From 1e1689536f346a431b748dc8ad9ac0828d2c065d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 1 Jul 2009 08:34:32 +0200
Subject: ALSA: hda - Add missing static to patch_ca0110()

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0110.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 392d108c355..019ca7cb56d 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec)
 }
 
 
-int patch_ca0110(struct hda_codec *codec)
+static int patch_ca0110(struct hda_codec *codec)
 {
 	struct ca0110_spec *spec;
 	int err;
-- 
cgit v1.2.3


From ff84847171508a3c76eb7e483204d1be7738729b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 1 Jul 2009 18:08:01 +0200
Subject: ALSA: hda - Add quirk for HP 6930p

Added a quirk model=laptop for HP 6930p (103c:30dc) with AD1984A codec.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 84cc49ca914..85e8618e849 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3966,6 +3966,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
 	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
+	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
-- 
cgit v1.2.3


From 826390796d09444b93e1f957582f8970ddfd9b3d Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 2 Jul 2009 08:31:30 +0200
Subject: sound: virtuoso: fix Xonar D1/DX silence after resume

When resuming, we better take the DACs out of the reset state before
trying to use them.

Reference: kernel bug #13599
	http://bugzilla.kernel.org/show_bug.cgi?id=13599

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 | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index bf971f7cfdc..6ebcb6bdd71 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip)
 
 static void xonar_d1_resume(struct oxygen *chip)
 {
+	oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC);
+	msleep(1);
 	cs43xx_init(chip);
 	xonar_enable_output(chip);
 }
-- 
cgit v1.2.3


From 099db17e66294b02814dee01c81d9abbbeece93e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 2 Jul 2009 16:10:23 +0200
Subject: ALSA: hda - Add GPIO1 control at muting with HP laptops

HP laptops with AD1984A codecs (at least mobile models) need to set
GPIO1 appropriately to indicate the mute state.  The BIOS checks this
bit to judge whether the mute on or off is sent via F8 key.
Without changing this bit, the BIOS can be confused and may toggle
the mute wrongly.

Reference: Novell bnc#515266
	https://bugzilla.novell.com/show_bug.cgi?id=515266

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 27 ++++++++++++++++++++++++++-
 1 file changed, 26 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 85e8618e849..f795ee588cc 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3734,9 +3734,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	{ } /* end */
 };
 
+static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
+				        struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+	int mute = (!ucontrol->value.integer.value[0] &&
+		    !ucontrol->value.integer.value[1]);
+	/* toggle GPIO1 according to the mute state */
+	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+			    mute ? 0x02 : 0x0);
+	return ret;
+}
+
 static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+	/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_amp_switch_get,
+		.put = ad1884a_mobile_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+	},
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
@@ -3857,6 +3878,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = {
 	/* unsolicited event for pin-sense */
 	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+	/* allow to touch GPIO1 (for mute control) */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
 	{ } /* end */
 };
 
-- 
cgit v1.2.3


From aa202455eec51699e44f658530728162cefa1307 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 15:00:54 +0200
Subject: ALSA: hda - Improve ASUS eeePC 1000 mixer

The mixer elements created for ASUS eeePC 1000 with ALC269 aren't
standard but strange words like "LineOut".  Rename the element names
to follow the standard one like "Headphone" and "Speaker".
Also, split the volumes to each so that the virtual master can control
them.

The alc269_fujitsu_mixer is removed because it's now identical with
the new eeepc mixer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 24 +++++-------------------
 1 file changed, 5 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a8e58c483d..e661b21354b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
 	{ }
 };
 
-/* bind volumes of both NID 0x0c and 0x0d */
-static struct hda_bind_ctls alc269_epc_bind_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
 static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
-	HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol),
-	HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = {
 };
 
 /* FSC amilo */
-static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol),
-	{ } /* end */
-};
+#define alc269_fujitsu_mixer	alc269_eeepc_mixer
 
 static struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-- 
cgit v1.2.3


From 022b466fc353d3dc7a152451144be656248666ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 23:03:30 +0200
Subject: ALSA: hda - Avoid invalid formats and rates with shared SPDIF

Check whether formats and rates don't result in zero due to the
restriction of SPDIF sharing.  If any of them can be zero, disable
the SPDIF sharing mode instead.  Otherwise it will lead to a PCM
configuration error.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 462e2cedaa6..26d255de6be 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec,
 		}
 		mutex_lock(&codec->spdif_mutex);
 		if (mout->share_spdif) {
-			runtime->hw.rates &= mout->spdif_rates;
-			runtime->hw.formats &= mout->spdif_formats;
-			if (mout->spdif_maxbps < hinfo->maxbps)
-				hinfo->maxbps = mout->spdif_maxbps;
+			if ((runtime->hw.rates & mout->spdif_rates) &&
+			    (runtime->hw.formats & mout->spdif_formats)) {
+				runtime->hw.rates &= mout->spdif_rates;
+				runtime->hw.formats &= mout->spdif_formats;
+				if (mout->spdif_maxbps < hinfo->maxbps)
+					hinfo->maxbps = mout->spdif_maxbps;
+			} else {
+				mout->share_spdif = 0;
+				/* FIXME: need notify? */
+			}
 		}
 		mutex_unlock(&codec->spdif_mutex);
 	}
-- 
cgit v1.2.3


From 70d321e6380f128096429d6e5b678f94ab0cef5d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 23:06:45 +0200
Subject: ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open
 callback

The PCM rates bit field may have been changed by the codec open callback.
In that case, we need to reset rate_min and rate_max.  So, simply call
snd_pcm_lib_hw_rates() again after the codec open callback.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4e9ea708027..b36dc46615a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 		mutex_unlock(&chip->open_mutex);
 		return err;
 	}
+	snd_pcm_limit_hw_rates(runtime);
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	azx_dev->substream = substream;
 	azx_dev->running = 0;
-- 
cgit v1.2.3


From c470331e69bd54d11a9ea3c27a0e4ad783d02d6b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 23:10:23 +0200
Subject: ALSA: hda - Add sanity check in PCM open callback

Add some sanity checks of struct snd_pcm_hardware fields in the PCM
open callback of hda driver.  This makes a bit easier to debug any PCM
setup errors in the codec side.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b36dc46615a..1877d95d4aa 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1464,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	snd_pcm_set_sync(substream);
 	mutex_unlock(&chip->open_mutex);
 
+	if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max))
+		return -EINVAL;
+	if (snd_BUG_ON(!runtime->hw.formats))
+		return -EINVAL;
+	if (snd_BUG_ON(!runtime->hw.rates))
+		return -EINVAL;
 	return 0;
 }
 
-- 
cgit v1.2.3


From 02358fcfa54ce018a0bb56ca9f5a898de574a9d3 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Sat, 4 Jul 2009 01:44:59 -0300
Subject: ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
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 14f3c3e0f62..41b5b3a18c1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1590,8 +1590,6 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_REF),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30,
-		      "SigmaTel",STAC_9205_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
 		      "DFI LanParty", STAC_REF),
 	/* Dell laptops have BIOS problem */
@@ -2344,6 +2342,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_9205_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30,
+		      "SigmaTel", STAC_9205_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
 		      "DFI LanParty", STAC_9205_REF),
 	/* Dell */
-- 
cgit v1.2.3


From aba6653617754e12763a0d3c9dda332b66190a50 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 5 Jul 2009 11:44:46 +0200
Subject: ALSA: hda - Fix error path in the sanity check in azx_pcm_open()

Release resources cleanly after errors in the sanity check in
azx_pcm_open().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 18 +++++++++++-------
 1 file changed, 11 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 1877d95d4aa..16e09d74057 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1455,6 +1455,17 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 		return err;
 	}
 	snd_pcm_limit_hw_rates(runtime);
+	/* sanity check */
+	if (snd_BUG_ON(!runtime->hw.channels_min) ||
+	    snd_BUG_ON(!runtime->hw.channels_max) ||
+	    snd_BUG_ON(!runtime->hw.formats) ||
+	    snd_BUG_ON(!runtime->hw.rates)) {
+		azx_release_device(azx_dev);
+		hinfo->ops.close(hinfo, apcm->codec, substream);
+		snd_hda_power_down(apcm->codec);
+		mutex_unlock(&chip->open_mutex);
+		return -EINVAL;
+	}
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	azx_dev->substream = substream;
 	azx_dev->running = 0;
@@ -1463,13 +1474,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	runtime->private_data = azx_dev;
 	snd_pcm_set_sync(substream);
 	mutex_unlock(&chip->open_mutex);
-
-	if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max))
-		return -EINVAL;
-	if (snd_BUG_ON(!runtime->hw.formats))
-		return -EINVAL;
-	if (snd_BUG_ON(!runtime->hw.rates))
-		return -EINVAL;
 	return 0;
 }
 
-- 
cgit v1.2.3


From 55d1d6c1ef630dddd3cb5354c32a5aca954399e8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jul 2009 13:39:03 +0200
Subject: ALSA: hda - Clean up VT170x dig-in initialization code

Minor clean up for initializing the digital-in pin.
No functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 24 +++++++-----------------
 1 file changed, 7 insertions(+), 17 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 8e004fb6961..c4ddbbc6231 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -211,6 +211,7 @@ struct via_spec {
 	unsigned int num_adc_nids;
 	hda_nid_t *adc_nids;
 	hda_nid_t dig_in_nid;
+	hda_nid_t dig_in_pin;
 
 	/* capture source */
 	const struct hda_input_mux *input_mux;
@@ -998,25 +999,11 @@ static int via_init(struct hda_codec *codec)
 
 	/* Lydia Add for EAPD enable */
 	if (!spec->dig_in_nid) { /* No Digital In connection */
-		if (IS_VT1708_VENDORID(codec->vendor_id)) {
-			snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0,
+		if (spec->dig_in_pin) {
+			snd_hda_codec_write(codec, spec->dig_in_pin, 0,
 					    AC_VERB_SET_PIN_WIDGET_CONTROL,
 					    PIN_OUT);
-			snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 0x02);
-		} else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) ||
-			   IS_VT1709_6CH_VENDORID(codec->vendor_id)) {
-			snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    PIN_OUT);
-			snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 0x02);
-		} else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) ||
-			   IS_VT1708B_4CH_VENDORID(codec->vendor_id)) {
-			snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    PIN_OUT);
-			snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0,
+			snd_hda_codec_write(codec, spec->dig_in_pin, 0,
 					    AC_VERB_SET_EAPD_BTLENABLE, 0x02);
 		}
 	} else /* enable SPDIF-input pin */
@@ -1326,6 +1313,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 
 	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708_DIGOUT_NID;
+	spec->dig_in_pin = VT1708_DIGIN_PIN;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708_DIGIN_NID;
 
@@ -1799,6 +1787,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 
 	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1709_DIGOUT_NID;
+	spec->dig_in_pin = VT1709_DIGIN_PIN;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1709_DIGIN_NID;
 
@@ -2344,6 +2333,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 
 	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID;
+	spec->dig_in_pin = VT1708B_DIGIN_PIN;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708B_DIGIN_NID;
 
-- 
cgit v1.2.3


From d3a11e601a51291fbdd40c47f6af6769b6e905ef Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jul 2009 13:43:35 +0200
Subject: ALSA: hda - Add missing EAPD initialization for VIA codecs

If the output pin is used and EAPD capability is present, turn on
the EAPD bit.  This fixes the silent output problem on ASUS laptops
with VT1708S codec.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index c4ddbbc6231..322e1027247 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -320,6 +320,9 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec,
 			    pin_type);
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 			    AMP_OUT_UNMUTE);
+	if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+		snd_hda_codec_write(codec, nid, 0, 
+				    AC_VERB_SET_EAPD_BTLENABLE, 0x02);
 }
 
 
-- 
cgit v1.2.3


From 337b9d02b4873ceac91565272545fb6fd446d939 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jul 2009 18:18:59 +0200
Subject: ALSA: hda - Fix capture source selection in patch_via.c

The fixed widget NIDs in patch_via.c seem wrong for some codecs,
and it resulted in the invalid capture source selection.

This patch adds the code to parse the topology instead of using
fixed numbers in order to get the right MUX widget id corresponding
to the ADCs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 54 ++++++++++++++++++++++++++++++-----------------
 1 file changed, 35 insertions(+), 19 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 322e1027247..38db4596422 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -210,6 +210,7 @@ struct via_spec {
 	/* capture */
 	unsigned int num_adc_nids;
 	hda_nid_t *adc_nids;
+	hda_nid_t mux_nids[3];
 	hda_nid_t dig_in_nid;
 	hda_nid_t dig_in_pin;
 
@@ -393,25 +394,11 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
 	unsigned int vendor_id = codec->vendor_id;
 
-	/* AIW0  lydia 060801 add for correct sw0 input select */
-	if (IS_VT1708_VENDORID(vendor_id) && (adc_idx == 0))
-		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-					     0x18, &spec->cur_mux[adc_idx]);
-	else if ((IS_VT1709_10CH_VENDORID(vendor_id) ||
-		  IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0))
-		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-					     0x19, &spec->cur_mux[adc_idx]);
-	else if ((IS_VT1708B_8CH_VENDORID(vendor_id) ||
-		  IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0))
-		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-					     0x17, &spec->cur_mux[adc_idx]);
-	else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0))
-		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-					     0x13, &spec->cur_mux[adc_idx]);
-	else
-		return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-					     spec->adc_nids[adc_idx],
-					     &spec->cur_mux[adc_idx]);
+	if (!spec->mux_nids[adc_idx])
+		return -EINVAL;
+	return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+				     spec->mux_nids[adc_idx],
+				     &spec->cur_mux[adc_idx]);
 }
 
 static int via_independent_hp_info(struct snd_kcontrol *kcontrol,
@@ -1343,6 +1330,29 @@ static int via_auto_init(struct hda_codec *codec)
 	return 0;
 }
 
+static int get_mux_nids(struct hda_codec *codec)
+{
+	struct via_spec *spec = codec->spec;
+	hda_nid_t nid, conn[8];
+	unsigned int type;
+	int i, n;
+
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		nid = spec->adc_nids[i];
+		while (nid) {
+			n = snd_hda_get_connections(codec, nid, conn,
+						    ARRAY_SIZE(conn));
+			if (n <= 0)
+				break;
+			if (n > 1) {
+				spec->mux_nids[i] = nid;
+				break;
+			}
+			nid = conn[0];
+		}
+	}
+}
+
 static int patch_vt1708(struct hda_codec *codec)
 {
 	struct via_spec *spec;
@@ -1851,6 +1861,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		spec->adc_nids = vt1709_adc_nids;
 		spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
+		get_mux_nids(codec);
 		spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
 		spec->num_mixers++;
 	}
@@ -1944,6 +1955,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		spec->adc_nids = vt1709_adc_nids;
 		spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids);
+		get_mux_nids(codec);
 		spec->mixers[spec->num_mixers] = vt1709_capture_mixer;
 		spec->num_mixers++;
 	}
@@ -2397,6 +2409,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		spec->adc_nids = vt1708B_adc_nids;
 		spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
+		get_mux_nids(codec);
 		spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
 		spec->num_mixers++;
 	}
@@ -2448,6 +2461,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		spec->adc_nids = vt1708B_adc_nids;
 		spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids);
+		get_mux_nids(codec);
 		spec->mixers[spec->num_mixers] = vt1708B_capture_mixer;
 		spec->num_mixers++;
 	}
@@ -2882,6 +2896,7 @@ static int patch_vt1708S(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		spec->adc_nids = vt1708S_adc_nids;
 		spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids);
+		get_mux_nids(codec);
 		spec->mixers[spec->num_mixers] = vt1708S_capture_mixer;
 		spec->num_mixers++;
 	}
@@ -3199,6 +3214,7 @@ static int patch_vt1702(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		spec->adc_nids = vt1702_adc_nids;
 		spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids);
+		get_mux_nids(codec);
 		spec->mixers[spec->num_mixers] = vt1702_capture_mixer;
 		spec->num_mixers++;
 	}
-- 
cgit v1.2.3


From 1c55d521f4e58be55735d7ac47e8197d6791fa9a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Jul 2009 07:45:46 +0200
Subject: ALSA: hda - Check widget types while parsing capture source in
 patch_via.c

Check the widget type and don't take invalid widgets while parsing
the capture source in patch_via.c.

Also, fixed some compile warnings introduced in the previous commit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 38db4596422..9008b4b013a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -392,7 +392,6 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct via_spec *spec = codec->spec;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	unsigned int vendor_id = codec->vendor_id;
 
 	if (!spec->mux_nids[adc_idx])
 		return -EINVAL;
@@ -1340,6 +1339,10 @@ static int get_mux_nids(struct hda_codec *codec)
 	for (i = 0; i < spec->num_adc_nids; i++) {
 		nid = spec->adc_nids[i];
 		while (nid) {
+			type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
+				>> AC_WCAP_TYPE_SHIFT;
+			if (type == AC_WID_PIN)
+				break;
 			n = snd_hda_get_connections(codec, nid, conn,
 						    ARRAY_SIZE(conn));
 			if (n <= 0)
@@ -1351,6 +1354,7 @@ static int get_mux_nids(struct hda_codec *codec)
 			nid = conn[0];
 		}
 	}
+	return 0;
 }
 
 static int patch_vt1708(struct hda_codec *codec)
-- 
cgit v1.2.3


From dc4c2e6bde77735071dbef7aca6bd6c0116102b3 Mon Sep 17 00:00:00 2001
From: Andiry Brienza <andiry.xu@amd.com>
Date: Wed, 8 Jul 2009 13:55:31 +0800
Subject: ALSA: hda - Disable AMD SB600 64bit address support only

HDA driver disabled HD audio 64bit address support for all AMD
SB600/SB700/SB800 platforms with commit
09240cf429505891d6123ce14a29f58f2a60121e due to one SB600 issue
reported by community, but we do not see the similar issue on
SB700/SB800 platforms.
This patch is to refine the workaround for SB600 only.

Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 16 +++++++++++++---
 1 file changed, 13 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 16e09d74057..77c1b840ca8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2333,9 +2333,19 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	gcap = azx_readw(chip, GCAP);
 	snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap);
 
-	/* ATI chips seems buggy about 64bit DMA addresses */
-	if (chip->driver_type == AZX_DRIVER_ATI)
-		gcap &= ~ICH6_GCAP_64OK;
+	/* disable SB600 64bit support for safety */
+	if ((chip->driver_type == AZX_DRIVER_ATI) ||
+	    (chip->driver_type == AZX_DRIVER_ATIHDMI)) {
+		struct pci_dev *p_smbus;
+		p_smbus = pci_get_device(PCI_VENDOR_ID_ATI,
+					 PCI_DEVICE_ID_ATI_SBX00_SMBUS,
+					 NULL);
+		if (p_smbus) {
+			if (p_smbus->revision < 0x30)
+				gcap &= ~ICH6_GCAP_64OK;
+			pci_dev_put(p_smbus);
+		}
+	}
 
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
-- 
cgit v1.2.3


From 508f711090e06477081fd94cb9298b1b14dda9ff Mon Sep 17 00:00:00 2001
From: Darren Salt <linux@youmustbejoking.demon.co.uk>
Date: Wed, 8 Jul 2009 15:29:49 +0100
Subject: ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC)

There is a regression, introduced in aa202455eec51699e44f658530728162cefa1307
(in alsa-kernel) which I noticed when trying to use the headphone socket on
my EeeCPC 901: the output was *very* quiet, practically silent.

This patch corrects the control types to that which was obviously intended in
the referenced commit.

Signed-off-by: Darren Salt <linux@youmustbejoking.demon.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e661b21354b..c6c3d4a4d64 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12878,9 +12878,9 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
 
 static struct snd_kcontrol_new alc269_eeepc_mixer[] = {
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
-- 
cgit v1.2.3


From ad361c9884e809340f6daca80d56a9e9c871690a Mon Sep 17 00:00:00 2001
From: Joe Perches <joe@perches.com>
Date: Mon, 6 Jul 2009 13:05:40 -0700
Subject: Remove multiple KERN_ prefixes from printk formats

Commit 5fd29d6ccbc98884569d6f3105aeca70858b3e0f ("printk: clean up
handling of log-levels and newlines") changed printk semantics.  printk
lines with multiple KERN_<level> prefixes are no longer emitted as
before the patch.

<level> is now included in the output on each additional use.

Remove all uses of multiple KERN_<level>s in formats.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
---
 sound/pci/emu10k1/p16v.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index e617acaf10e..61b8ab39800 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -644,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 	int err;
         int capture=1;
   
-	/* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */
+	/* snd_printk(KERN_DEBUG "snd_p16v_pcm called. device=%d\n", device); */
 	emu->p16v_device_offset = device;
 	if (rpcm)
 		*rpcm = NULL;
-- 
cgit v1.2.3


From 369693dc93533097c0ca7243affb4f3244c336e8 Mon Sep 17 00:00:00 2001
From: Paul Vojta <vojta@math.berkeley.edu>
Date: Wed, 8 Jul 2009 23:57:46 -0700
Subject: ALSA: hda - fix beep tone calculation for IDT/STAC codecs

In the beep tone calculation for IDT/STAC codecs, lower numbers correspond
to higher frequencies and vice versa.  The current code has this backwards,
resulting in beep frequencies which are way too high (and sound bad on
tinny laptop speakers, resulting in complaints).

[Also added hz <= 0 check by tiwai]

Signed-off-by: Paul Vojta <vojta@math.berkeley.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 29272f2e95a..b0275a05087 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -50,19 +50,22 @@ static void snd_hda_generate_beep(struct work_struct *work)
  * The tone frequency of beep generator on IDT/STAC codecs is
  * defined from the 8bit tone parameter, in Hz,
  *    freq = 48000 * (257 - tone) / 1024
- * that is from 12kHz to 93.75kHz in step of 46.875 hz
+ * that is from 12kHz to 93.75Hz in steps of 46.875 Hz
  */
 static int beep_linear_tone(struct hda_beep *beep, int hz)
 {
+	if (hz <= 0)
+		return 0;
 	hz *= 1000; /* fixed point */
-	hz = hz - DIGBEEP_HZ_MIN;
+	hz = hz - DIGBEEP_HZ_MIN
+		+ DIGBEEP_HZ_STEP / 2; /* round to nearest step */
 	if (hz < 0)
 		hz = 0; /* turn off PC beep*/
 	else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
-		hz = 0xff;
+		hz = 1; /* max frequency */
 	else {
 		hz /= DIGBEEP_HZ_STEP;
-		hz++;
+		hz = 255 - hz;
 	}
 	return hz;
 }
-- 
cgit v1.2.3


From 005b10769c05fb16db70f7689ffb5ba17e3fc324 Mon Sep 17 00:00:00 2001
From: David Heidelberger <d.okias@gmail.com>
Date: Thu, 9 Jul 2009 18:45:46 +0200
Subject: ALSA: hda - targa and targa-2ch fix

Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and
ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id
64a8be74357477558183b43156c5536b642de134

Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 12 ++++++------
 1 file changed, 6 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c6c3d4a4d64..bbb9b42e260 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6919,9 +6919,6 @@ static struct hda_verb alc882_targa_verbs[] = {
 	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
 
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x03},
 	{ } /* end */
 };
 
@@ -7241,7 +7238,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_TARGA] = {
 		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_targa_verbs},
+		.init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs,
+				alc882_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
@@ -9238,7 +9236,8 @@ static struct alc_config_preset alc883_presets[] = {
 	},
 	[ALC883_TARGA_DIG] = {
 		.mixers = { alc883_targa_mixer, alc883_chmode_mixer },
-		.init_verbs = { alc883_init_verbs, alc883_targa_verbs},
+		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+				alc883_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
@@ -9251,7 +9250,8 @@ static struct alc_config_preset alc883_presets[] = {
 	},
 	[ALC883_TARGA_2ch_DIG] = {
 		.mixers = { alc883_targa_2ch_mixer},
-		.init_verbs = { alc883_init_verbs, alc883_targa_verbs},
+		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+				alc883_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.adc_nids = alc883_adc_nids_alt,
-- 
cgit v1.2.3


From 9d30937accf2c01e8b0bd59787409a7348cbbcb7 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Fri, 10 Jul 2009 12:27:31 +0200
Subject: ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid
 checking

On some IbexPeak systems with ALC889A errors like "azx_get_response
timeout, switching to polling mode: last cmd=0xaf9f000b" are produced,
because non-existent codec #10 is wrongly accessed.

The problem is that snd_hda_get_connections() returns out-of-range result
for NID 0x1c (something like 0xf8f9 or 0xffff).

This patch adds a check to alc880_parse_auto_config() to avoid using
of this out-of-range NIDs. A better fix maybe to improve
snd_hda_get_connections() routine to check for valid NID ranges if
NIDs are expected as result.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bbb9b42e260..7e99763ca52 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4505,6 +4505,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 					      &dig_nid, 1);
 		if (err < 0)
 			continue;
+		if (dig_nid > 0x7f) {
+			printk(KERN_ERR "alc880_auto: invalid dig_nid "
+				"connection 0x%x for NID 0x%x\n", dig_nid,
+				spec->autocfg.dig_out_pins[i]);
+			continue;
+		}
 		if (!i)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
-- 
cgit v1.2.3


From cb65c8732a50f8a145d36dbdac026a1789ad1587 Mon Sep 17 00:00:00 2001
From: Jaswinder Singh Rajput <jaswinder@kernel.org>
Date: Wed, 15 Jul 2009 16:45:40 +0530
Subject: ALSA: riptide -  proper handling of pci_register_driver for joystick
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

We need to check returning error for pci_register_driver(&joystick_driver)

On failure, we should unregister formerly registered audio drivers

This also fixed the compiler warning :

  CC [M]  sound/pci/riptide/riptide.o
 sound/pci/riptide/riptide.c: In function ‘alsa_card_riptide_init’:
 sound/pci/riptide/riptide.c:2200: warning: ignoring return value of ‘__pci_register_driver’, declared with attribute warn_unused_result

Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/riptide/riptide.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 235a71e5ac8..b5ca02e2038 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2197,9 +2197,12 @@ static int __init alsa_card_riptide_init(void)
 	if (err < 0)
 		return err;
 #if defined(SUPPORT_JOYSTICK)
-	pci_register_driver(&joystick_driver);
+	err = pci_register_driver(&joystick_driver);
+	/* On failure unregister formerly registered audio driver */
+	if (err < 0)
+		pci_unregister_driver(&driver);
 #endif
-	return 0;
+	return err;
 }
 
 static void __exit alsa_card_riptide_exit(void)
-- 
cgit v1.2.3


From 2e9bf247066a293ebcd4672ddd487808ab5f2d1b Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Sat, 18 Jul 2009 11:48:19 +0200
Subject: ALSA: hda_codec: Check for invalid zero connections

To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 26d255de6be..88480c0c58a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -332,6 +332,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 						  AC_VERB_GET_CONNECT_LIST, i);
 		range_val = !!(parm & (1 << (shift-1))); /* ranges */
 		val = parm & mask;
+		if (val == 0) {
+			snd_printk(KERN_WARNING "hda_codec: "
+				   "invalid CONNECT_LIST verb %x[%i]:%x\n",
+				    nid, i, parm);
+			return 0;
+		}
 		parm >>= shift;
 		if (range_val) {
 			/* ranges between the previous and this one */
-- 
cgit v1.2.3


From 42b95f0c6b524b5a670dd17533a3522db368f600 Mon Sep 17 00:00:00 2001
From: Hao Song <baritono.tux@gmail.com>
Date: Mon, 20 Jul 2009 15:01:16 +0800
Subject: ALSA: hda - Add quirk for Gateway T6834c laptop

Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a18c1..d9b89ba2b65 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2378,6 +2378,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
 		      "Dell Vostro 1500", STAC_9205_DELL_M42),
 	/* Gateway */
+	SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD),
 	SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
 	{} /* terminator */
 };
-- 
cgit v1.2.3


From b04add956616b6d89ff21da749b46ad2bd58ef32 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Jul 2009 08:01:36 +0200
Subject: ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs

The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS.  Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.

This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides.  Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
 sound/pci/hda/patch_sigmatel.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d9b89ba2b65..da7f9f65c04 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5855,6 +5855,8 @@ static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
 };
 
 static struct snd_pci_quirk stac9872_cfg_tbl[] = {
+	SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0,
+			   "Sony VAIO F/S", STAC_9872_VAIO),
 	{} /* terminator */
 };
 
@@ -5867,6 +5869,8 @@ static int patch_stac9872(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 	codec->spec = spec;
+	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+	spec->pin_nids = stac9872_pin_nids;
 
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
 							stac9872_models,
@@ -5878,8 +5882,6 @@ static int patch_stac9872(struct hda_codec *codec)
 		stac92xx_set_config_regs(codec,
 					 stac9872_brd_tbl[spec->board_config]);
 
-	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
-	spec->pin_nids = stac9872_pin_nids;
 	spec->multiout.dac_nids = spec->dac_nids;
 	spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
 	spec->adc_nids = stac9872_adc_nids;
-- 
cgit v1.2.3


From 34fdeb2d07102e07ecafe79dec170bd6733f2e56 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Jul 2009 15:42:51 +0200
Subject: ALSA: ca0106 - Fix the max capture buffer size

The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.

This patch restricts the max buffer size less than that to make life
a bit easier.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
 sound/pci/ca0106/ca0106_main.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 57b992a5c05..700f15ea16d 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -325,9 +325,9 @@ static struct snd_pcm_hardware snd_ca0106_capture_hw = {
 	.rate_max =		192000,
 	.channels_min =		2,
 	.channels_max =		2,
-	.buffer_bytes_max =	((65536 - 64) * 8),
+	.buffer_bytes_max =	65536 - 128,
 	.period_bytes_min =	64,
-	.period_bytes_max =	(65536 - 64),
+	.period_bytes_max =	32768 - 64,
 	.periods_min =		2,
 	.periods_max =		2,
 	.fifo_size =		0,
-- 
cgit v1.2.3


From 55fe27f7e2c9d24ce870136bd99ae67b020122d1 Mon Sep 17 00:00:00 2001
From: Frank Roth <frashman@freenet.de>
Date: Mon, 20 Jul 2009 17:00:14 +0200
Subject: ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2

On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong.
I double checked it with connector colors and creative soundblaster
windows drivers.

So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.

Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctdaio.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 082e35c08c0..deb6cfa7360 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -57,9 +57,9 @@ struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
 
 struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
 	[LINEO1] = {.left = 0x40, .right = 0x41},
-	[LINEO2] = {.left = 0x70, .right = 0x71},
+	[LINEO2] = {.left = 0x60, .right = 0x61},
 	[LINEO3] = {.left = 0x50, .right = 0x51},
-	[LINEO4] = {.left = 0x60, .right = 0x61},
+	[LINEO4] = {.left = 0x70, .right = 0x71},
 	[LINEIM] = {.left = 0x45, .right = 0xc5},
 	[SPDIFOO] = {.left = 0x00, .right = 0x01},
 	[SPDIFIO] = {.left = 0x05, .right = 0x85},
-- 
cgit v1.2.3


From 86de7416600e93835eeacee379aea939b6a0917a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 16:02:46 +0200
Subject: ALSA: hda - Use snprintf() to be safer

Use snprint() for creating the jack name string instead of sprintf()
in patch_sigmatel.c.

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 da7f9f65c04..512f3b9b9a4 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4066,7 +4066,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
 	jack->nid = nid;
 	jack->type = type;
 
-	sprintf(name, "%s at %s %s Jack",
+	snprintf(name, sizeof(name), "%s at %s %s Jack",
 		snd_hda_get_jack_type(def_conf),
 		snd_hda_get_jack_connectivity(def_conf),
 		snd_hda_get_jack_location(def_conf));
-- 
cgit v1.2.3


From 68110661e86868cd107955ec7c077e1f34519f78 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 17:05:15 +0200
Subject: ALSA: ctxfi - Fix uninitialized error checks
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
  sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
  sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
  sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
  sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
  sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
  sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ctxfi/ctamixer.c | 14 ++++++--------
 sound/pci/ctxfi/ctsrc.c    |  7 +++----
 2 files changed, 9 insertions(+), 12 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index a1db51b3ead..a7f4a671f7b 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -242,13 +242,12 @@ static int get_amixer_rsc(struct amixer_mgr *mgr,
 
 	/* Allocate mem for amixer resource */
 	amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
-	if (NULL == amixer) {
-		err = -ENOMEM;
-		return err;
-	}
+	if (!amixer)
+		return -ENOMEM;
 
 	/* Check whether there are sufficient
 	 * amixer resources to meet request. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
@@ -397,12 +396,11 @@ static int get_sum_rsc(struct sum_mgr *mgr,
 
 	/* Allocate mem for sum resource */
 	sum = kzalloc(sizeof(*sum), GFP_KERNEL);
-	if (NULL == sum) {
-		err = -ENOMEM;
-		return err;
-	}
+	if (!sum)
+		return -ENOMEM;
 
 	/* Check whether there are sufficient sum resources to meet request. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index e1c145d8b70..df43a5cd393 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -724,12 +724,11 @@ static int get_srcimp_rsc(struct srcimp_mgr *mgr,
 
 	/* Allocate mem for SRCIMP resource */
 	srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
-	if (NULL == srcimp) {
-		err = -ENOMEM;
-		return err;
-	}
+	if (!srcimp)
+		return -ENOMEM;
 
 	/* Check whether there are sufficient SRCIMP resources. */
+	err = 0;
 	spin_lock_irqsave(&mgr->mgr_lock, flags);
 	for (i = 0; i < desc->msr; i++) {
 		err = mgr_get_resource(&mgr->mgr, 1, &idx);
-- 
cgit v1.2.3


From 4012ade9338c05428162e85cc9b149dcadf1ce85 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 18:15:10 +0200
Subject: ALSA: hda - Restore GPIO1 properly at resume with AD1984A

The commit 099db17e66294b02814dee01c81d9abbbeece93e introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.

The fix is simple, use the cached write for storing GPIO data.

Reference: Novell bnc#522764
	https://bugzilla.novell.com/show_bug.cgi?id=522764

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index f795ee588cc..e8e6a43865c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3742,7 +3742,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
 	int mute = (!ucontrol->value.integer.value[0] &&
 		    !ucontrol->value.integer.value[1]);
 	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
 			    mute ? 0x02 : 0x0);
 	return ret;
 }
-- 
cgit v1.2.3


From 8de56b7deb2534a586839eda52843c1dae680dc5 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 24 Jul 2009 16:51:47 +0200
Subject: ALSA: hda - Fix mute control with some ALC262 models

The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted.  This patch fixes
the issue.

Reference: Novell bnc#404873
	https://bugzilla.novell.com/show_bug.cgi?id=404873

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
 sound/pci/hda/patch_realtek.c | 33 ++++++++++++++++-----------------
 1 file changed, 16 insertions(+), 17 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7e99763ca52..8c8b273116f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10631,6 +10631,18 @@ static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec,
 	alc262_lenovo_3000_automute(codec, 1);
 }
 
+static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid,
+				  int dir, int idx, long *valp)
+{
+	int i, change = 0;
+
+	for (i = 0; i < 2; i++, valp++)
+		change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx,
+						   HDA_AMP_MUTE,
+						   *valp ? 0 : HDA_AMP_MUTE);
+	return change;
+}
+
 /* bind hp and internal speaker mute (with plug check) */
 static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
 					 struct snd_ctl_elem_value *ucontrol)
@@ -10639,13 +10651,8 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-	change |= snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-
+	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
+	change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
 	if (change)
 		alc262_fujitsu_automute(codec, 0);
 	return change;
@@ -10680,10 +10687,7 @@ static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-						 HDA_AMP_MUTE,
-						 valp ? 0 : HDA_AMP_MUTE);
-
+	change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp);
 	if (change)
 		alc262_lenovo_3000_automute(codec, 0);
 	return change;
@@ -11854,12 +11858,7 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 	int change;
 
-	change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-					  HDA_AMP_MUTE,
-					  valp[0] ? 0 : HDA_AMP_MUTE);
-	change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-					   HDA_AMP_MUTE,
-					   valp[1] ? 0 : HDA_AMP_MUTE);
+	change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp);
 	if (change)
 		alc268_acer_automute(codec, 0);
 	return change;
-- 
cgit v1.2.3


From 626f5cefc60b281a00db1402b82deff82080c70a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jul 2009 00:54:39 +0200
Subject: ALSA: hda - Add quirk for Dell Studio 1555

Added a quirk entry for Dell Studio 1555.

Reference: Novell bnc#525244
	https://bugzilla.novell.com/show_bug.cgi?id=525244

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 512f3b9b9a4..5383d8cff88 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1809,6 +1809,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 1537", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a0,
 				"Dell Studio 17", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
+				"Dell Studio 1555", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From 78735cffc2d9ab0dec32f1ba7cbc1d84b45bbf29 Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Wed, 29 Jul 2009 14:35:20 +0200
Subject: ALSA: hda: fix out-of-bound hdmi_eld.sad[] write

e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_eld.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index fcad5ec3177..9446a5abea1 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -508,7 +508,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
 	char name[64];
 	char *sname;
 	long long val;
-	int n;
+	unsigned int n;
 
 	while (!snd_info_get_line(buffer, line, sizeof(line))) {
 		if (sscanf(line, "%s %llx", name, &val) != 2)
@@ -539,7 +539,7 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry,
 				sname++;
 				n = 10 * n + name[4] - '0';
 			}
-			if (n < 0 || n > 31) /* double the CEA limit */
+			if (n >= ELD_MAX_SAD)
 				continue;
 			if (!strcmp(sname, "_coding_type"))
 				e->sad[n].format = val;
-- 
cgit v1.2.3


From aa563af763373a7e67a7b8fdb427d2a2fcbeab3b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 31 Jul 2009 10:05:11 +0200
Subject: ALSA: hda - Increase PCM stream name buf in patch_realtek.c

The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo".  Now the size is doubled.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c8b273116f..b95df5d5dcc 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -275,13 +275,13 @@ struct alc_spec {
 						 */
 	unsigned int num_init_verbs;
 
-	char stream_name_analog[16];	/* analog PCM stream */
+	char stream_name_analog[32];	/* analog PCM stream */
 	struct hda_pcm_stream *stream_analog_playback;
 	struct hda_pcm_stream *stream_analog_capture;
 	struct hda_pcm_stream *stream_analog_alt_playback;
 	struct hda_pcm_stream *stream_analog_alt_capture;
 
-	char stream_name_digital[16];	/* digital PCM stream */
+	char stream_name_digital[32];	/* digital PCM stream */
 	struct hda_pcm_stream *stream_digital_playback;
 	struct hda_pcm_stream *stream_digital_capture;
 
-- 
cgit v1.2.3


From ce577e8cf5ddb4216553c9d563a9835d6de70ffa Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 3 Aug 2009 08:23:52 +0200
Subject: ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527

Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.

Reference: Novell bnc#526325
	https://bugzilla.novell.com/show_bug.cgi?id=526325

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b95df5d5dcc..f6b4cbf1ead 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15157,7 +15157,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
 	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
 	/*SND_PCI_QUIRK(0x1179, 0xff00, "DALLAS", ALC861VD_DALLAS),*/ /*lenovo*/
-	SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
+	SND_PCI_QUIRK(0x1179, 0xff01, "Toshiba A135", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
 	SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
-- 
cgit v1.2.3


From deadff1665491afce124a8ff83f00f784161f660 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:45:16 +0800
Subject: ALSA: hda: track CIRB/CORB command/response states for each codec

Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.

Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c |  2 +-
 sound/pci/hda/hda_codec.h |  2 +-
 sound/pci/hda/hda_intel.c | 76 +++++++++++++++++++++++++++++++++--------------
 3 files changed, 56 insertions(+), 24 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88480c0c58a..c7df01b72ca 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -174,7 +174,7 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
 	mutex_lock(&bus->cmd_mutex);
 	err = bus->ops.command(bus, cmd);
 	if (!err && res)
-		*res = bus->ops.get_response(bus);
+		*res = bus->ops.get_response(bus, codec->addr);
 	mutex_unlock(&bus->cmd_mutex);
 	snd_hda_power_down(codec);
 	if (res && *res == -1 && bus->rirb_error) {
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index cad79efaabc..1b75f28ed09 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -568,7 +568,7 @@ struct hda_bus_ops {
 	/* send a single command */
 	int (*command)(struct hda_bus *bus, unsigned int cmd);
 	/* get a response from the last command */
-	unsigned int (*get_response)(struct hda_bus *bus);
+	unsigned int (*get_response)(struct hda_bus *bus, unsigned int addr);
 	/* free the private data */
 	void (*private_free)(struct hda_bus *);
 	/* attach a PCM stream */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 77c1b840ca8..19e67a1b602 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -253,7 +253,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 
 /* STATESTS int mask: S3,SD2,SD1,SD0 */
 #define AZX_MAX_CODECS		4
-#define STATESTS_INT_MASK	0x0f
+#define STATESTS_INT_MASK	((1 << AZX_MAX_CODECS) - 1)
 
 /* SD_CTL bits */
 #define SD_CTL_STREAM_RESET	0x01	/* stream reset bit */
@@ -361,8 +361,8 @@ struct azx_rb {
 	dma_addr_t addr;	/* physical address of CORB/RIRB buffer */
 	/* for RIRB */
 	unsigned short rp, wp;	/* read/write pointers */
-	int cmds;		/* number of pending requests */
-	u32 res;		/* last read value */
+	int cmds[AZX_MAX_CODECS];	/* number of pending requests */
+	u32 res[AZX_MAX_CODECS];	/* last read value */
 };
 
 struct azx {
@@ -531,7 +531,8 @@ static void azx_init_cmd_io(struct azx *chip)
 	/* RIRB set up */
 	chip->rirb.addr = chip->rb.addr + 2048;
 	chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
-	chip->rirb.wp = chip->rirb.rp = chip->rirb.cmds = 0;
+	chip->rirb.wp = chip->rirb.rp = 0;
+	memset(chip->rirb.cmds, 0, sizeof(chip->rirb.cmds));
 	azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr);
 	azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr));
 
@@ -552,10 +553,35 @@ static void azx_free_cmd_io(struct azx *chip)
 	azx_writeb(chip, CORBCTL, 0);
 }
 
+static unsigned int azx_command_addr(u32 cmd)
+{
+	unsigned int addr = cmd >> 28;
+
+	if (addr >= AZX_MAX_CODECS) {
+		snd_BUG();
+		addr = 0;
+	}
+
+	return addr;
+}
+
+static unsigned int azx_response_addr(u32 res)
+{
+	unsigned int addr = res & 0xf;
+
+	if (addr >= AZX_MAX_CODECS) {
+		snd_BUG();
+		addr = 0;
+	}
+
+	return addr;
+}
+
 /* send a command */
 static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
 {
 	struct azx *chip = bus->private_data;
+	unsigned int addr = azx_command_addr(val);
 	unsigned int wp;
 
 	/* add command to corb */
@@ -564,7 +590,7 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
 	wp %= ICH6_MAX_CORB_ENTRIES;
 
 	spin_lock_irq(&chip->reg_lock);
-	chip->rirb.cmds++;
+	chip->rirb.cmds[addr]++;
 	chip->corb.buf[wp] = cpu_to_le32(val);
 	azx_writel(chip, CORBWP, wp);
 	spin_unlock_irq(&chip->reg_lock);
@@ -578,13 +604,14 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
 static void azx_update_rirb(struct azx *chip)
 {
 	unsigned int rp, wp;
+	unsigned int addr;
 	u32 res, res_ex;
 
 	wp = azx_readb(chip, RIRBWP);
 	if (wp == chip->rirb.wp)
 		return;
 	chip->rirb.wp = wp;
-		
+
 	while (chip->rirb.rp != wp) {
 		chip->rirb.rp++;
 		chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES;
@@ -592,18 +619,20 @@ static void azx_update_rirb(struct azx *chip)
 		rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
 		res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
 		res = le32_to_cpu(chip->rirb.buf[rp]);
+		addr = azx_response_addr(res_ex);
 		if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
 			snd_hda_queue_unsol_event(chip->bus, res, res_ex);
-		else if (chip->rirb.cmds) {
-			chip->rirb.res = res;
+		else if (chip->rirb.cmds[addr]) {
+			chip->rirb.res[addr] = res;
 			smp_wmb();
-			chip->rirb.cmds--;
+			chip->rirb.cmds[addr]--;
 		}
 	}
 }
 
 /* receive a response */
-static unsigned int azx_rirb_get_response(struct hda_bus *bus)
+static unsigned int azx_rirb_get_response(struct hda_bus *bus,
+					  unsigned int addr)
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
@@ -616,10 +645,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
 			azx_update_rirb(chip);
 			spin_unlock_irq(&chip->reg_lock);
 		}
-		if (!chip->rirb.cmds) {
+		if (!chip->rirb.cmds[addr]) {
 			smp_rmb();
 			bus->rirb_error = 0;
-			return chip->rirb.res; /* the last value */
+			return chip->rirb.res[addr]; /* the last value */
 		}
 		if (time_after(jiffies, timeout))
 			break;
@@ -692,7 +721,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
  */
 
 /* receive a response */
-static int azx_single_wait_for_response(struct azx *chip)
+static int azx_single_wait_for_response(struct azx *chip, unsigned int addr)
 {
 	int timeout = 50;
 
@@ -700,7 +729,7 @@ static int azx_single_wait_for_response(struct azx *chip)
 		/* check IRV busy bit */
 		if (azx_readw(chip, IRS) & ICH6_IRS_VALID) {
 			/* reuse rirb.res as the response return value */
-			chip->rirb.res = azx_readl(chip, IR);
+			chip->rirb.res[addr] = azx_readl(chip, IR);
 			return 0;
 		}
 		udelay(1);
@@ -708,7 +737,7 @@ static int azx_single_wait_for_response(struct azx *chip)
 	if (printk_ratelimit())
 		snd_printd(SFX "get_response timeout: IRS=0x%x\n",
 			   azx_readw(chip, IRS));
-	chip->rirb.res = -1;
+	chip->rirb.res[addr] = -1;
 	return -EIO;
 }
 
@@ -716,6 +745,7 @@ static int azx_single_wait_for_response(struct azx *chip)
 static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
 {
 	struct azx *chip = bus->private_data;
+	unsigned int addr = azx_command_addr(val);
 	int timeout = 50;
 
 	bus->rirb_error = 0;
@@ -728,7 +758,7 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
 			azx_writel(chip, IC, val);
 			azx_writew(chip, IRS, azx_readw(chip, IRS) |
 				   ICH6_IRS_BUSY);
-			return azx_single_wait_for_response(chip);
+			return azx_single_wait_for_response(chip, addr);
 		}
 		udelay(1);
 	}
@@ -739,10 +769,11 @@ static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
 }
 
 /* receive a response */
-static unsigned int azx_single_get_response(struct hda_bus *bus)
+static unsigned int azx_single_get_response(struct hda_bus *bus,
+					    unsigned int addr)
 {
 	struct azx *chip = bus->private_data;
-	return chip->rirb.res;
+	return chip->rirb.res[addr];
 }
 
 /*
@@ -765,13 +796,14 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
 }
 
 /* get a response */
-static unsigned int azx_get_response(struct hda_bus *bus)
+static unsigned int azx_get_response(struct hda_bus *bus,
+				     unsigned int addr)
 {
 	struct azx *chip = bus->private_data;
 	if (chip->single_cmd)
-		return azx_single_get_response(bus);
+		return azx_single_get_response(bus, addr);
 	else
-		return azx_rirb_get_response(bus);
+		return azx_rirb_get_response(bus, addr);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -1245,7 +1277,7 @@ static int probe_codec(struct azx *chip, int addr)
 
 	chip->probing = 1;
 	azx_send_cmd(chip->bus, cmd);
-	res = azx_get_response(chip->bus);
+	res = azx_get_response(chip->bus, addr);
 	chip->probing = 0;
 	if (res == -1)
 		return -EIO;
-- 
cgit v1.2.3


From a678cdee25a387c8fc3b2754974695412baf1d85 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:46:46 +0800
Subject: ALSA: hda: take cmd_mutex in probe_codec()

Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.

So cmd_mutex would be a safe addition to probe_codec().

Signed-off-by: Wu Fengguang <fengguang.wu@intel.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 19e67a1b602..ddabc827ac4 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1275,10 +1275,12 @@ static int probe_codec(struct azx *chip, int addr)
 		(AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID;
 	unsigned int res;
 
+	mutex_lock(&chip->bus->cmd_mutex);
 	chip->probing = 1;
 	azx_send_cmd(chip->bus, cmd);
 	res = azx_get_response(chip->bus, addr);
 	chip->probing = 0;
+	mutex_unlock(&chip->bus->cmd_mutex);
 	if (res == -1)
 		return -EIO;
 	snd_printdd(SFX "codec #%d probed OK\n", addr);
-- 
cgit v1.2.3


From cdb1fbf23181c133fb24f12ad14ccea7dc399599 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:47:41 +0800
Subject: ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io

Just for safety.  azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ddabc827ac4..b6e6314d006 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -513,6 +513,7 @@ static int azx_alloc_cmd_io(struct azx *chip)
 
 static void azx_init_cmd_io(struct azx *chip)
 {
+	spin_lock_irq(&chip->reg_lock);
 	/* CORB set up */
 	chip->corb.addr = chip->rb.addr;
 	chip->corb.buf = (u32 *)chip->rb.area;
@@ -544,13 +545,16 @@ static void azx_init_cmd_io(struct azx *chip)
 	azx_writew(chip, RINTCNT, 1);
 	/* enable rirb dma and response irq */
 	azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN);
+	spin_unlock_irq(&chip->reg_lock);
 }
 
 static void azx_free_cmd_io(struct azx *chip)
 {
+	spin_lock_irq(&chip->reg_lock);
 	/* disable ringbuffer DMAs */
 	azx_writeb(chip, RIRBCTL, 0);
 	azx_writeb(chip, CORBCTL, 0);
+	spin_unlock_irq(&chip->reg_lock);
 }
 
 static unsigned int azx_command_addr(u32 cmd)
-- 
cgit v1.2.3


From c32649feb4573b31f0a2bfdf35cbe1351256c764 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 18:48:12 +0800
Subject: ALSA: hda: read CORBWP inside reg_lock

This converts the last CORBWP access outside of reg_lock.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b6e6314d006..df6d9820efa 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -588,15 +588,17 @@ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
 	unsigned int addr = azx_command_addr(val);
 	unsigned int wp;
 
+	spin_lock_irq(&chip->reg_lock);
+
 	/* add command to corb */
 	wp = azx_readb(chip, CORBWP);
 	wp++;
 	wp %= ICH6_MAX_CORB_ENTRIES;
 
-	spin_lock_irq(&chip->reg_lock);
 	chip->rirb.cmds[addr]++;
 	chip->corb.buf[wp] = cpu_to_le32(val);
 	azx_writel(chip, CORBWP, wp);
+
 	spin_unlock_irq(&chip->reg_lock);
 
 	return 0;
-- 
cgit v1.2.3


From feb273404f15d86098cb0e81e46330d5c1e22b1b Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 19:17:14 +0800
Subject: ALSA: hda: remember last command for each codec

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index df6d9820efa..7c43f92de2f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -418,7 +418,7 @@ struct azx {
 	unsigned int probing :1; /* codec probing phase */
 
 	/* for debugging */
-	unsigned int last_cmd;	/* last issued command (to sync) */
+	unsigned int last_cmd[AZX_MAX_CODECS];
 
 	/* for pending irqs */
 	struct work_struct irq_pending_work;
@@ -668,7 +668,8 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 
 	if (chip->msi) {
 		snd_printk(KERN_WARNING SFX "No response from codec, "
-			   "disabling MSI: last cmd=0x%08x\n", chip->last_cmd);
+			   "disabling MSI: last cmd=0x%08x\n",
+			   chip->last_cmd[addr]);
 		free_irq(chip->irq, chip);
 		chip->irq = -1;
 		pci_disable_msi(chip->pci);
@@ -683,7 +684,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 	if (!chip->polling_mode) {
 		snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
 			   "switching to polling mode: last cmd=0x%08x\n",
-			   chip->last_cmd);
+			   chip->last_cmd[addr]);
 		chip->polling_mode = 1;
 		goto again;
 	}
@@ -707,7 +708,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 
 	snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
 		   "switching to single_cmd mode: last cmd=0x%08x\n",
-		   chip->last_cmd);
+		   chip->last_cmd[addr]);
 	chip->single_cmd = 1;
 	bus->response_reset = 0;
 	/* re-initialize CORB/RIRB */
@@ -794,7 +795,7 @@ static int azx_send_cmd(struct hda_bus *bus, unsigned int val)
 {
 	struct azx *chip = bus->private_data;
 
-	chip->last_cmd = val;
+	chip->last_cmd[azx_command_addr(val)] = val;
 	if (chip->single_cmd)
 		return azx_single_send_cmd(bus, val);
 	else
-- 
cgit v1.2.3


From e310bb0646e57a4f9182865115c5780931456c65 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sat, 1 Aug 2009 19:18:45 +0800
Subject: ALSA: hda: warn on spurious response

To help disclose hardware bugs.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7c43f92de2f..175f07a381b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -632,7 +632,11 @@ static void azx_update_rirb(struct azx *chip)
 			chip->rirb.res[addr] = res;
 			smp_wmb();
 			chip->rirb.cmds[addr]--;
-		}
+		} else
+			snd_printk(KERN_ERR SFX "spurious response %#x:%#x, "
+				   "last cmd=%#08x\n",
+				   res, res_ex,
+				   chip->last_cmd[addr]);
 	}
 }
 
-- 
cgit v1.2.3


From 84d3dc200fc8b878acf7c1840b238e6a0450e4d0 Mon Sep 17 00:00:00 2001
From: Chengu Wang <wangchengu@gmail.com>
Date: Thu, 30 Jul 2009 19:43:55 +0800
Subject: ALSA: hda: Correct EAPD for Dell Inspiron 1525

The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.

The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.

Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 5383d8cff88..456ef6ac12e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2266,7 +2266,7 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0227, "Dell Vostro 1400  ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022e, "Dell     ", STAC_DELL_BIOS),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_3ST),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0242, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0243, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x02ff, "Dell     ", STAC_DELL_BIOS),
@@ -5645,6 +5645,13 @@ static int patch_stac927x(struct hda_codec *codec)
 		/* GPIO2 High = Enable EAPD */
 		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04;
 		spec->gpio_data = 0x04;
+		switch (codec->subsystem_id) {
+		case 0x1028022f:
+			/* correct EAPD to be GPIO0 */
+			spec->eapd_mask = spec->gpio_mask = 0x01;
+			spec->gpio_dir = spec->gpio_data = 0x01;
+			break;
+		};
 		spec->dmic_nids = stac927x_dmic_nids;
 		spec->num_dmics = STAC927X_NUM_DMICS;
 
-- 
cgit v1.2.3


From 4b35d2ca2307d40ccb6b3b6f9cc25ac9178b2a6c Mon Sep 17 00:00:00 2001
From: Roel Kluin <roel.kluin@gmail.com>
Date: Sun, 2 Aug 2009 13:30:45 +0200
Subject: ALSA: hda - Read buffer overflow

Check whether index is within bounds before testing the element.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f6b4cbf1ead..51c44fdbc0f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -559,7 +559,7 @@ static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
 
 	/* Find enumerated value for current pinctl setting */
 	i = alc_pin_mode_min(dir);
-	while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir))
+	while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
 		i++;
 	*valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
 	return 0;
-- 
cgit v1.2.3


From 100d5eb36ba20dc0b99a17ea2b9800c567bfc3d1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Aug 2009 11:55:51 +0200
Subject: ALSA: hda - Add missing vmaster initialization for ALC269

Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.

Reference: Novell bnc#527361
	https://bugzilla.novell.com/show_bug.cgi?id=527361

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
---
 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 51c44fdbc0f..5cc927f4783 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13563,6 +13563,8 @@ static int patch_alc269(struct hda_codec *codec)
 		set_capture_mixer(spec);
 	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
+	spec->vmaster_nid = 0x02;
+
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC269_AUTO)
 		spec->init_hook = alc269_auto_init;
-- 
cgit v1.2.3


From dd704698f56c1451fc9c5daadcd6e3a089de2c40 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Aug 2009 08:45:11 +0200
Subject: ALSA: hda - Don't override ADC definitions for ALC codecs

ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition.  This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).

This patch avoids the overriding by adding the proper checks.

Reference: Novell bnc#529467
	https://bugzilla.novell.com/show_bug.cgi?id=529467

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 18 ++++++++++++------
 1 file changed, 12 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5cc927f4783..fea976793ae 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15579,9 +15579,12 @@ static int patch_alc861vd(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861vd_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861vd_pcm_digital_capture;
 
-	spec->adc_nids = alc861vd_adc_nids;
-	spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
-	spec->capsrc_nids = alc861vd_capsrc_nids;
+	if (!spec->adc_nids) {
+		spec->adc_nids = alc861vd_adc_nids;
+		spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
+	}
+	if (!spec->capsrc_nids)
+		spec->capsrc_nids = alc861vd_capsrc_nids;
 
 	set_capture_mixer(spec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
@@ -17498,9 +17501,12 @@ static int patch_alc662(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc662_pcm_digital_playback;
 	spec->stream_digital_capture = &alc662_pcm_digital_capture;
 
-	spec->adc_nids = alc662_adc_nids;
-	spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
-	spec->capsrc_nids = alc662_capsrc_nids;
+	if (!spec->adc_nids) {
+		spec->adc_nids = alc662_adc_nids;
+		spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
+	}
+	if (!spec->capsrc_nids)
+		spec->capsrc_nids = alc662_capsrc_nids;
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
-- 
cgit v1.2.3