From dfb12eeb0f04b37e5eb3858864d074af4ecd2ac7 Mon Sep 17 00:00:00 2001
From: Daniel T Chen
Date: Sun, 27 Dec 2009 15:48:40 -0500
Subject: ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2

BugLink: https://bugs.launchpad.net/ubuntu/+bug/498863

This mainboard needs ac97_codec=0.

Cc: stable@kernel.org
Tested-by: Apoorv Parle <apparle@yahoo.co.in>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/atiixp.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index d6752dff2a44..42b4fbbd8e2b 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = {
 MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
 
 static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+	SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0),
 	SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
 	{ } /* terminator */
 };
-- 
cgit v1.2.3


From 9980c6209ebc2a02eb3ca51a4fae1e17f645c868 Mon Sep 17 00:00:00 2001
From: Roel Kluin
Date: Sun, 27 Dec 2009 22:26:47 +0100
Subject: ALSA: test off by one in setsamplerate()

With `while (i++ < MAX_WRITE_RETRY)' i reaches MAX_WRITE_RETRY + 1 after the loop

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

(limited to 'sound')

diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index b5ca02e2038c..e66ef2b69b5d 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
 				 rptr.retwords[2] != M &&
 				 rptr.retwords[3] != N &&
 				 i++ < MAX_WRITE_RETRY);
-			if (i == MAX_WRITE_RETRY) {
+			if (i > MAX_WRITE_RETRY) {
 				snd_printdd("sent samplerate %d: %d failed\n",
 					    *intdec, rate);
 				return -EIO;
-- 
cgit v1.2.3


From 78b8d5d2ee280c463908fd75f3bdf246bcb6ac8d Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Mon, 28 Dec 2009 12:24:22 +0100
Subject: ALSA: usb-audio - Avoid Oops after disconnect

As the release of substreams may be done asynchronously from the
disconnection, close callback needs to check the shutdown flag before
actually accessing the usb interface.

Reference: Novell bnc#505027
	http://bugzilla.novell.com/show_bug.cgi?id=565027

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

(limited to 'sound')

diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 4963defee18a..9edef4684978 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -1936,7 +1936,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
 	struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
 	struct snd_usb_substream *subs = &as->substream[direction];
 
-	if (subs->interface >= 0) {
+	if (!as->chip->shutdown && subs->interface >= 0) {
 		usb_set_interface(subs->dev, subs->interface, 0);
 		subs->interface = -1;
 	}
-- 
cgit v1.2.3


From 5ee518ecbcb5934e284ea51a19a939c891f5f7ea Mon Sep 17 00:00:00 2001
From: Mark Brown
Date: Thu, 7 Jan 2010 16:29:20 +0000
Subject: ASoC: Fix WM8350 DSP mode B configuration

We need to set the LRCLK inversion bit to select DSP mode.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8350.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ebbf11b653a4..718ef912e758 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		iface |= 0x3 << 8;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x3 << 8;	/* lg not sure which mode */
+		iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
 		break;
 	default:
 		return -EINVAL;
-- 
cgit v1.2.3


From 9c0afc861a7228f718cb6a79fa7f9d46bf9ff300 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 12 Jan 2010 14:00:11 +0100
Subject: ALSA: hda - Fix ALC861-VD capture source mixer

The capture source or input source mixer element wasn't created properly
for ALC861-VD codec due to the wrong NID passed to
alc_auto_create_input_ctls().

References: Novell bnc#568305
	http://bugzilla.novell.com/show_bug.cgi?id=568305

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

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c7465053d6bb..e3caa78ccd54 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15493,7 +15493,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0);
 }
 
 
-- 
cgit v1.2.3


From 4dee8baa18d611b6dc854e1cc193550ff6f687be Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 13 Jan 2010 17:20:08 +0100
Subject: ALSA: hda - Fix Toshiba NB20x quirk entry

The alc664-mode4 model doesn't seem to fit with Toshiba NB205 correctly.
NB205 uses the pin 0x17 connected with the mixer 0x0f for the speaker
output, which isn't controlled by mode4 model at all.
Rather model=auto works fine as is on the latest driver, so let it back
again.

Tested-by: Nickolas Lloyd <ultrageek.lloyd@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')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e3caa78ccd54..bff60cea7777 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -17251,7 +17251,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
 		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
+	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
-- 
cgit v1.2.3


From a76221d47ef2b73ff16c0fef00a784026308ea02 Mon Sep 17 00:00:00 2001
From: Alex Murray
Date: Wed, 13 Jan 2010 23:15:03 +1030
Subject: ALSA: hda - Improved MacBook (Pro) 5,1 / 5,2 support

This patch adds support for automatically muting the speakers when headphones
are inserted, as well as relabelling the headphone widgets from the
non-standard "HP" to the standard "Headphone" for the mb5 model.

Signed-off-by: Alex Murray <murray.alex@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++--
 1 file changed, 26 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index bff60cea7777..11b989bacd3c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7094,8 +7094,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
-	HDA_BIND_MUTE   ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7496,6 +7496,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	/* Front Mic pin: input vref at 80% */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7680,6 +7681,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	/* Headphone insertion or removal. */
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc885_mb5_automute(codec);
+}
+
 static void alc885_imac91_automute(struct hda_codec *codec)
 {
  	unsigned int present;
@@ -9126,6 +9148,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc885_mb5_unsol_event,
+		.init_hook = alc885_mb5_automute,
 	},
 	[ALC885_MACPRO] = {
 		.mixers = { alc882_macpro_mixer },
-- 
cgit v1.2.3


From c7a8eb103248a110cdbe0530d8c5ce987f099eee Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Thu, 14 Jan 2010 12:39:02 +0100
Subject: ALSA: hda - Fix missing capture mixer for ALC861/660 codecs

The capture-related mixer elements are missing with ALC861/ALC660 codecs
when quirks are present, due to missing call of set_capture_mixer().

Reference: Novell bnc#567340
	http://bugzilla.novell.com/show_bug.cgi?id=567340

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')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 11b989bacd3c..abae1007cea2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14879,6 +14879,8 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+	if (!spec->cap_mixer)
+		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
 
 	spec->vmaster_nid = 0x03;
-- 
cgit v1.2.3


From c181a13a41ef32c9037393f4b42b780e1a36eb91 Mon Sep 17 00:00:00 2001
From: Thadeu Lima de Souza Cascardo
Date: Tue, 12 Jan 2010 20:20:39 -0200
Subject: ALSA: use subsys_initcall for sound core instead of module_init

This is needed for built-in drivers which are built before the sound directory,
like thinkpad_acpi.

Otherwise, registering a card fails.

Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/sound.c | 4 ++--
 sound/sound_core.c | 2 +-
 2 files changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02f6ca9..563d1967a0ad 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
 	unregister_chrdev(major, "alsa");
 }
 
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c909a31..7c2d677a2df5 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
 	class_destroy(sound_class);
 }
 
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
 module_exit(cleanup_soundcore);
 
 
-- 
cgit v1.2.3


From d38cce7046cfd0011f69d5dcf6a22525438154f6 Mon Sep 17 00:00:00 2001
From: Kunal Gangakhedkar
Date: Fri, 15 Jan 2010 21:01:47 +0530
Subject: ALSA: hda - Fix mute led GPIO on HP dv-series notebooks

On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type
"HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set
properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO)
either.

As per the documentation of find_mute_led_gpio(), these strings occur
in HP B-series systems - so, before scanning the SMBIOS strings, we need to
check if we're dealing with a B-series system.
Need to get confirmation from HP if this logic takes care of all the
systems. I'm trying to poke a friend there.

Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 61 +++++++++++++++++++++++++++++++-----------
 1 file changed, 45 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2291a8396817..799ba2570902 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	switch (codec->vendor_id) {
+	case 0x111d7608:
+		/* GPIO 0 */
+		spec->gpio_led = 0x01;
+		break;
+	case 0x111d7600:
+	case 0x111d7601:
+	case 0x111d7602:
+	case 0x111d7603:
+		/* GPIO 3 */
+		spec->gpio_led = 0x08;
+		break;
+	}
+}
+
 /*
  * This method searches for the mute LED GPIO configuration
  * provided as OEM string in SMBIOS. The format of that string
@@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
  *
  * So, HP B-series like systems may have HP_Mute_LED_0 (current models)
  * or  HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
  */
 static int find_mute_led_gpio(struct hda_codec *codec)
 {
@@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
 		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
 								NULL, dev))) {
 			if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
-			      &spec->gpio_led_polarity,
-			      &spec->gpio_led) == 2) {
+				  &spec->gpio_led_polarity,
+				  &spec->gpio_led) == 2) {
 				spec->gpio_led = 1 << spec->gpio_led;
 				return 1;
 			}
 			if (sscanf(dev->name, "HP_Mute_LED_%d",
-			      &spec->gpio_led_polarity) == 1) {
-				switch (codec->vendor_id) {
-				case 0x111d7608:
-					/* GPIO 0 */
-					spec->gpio_led = 0x01;
-					return 1;
-				case 0x111d7600:
-				case 0x111d7601:
-				case 0x111d7602:
-				case 0x111d7603:
-					/* GPIO 3 */
-					spec->gpio_led = 0x08;
-					return 1;
-				}
+				  &spec->gpio_led_polarity) == 1) {
+				set_hp_led_gpio(codec);
+				return 1;
 			}
 		}
+
+		/*
+		 * Fallback case - if we don't find the DMI strings,
+		 * we statically set the GPIO - if not a B-series system.
+		 */
+		if (!hp_blike_system(codec->subsystem_id)) {
+			set_hp_led_gpio(codec);
+			spec->gpio_led_polarity = 1;
+			return 1;
+		}
 	}
 	return 0;
 }
@@ -5548,6 +5575,8 @@ again:
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
+	snd_printdd("Found board config: %d\n", spec->board_config);
+
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
-- 
cgit v1.2.3


From eaa9b3a748539651f50e3a234c8854e1b42a839a Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Sun, 17 Jan 2010 13:09:33 +0100
Subject: ALSA: hda - Fix capture on Sony VAIO with single input

Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the
recording doesn't work with model=auto because ALC262 parser sets the
wrong cap NIDs to choose the route and the default route for the sole
input pin wasn't initialized properly.  This patch solves these issues.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++------
 1 file changed, 54 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index abae1007cea2..3f92def752fd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1230,6 +1230,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 			return; /* invalid entry */
 		}
 	}
+	if (!ext || !fixed)
+		return;
 	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
 		return; /* no unsol support */
 	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -4812,6 +4814,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
 	spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin;
+	int i;
+
+	/* search for the input pin; there must be only one */
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (spec->autocfg.input_pins[i]) {
+			pin = spec->autocfg.input_pins[i];
+			break;
+		}
+	}
+	if (!pin)
+		return;
+
+	/* set the default connection to that pin */
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t cap = spec->capsrc_nids ?
+			spec->capsrc_nids[i] : spec->adc_nids[i];
+		int idx;
+
+		idx = get_connection_index(codec, cap, pin);
+		if (idx < 0)
+			continue;
+		/* use only this ADC */
+		if (spec->capsrc_nids)
+			spec->capsrc_nids += i;
+		spec->adc_nids += i;
+		spec->num_adc_nids = 1;
+		/* select or unmute this route */
+		if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+			snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+						 HDA_AMP_MUTE, 0);
+		} else {
+			snd_hda_codec_write_cache(codec, cap, 0,
+					  AC_VERB_SET_CONNECT_SEL, idx);
+		}
+		return;
+	}
+}
+
 static void set_capture_mixer(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4824,14 +4869,15 @@ static void set_capture_mixer(struct hda_codec *codec)
 		  alc_capture_mixer3 },
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
-		int mux;
-		if (spec->auto_mic) {
-			mux = 0;
+		int mux = 0;
+		if (spec->auto_mic)
 			fixup_automic_adc(codec);
-		} else if (spec->input_mux && spec->input_mux->num_items > 1)
-			mux = 1;
-		else
-			mux = 0;
+		else if (spec->input_mux) {
+			if (spec->input_mux->num_items > 1)
+				mux = 1;
+			else if (spec->input_mux->num_items == 1)
+				fixup_single_adc(codec);
+		}
 		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
 	}
 }
@@ -11203,7 +11249,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 #define alc262_auto_create_input_ctls \
-	alc880_auto_create_input_ctls
+	alc882_auto_create_input_ctls
 
 /*
  * generic initialization of ADC, input mixers and output mixers
-- 
cgit v1.2.3


From 808c569f3609b37642d1e08373e3de829b99d0f8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Mon, 18 Jan 2010 14:16:24 +0100
Subject: ALSA: Remove warning message for invalid OSS minor ranges

When a card instance with a higher card number is registered, warning
messages are spewed eventually with stack traces due to the invalid minor
number for OSS device registration.  For example, thinkpad-acpi registers
the card number 29 as default, and you'll see always these messages.
This is rather confusing (and worries users), thus better to return
simply the error code.

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

(limited to 'sound')

diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe12264ff80..0c164e5e4322 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
 	default:
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+	if (minor < 0 || minor >= SNDRV_OSS_MINORS)
 		return -EINVAL;
 	return minor;
 }
-- 
cgit v1.2.3


From 4feabefe53eb3742f0b2773a43200d1686f3a288 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 19 Jan 2010 15:38:44 +0100
Subject: ALSA: hda - Fix parsing pin node 0x21 on ALC259

ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled
properly in alc268_new_analog_output().

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 3f92def752fd..79cdae324c5e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12541,6 +12541,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x21:
 		dac = 0x03;
 		break;
 	default:
-- 
cgit v1.2.3


From 3fb4a508b8e7957aa899f32cd6d9d462e102c7ca Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Tue, 19 Jan 2010 15:46:37 +0100
Subject: ALSA: hda - Turn on EAPD only if available for Realtek codecs

Some codecs disable widgets used for output pins and reserve as vendor-
spec widgets.  Thus we need to check the widget type and pin cap before
actually sending SET_EAPD verbs in the auto-configuration mode.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++------------
 1 file changed, 17 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 79cdae324c5e..6ae610c0111e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
 }
 
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+	if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+		return;
+	if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+				    on ? 2 : 0);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	unsigned int tmp;
@@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 	case ALC_INIT_DEFAULT:
 		switch (codec->vendor_id) {
 		case 0x10ec0260:
-			snd_hda_codec_write(codec, 0x0f, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x10, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x0f, 1);
+			set_eapd(codec, 0x10, 1);
 			break;
 		case 0x10ec0262:
 		case 0x10ec0267:
 		case 0x10ec0268:
 		case 0x10ec0269:
+		case 0x10ec0270:
 		case 0x10ec0272:
 		case 0x10ec0660:
 		case 0x10ec0662:
 		case 0x10ec0663:
 		case 0x10ec0862:
 		case 0x10ec0889:
-			snd_hda_codec_write(codec, 0x14, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x15, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x14, 1);
+			set_eapd(codec, 0x15, 1);
 			break;
 		}
 		switch (codec->vendor_id) {
@@ -1836,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void alc889_power_eapd(struct hda_codec *codec, int power)
 {
-	snd_hda_codec_write(codec, 0x14, 0,
-			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
-	snd_hda_codec_write(codec, 0x15, 0,
-			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
+	set_eapd(codec, 0x14, power);
+	set_eapd(codec, 0x15, power);
 }
 #endif
 
-- 
cgit v1.2.3


From dc99be47667c56046555e89e62f1ac17fa06329a Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 20 Jan 2010 08:35:06 +0100
Subject: ALSA: hda - Fix HP T5735 automute

This patch fixes the aut-mute setup on HP T5735 with ALC262 codec.
Instead of wrong amp, use pin control toggling for muting the speaker now.

Tested-by: Lee Trager <lee.trager@hp.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6ae610c0111e..d00e6d1da085 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10382,7 +10382,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
 static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11793,9 +11793,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hp_t5735_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_RP5700] = {
 		.mixers = { alc262_hp_rp5700_mixer },
-- 
cgit v1.2.3


From 973b8cb0ead3e0b1dd3ee7b2df52e4dff1ffc707 Mon Sep 17 00:00:00 2001
From: Łukasz Wojniłowicz
Date: Sun, 24 Jan 2010 14:12:37 +0100
Subject: ALSA: hda - add possibility to choose speakers configuration for
 4930g

Now one can choose speaker configuration in e.g. PulseAudio mixer

Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.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 d00e6d1da085..da34095c707f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9478,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
 		.channel_mode = alc883_3ST_6ch_modes,
 		.need_dac_fix = 1,
+		.const_channel_count = 6,
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
-- 
cgit v1.2.3


From 40aa7030e5213a43e9e0554fd7f95534ea310bf3 Mon Sep 17 00:00:00 2001
From: Guennadi Liakhovetski
Date: Fri, 22 Jan 2010 18:00:03 +0100
Subject: ASoC: fix a memory-leak in wm8903

Remember to free the temporary register-cache.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8903.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e3f2b0..3595bd57c4eb 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
-	u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+	u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
 				 GFP_KERNEL);
 
 	/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
 		for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
 			if (tmp_cache[i] != reg_cache[i])
 				snd_soc_write(codec, i, tmp_cache[i]);
+		kfree(tmp_cache);
 	} else {
 		dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
 	}
-- 
cgit v1.2.3


From 8ce28d6abff34886d3797b25324c940471b99164 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Wed, 27 Jan 2010 20:26:08 +0100
Subject: ALSA: hda - Add an ASUS mobo to MSI blacklist

Sid Boyce reported that his machine locks up without enable_msi=0 option.
This looks like another ASUS mobo with Nvidia combo.

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

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ec9c348336cc..565de38a3fc7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2332,6 +2332,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  */
 static struct snd_pci_quirk msi_black_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+	SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */
 	{}
 };
 
-- 
cgit v1.2.3


From 3e59aaa7ae9de49af8810102f12857860d5bd0ed Mon Sep 17 00:00:00 2001
From: Anuj Aggarwal
Date: Fri, 29 Jan 2010 13:58:55 +0530
Subject: ASoC: AIC23: Fixing writes to non-existing registers in resume
 function

Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23
register in resume function because of which register writes happen
on some non-existing registers.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/tlv320aic23.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb54774..da589d8664d0 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 	u16 reg;
 
 	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+	for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
-- 
cgit v1.2.3


From 5bbd4953a4fb5d8d597b4a53b8da97eee320b634 Mon Sep 17 00:00:00 2001
From: Anuj Aggarwal
Date: Fri, 29 Jan 2010 15:49:22 +0530
Subject: ASoC: AM3517: ASoC driver not getting compiled

Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes
CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the
Makefile. Whereas the config option defined in Kconfig is
SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517
was not getting compiled.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/Makefile | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c523f4..19283e5edfbf 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
-- 
cgit v1.2.3


From 1eb6dc7dabcb4aa762d96f4f6978f3ef86321d68 Mon Sep 17 00:00:00 2001
From: Maxim Levitsky
Date: Thu, 4 Feb 2010 22:21:47 +0200
Subject: ALSA: hda - Delay switching to polling mode if an interrupt was
 missing

My sound codec seems sometimes (very rarely) to omit interrupts (ALC268)
However, interrupt mode still works.
Thus if we get timeout, poll the codec once.

If we get 3 such polls in a row, then switch to polling mode.

This patch is maybe an bandaid, but this might be a workaround for hardware bug.

Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 19 +++++++++++++++++--
 1 file changed, 17 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 565de38a3fc7..d853e2c33bb7 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -426,6 +426,7 @@ struct azx {
 
 	/* flags */
 	int position_fix;
+	int poll_count;
 	unsigned int running :1;
 	unsigned int initialized :1;
 	unsigned int single_cmd :1;
@@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = {
 #define get_azx_dev(substream) (substream->runtime->private_data)
 
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
-
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
 /*
  * Interface for HD codec
  */
@@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
+	int do_poll = 0;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
 	for (;;) {
-		if (chip->polling_mode) {
+		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
 			spin_unlock_irq(&chip->reg_lock);
@@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		if (!chip->rirb.cmds[addr]) {
 			smp_rmb();
 			bus->rirb_error = 0;
+
+			if (!do_poll)
+				chip->poll_count = 0;
 			return chip->rirb.res[addr]; /* the last value */
 		}
 		if (time_after(jiffies, timeout))
@@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 	}
 
+	if (!chip->polling_mode && chip->poll_count < 2) {
+		snd_printdd(SFX "azx_get_response timeout, "
+			   "polling the codec once: last cmd=0x%08x\n",
+			   chip->last_cmd[addr]);
+		do_poll = 1;
+		chip->poll_count++;
+		goto again;
+	}
+
+
 	if (!chip->polling_mode) {
 		snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
 			   "switching to polling mode: last cmd=0x%08x\n",
-- 
cgit v1.2.3


From 9492837a6f54b069e13e40e3c89898bb8837a386 Mon Sep 17 00:00:00 2001
From: Maxim Levitsky
Date: Thu, 4 Feb 2010 22:26:37 +0200
Subject: ALSA: cosmetic: make hda intel interrupt name consistent with others

This renames the interrupt name in /proc/interrupt.
HDA Intel -> hda_intel

This also eliminates space from the name, probably helping some
parsers.
Don't think anybody depends on this name in userspace

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

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index d853e2c33bb7..b8faa6dc5abe 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2058,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
 {
 	if (request_irq(chip->pci->irq, azx_interrupt,
 			chip->msi ? 0 : IRQF_SHARED,
-			"HDA Intel", chip)) {
+			"hda_intel", chip)) {
 		printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
 		       "disabling device\n", chip->pci->irq);
 		if (do_disconnect)
-- 
cgit v1.2.3


From 3b9447fb7fa1829731290e64ef928d4f6461310a Mon Sep 17 00:00:00 2001
From: Grazvydas Ignotas
Date: Fri, 5 Feb 2010 00:55:33 +0200
Subject: ASoC: pandora: Add APLL supply to fix audio output

Pandora's external DAC is using 256*Fs output from the TWL4030
codec, and TWL4030 needs to have APLL enabled for it's 256*Fs
output to function.

Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/omap3pandora.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 71b2c161158d..68980c19a3bc 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"PCM DAC", NULL, "APLL Enable"},
 	{"Headphone Amplifier", NULL, "PCM DAC"},
 	{"Line Out", NULL, "PCM DAC"},
 	{"Headphone Jack", NULL, "Headphone Amplifier"},
-- 
cgit v1.2.3


From fed08d036f2aabd8d0c684439de37f8ebec2bbc2 Mon Sep 17 00:00:00 2001
From: Jody Bruchon
Date: Sat, 6 Feb 2010 10:46:26 -0500
Subject: ALSA: hda-intel: Avoid divide by zero crash

On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.

Signed-off-by: Jody Bruchon <jody@nctritech.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index b8faa6dc5abe..e767c3f395ab 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1893,6 +1893,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 
 	if (!bdl_pos_adj[chip->dev_index])
 		return 1; /* no delayed ack */
+	if (azx_dev->period_bytes == 0) {
+		printk(KERN_WARNING
+		       "hda-intel: Divide by zero was avoided "
+		       "in azx_dev->period_bytes.\n");
+		return 0;
+	}
 	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		return 0; /* NG - it's below the period boundary */
 	return 1; /* OK, it's fine */
-- 
cgit v1.2.3


From d6d8bf549393484e906913f02fa3c9518a2819b6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai
Date: Fri, 12 Feb 2010 18:17:06 +0100
Subject: ALSA: hda - use WARN_ON_ONCE() for zero-division detection

Replace the zero-division warning message with WARN_ON_ONCE() per the
advice by Linus.  This shouldn't happen, but if it happens, it's
possible that the bug happens often due to buggy IRQs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e767c3f395ab..3600e9cc9bc6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1893,12 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 
 	if (!bdl_pos_adj[chip->dev_index])
 		return 1; /* no delayed ack */
-	if (azx_dev->period_bytes == 0) {
-		printk(KERN_WARNING
-		       "hda-intel: Divide by zero was avoided "
-		       "in azx_dev->period_bytes.\n");
-		return 0;
-	}
+	if (WARN_ONCE(!azx_dev->period_bytes,
+		      "hda-intel: zero azx_dev->period_bytes"))
+		return 0; /* this shouldn't happen! */
 	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		return 0; /* NG - it's below the period boundary */
 	return 1; /* OK, it's fine */
-- 
cgit v1.2.3


From 3be522a9514f58e0596db34898a514df206cadc5 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela
Date: Tue, 16 Feb 2010 11:55:43 +0100
Subject: ALSA: pcm core - fix fifo_size channels interval check

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
---
 sound/core/pcm_native.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 25b0641e6b8c..f7e1c9f0d3ed 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -315,10 +315,10 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
 	if (!params->info)
 		params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES;
 	if (!params->fifo_size) {
-		if (snd_mask_min(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT]) ==
-		    snd_mask_max(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT]) &&
-                    snd_mask_min(&params->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) ==
-                    snd_mask_max(&params->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) {
+		m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+		i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+		if (snd_mask_min(m) == snd_mask_max(m) &&
+                    snd_interval_min(i) == snd_interval_max(i)) {
 			changed = substream->ops->ioctl(substream,
 					SNDRV_PCM_IOCTL1_FIFO_SIZE, params);
 			if (changed < 0)
-- 
cgit v1.2.3


From b721e68bdc5b39c51bf6a1469f8d3663fbe03243 Mon Sep 17 00:00:00 2001
From: Giuliano Pochini
Date: Wed, 17 Feb 2010 00:57:44 +0100
Subject: ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50

This patch fixes a division by zero error in the irq handler.

There is a small window between the hw_params() callback and when
runtime->frame_bits is set by ALSA middle layer. When another substream is
already running, if an interrupt is delivered during that window the irq
handler calls pcm_pointer() which does a division by zero. The patch below
makes the irq handler skip substreams that are initialized but not started
yet. Cc to Clemens Ladisch because he proposed an alternate fix.

For more information, please read the original thread in the linux-kernel
mailing list: http://lkml.org/lkml/2010/2/2/187

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 1305f7ca02c3..641d7f07392c 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1821,7 +1821,9 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 	/* The hardware doesn't tell us which substream caused the irq,
 	thus we have to check all running substreams. */
 	for (ss = 0; ss < DSP_MAXPIPES; ss++) {
-		if ((substream = chip->substream[ss])) {
+		substream = chip->substream[ss];
+		if (substream && ((struct audiopipe *)substream->runtime->
+				private_data)->state == PIPE_STATE_STARTED) {
 			period = pcm_pointer(substream) /
 				substream->runtime->period_size;
 			if (period != chip->last_period[ss]) {
-- 
cgit v1.2.3