From 92fd918c2416404c2ec09829b25243b9a785dc9b Mon Sep 17 00:00:00 2001
From: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Date: Fri, 30 Mar 2012 09:52:25 +1300
Subject: ALSA: asihpi - fix return value of hpios_locked_mem_alloc()

Make this function consistent with others in this module by
returning 1 for error, instead of -ENOMEM
(reverts function signature change from a938fb1e)

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/asihpi/hpi_internal.h |  4 ++--
 sound/pci/asihpi/hpios.c        | 10 +++++-----
 2 files changed, 7 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index 8c63200cf339..bc86cb726d79 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned.
 If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and
 HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle.
 */
-int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
+u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
 							   /**< memory handle */
 	u32 size, /**< Size in bytes to allocate */
 	struct pci_dev *p_os_reference
diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 87f4385fe8c7..5ef4fe964366 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -39,11 +39,11 @@ void hpios_delay_micro_seconds(u32 num_micro_sec)
 
 }
 
-/** Allocated an area of locked memory for bus master DMA operations.
+/** Allocate an area of locked memory for bus master DMA operations.
 
-On error, return -ENOMEM, and *pMemArea.size = 0
+If allocation fails, return 1, and *pMemArea.size = 0
 */
-int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
+u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 	struct pci_dev *pdev)
 {
 	/*?? any benefit in using managed dmam_alloc_coherent? */
@@ -62,7 +62,7 @@ int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
 		HPI_DEBUG_LOG(WARNING,
 			"failed to allocate %d bytes locked memory\n", size);
 		p_mem_area->size = 0;
-		return -ENOMEM;
+		return 1;
 	}
 }
 
-- 
cgit v1.2.3


From 4f32456e5ed4852abc9b555c887dfb3481ea9cab Mon Sep 17 00:00:00 2001
From: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Date: Fri, 6 Apr 2012 15:34:15 +0200
Subject: ALSA: hda - Fix proc output for ADC amp values of CX20549

The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.h      |  3 +++
 sound/pci/hda/hda_proc.c       | 13 ++++++++++---
 sound/pci/hda/patch_conexant.c |  8 ++++----
 3 files changed, 17 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 9a9f372e1be4..56b4f74c0b13 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -851,6 +851,9 @@ struct hda_codec {
 	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
 					    * (e.g. Conexant codecs)
 					    */
+	unsigned int single_adc_amp:1; /* adc in-amp takes no index
+					* (e.g. CX20549 codec)
+					*/
 	unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */
 	unsigned int pins_shutup:1;	/* pins are shut up */
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 254ab5204603..e59e2f059b6e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -651,9 +651,16 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_iprintf(buffer, "  Amp-In caps: ");
 			print_amp_caps(buffer, codec, nid, HDA_INPUT);
 			snd_iprintf(buffer, "  Amp-In vals: ");
-			print_amp_vals(buffer, codec, nid, HDA_INPUT,
-				       wid_caps & AC_WCAP_STEREO,
-				       wid_type == AC_WID_PIN ? 1 : conn_len);
+			if (wid_type == AC_WID_PIN ||
+			    (codec->single_adc_amp &&
+			     wid_type == AC_WID_AUD_IN))
+				print_amp_vals(buffer, codec, nid, HDA_INPUT,
+					       wid_caps & AC_WCAP_STEREO,
+					       1);
+			else
+				print_amp_vals(buffer, codec, nid, HDA_INPUT,
+					       wid_caps & AC_WCAP_STEREO,
+					       conn_len);
 		}
 		if (wid_caps & AC_WCAP_OUT_AMP) {
 			snd_iprintf(buffer, "  Amp-Out caps: ");
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e6eafb18c8f5..368617abab4c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -141,7 +141,6 @@ struct conexant_spec {
 	unsigned int hp_laptop:1;
 	unsigned int asus:1;
 	unsigned int pin_eapd_ctrls:1;
-	unsigned int single_adc_amp:1;
 
 	unsigned int adc_switching:1;
 
@@ -1111,6 +1110,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 		return -ENOMEM;
 	codec->spec = spec;
 	codec->pin_amp_workaround = 1;
+	codec->single_adc_amp = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -4220,7 +4220,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
 		int idx = get_input_connection(codec, adc_nid, nid);
 		if (idx < 0)
 			continue;
-		if (spec->single_adc_amp)
+		if (codec->single_adc_amp)
 			idx = 0;
 		return cx_auto_add_volume_idx(codec, label, pfx,
 					      cidx, adc_nid, HDA_INPUT, idx);
@@ -4275,7 +4275,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 		if (cidx < 0)
 			continue;
 		input_conn[i] = spec->imux_info[i].adc;
-		if (!spec->single_adc_amp)
+		if (!codec->single_adc_amp)
 			input_conn[i] |= cidx << 8;
 		if (i > 0 && input_conn[i] != input_conn[0])
 			multi_connection = 1;
@@ -4470,7 +4470,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	switch (codec->vendor_id) {
 	case 0x14f15045:
-		spec->single_adc_amp = 1;
+		codec->single_adc_amp = 1;
 		break;
 	case 0x14f15051:
 		add_cx5051_fake_mutes(codec);
-- 
cgit v1.2.3


From 3edbbb9ec5621478dc3c3b1c66ecb7d177b35c20 Mon Sep 17 00:00:00 2001
From: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Date: Fri, 6 Apr 2012 15:34:16 +0200
Subject: ALSA: hda - Rename capture sources of CX20549 to match common
 conventions

This includes renaming "Line In" to line, also in the mixer settings.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 26 +++++++++++++-------------
 1 file changed, 13 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 368617abab4c..c0a3a17edd86 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -686,27 +686,27 @@ static const struct hda_channel_mode cxt5045_modes[1] = {
 static const struct hda_input_mux cxt5045_capture_source = {
 	.num_items = 2,
 	.items = {
-		{ "IntMic", 0x1 },
-		{ "ExtMic", 0x2 },
+		{ "Internal Mic", 0x1 },
+		{ "Mic",          0x2 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_benq = {
 	.num_items = 5,
 	.items = {
-		{ "IntMic", 0x1 },
-		{ "ExtMic", 0x2 },
-		{ "LineIn", 0x3 },
-		{ "CD",     0x4 },
-		{ "Mixer",  0x0 },
+		{ "CD",           0x4 },
+		{ "Internal Mic", 0x1 },
+		{ "Mic",          0x2 },
+		{ "Line",         0x3 },
+		{ "Mixer",        0x0 },
 	}
 };
 
 static const struct hda_input_mux cxt5045_capture_source_hp530 = {
 	.num_items = 2,
 	.items = {
-		{ "ExtMic", 0x1 },
-		{ "IntMic", 0x2 },
+		{ "Mic",          0x1 },
+		{ "Internal Mic", 0x2 },
 	}
 };
 
@@ -826,10 +826,10 @@ static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Capture Volume", 0x1a, 0x03, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Capture Switch", 0x1a, 0x03, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
 
 	HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
-- 
cgit v1.2.3


From cbf2d28e83d47792bd7af000017042dbc59f5df6 Mon Sep 17 00:00:00 2001
From: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Date: Fri, 6 Apr 2012 15:34:17 +0200
Subject: ALSA: hda - fix record volume controls of CX20459 ("Venice")

The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 31 ++++++-------------------------
 1 file changed, 6 insertions(+), 25 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index c0a3a17edd86..4b51c8f2fda2 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -797,10 +797,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static const struct snd_kcontrol_new cxt5045_mixers[] = {
-	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
@@ -821,27 +819,18 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
-	HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Line Capture Volume", 0x1a, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Capture Switch", 0x1a, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
-
 	{}
 };
 
 static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
@@ -977,16 +966,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 		.put = conexant_mux_enum_put,
 	},
 	/* Audio input controls */
-	HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
-	HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
 	{ } /* end */
 };
 
-- 
cgit v1.2.3


From e6e03daecd2c82437b550ad1a62052c22fdb2b5b Mon Sep 17 00:00:00 2001
From: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Date: Fri, 6 Apr 2012 15:34:18 +0200
Subject: ALSA: hda - Remove CD control from model=benq for CX20549

The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.

Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4b51c8f2fda2..4b365488c58b 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -692,9 +692,8 @@ static const struct hda_input_mux cxt5045_capture_source = {
 };
 
 static const struct hda_input_mux cxt5045_capture_source_benq = {
-	.num_items = 5,
+	.num_items = 4,
 	.items = {
-		{ "CD",           0x4 },
 		{ "Internal Mic", 0x1 },
 		{ "Mic",          0x2 },
 		{ "Line",         0x3 },
@@ -819,9 +818,6 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
-
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
 
-- 
cgit v1.2.3


From 51969d62c3b26e887dae734de421b320a296ac58 Mon Sep 17 00:00:00 2001
From: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Date: Fri, 6 Apr 2012 15:34:19 +0200
Subject: ALSA: hda - CX20549 doesn't need pin_amp_workaround.

CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:

  $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
  nid = 0x10, verb = 0x300, param = 0xb126
  value = 0x0
  $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
  nid = 0x10, verb = 0xb00, param = 0x8001
  value = 0x0

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4b365488c58b..84337e63fadf 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1086,7 +1086,6 @@ static int patch_cxt5045(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
-	codec->pin_amp_workaround = 1;
 	codec->single_adc_amp = 1;
 
 	spec->multiout.max_channels = 2;
@@ -4443,7 +4442,6 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
-	codec->pin_amp_workaround = 1;
 
 	switch (codec->vendor_id) {
 	case 0x14f15045:
@@ -4451,7 +4449,10 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		break;
 	case 0x14f15051:
 		add_cx5051_fake_mutes(codec);
+		codec->pin_amp_workaround = 1;
 		break;
+	default:
+		codec->pin_amp_workaround = 1;
 	}
 
 	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
-- 
cgit v1.2.3


From 250f32747e62cb415b85083e247184188f24e566 Mon Sep 17 00:00:00 2001
From: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Date: Fri, 6 Apr 2012 15:34:20 +0200
Subject: ALSA: hda - clean up CX20549 test mixer setup

name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.

remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1

really select MIC1, not Mixer out for recording

"Mixer out" for recording is not a "pin", adjust comment

Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 38 +++++++++++++++++---------------------
 1 file changed, 17 insertions(+), 21 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 84337e63fadf..3848711d89f7 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -930,10 +930,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 	/* Output controls */
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
 	
 	/* Modes for retasking pin widgets */
 	CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
@@ -944,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 
 	/* Loopback mixer controls */
 
-	HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT),
-	HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT),
+	HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Input Source",
@@ -985,10 +985,6 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
 
-	/* Start with output sum widgets muted and their output gains at min */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
 	/* Unmute retasking pin widget output buffers since the default
 	 * state appears to be output.  As the pin mode is changed by the
 	 * user the pin mode control will take care of enabling the pin's
@@ -1003,11 +999,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
 	/* Set ADC connection select to match default mixer setting (mic1
 	 * pin)
 	 */
-	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x01},
 
 	/* Mute all inputs to mixer widget (even unconnected ones) */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */
-- 
cgit v1.2.3


From d3a92d624806a7964ca3122f917ff2ba69e4cdd8 Mon Sep 17 00:00:00 2001
From: Hans Verkuil <hverkuil@xs4all.nl>
Date: Sun, 1 Apr 2012 15:24:48 -0300
Subject: [media] Drivers/media/radio: Fix build error

On Sunday, April 01, 2012 21:09:34 Tracey Dent wrote:
> radio-maxiradio depends on SND_FM801_TEA575X_BOOL to build or will
> result in an build error such as:
>
> Kernel: arch/x86/boot/bzImage is ready  (#1)
> ERROR: "snd_tea575x_init" [drivers/media/radio/radio-maxiradio.ko] undefined!
> ERROR: "snd_tea575x_exit" [drivers/media/radio/radio-maxiradio.ko] undefined!
> WARNING: modpost: Found 6 section mismatch(es).
> To see full details build your kernel with:
> 'make CONFIG_DEBUG_SECTION_MISMATCH=y'
> make[1]: *** [__modpost] Error 1
> make: *** [modules] Error 2
>
> Select CONFIG_SND_TEA575X to fixes problem and enable
> the driver to be built as desired.
>
> v2:
> instead of selecting CONFIG_SND_FM801_TEA575X_BOOL, select
> CONFIG_SND_TEA575X, which in turns selects CONFIG_SND_FM801_TEA575X_BOOL
> and any other dependencies for it to build.

No, this is the correct patch:

RADIO_MAXIRADIO should be treated just like RADIO_SF16FMR2, I just didn't
realize at the time that it had to be added as a SND_TEA575X dependency.

Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Tested-by: Shea Levy <shea@shealevy.com>
Acked-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
---
 sound/pci/Kconfig | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 88168044375f..5ca0939e4223 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -2,8 +2,8 @@
 
 config SND_TEA575X
 	tristate
-	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2
-	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2
+	depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO
+	default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO
 
 menuconfig SND_PCI
 	bool "PCI sound devices"
-- 
cgit v1.2.3


From fae3d88a5c56c3f836e95c4516da883a48612437 Mon Sep 17 00:00:00 2001
From: Fengguang Wu <fengguang.wu@intel.com>
Date: Tue, 10 Apr 2012 17:00:35 +0800
Subject: ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Also remove two warnings when CONFIG_SND_DEBUG is not set:

sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/core.h       | 10 ++++++++++
 sound/pci/hda/hda_eld.c    |  6 +++---
 sound/pci/hda/patch_hdmi.c |  9 ++++-----
 3 files changed, 17 insertions(+), 8 deletions(-)

(limited to 'sound/pci')

diff --git a/include/sound/core.h b/include/sound/core.h
index b6e0f57d451d..bc056687f647 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -325,6 +325,13 @@ void release_and_free_resource(struct resource *res);
 
 /* --- */
 
+/* sound printk debug levels */
+enum {
+	SND_PR_ALWAYS,
+	SND_PR_DEBUG,
+	SND_PR_VERBOSE,
+};
+
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
 __printf(4, 5)
 void __snd_printk(unsigned int level, const char *file, int line,
@@ -354,6 +361,8 @@ void __snd_printk(unsigned int level, const char *file, int line,
  */
 #define snd_printd(fmt, args...) \
 	__snd_printk(1, __FILE__, __LINE__, fmt, ##args)
+#define _snd_printd(level, fmt, args...) \
+	__snd_printk(level, __FILE__, __LINE__, fmt, ##args)
 
 /**
  * snd_BUG - give a BUG warning message and stack trace
@@ -383,6 +392,7 @@ void __snd_printk(unsigned int level, const char *file, int line,
 #else /* !CONFIG_SND_DEBUG */
 
 #define snd_printd(fmt, args...)	do { } while (0)
+#define _snd_printd(level, fmt, args...) do { } while (0)
 #define snd_BUG()			do { } while (0)
 static inline int __snd_bug_on(int cond)
 {
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b58b4b1687fa..4c054f4486b9 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
 	else
 		buf2[0] = '\0';
 
-	printk(KERN_INFO "HDMI: supports coding type %s:"
+	_snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:"
 			" channels = %d, rates =%s%s\n",
 			cea_audio_coding_type_names[a->format],
 			a->channels,
@@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e)
 {
 	int i;
 
-	printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
+	_snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n",
 			e->monitor_name,
 			eld_connection_type_names[e->conn_type]);
 
 	if (e->spk_alloc) {
 		char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
 		snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
-		printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
+		_snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf);
 	}
 
 	for (i = 0; i < e->sad_count; i++)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 540cd13f7f15..83f345f3c961 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -757,8 +757,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	struct hdmi_spec *spec = codec->spec;
 	int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
 	int pin_nid;
-	int pd = !!(res & AC_UNSOL_RES_PD);
-	int eldv = !!(res & AC_UNSOL_RES_ELDV);
 	int pin_idx;
 	struct hda_jack_tbl *jack;
 
@@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	pin_nid = jack->nid;
 	jack->jack_dirty = 1;
 
-	printk(KERN_INFO
+	_snd_printd(SND_PR_VERBOSE,
 		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid, pd, eldv);
+		codec->addr, pin_nid,
+		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
 	if (pin_idx < 0)
@@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
 	if (eld->monitor_present)
 		eld_valid	= !!(present & AC_PINSENSE_ELDV);
 
-	printk(KERN_INFO
+	_snd_printd(SND_PR_VERBOSE,
 		"HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
 		codec->addr, pin_nid, eld->monitor_present, eld_valid);
 
-- 
cgit v1.2.3


From 912093bc7c08f59e97faed2c0269e1e5429dcd58 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 11 Apr 2012 14:03:41 +0200
Subject: ALSA: hda/realtek - Add a few ALC882 model strings back

Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/HD-Audio-Models.txt |  4 +++-
 sound/pci/hda/patch_realtek.c                | 10 +++++++++-
 2 files changed, 12 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index d97d992ced14..03f7897c6414 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -43,7 +43,9 @@ ALC680
 
 ALC882/883/885/888/889
 ======================
-  N/A
+  acer-aspire-4930g	Acer Aspire 4930G/5930G/6530G/6930G/7730G
+  acer-aspire-8930g	Acer Aspire 8330G/6935G
+  acer-aspire		Acer Aspire others
 
 ALC861/660
 ==========
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9917e55d6f11..e7b2b839a539 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5399,6 +5399,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	{}
 };
 
+static const struct alc_model_fixup alc882_fixup_models[] = {
+	{.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"},
+	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
+	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
+	{}
+};
+
 /*
  * BIOS auto configuration
  */
@@ -5439,7 +5446,8 @@ static int patch_alc882(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
-	alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
+	alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
+		       alc882_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
-- 
cgit v1.2.3


From 038d4fef376bc494d4f11072d2ab248414b7d568 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 11 Apr 2012 17:18:12 +0200
Subject: ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co

Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 8 ++++++--
 1 file changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e7b2b839a539..4eec2150312b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5269,7 +5269,9 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ 0x16, 0x99130111 }, /* CLFE speaker */
 			{ 0x17, 0x99130112 }, /* surround speaker */
 			{ }
-		}
+		},
+		.chained = true,
+		.chain_id = ALC882_FIXUP_GPIO1,
 	},
 	[ALC882_FIXUP_ACER_ASPIRE_8930G] = {
 		.type = ALC_FIXUP_PINS,
@@ -5312,7 +5314,9 @@ static const struct alc_fixup alc882_fixups[] = {
 			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
 			{ 0x20, AC_VERB_SET_PROC_COEF, 0x3050 },
 			{ }
-		}
+		},
+		.chained = true,
+		.chain_id = ALC882_FIXUP_GPIO1,
 	},
 	[ALC885_FIXUP_MACPRO_GPIO] = {
 		.type = ALC_FIXUP_FUNC,
-- 
cgit v1.2.3


From fe97da1f7001ca0f572358462606eb3d1bde3f23 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 12 Apr 2012 08:00:19 +0200
Subject: ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G

It's compatible with 8930G.
Using the same fixup gives the proper 5.1 sound back.

Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4eec2150312b..d25a6f90a37b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5363,6 +5363,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
 	SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
+	SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
 	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
-- 
cgit v1.2.3


From 29ebe40284c75a5888c601872059fca7e258528d Mon Sep 17 00:00:00 2001
From: Josh Boyer <jwboyer@redhat.com>
Date: Thu, 12 Apr 2012 13:55:36 -0400
Subject: ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines

A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine.  Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option.  All Mac machines are now explicitly
handled with a quirk and the auto-parser.  This adds a quirk for the device
found on the Mac Pro 5,1 machines.

This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559

[sorted the new entry in the ID number order by tiwai]

Reported-by: Gabriel Somlo <somlo@cmu.edu>
Signed-off-by: Josh Boyer <jwboyer@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d25a6f90a37b..8f4a48463fad 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5389,6 +5389,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF),
+	SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO),
 	SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF),
 	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF),
-- 
cgit v1.2.3


From 7d7eb9ea314e992413620610b4d09c9cd5fa8959 Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Thu, 12 Apr 2012 22:11:25 +0200
Subject: ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).

In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.

While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 16 +++++++++-------
 1 file changed, 9 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8f4a48463fad..2508f8109f11 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3398,8 +3398,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 	for (;;) {
 		badness = fill_and_eval_dacs(codec, fill_hardwired,
 					     fill_mio_first);
-		if (badness < 0)
+		if (badness < 0) {
+			kfree(best_cfg);
 			return badness;
+		}
 		debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n",
 			      cfg->line_out_type, fill_hardwired, fill_mio_first,
 			      badness);
@@ -3434,7 +3436,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
 			fill_hardwired = true;
 			continue;
-		} 
+		}
 		if (cfg->hp_outs > 0 &&
 		    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
 			cfg->speaker_outs = cfg->line_outs;
@@ -3448,7 +3450,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 			cfg->line_out_type = AUTO_PIN_HP_OUT;
 			fill_hardwired = true;
 			continue;
-		} 
+		}
 		break;
 	}
 
@@ -4423,7 +4425,7 @@ static int alc_parse_auto_config(struct hda_codec *codec,
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	static const hda_nid_t alc880_ignore[] = { 0x1d, 0 };
-	static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; 
+	static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 };
 	return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids);
 }
 
@@ -6093,7 +6095,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	 * Basically the device should work as is without the fixup table.
 	 * If BIOS doesn't give a proper info, enable the corresponding
 	 * fixup entry.
-	 */ 
+	 */
 	SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
 		      ALC269_FIXUP_AMIC),
 	SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
@@ -6310,7 +6312,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec,
 {
 	if (action == ALC_FIXUP_ACT_PRE_PROBE)
 		codec->no_jack_detect = 1;
-}	
+}
 
 static const struct alc_fixup alc861_fixups[] = {
 	[ALC861_FIXUP_FSC_AMILO_PI1505] = {
@@ -6728,7 +6730,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	 * Basically the device should work as is without the fixup table.
 	 * If BIOS doesn't give a proper info, enable the corresponding
 	 * fixup entry.
-	 */ 
+	 */
 	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1),
-- 
cgit v1.2.3


From 118cb4a408e1c4021ac85d6c05da66bb6f57e556 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Apr 2012 07:33:27 +0200
Subject: ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1

Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken.  There are two problems behind it:

- This machine doesn't like the default COEF setup for ALC260 we take
  now as default

- BIOS doesn't set the pins correctly at all; especially the machine
  uses only the pin 0x0f for both headphone and speaker

This patch adds the fixup as a workaround for these issues.

Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 49 +++++++++++++++++++++++++++++++++++++++----
 1 file changed, 45 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2508f8109f11..e65e35433055 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1445,6 +1445,13 @@ enum {
 	ALC_FIXUP_ACT_BUILD,
 };
 
+static void alc_apply_pincfgs(struct hda_codec *codec,
+			      const struct alc_pincfg *cfg)
+{
+	for (; cfg->nid; cfg++)
+		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+
 static void alc_apply_fixup(struct hda_codec *codec, int action)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1478,9 +1485,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 			snd_printdd(KERN_INFO "hda_codec: %s: "
 				    "Apply pincfg for %s\n",
 				    codec->chip_name, modelname);
-			for (; cfg->nid; cfg++)
-				snd_hda_codec_set_pincfg(codec, cfg->nid,
-							 cfg->val);
+			alc_apply_pincfgs(codec, cfg);
 			break;
 		case ALC_FIXUP_VERBS:
 			if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
@@ -4861,6 +4866,7 @@ enum {
 	ALC260_FIXUP_GPIO1_TOGGLE,
 	ALC260_FIXUP_REPLACER,
 	ALC260_FIXUP_HP_B1900,
+	ALC260_FIXUP_KN1,
 };
 
 static void alc260_gpio1_automute(struct hda_codec *codec)
@@ -4888,6 +4894,36 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
 	}
 }
 
+static void alc260_fixup_kn1(struct hda_codec *codec,
+			     const struct alc_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	static const struct alc_pincfg pincfgs[] = {
+		{ 0x0f, 0x02214000 }, /* HP/speaker */
+		{ 0x12, 0x90a60160 }, /* int mic */
+		{ 0x13, 0x02a19000 }, /* ext mic */
+		{ 0x18, 0x01446000 }, /* SPDIF out */
+		/* disable bogus I/O pins */
+		{ 0x10, 0x411111f0 },
+		{ 0x11, 0x411111f0 },
+		{ 0x14, 0x411111f0 },
+		{ 0x15, 0x411111f0 },
+		{ 0x16, 0x411111f0 },
+		{ 0x17, 0x411111f0 },
+		{ 0x19, 0x411111f0 },
+		{ }
+	};
+
+	switch (action) {
+	case ALC_FIXUP_ACT_PRE_PROBE:
+		alc_apply_pincfgs(codec, pincfgs);
+		break;
+	case ALC_FIXUP_ACT_PROBE:
+		spec->init_amp = ALC_INIT_NONE;
+		break;
+	}
+}
+
 static const struct alc_fixup alc260_fixups[] = {
 	[ALC260_FIXUP_HP_DC5750] = {
 		.type = ALC_FIXUP_PINS,
@@ -4938,7 +4974,11 @@ static const struct alc_fixup alc260_fixups[] = {
 		.v.func = alc260_fixup_gpio1_toggle,
 		.chained = true,
 		.chain_id = ALC260_FIXUP_COEF,
-	}
+	},
+	[ALC260_FIXUP_KN1] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc260_fixup_kn1,
+	},
 };
 
 static const struct snd_pci_quirk alc260_fixup_tbl[] = {
@@ -4948,6 +4988,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750),
 	SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1),
+	SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1),
 	SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER),
 	SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF),
 	{}
-- 
cgit v1.2.3


From 3e843196c697ee2c319d96e861980fb4c3e04e24 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Apr 2012 12:04:03 +0200
Subject: ALSA: hda/sigmatel - Fix inverted mute LED

While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way.  The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.

Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 33a9946b492c..4742cac26aa9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5063,12 +5063,11 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled)
 	if (spec->gpio_led_polarity)
 		muted = !muted;
 
-	/*polarity defines *not* muted state level*/
 	if (!spec->vref_mute_led_nid) {
 		if (muted)
-			spec->gpio_data &= ~spec->gpio_led; /* orange */
+			spec->gpio_data |= spec->gpio_led;
 		else
-			spec->gpio_data |= spec->gpio_led; /* white */
+			spec->gpio_data &= ~spec->gpio_led;
 		stac_gpio_set(codec, spec->gpio_mask,
 				spec->gpio_dir, spec->gpio_data);
 	} else {
-- 
cgit v1.2.3


From ca3649de026ff95c6f2847e8d096cf2f411c02b3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Apr 2012 15:15:25 +0200
Subject: ALSA: hda/conexant - Don't set HP pin-control bit unconditionally

Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.

Check the pin-capability and avoid the HP bit if not supported.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index d29d6d377904..f52c9ef3cc8c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3951,9 +3951,14 @@ static void cx_auto_init_output(struct hda_codec *codec)
 	int i;
 
 	mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids);
-	for (i = 0; i < cfg->hp_outs; i++)
+	for (i = 0; i < cfg->hp_outs; i++) {
+		unsigned int val = PIN_OUT;
+		if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) &
+		    AC_PINCAP_HP_DRV)
+			val |= AC_PINCTL_HP_EN;
 		snd_hda_codec_write(codec, cfg->hp_pins[i], 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+	}
 	mute_outputs(codec, cfg->hp_outs, cfg->hp_pins);
 	mute_outputs(codec, cfg->line_outs, cfg->line_out_pins);
 	mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins);
-- 
cgit v1.2.3


From d70f363222ef373c2037412f09a600357cfa1c7a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 19 Apr 2012 15:18:08 +0200
Subject: ALSA: hda/conexant - Set up the missing docking-station pins

ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.

The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.

Bugzilla entries:
	https://bugzilla.redhat.com/show_bug.cgi?id=808559
	https://bugzilla.redhat.com/show_bug.cgi?id=806217
	https://bugzilla.redhat.com/show_bug.cgi?id=810697

Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 28 +++++++++++++++++++++++++---
 1 file changed, 25 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f52c9ef3cc8c..58b5de4a6eed 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -4367,8 +4367,10 @@ static void apply_pin_fixup(struct hda_codec *codec,
 
 enum {
 	CXT_PINCFG_LENOVO_X200,
+	CXT_PINCFG_LENOVO_TP410,
 };
 
+/* ThinkPad X200 & co with cxt5051 */
 static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
 	{ 0x17, 0x21a11000 }, /* dock-mic */
@@ -4376,15 +4378,33 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
 	{}
 };
 
+/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */
+static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = {
+	{ 0x19, 0x042110ff }, /* HP (seq# overridden) */
+	{ 0x1a, 0x21a190f0 }, /* dock-mic */
+	{ 0x1c, 0x212140ff }, /* dock-HP */
+	{}
+};
+
 static const struct cxt_pincfg *cxt_pincfg_tbl[] = {
 	[CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200,
+	[CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410,
 };
 
-static const struct snd_pci_quirk cxt_fixups[] = {
+static const struct snd_pci_quirk cxt5051_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200),
 	{}
 };
 
+static const struct snd_pci_quirk cxt5066_fixups[] = {
+	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
+	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
+	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
+	SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
+	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
+	{}
+};
+
 /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
  * can be created (bko#42825)
  */
@@ -4421,11 +4441,13 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		break;
 	case 0x14f15051:
 		add_cx5051_fake_mutes(codec);
+		apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl);
+		break;
+	default:
+		apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl);
 		break;
 	}
 
-	apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
-
 	err = cx_auto_search_adcs(codec);
 	if (err < 0)
 		return err;
-- 
cgit v1.2.3


From 5ac57550f279c3d991ef0b398681bcaca18169f7 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 20 Apr 2012 10:01:46 +0200
Subject: ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E

According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.

Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/950490
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e65e35433055..818f90bc7d57 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6109,6 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = {
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
+	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
-- 
cgit v1.2.3


From c914f55f7cdfafe9d7d5b248751902c7ab57691e Mon Sep 17 00:00:00 2001
From: Mark Hills <mark@pogo.org.uk>
Date: Mon, 30 Apr 2012 19:39:22 +0100
Subject: ALSA: echoaudio: Remove incorrect part of assertion

This assertion seems to imply that chip->dsp_code_to_load is a pointer.
It's actually an integer handle on the actual firmware, and 0 has no
special meaning.

The assertion prevents initialisation of a Darla20 card, but would also
affect other models. It seems it was introduced in commit dd7b254d.

ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting...
ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000
ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware...
ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20
------------[ cut here ]------------
WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]()
Hardware name: Dell DM051
BUG? (!chip->dsp_code_to_load || !chip->comm_page)

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/echoaudio/echoaudio_dsp.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index 64417a733220..d8c670c9d62c 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip)
 	const struct firmware *fw;
 	int box_type, err;
 
-	if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page))
+	if (snd_BUG_ON(!chip->comm_page))
 		return -EPERM;
 
 	/* See if the ASIC is present and working - only if the DSP is already loaded */
-- 
cgit v1.2.3


From f5c53d898cc34079373c63a290528963db31d681 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 7 May 2012 10:07:33 +0200
Subject: ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G

Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 818f90bc7d57..27d0f637864a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5405,6 +5405,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
 		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+	SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G",
+		      ALC882_FIXUP_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
 	SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G),
 	SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
-- 
cgit v1.2.3


From bca40138558f0b39357fd1ca477868e4f52f4b1e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 7 May 2012 11:13:14 +0200
Subject: ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo

Reported-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 27d0f637864a..8ea613eb73f0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5440,6 +5440,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF),
 
 	SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
+	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
-- 
cgit v1.2.3


From 42eb92380f73f28e3a5a51973af1183fdbac82f2 Mon Sep 17 00:00:00 2001
From: Andre Schramm <andre.schramm@iosono-sound.com>
Date: Mon, 7 May 2012 18:52:51 +0200
Subject: ALSA: hdsp - Provide ioctl_compat

snd_hdsp uses its own ioctls to acquire config- and status information.
Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels.

Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Reviewed-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/rme9652/hdsp.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b68cdec03b9e..0b2aea2ce172 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 	strcpy(hw->name, "HDSP hwdep interface");
 
 	hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
+	hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl;
 
 	return 0;
 }
-- 
cgit v1.2.3


From af741c150f66db8d1da6f82ac75e2571f7f1dd38 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 7 May 2012 18:09:48 +0200
Subject: ALSA: hda/realtek - Call alc_auto_parse_customize_define() always
 after fixup

The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization.  Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 13 +++++++------
 1 file changed, 7 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8ea613eb73f0..7810913d07a0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5641,13 +5641,13 @@ static int patch_alc262(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80);
 	}
 #endif
-	alc_auto_parse_customize_define(codec);
-
 	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
 
 	alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc262_parse_auto_config(codec);
 	if (err < 0)
@@ -6252,8 +6252,6 @@ static int patch_alc269(struct hda_codec *codec)
 
 	spec->mixer_nid = 0x0b;
 
-	alc_auto_parse_customize_define(codec);
-
 	err = alc_codec_rename_from_preset(codec);
 	if (err < 0)
 		goto error;
@@ -6286,6 +6284,8 @@ static int patch_alc269(struct hda_codec *codec)
 		       alc269_fixup_tbl, alc269_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc269_parse_auto_config(codec);
 	if (err < 0)
@@ -6862,8 +6862,6 @@ static int patch_alc662(struct hda_codec *codec)
 	/* handle multiple HPs as is */
 	spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
 
-	alc_auto_parse_customize_define(codec);
-
 	alc_fix_pll_init(codec, 0x20, 0x04, 15);
 
 	err = alc_codec_rename_from_preset(codec);
@@ -6880,6 +6878,9 @@ static int patch_alc662(struct hda_codec *codec)
 	alc_pick_fixup(codec, alc662_fixup_models,
 		       alc662_fixup_tbl, alc662_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
+	alc_auto_parse_customize_define(codec);
+
 	/* automatic parse from the BIOS config */
 	err = alc662_parse_auto_config(codec);
 	if (err < 0)
-- 
cgit v1.2.3


From 619a341b78f17fb86d92e89c04612676cd05e26f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 8 May 2012 16:30:59 +0200
Subject: Revert "ALSA: hda - Set codec to D3 forcibly even if not used"

This reverts commit 785f857d1cb0856b612b46a0545b74aa2596e44a.

The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.

Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence.  But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.

In short, the safest option now is to revert this affecting commit.

Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.

Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c |  4 ----
 sound/pci/hda/hda_intel.c | 14 +++++++++++++-
 2 files changed, 13 insertions(+), 5 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 7a8fcc4c15f8..841475cc13b6 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -5444,10 +5444,6 @@ int snd_hda_suspend(struct hda_bus *bus)
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		if (hda_codec_is_power_on(codec))
 			hda_call_codec_suspend(codec);
-		else /* forcibly change the power to D3 even if not used */
-			hda_set_power_state(codec,
-					    codec->afg ? codec->afg : codec->mfg,
-					    AC_PWRST_D3);
 		if (codec->patch_ops.post_suspend)
 			codec->patch_ops.post_suspend(codec);
 	}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c19e71a94e1b..6e958bf94191 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2351,6 +2351,17 @@ static void azx_power_notify(struct hda_bus *bus)
  * power management
  */
 
+static int snd_hda_codecs_inuse(struct hda_bus *bus)
+{
+	struct hda_codec *codec;
+
+	list_for_each_entry(codec, &bus->codec_list, list) {
+		if (snd_hda_codec_needs_resume(codec))
+			return 1;
+	}
+	return 0;
+}
+
 static int azx_suspend(struct pci_dev *pci, pm_message_t state)
 {
 	struct snd_card *card = pci_get_drvdata(pci);
@@ -2397,7 +2408,8 @@ static int azx_resume(struct pci_dev *pci)
 		return -EIO;
 	azx_init_pci(chip);
 
-	azx_init_chip(chip, 1);
+	if (snd_hda_codecs_inuse(chip->bus))
+		azx_init_chip(chip, 1);
 
 	snd_hda_resume(chip->bus);
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
-- 
cgit v1.2.3


From 32cf4023e689ad5b3a81a749d8cc99d7f184cb99 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 4 May 2012 11:05:55 +0200
Subject: ALSA: HDA: Lessen CPU usage when waiting for chip to respond

When an IRQ for some reason gets lost, we wait up to a second using
udelay, which is CPU intensive. This patch improves the situation by
waiting about 30 ms in the CPU intensive mode, then stepping down to
using msleep(2) instead. In essence, we trade some granularity in
exchange for less CPU consumption when the waiting time is a bit longer.

As a result, PulseAudio should no longer be killed by the kernel
for taking up to much RT-prio CPU time. At least not for *this* reason.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 6e958bf94191..1f350522bed4 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -783,11 +783,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
+	unsigned long loopcounter;
 	int do_poll = 0;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
-	for (;;) {
+
+	for (loopcounter = 0;; loopcounter++) {
 		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
@@ -803,7 +805,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 		if (time_after(jiffies, timeout))
 			break;
-		if (bus->needs_damn_long_delay)
+		if (bus->needs_damn_long_delay || loopcounter > 3000)
 			msleep(2); /* temporary workaround */
 		else {
 			udelay(10);
-- 
cgit v1.2.3


From b0791dda813c179e539b0fc1ecd3f5f30f2571e2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 15 May 2012 08:07:31 +0200
Subject: ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops

BIOS on some HP laptops don't set the speaker-pins as fixed but expose
as jacks, and this confuses the driver as if these pins are
jack-detectable.  As a result, the machine doesn't get sounds from
speakers because the driver prepares the power-map update via jack
unsol events which never come up in reality.  The bug was introduced
in some time in 3.2 for enabling the power-mapping feature.

This patch fixes the problem by replacing the check of the persistent
power-map bits with a proper is_jack_detectable() call.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43240

Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4742cac26aa9..2cb1e08f962a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4415,9 +4415,9 @@ static int stac92xx_init(struct hda_codec *codec)
 		def_conf = get_defcfg_connect(def_conf);
 		/* skip any ports that don't have jacks since presence
  		 * detection is useless */
-		if (def_conf != AC_JACK_PORT_COMPLEX) {
-			if (def_conf != AC_JACK_PORT_NONE)
-				stac_toggle_power_map(codec, nid, 1);
+		if (def_conf != AC_JACK_PORT_NONE &&
+		    !is_jack_detectable(codec, nid)) {
+			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
 		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) {
-- 
cgit v1.2.3