max = 0;
 		for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) {
 			unsigned int rate = combine_triple(&fmt[idx]);
+			if (!rate)
+				continue;
 			/* C-Media CM6501 mislabels its 96 kHz altsetting */
 			if (rate == 48000 && nr_rates == 1 &&
 			    chip->usb_id == USB_ID(0x0d8c, 0x0201) &&
 			    fp->altsetting == 5 && fp->maxpacksize == 392)
 				rate = 96000;
-			fp->rate_table[r] = rate;
-			nonzero_rates |= rate;
-			if (rate < fp->rate_min)
+			fp->rate_table[fp->nr_rates] = rate;
+			if (!fp->rate_min || rate < fp->rate_min)
 				fp->rate_min = rate;
-			else if (rate > fp->rate_max)
+			if (!fp->rate_max || rate > fp->rate_max)
 				fp->rate_max = rate;
 			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+			fp->nr_rates++;
 		}
-		if (!nonzero_rates) {
+		if (!fp->nr_rates) {
 			hwc_debug("All rates were zero. Skipping format!\n");
 			return -1;
 		}
-- 
cgit v1.2.3


From 3b03cc5b86e2052295b9b484f37226ee15c87924 Mon Sep 17 00:00:00 2001
From: Joris van Rantwijk <jorispubl@xs4all.nl>
Date: Mon, 16 Feb 2009 22:58:23 +0100
Subject: ALSA: usb-audio - Workaround for misdetected sample rate with CM6207

The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz
in its USB device descriptor. This patch extends an existing workaround
in usbaudio.c to also cover the CM6207.

This resolves issue 0004249 in the ALSA bug tracker.

Signed-off-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/usbaudio.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 80863093d2c..19e37451c21 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2539,7 +2539,8 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform
 				continue;
 			/* C-Media CM6501 mislabels its 96 kHz altsetting */
 			if (rate == 48000 && nr_rates == 1 &&
-			    chip->usb_id == USB_ID(0x0d8c, 0x0201) &&
+			    (chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
+			     chip->usb_id == USB_ID(0x0d8c, 0x0102)) &&
 			    fp->altsetting == 5 && fp->maxpacksize == 392)
 				rate = 96000;
 			fp->rate_table[fp->nr_rates] = rate;
-- 
cgit v1.2.3


From 2678f60d2bc05a12580b93eb36f089f0e55693e0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 18 Feb 2009 16:46:27 +0100
Subject: ALSA: jack - Use card->shortname for input name

Currently the jack layer refers to card->longname as a part of
its input device name string.  However, longname is often really long
and way too ugly as an identifier, such as,
"HDA Intel at 0xf8400000 irq 21".

This patch changes the code to use card->shortname instead.
The shortname string contains usually the h/w vendor and product
names but without messy I/O port or IRQ numbers.

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

(limited to 'sound')

diff --git a/sound/core/jack.c b/sound/core/jack.c
index dd4a12dc09a..077a85262c1 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -47,7 +47,7 @@ static int snd_jack_dev_register(struct snd_device *device)
 	int err;
 
 	snprintf(jack->name, sizeof(jack->name), "%s %s",
-		 card->longname, jack->id);
+		 card->shortname, jack->id);
 	jack->input_dev->name = jack->name;
 
 	/* Default to the sound card device. */
-- 
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')

diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 18c7c91786b..6c870c12a17 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')

diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 84131a916c9..69d87dee699 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')

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 482fb0304ca..4ae51dcb81a 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 3d92e8f3ae9ba21cac30370eb254ed9dc20df043 Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert@linux-m68k.org>
Date: Sun, 22 Feb 2009 09:38:47 +0100
Subject: m68k: atari - Rename "mfp" to "st_mfp"

http://kisskb.ellerman.id.au/kisskb/buildresult/72115/:
| net/mac80211/ieee80211_i.h:327: error: syntax error before 'volatile'
| net/mac80211/ieee80211_i.h:350: error: syntax error before '}' token
| net/mac80211/ieee80211_i.h:455: error: field 'sta' has incomplete type
| distcc[19430] ERROR: compile net/mac80211/main.c on sprygo/32 failed

This is caused by

| # define mfp ((*(volatile struct MFP*)MFP_BAS))

in arch/m68k/include/asm/atarihw.h, which conflicts with the new "mfp" enum in
net/mac80211/ieee80211_i.h.

Rename "mfp" to "st_mfp", as it's a way too generic name for a global #define.

Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
---
 sound/oss/dmasound/dmasound_atari.c | 16 ++++++++--------
 1 file changed, 8 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 57d9f154c88..38931f2f696 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -847,23 +847,23 @@ static int __init AtaIrqInit(void)
 	   of events. So all we need to keep the music playing is
 	   to provide the sound hardware with new data upon
 	   an interrupt from timer A. */
-	mfp.tim_ct_a = 0;	/* ++roman: Stop timer before programming! */
-	mfp.tim_dt_a = 1;	/* Cause interrupt after first event. */
-	mfp.tim_ct_a = 8;	/* Turn on event counting. */
+	st_mfp.tim_ct_a = 0;	/* ++roman: Stop timer before programming! */
+	st_mfp.tim_dt_a = 1;	/* Cause interrupt after first event. */
+	st_mfp.tim_ct_a = 8;	/* Turn on event counting. */
 	/* Register interrupt handler. */
 	if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound",
 			AtaInterrupt))
 		return 0;
-	mfp.int_en_a |= 0x20;	/* Turn interrupt on. */
-	mfp.int_mk_a |= 0x20;
+	st_mfp.int_en_a |= 0x20;	/* Turn interrupt on. */
+	st_mfp.int_mk_a |= 0x20;
 	return 1;
 }
 
 #ifdef MODULE
 static void AtaIrqCleanUp(void)
 {
-	mfp.tim_ct_a = 0;	/* stop timer */
-	mfp.int_en_a &= ~0x20;	/* turn interrupt off */
+	st_mfp.tim_ct_a = 0;		/* stop timer */
+	st_mfp.int_en_a &= ~0x20;	/* turn interrupt off */
 	free_irq(IRQ_MFP_TIMA, AtaInterrupt);
 }
 #endif /* MODULE */
@@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void)
 		is_falcon = 0;
 	    } else
 		return -ENODEV;
-	    if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0)
+	    if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0)
 		return dmasound_init();
 	    else {
 		printk("DMA sound driver: Timer A interrupt already in use\n");
-- 
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')

diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 3f00ddf450f..c7c54e7748e 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 5370d96f85962769ea3df3a81cc885f257c51589 Mon Sep 17 00:00:00 2001
From: Steve Chen <schen@mvista.com>
Date: Sat, 21 Feb 2009 08:05:04 -0600
Subject: ALSA: fix excessive background noise introduced by OSS emulation rate
 shrink

Incorrect variable was used to get the next sample which caused S2
to be stuck with the same value resulting in loud background noise.

Signed-off-by: Steve Chen <schen at mvista.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/oss/rate.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c
index a466443c4a2..2fa9299a440 100644
--- a/sound/core/oss/rate.c
+++ b/sound/core/oss/rate.c
@@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin,
 		while (dst_frames1 > 0) {
 			S1 = S2;
 			if (src_frames1-- > 0) {
-				S1 = *src;
+				S2 = *src;
 				src += src_step;
 			}
 			if (pos & ~R_MASK) {
-- 
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')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ed8fcbd6000..f6571224b34 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')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f6571224b34..a680be0d453 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')

diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 7958006a1d6..101a1c13a20 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')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8027edf3c8f..3bc427645da 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')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a680be0d453..6c26afcb826 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')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8d9178f47e..5e909e0da04 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')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3bc427645da..995b413078f 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')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 995b413078f..6094344fb22 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