From 02b6b5b640e773eb4d4d0685fa6c1fbc660b2834 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 21 Jan 2011 13:27:39 +0100
Subject: ALSA: HDA: Refactor some redundant code for Conexant 5066/205xx

Four very similar procedures - one for each model - now
refactored into one. This isn't all duplicated code, but a step
in the right direction.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 81 ++++++++++++------------------------------
 1 file changed, 23 insertions(+), 58 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9bb030a469cd..7cd59b9f0e97 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2387,79 +2387,53 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
 	cxt5066_update_speaker(codec);
 }
 
-/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
+/* Dispatch the right mic autoswitch function */
+static void cxt5066_automic(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
-	switch (res >> 26) {
-	case CONEXANT_HP_EVENT:
-		cxt5066_hp_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
-		/* ignore mic events in DC mode; we're always using the jack */
-		if (!spec->dc_enable)
-			cxt5066_olpc_automic(codec);
-		break;
-	}
-}
 
-/* unsolicited event for jack sensing */
-static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res)
-{
-	snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26);
-	switch (res >> 26) {
-	case CONEXANT_HP_EVENT:
-		cxt5066_hp_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
+	if (spec->dell_vostro)
 		cxt5066_vostro_automic(codec);
-		break;
-	}
-}
-
-/* unsolicited event for jack sensing */
-static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res)
-{
-	snd_printdd("CXT5066_ideapad: unsol event %x (%x)\n", res, res >> 26);
-	switch (res >> 26) {
-	case CONEXANT_HP_EVENT:
-		cxt5066_hp_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
+	else if (spec->ideapad)
 		cxt5066_ideapad_automic(codec);
-		break;
-	}
+	else if (spec->thinkpad)
+		cxt5066_thinkpad_automic(codec);
+	else if (spec->hp_laptop)
+		cxt5066_hp_laptop_automic(codec);
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-	snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26);
+	struct conexant_spec *spec = codec->spec;
+	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cxt5066_hp_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
-		cxt5066_hp_laptop_automic(codec);
+		/* ignore mic events in DC mode; we're always using the jack */
+		if (!spec->dc_enable)
+			cxt5066_olpc_automic(codec);
 		break;
 	}
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-	snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26);
+	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cxt5066_hp_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
-		cxt5066_thinkpad_automic(codec);
+		cxt5066_automic(codec);
 		break;
 	}
 }
 
+
 static const struct hda_input_mux cxt5066_analog_mic_boost = {
 	.num_items = 5,
 	.items = {
@@ -3039,20 +3013,11 @@ static struct hda_verb cxt5066_init_verbs_hp_laptop[] = {
 /* initialize jack-sensing, too */
 static int cxt5066_init(struct hda_codec *codec)
 {
-	struct conexant_spec *spec = codec->spec;
-
 	snd_printdd("CXT5066: init\n");
 	conexant_init(codec);
 	if (codec->patch_ops.unsol_event) {
 		cxt5066_hp_automute(codec);
-		if (spec->dell_vostro)
-			cxt5066_vostro_automic(codec);
-		else if (spec->ideapad)
-			cxt5066_ideapad_automic(codec);
-		else if (spec->thinkpad)
-			cxt5066_thinkpad_automic(codec);
-		else if (spec->hp_laptop)
-			cxt5066_hp_laptop_automic(codec);
+		cxt5066_automic(codec);
 	}
 	cxt5066_set_mic_boost(codec);
 	return 0;
@@ -3169,7 +3134,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_HP_LAPTOP:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_hp_laptop_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->init_verbs[spec->num_init_verbs] =
 			cxt5066_init_verbs_hp_laptop;
 		spec->num_init_verbs++;
@@ -3207,7 +3172,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_DELL_VOSTRO:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_vostro_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->init_verbs[0] = cxt5066_init_verbs_vostro;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
@@ -3224,7 +3189,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_IDEAPAD:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_ideapad_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->init_verbs[0] = cxt5066_init_verbs_ideapad;
@@ -3240,7 +3205,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		break;
 	case CXT5066_THINKPAD:
 		codec->patch_ops.init = cxt5066_init;
-		codec->patch_ops.unsol_event = cxt5066_thinkpad_event;
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->init_verbs[0] = cxt5066_init_verbs_thinkpad;
-- 
cgit v1.2.3


From a1d6906e2d2b4655e248f490ab088c27876a600a Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Fri, 21 Jan 2011 13:33:28 +0100
Subject: ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx

BugLink: http://bugs.launchpad.net/bugs/701271

This new model, named "asus", is identical to the "hp_laptop" model,
except for the location of the internal mic, which is at pin 0x1a.
It is used for Asus K52JU and Lenovo G560.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 Documentation/sound/alsa/HD-Audio-Models.txt |  1 +
 sound/pci/hda/patch_conexant.c               | 24 +++++++++++++++++++++++-
 2 files changed, 24 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16ae4300c747..0caf77e59be4 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -296,6 +296,7 @@ Conexant 5066
 =============
   laptop	Basic Laptop config (default)
   hp-laptop	HP laptops, e g G60
+  asus		Asus K52JU, Lenovo G560
   dell-laptop	Dell laptops
   dell-vostro	Dell Vostro
   olpc-xo-1_5	OLPC XO 1.5
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7cd59b9f0e97..19f0daf6497d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -127,6 +127,7 @@ struct conexant_spec {
 	unsigned int ideapad:1;
 	unsigned int thinkpad:1;
 	unsigned int hp_laptop:1;
+	unsigned int asus:1;
 
 	unsigned int ext_mic_present;
 	unsigned int recording;
@@ -2312,6 +2313,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec)
 	}
 }
 
+
+/* toggle input of built-in digital mic and mic jack appropriately */
+static void cxt5066_asus_automic(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_jack_detect(codec, 0x1b);
+	snd_printdd("CXT5066: external microphone present=%d\n", present);
+	snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+			    present ? 1 : 0);
+}
+
+
 /* toggle input of built-in digital mic and mic jack appropriately */
 static void cxt5066_hp_laptop_automic(struct hda_codec *codec)
 {
@@ -2400,6 +2414,8 @@ static void cxt5066_automic(struct hda_codec *codec)
 		cxt5066_thinkpad_automic(codec);
 	else if (spec->hp_laptop)
 		cxt5066_hp_laptop_automic(codec);
+	else if (spec->asus)
+		cxt5066_asus_automic(codec);
 }
 
 /* unsolicited event for jack sensing */
@@ -3045,6 +3061,7 @@ enum {
 	CXT5066_DELL_VOSTRO,	/* Dell Vostro 1015i */
 	CXT5066_IDEAPAD,	/* Lenovo IdeaPad U150 */
 	CXT5066_THINKPAD,	/* Lenovo ThinkPad T410s, others? */
+	CXT5066_ASUS,		/* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */
 	CXT5066_HP_LAPTOP,      /* HP Laptop */
 	CXT5066_MODELS
 };
@@ -3056,6 +3073,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 	[CXT5066_DELL_VOSTRO]	= "dell-vostro",
 	[CXT5066_IDEAPAD]	= "ideapad",
 	[CXT5066_THINKPAD]	= "thinkpad",
+	[CXT5066_ASUS]		= "asus",
 	[CXT5066_HP_LAPTOP]	= "hp-laptop",
 };
 
@@ -3068,6 +3086,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
 	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
+	SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3077,6 +3096,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
 	{}
 };
@@ -3132,13 +3152,15 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->num_init_verbs++;
 		spec->dell_automute = 1;
 		break;
+	case CXT5066_ASUS:
 	case CXT5066_HP_LAPTOP:
 		codec->patch_ops.init = cxt5066_init;
 		codec->patch_ops.unsol_event = cxt5066_unsol_event;
 		spec->init_verbs[spec->num_init_verbs] =
 			cxt5066_init_verbs_hp_laptop;
 		spec->num_init_verbs++;
-		spec->hp_laptop = 1;
+		spec->hp_laptop = board_config == CXT5066_HP_LAPTOP;
+		spec->asus = board_config == CXT5066_ASUS;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		/* no S/PDIF out */
-- 
cgit v1.2.3


From f6a2491ca23d26d829730e33fbdd9e44fc5d1d53 Mon Sep 17 00:00:00 2001
From: Andy Robinson <ajr55555@gmail.com>
Date: Mon, 24 Jan 2011 10:12:37 -0500
Subject: ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct
 SPDIF output

Changed the Asus A52J quirk to use the asus model instead of the
hp_laptop model, which fixes the external mic input. Added an Asus
U50F quirk to use the asus model. For the cxt5066 codecs, added
checking of the digital output pins to determine which digital output
nodes to use instead of always using node 0x21, since some systems
have node 0x12 connected to a SPDIF out jack.

[A slight modification for better readability by tiwai]

Signed-off-by: Andy Robinson <ajr55555@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 35 +++++++++++++++++++++++++++++++----
 1 file changed, 31 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 19f0daf6497d..9867afc7895e 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -85,6 +85,7 @@ struct conexant_spec {
 	unsigned int auto_mic;
 	int auto_mic_ext;		/* autocfg.inputs[] index for ext mic */
 	unsigned int need_dac_fix;
+	hda_nid_t slave_dig_outs[2];
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -353,6 +354,8 @@ static int conexant_build_pcms(struct hda_codec *codec)
 			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
 				spec->dig_in_nid;
 		}
+		if (spec->slave_dig_outs[0])
+			codec->slave_dig_outs = spec->slave_dig_outs;
 	}
 
 	return 0;
@@ -2101,7 +2104,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
 static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
 static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
-#define CXT5066_SPDIF_OUT	0x21
+static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
 
 /* OLPC's microphone port is DC coupled for use with external sensors,
  * therefore we use a 50% mic bias in order to center the input signal with
@@ -2623,6 +2626,27 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
 	spec->recording = 0;
 }
 
+static void conexant_check_dig_outs(struct hda_codec *codec,
+				    hda_nid_t *dig_pins,
+				    int num_pins)
+{
+	struct conexant_spec *spec = codec->spec;
+	hda_nid_t *nid_loc = &spec->multiout.dig_out_nid;
+	int i;
+
+	for (i = 0; i < num_pins; i++, dig_pins++) {
+		unsigned int cfg = snd_hda_codec_get_pincfg(codec, *dig_pins);
+		if (get_defcfg_connect(cfg) == AC_JACK_PORT_NONE)
+			continue;
+		if (snd_hda_get_connections(codec, *dig_pins, nid_loc, 1) != 1)
+			continue;
+		if (spec->slave_dig_outs[0])
+			nid_loc++;
+		else
+			nid_loc = spec->slave_dig_outs;
+	}
+}
+
 static struct hda_input_mux cxt5066_capture_source = {
 	.num_items = 4,
 	.items = {
@@ -3085,8 +3109,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP),
+	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
+	SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
@@ -3118,7 +3143,8 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
 	spec->multiout.dac_nids = cxt5066_dac_nids;
-	spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+	conexant_check_dig_outs(codec, cxt5066_digout_pin_nids,
+	    ARRAY_SIZE(cxt5066_digout_pin_nids));
 	spec->num_adc_nids = 1;
 	spec->adc_nids = cxt5066_adc_nids;
 	spec->capsrc_nids = cxt5066_capsrc_nids;
@@ -3164,7 +3190,8 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		/* no S/PDIF out */
-		spec->multiout.dig_out_nid = 0;
+		if (board_config == CXT5066_HP_LAPTOP)
+			spec->multiout.dig_out_nid = 0;
 		/* input source automatically selected */
 		spec->input_mux = NULL;
 		spec->port_d_mode = 0;
-- 
cgit v1.2.3


From c9ba374d24882c04e7cc000d8cf3b0fe56511b84 Mon Sep 17 00:00:00 2001
From: Andreas Mohr <andi@lisas.de>
Date: Tue, 25 Jan 2011 06:46:31 +0100
Subject: ALSA: azt3328 -  fix broken AZF_FMT_XLATE macro

Cleanly revert to non-macro implementation of
snd_azf3328_codec_setfmt(), to fix last-minute functionality breakage
induced by following checkpatch.pl recommendations without giving them
their due full share of thought ("revolting computer, ensuing PEBKAC").

I would like to thank Jiri Slaby for his very timely (in -rc1 even)
and unexpected (uncommon hardware) "recognition of the dangerous situation"
due to his very commendable static parser use. :)

Reported-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/azt3328.c | 38 ++++++++++++++++----------------------
 1 file changed, 16 insertions(+), 22 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 6117595fc075..573594bf3225 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
 
 	snd_azf3328_dbgcallenter();
 	switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
-	do { \
-		case AZF_FREQ_ ## in_freq: \
-			freq = SOUNDFORMAT_FREQ_ ## out_bits; \
-			break; \
-	} while (0);
-	AZF_FMT_XLATE(4000, SUSPECTED_4000)
-	AZF_FMT_XLATE(4800, SUSPECTED_4800)
-	/* the AZF3328 names it "5510" for some strange reason: */
-	AZF_FMT_XLATE(5512, 5510)
-	AZF_FMT_XLATE(6620, 6620)
-	AZF_FMT_XLATE(8000, 8000)
-	AZF_FMT_XLATE(9600, 9600)
-	AZF_FMT_XLATE(11025, 11025)
-	AZF_FMT_XLATE(13240, SUSPECTED_13240)
-	AZF_FMT_XLATE(16000, 16000)
-	AZF_FMT_XLATE(22050, 22050)
-	AZF_FMT_XLATE(32000, 32000)
+	case AZF_FREQ_4000:  freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+	case AZF_FREQ_4800:  freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+	case AZF_FREQ_5512:
+		/* the AZF3328 names it "5510" for some strange reason */
+			     freq = SOUNDFORMAT_FREQ_5510; break;
+	case AZF_FREQ_6620:  freq = SOUNDFORMAT_FREQ_6620; break;
+	case AZF_FREQ_8000:  freq = SOUNDFORMAT_FREQ_8000; break;
+	case AZF_FREQ_9600:  freq = SOUNDFORMAT_FREQ_9600; break;
+	case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+	case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+	case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+	case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+	case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
 	default:
 		snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
 		/* fall-through */
-	AZF_FMT_XLATE(44100, 44100)
-	AZF_FMT_XLATE(48000, 48000)
-	AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+	case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+	case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+	case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
 	}
 	/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
 	/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
-- 
cgit v1.2.3


From d757534ed15387202e322854cd72dc58bbb975de Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 25 Jan 2011 19:44:26 +0100
Subject: ALSA: HDA: Fix dmesg output of HDMI supported bits

This typo caused the dmesg output of the supported bits of HDMI
to be cut off early.

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a663471dadc..74b0560289c0 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
 	snd_print_pcm_rates(a->rates, buf, sizeof(buf));
 
 	if (a->format == AUDIO_CODING_TYPE_LPCM)
-		snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8));
+		snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8);
 	else if (a->max_bitrate)
 		snprintf(buf2, sizeof(buf2),
 				", max bitrate = %d", a->max_bitrate);
-- 
cgit v1.2.3


From ded9f5238bb719737f82b0b5b957937cb0203804 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 26 Jan 2011 11:46:12 +0100
Subject: ALSA: HDA: Fix automute on Thinkpad L412/L512

BugLink: http://bugs.launchpad.net/bugs/707902

More Thinkpad machines with invalid SKU found, that disables
automute between speakers and headphones on these machines.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index be4df4c6fd56..2fa9ed99c32f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14954,9 +14954,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
 	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
-	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
-	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	{}
-- 
cgit v1.2.3


From 195938753951e70e85303301c37906c7ad72645e Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 27 Jan 2011 10:28:46 +0100
Subject: ALSA: HDA: Fix microphone(s) on Lenovo Edge 13

BugLink: http://bugs.launchpad.net/bugs/708521

This Edge 13 model has an internal mic at 0x1a and should
therefore use the asus quirk.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9867afc7895e..7e1ca43bd660 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3120,6 +3120,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
 	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
-- 
cgit v1.2.3


From efbeb0718126d277c9d7e902eec8da956acf4bd6 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 31 Jan 2011 11:47:52 +0100
Subject: ALSA: oxygen: fix output routing on Xonar DG

This card uses separate I2S outputs for the front speakers and
headphones, and reverses the order of the three speaker outputs.
To work around this, add a model-specific callback to adjust the
controller's playback routing.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen.h       |  2 ++
 sound/pci/oxygen/oxygen_mixer.c |  2 ++
 sound/pci/oxygen/xonar_dg.c     | 36 ++++++++++++++++++++++++++++++++++++
 3 files changed, 40 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d17cd2..f53897a708b4 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
 	void (*update_dac_volume)(struct oxygen *chip);
 	void (*update_dac_mute)(struct oxygen *chip);
 	void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+	unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+					   unsigned int play_routing);
 	void (*gpio_changed)(struct oxygen *chip);
 	void (*uart_input)(struct oxygen *chip);
 	void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d5895d..26c7e8bcb229 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
 			    (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
 			    (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
 			    (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+	if (chip->model.adjust_dac_routing)
+		reg_value = chip->model.adjust_dac_routing(chip, reg_value);
 	oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
 			      OXYGEN_PLAY_DAC0_SOURCE_MASK |
 			      OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602eba79..bc6eb58be380 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
  *
  *   SPI 0 -> CS4245
  *
+ *   I²S 1 -> CS4245
+ *   I²S 2 -> CS4361 (center/LFE)
+ *   I²S 3 -> CS4361 (surround)
+ *   I²S 4 -> CS4361 (front)
+ *
  *   GPIO 3 <- ?
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
  *   input 1 <- aux
  *   input 2 <- front mic
  *   input 4 <- line/mic
+ *   DAC out -> headphones
  *   aux out -> front panel headphones
  */
 
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
 	cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
 }
 
+static inline unsigned int shift_bits(unsigned int value,
+				      unsigned int shift_from,
+				      unsigned int shift_to,
+				      unsigned int mask)
+{
+	if (shift_from < shift_to)
+		return (value << (shift_to - shift_from)) & mask;
+	else
+		return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+					  unsigned int play_routing)
+{
+	return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
 static int output_switch_info(struct snd_kcontrol *ctl,
 			      struct snd_ctl_elem_info *info)
 {
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
 	.resume = dg_resume,
 	.set_dac_params = set_cs4245_dac_params,
 	.set_adc_params = set_cs4245_adc_params,
+	.adjust_dac_routing = adjust_dg_dac_routing,
 	.dump_registers = dump_cs4245_registers,
 	.model_data_size = sizeof(struct dg),
 	.device_config = PLAYBACK_0_TO_I2S |
-- 
cgit v1.2.3


From 70f7db11c45a313b23922cacf248c613c3b2144c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Feb 2011 17:16:38 +0100
Subject: ALSA: hda - Fix memory leaks in conexant jack arrays

The Conexant codec driver adds the jack arrays in init callback which
may be called also in each PM resume.  This results in the addition of
new jack element at each time.

The fix is to check whether the requested jack is already present in
the array.

Reference: Novell bug 668929
	https://bugzilla.novell.com/show_bug.cgi?id=668929

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7e1ca43bd660..fbe97d32140d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -407,10 +407,16 @@ static int conexant_add_jack(struct hda_codec *codec,
 	struct conexant_spec *spec;
 	struct conexant_jack *jack;
 	const char *name;
-	int err;
+	int i, err;
 
 	spec = codec->spec;
 	snd_array_init(&spec->jacks, sizeof(*jack), 32);
+
+	jack = spec->jacks.list;
+	for (i = 0; i < spec->jacks.used; i++, jack++)
+		if (jack->nid == nid)
+			return 0 ; /* already present */
+
 	jack = snd_array_new(&spec->jacks);
 	name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ;
 
-- 
cgit v1.2.3


From 460c92fa38ff140f83c269e948e2aaab071d0af0 Mon Sep 17 00:00:00 2001
From: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Date: Mon, 7 Feb 2011 13:13:27 +0100
Subject: ALSA: hda - switch lfe with side in mixer for 4930g
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Built-in sub-woofer can now be controlled by lfe slider instead of
side slider on Acer Aspire 5930g

Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++-
 1 file changed, 24 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2fa9ed99c32f..2571d977df22 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2290,6 +2290,29 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0,
+		HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, 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, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
+
 static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -10359,7 +10382,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ACER_ASPIRE_4930G] = {
-		.mixers = { alc888_base_mixer,
+		.mixers = { alc888_acer_aspire_4930g_mixer,
 				alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
 				alc888_acer_aspire_4930g_verbs },
-- 
cgit v1.2.3


From 11839aed21881d7edd65dd79f22a8eb18426f672 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 8 Feb 2011 17:25:49 +0100
Subject: ALSA: hda - Fix missing CA initialization for HDMI/DP

The commit 53d7d69d8ffdfa60c5b66cc2e9ee0774aaaef5c0
    ALSA: hdmi - support infoframe for DisplayPort
dropped the initialization of CA field accidentally.
This resulted in only two-channel LPCM mode on Nvidia machines.

Reference: kernel bug 28592
	https://bugzilla.kernel.org/show_bug.cgi?id=28592

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

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d5b83fa8d24..a58767736727 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -642,6 +642,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
 			hdmi_ai->ver		= 0x01;
 			hdmi_ai->len		= 0x0a;
 			hdmi_ai->CC02_CT47	= channels - 1;
+			hdmi_ai->CA		= ca;
 			hdmi_checksum_audio_infoframe(hdmi_ai);
 		} else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */
 			struct dp_audio_infoframe *dp_ai;
@@ -651,6 +652,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
 			dp_ai->len		= 0x1b;
 			dp_ai->ver		= 0x11 << 2;
 			dp_ai->CC02_CT47	= channels - 1;
+			dp_ai->CA		= ca;
 		} else {
 			snd_printd("HDMI: unknown connection type at pin %d\n",
 				   pin_nid);
-- 
cgit v1.2.3


From 41a63f18d339ae6aefe73d45a8147f63f3439b30 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 10 Feb 2011 17:39:20 +0100
Subject: ALSA: hda - Don't handle empty patch files

When an empty string is passed to patch option, the driver should
ignore it.  Otherwise it gets an error by trying to load it.

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/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a991eb15..0baffcdee8f9 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2703,7 +2703,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	if (err < 0)
 		goto out_free;
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
-	if (patch[dev]) {
+	if (patch[dev] && *patch[dev]) {
 		snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
 			   patch[dev]);
 		err = snd_hda_load_patch(chip->bus, patch[dev]);
-- 
cgit v1.2.3


From a6c47a85b8e7e4a8c47394607c5e5c43224b0892 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Thu, 10 Feb 2011 15:39:19 +0100
Subject: ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G

According to the reporter, node 0x15 needs to be muted for subwoofer
to stop sounding. This pin is marked as unused by BIOS, so fix that.

BugLink: http://bugs.launchpad.net/bugs/715877

Cc: stable@kernel.org (2.6.37+)
Reported-by: Hans Peter
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 2571d977df22..089a7de2439e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19517,6 +19517,7 @@ static const struct alc_fixup alc662_fixups[] = {
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
-- 
cgit v1.2.3


From 965b76d23ea354848dea8d34059d04e150dcd464 Mon Sep 17 00:00:00 2001
From: Anisse Astier <anisse@astier.eu>
Date: Thu, 10 Feb 2011 13:14:44 +0100
Subject: ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662

This netbook has a only one jack output and an internal mic.

By default, mic and jack sense aren't working. Using lenovo-101e
parameters makes both work.

The device seems based on a Sharetronic Q70, so this should fix audio for
this model too.

Signed-off-by: Anisse Astier <anisse@astier.eu>
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 089a7de2439e..3328a259a242 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -18825,6 +18825,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 					ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
 			   ALC663_ASUS_H13),
+	SND_PCI_QUIRK(0x1991, 0x5628, "Ordissimo EVE", ALC662_LENOVO_101E),
 	{}
 };
 
-- 
cgit v1.2.3


From 5e5677f239ba69fc718ec9a87ac4ba035dafe2c0 Mon Sep 17 00:00:00 2001
From: Raymond Yau <superquad.vortex2@gmail.com>
Date: Mon, 14 Feb 2011 07:33:24 +0800
Subject: ALSA: au88x0 - Modify pointer callback to give accurate playback
 position

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0_core.c | 14 +++++++++++---
 1 file changed, 11 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f356e0f..16c0bdfbb164 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
 static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
 {
 	stream_t *dma = &vortex->dma_adb[adbdma];
-	int temp;
+	int temp, page, delta;
 
 	temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
-	temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
-	return temp;
+	page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+	if (dma->nr_periods >= 4)
+		delta = (page - dma->period_real) & 3;
+	else {
+		delta = (page - dma->period_real);
+		if (delta < 0)
+			delta += dma->nr_periods;
+	}
+	return (dma->period_virt + delta) * dma->period_bytes
+		+ (temp & (dma->period_bytes - 1));
 }
 
 static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
-- 
cgit v1.2.3


From b540afc2b3d6e4cd1d1f137ef6d9e9c78d67fecd Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Mon, 14 Feb 2011 20:27:44 +0100
Subject: ALSA: HDA: Add position_fix quirk for an Asus device

The bug reporter claims that position_fix=1 is needed for his
microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40).

Reported-by: Kjell L.
BugLink: http://bugs.launchpad.net/bugs/718402
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0baffcdee8f9..fcedad9a5fef 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
-- 
cgit v1.2.3


From 983345e51e0de144775c7449e5cb01ce6cdd1346 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 15 Feb 2011 19:57:09 +0100
Subject: ALSA: HDA: Conexant auto: Handle multiple connections to ADC node

Conexant 20641 has several inputs to its ADC node, with one selector
and individual amps for all inputs. This patch adds support in the
Conexant auto parser to handle that case.

It also means that the pin node's volume is being renamed to "Boost"
to avoid name clash with the new volume controls on the ADC node.

BugLink: http://bugs.launchpad.net/bugs/719524
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 61 +++++++++++++++++++++++++++++++++---------
 1 file changed, 48 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fbe97d32140d..cd29eafdc0ed 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3729,9 +3729,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	return 0;
 }
 
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 			      const char *dir, int cidx,
-			      hda_nid_t nid, int hda_dir)
+			      hda_nid_t nid, int hda_dir, int amp_idx)
 {
 	static char name[32];
 	static struct snd_kcontrol_new knew[] = {
@@ -3743,7 +3743,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 
 	for (i = 0; i < 2; i++) {
 		struct snd_kcontrol *kctl;
-		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+							    hda_dir);
 		knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
 		knew[i].index = cidx;
 		snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3759,6 +3760,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 	return 0;
 }
 
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir)		\
+	cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
 #define cx_auto_add_pb_volume(codec, nid, str, idx)			\
 	cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
 
@@ -3808,29 +3812,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	static const char *prev_label;
-	int i, err, cidx;
+	int i, err, cidx, conn_len;
+	hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+	int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+	int adc_nid = spec->adc_nids[0];
+
+	conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+					   HDA_MAX_CONNECTIONS);
+	if (conn_len < 0)
+		return conn_len;
+
+	multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+	if (!multi_adc_volume) {
+		err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+					 HDA_INPUT);
+		if (err < 0)
+			return err;
+	}
 
-	err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
-				 HDA_INPUT);
-	if (err < 0)
-		return err;
 	prev_label = NULL;
 	cidx = 0;
 	for (i = 0; i < cfg->num_inputs; i++) {
 		hda_nid_t nid = cfg->inputs[i].pin;
 		const char *label;
-		if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+		int j;
+		int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+		if (!pin_amp && !multi_adc_volume)
 			continue;
+
 		label = hda_get_autocfg_input_label(codec, cfg, i);
 		if (label == prev_label)
 			cidx++;
 		else
 			cidx = 0;
 		prev_label = label;
-		err = cx_auto_add_volume(codec, label, " Capture", cidx,
-					 nid, HDA_INPUT);
-		if (err < 0)
-			return err;
+
+		if (pin_amp) {
+			err = cx_auto_add_volume(codec, label, " Boost", cidx,
+						 nid, HDA_INPUT);
+			if (err < 0)
+				return err;
+		}
+
+		if (!multi_adc_volume)
+			continue;
+		for (j = 0; j < conn_len; j++) {
+			if (conn[j] == nid) {
+				err = cx_auto_add_volume_idx(codec, label,
+				    " Capture", cidx, adc_nid, HDA_INPUT, j);
+				if (err < 0)
+					return err;
+				break;
+			}
+		}
 	}
 	return 0;
 }
-- 
cgit v1.2.3


From 89724958e5d596bb91328644c97dd80399443e87 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 16 Feb 2011 21:34:04 +0100
Subject: ALSA: HDA: Do not announce false surround in Conexant auto

Without this patch, one line-out and one speaker and
Conexant's auto parser would announce (non-working) surround
capabilities.

BugLink: http://bugs.launchpad.net/bugs/721126
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index cd29eafdc0ed..dd7c5c12225d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3410,7 +3410,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 		}
 	}
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.max_channels = nums * 2;
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
 	if (cfg->hp_outs > 0)
 		spec->auto_mute = 1;
-- 
cgit v1.2.3


From 306496761745942d8167e9193a738b559a7fb0b3 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Mon, 21 Feb 2011 10:23:18 +0100
Subject: ALSA: HDA: Fix mic initialization in VIA auto parser

This typo caused some microphone inputs not to be correctly
initialized on VIA codecs.

Reported-By: Mark Goldstein <goldstein.mark@gmail.com>
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a76c3260d941..63b0054200a8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = cfg->inputs[i].pin;
 		if (spec->smart51_enabled && is_smart51_pins(spec, nid))
 			ctl = PIN_OUT;
-		else if (i == AUTO_PIN_MIC)
+		else if (cfg->inputs[i].type == AUTO_PIN_MIC)
 			ctl = PIN_VREF50;
 		else
 			ctl = PIN_IN;
-- 
cgit v1.2.3