From 74dc8909c1ce38098e6689239ed6ae6b6bf9f92b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Jan 2011 14:14:41 +0100
Subject: ALSA: hda - Remove unused fixup entry for ALC262

... and a minor cleanup.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 12 +++---------
 1 file changed, 3 insertions(+), 9 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 98c9cd8f647..738d5d8962d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12453,12 +12453,6 @@ static const struct alc_fixup alc262_fixups[] = {
 			{ }
 		}
 	},
-	[PINFIX_PB_M5210] = {
-		.verbs = (const struct hda_verb[]) {
-			{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
-			{}
-		}
-	},
 };
 
 static struct snd_pci_quirk alc262_fixup_tbl[] = {
@@ -14826,7 +14820,7 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
 
 enum {
 	ALC269_FIXUP_SONY_VAIO,
-	ALC275_FIX_SONY_VAIO_GPIO2,
+	ALC275_FIXUP_SONY_VAIO_GPIO2,
 	ALC269_FIXUP_DELL_M101Z,
 	ALC269_FIXUP_SKU_IGNORE,
 	ALC269_FIXUP_ASUS_G73JW,
@@ -14841,7 +14835,7 @@ static const struct alc_fixup alc269_fixups[] = {
 			{}
 		}
 	},
-	[ALC275_FIX_SONY_VAIO_GPIO2] = {
+	[ALC275_FIXUP_SONY_VAIO_GPIO2] = {
 		.verbs = (const struct hda_verb[]) {
 			{0x01, AC_VERB_SET_GPIO_MASK, 0x04},
 			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04},
@@ -14886,7 +14880,7 @@ static const struct alc_fixup alc269_fixups[] = {
 };
 
 static struct snd_pci_quirk alc269_fixup_tbl[] = {
-	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2),
+	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
-- 
cgit v1.2.3-70-g09d2


From 6fc398cb306b0441436c93d6ddead3109b99f884 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Jan 2011 14:36:37 +0100
Subject: ALSA: hda - Apply mario fixup only once

The amp-override is necessary only once at initialization time.
Also fixed a coding style issue.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 738d5d8962d..f13920e5384 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19366,7 +19366,10 @@ static void alc662_auto_init(struct hda_codec *codec)
 }
 
 static void alc272_fixup_mario(struct hda_codec *codec,
-			       const struct alc_fixup *fix, int pre_init) {
+			       const struct alc_fixup *fix, int pre_init)
+{
+	if (!pre_init)
+		return;
 	if (snd_hda_override_amp_caps(codec, 0x2, HDA_OUTPUT,
 				      (0x3b << AC_AMPCAP_OFFSET_SHIFT) |
 				      (0x3b << AC_AMPCAP_NUM_STEPS_SHIFT) |
-- 
cgit v1.2.3-70-g09d2


From 9fb1ef25f4d31f07cdaf7c6075b40bbcb00c1f92 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Jan 2011 14:40:43 +0100
Subject: ALSA: hda - Apply Sony VAIO hweq fixup only once

This should be applied also only once as a part of the initialization.

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f13920e5384..98b4b2e1b93 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14814,6 +14814,8 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
 {
 	int coef;
 
+	if (pre_init)
+		return;
 	coef = alc_read_coef_idx(codec, 0x1e);
 	alc_write_coef_idx(codec, 0x1e, coef | 0x80);
 }
-- 
cgit v1.2.3-70-g09d2


From b5bfbc670283d1ff21df4cd3f9f036cc47e34ce4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Jan 2011 14:22:32 +0100
Subject: ALSA: hda - Reorganize fixup structure for Realtek

Instead of keeping various data types in a single record, put the
type field and keep a single value in each entry, but allows chaining
multiple fixup entries.  This allows more flexible data management
(see ALC275_FIXUP_SONY_HWEQ for example).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 317 +++++++++++++++++++++++++-----------------
 1 file changed, 193 insertions(+), 124 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 98b4b2e1b93..a06c9437cde 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -303,6 +303,8 @@ struct alc_customize_define {
 	unsigned int  fixup:1; /* Means that this sku is set by driver, not read from hw */
 };
 
+struct alc_fixup;
+
 struct alc_spec {
 	/* codec parameterization */
 	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -404,6 +406,11 @@ struct alc_spec {
 	/* for PLL fix */
 	hda_nid_t pll_nid;
 	unsigned int pll_coef_idx, pll_coef_bit;
+
+	/* fix-up list */
+	int fixup_id;
+	const struct alc_fixup *fixup_list;
+	const char *fixup_name;
 };
 
 /*
@@ -1683,88 +1690,130 @@ struct alc_model_fixup {
 };
 
 struct alc_fixup {
-	unsigned int sku;
-	const struct alc_pincfg *pins;
-	const struct hda_verb *verbs;
-	void (*func)(struct hda_codec *codec, const struct alc_fixup *fix,
-		     int pre_init);
+	int type;
+	union {
+		unsigned int sku;
+		const struct alc_pincfg *pins;
+		const struct hda_verb *verbs;
+		void (*func)(struct hda_codec *codec,
+			     const struct alc_fixup *fix,
+			     int action);
+	} v;
+	bool chained;
+	int chain_id;
 };
 
-static void __alc_pick_fixup(struct hda_codec *codec,
-			     const struct alc_fixup *fix,
-			     const char *modelname,
-			     int pre_init)
+enum {
+	ALC_FIXUP_INVALID,
+	ALC_FIXUP_SKU,
+	ALC_FIXUP_PINS,
+	ALC_FIXUP_VERBS,
+	ALC_FIXUP_FUNC,
+};
+
+enum {
+	ALC_FIXUP_ACT_PRE_PROBE,
+	ALC_FIXUP_ACT_PROBE,
+};
+
+static void alc_apply_fixup(struct hda_codec *codec, int action)
 {
-	const struct alc_pincfg *cfg;
-	struct alc_spec *spec;
+	struct alc_spec *spec = codec->spec;
+	int id = spec->fixup_id;
+	const char *modelname = spec->fixup_name;
+	int depth = 0;
 
-	cfg = fix->pins;
-	if (pre_init && fix->sku) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-		snd_printdd(KERN_INFO "hda_codec: %s: Apply sku override for %s\n",
-			    codec->chip_name, modelname);
-#endif
-		spec = codec->spec;
-		spec->cdefine.sku_cfg = fix->sku;
-		spec->cdefine.fixup = 1;
-	}
-	if (pre_init && cfg) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-		snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n",
-			    codec->chip_name, modelname);
-#endif
-		for (; cfg->nid; cfg++)
-			snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
-	}
-	if (!pre_init && fix->verbs) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-		snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n",
-			    codec->chip_name, modelname);
-#endif
-		add_verb(codec->spec, fix->verbs);
-	}
-	if (fix->func) {
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-		snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-func for %s\n",
-			    codec->chip_name, modelname);
-#endif
-		fix->func(codec, fix, pre_init);
+	if (!spec->fixup_list)
+		return;
+
+	while (id >= 0) {
+		const struct alc_fixup *fix = spec->fixup_list + id;
+		const struct alc_pincfg *cfg;
+
+		switch (fix->type) {
+		case ALC_FIXUP_SKU:
+			if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku)
+				break;;
+			snd_printdd(KERN_INFO "hda_codec: %s: "
+				    "Apply sku override for %s\n",
+				    codec->chip_name, modelname);
+			spec->cdefine.sku_cfg = fix->v.sku;
+			spec->cdefine.fixup = 1;
+			break;
+		case ALC_FIXUP_PINS:
+			cfg = fix->v.pins;
+			if (action != ALC_FIXUP_ACT_PRE_PROBE || !cfg)
+				break;
+			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);
+			break;
+		case ALC_FIXUP_VERBS:
+			if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs)
+				break;
+			snd_printdd(KERN_INFO "hda_codec: %s: "
+				    "Apply fix-verbs for %s\n",
+				    codec->chip_name, modelname);
+			add_verb(codec->spec, fix->v.verbs);
+			break;
+		case ALC_FIXUP_FUNC:
+			if (!fix->v.func)
+				break;
+			snd_printdd(KERN_INFO "hda_codec: %s: "
+				    "Apply fix-func for %s\n",
+				    codec->chip_name, modelname);
+			fix->v.func(codec, fix, action);
+			break;
+		default:
+			snd_printk(KERN_ERR "hda_codec: %s: "
+				   "Invalid fixup type %d\n",
+				   codec->chip_name, fix->type);
+			break;
+		}
+		if (!fix[id].chained)
+			break;
+		if (++depth > 10)
+			break;
+		id = fix[id].chain_id;
 	}
 }
 
 static void alc_pick_fixup(struct hda_codec *codec,
-				 const struct snd_pci_quirk *quirk,
-				 const struct alc_fixup *fix,
-				 int pre_init)
+			   const struct alc_model_fixup *models,
+			   const struct snd_pci_quirk *quirk,
+			   const struct alc_fixup *fixlist)
 {
-	quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
-	if (quirk) {
-		fix += quirk->value;
-#ifdef CONFIG_SND_DEBUG_VERBOSE
-		__alc_pick_fixup(codec, fix, quirk->name, pre_init);
-#else
-		__alc_pick_fixup(codec, fix, NULL, pre_init);
-#endif
-	}
-}
+	struct alc_spec *spec = codec->spec;
+	int id = -1;
+	const char *name = NULL;
 
-static void alc_pick_fixup_model(struct hda_codec *codec,
-				 const struct alc_model_fixup *models,
-				 const struct snd_pci_quirk *quirk,
-				 const struct alc_fixup *fix,
-				 int pre_init)
-{
 	if (codec->modelname && models) {
 		while (models->name) {
 			if (!strcmp(codec->modelname, models->name)) {
-				fix += models->id;
+				id = models->id;
+				name = models->name;
 				break;
 			}
 			models++;
 		}
-		__alc_pick_fixup(codec, fix, codec->modelname, pre_init);
-	} else {
-		alc_pick_fixup(codec, quirk, fix, pre_init);
+	}
+	if (id < 0) {
+		quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+		if (quirk) {
+			id = quirk->value;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
+			name = quirk->name;
+#endif
+		}
+	}
+
+	spec->fixup_id = id;
+	if (id >= 0) {
+		spec->fixup_list = fixlist;
+		spec->fixup_name = name;
 	}
 }
 
@@ -7090,7 +7139,8 @@ enum {
 
 static const struct alc_fixup alc260_fixups[] = {
 	[PINFIX_HP_DC5750] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x11, 0x90130110 }, /* speaker */
 			{ }
 		}
@@ -7301,8 +7351,10 @@ static int patch_alc260(struct hda_codec *codec)
 		board_config = ALC260_AUTO;
 	}
 
-	if (board_config == ALC260_AUTO)
-		alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 1);
+	if (board_config == ALC260_AUTO) {
+		alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	}
 
 	if (board_config == ALC260_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -7350,8 +7402,7 @@ static int patch_alc260(struct hda_codec *codec)
 	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
 
-	if (board_config == ALC260_AUTO)
-		alc_pick_fixup(codec, alc260_fixup_tbl, alc260_fixups, 0);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	spec->vmaster_nid = 0x08;
 
@@ -10678,7 +10729,8 @@ enum {
 
 static const struct alc_fixup alc882_fixups[] = {
 	[PINFIX_ABIT_AW9D_MAX] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x15, 0x01080104 }, /* side */
 			{ 0x16, 0x01011012 }, /* rear */
 			{ 0x17, 0x01016011 }, /* clfe */
@@ -10686,13 +10738,15 @@ static const struct alc_fixup alc882_fixups[] = {
 		}
 	},
 	[PINFIX_PB_M5210] = {
-		.verbs = (const struct hda_verb[]) {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
 			{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 },
 			{}
 		}
 	},
 	[PINFIX_ACER_ASPIRE_7736] = {
-		.sku = ALC_FIXUP_SKU_IGNORE,
+		.type = ALC_FIXUP_SKU,
+		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
 };
 
@@ -10978,8 +11032,10 @@ static int patch_alc882(struct hda_codec *codec)
 		board_config = ALC882_AUTO;
 	}
 
-	if (board_config == ALC882_AUTO)
-		alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1);
+	if (board_config == ALC882_AUTO) {
+		alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	}
 
 	alc_auto_parse_customize_define(codec);
 
@@ -11055,8 +11111,7 @@ static int patch_alc882(struct hda_codec *codec)
 	if (has_cdefine_beep(codec))
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
-	if (board_config == ALC882_AUTO)
-		alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -12446,7 +12501,8 @@ enum {
 
 static const struct alc_fixup alc262_fixups[] = {
 	[PINFIX_FSC_H270] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x14, 0x99130110 }, /* speaker */
 			{ 0x15, 0x0221142f }, /* front HP */
 			{ 0x1b, 0x0121141f }, /* rear HP */
@@ -12883,8 +12939,10 @@ static int patch_alc262(struct hda_codec *codec)
 		board_config = ALC262_AUTO;
 	}
 
-	if (board_config == ALC262_AUTO)
-		alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 1);
+	if (board_config == ALC262_AUTO) {
+		alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	}
 
 	if (board_config == ALC262_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -12954,8 +13012,7 @@ static int patch_alc262(struct hda_codec *codec)
 	if (!spec->no_analog && has_cdefine_beep(codec))
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
-	if (board_config == ALC262_AUTO)
-		alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 0);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -14810,11 +14867,11 @@ static int alc269_resume(struct hda_codec *codec)
 #endif /* SND_HDA_NEEDS_RESUME */
 
 static void alc269_fixup_hweq(struct hda_codec *codec,
-			       const struct alc_fixup *fix, int pre_init)
+			       const struct alc_fixup *fix, int action)
 {
 	int coef;
 
-	if (pre_init)
+	if (action != ALC_FIXUP_ACT_PROBE)
 		return;
 	coef = alc_read_coef_idx(codec, 0x1e);
 	alc_write_coef_idx(codec, 0x1e, coef | 0x80);
@@ -14832,22 +14889,26 @@ enum {
 
 static const struct alc_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_SONY_VAIO] = {
-		.verbs = (const struct hda_verb[]) {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
 			{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
 			{}
 		}
 	},
 	[ALC275_FIXUP_SONY_VAIO_GPIO2] = {
-		.verbs = (const struct hda_verb[]) {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
 			{0x01, AC_VERB_SET_GPIO_MASK, 0x04},
 			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04},
 			{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-			{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
 			{ }
-		}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_SONY_VAIO
 	},
 	[ALC269_FIXUP_DELL_M101Z] = {
-		.verbs = (const struct hda_verb[]) {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
 			/* Enables internal speaker */
 			{0x20, AC_VERB_SET_COEF_INDEX, 13},
 			{0x20, AC_VERB_SET_PROC_COEF, 0x4040},
@@ -14855,29 +14916,28 @@ static const struct alc_fixup alc269_fixups[] = {
 		}
 	},
 	[ALC269_FIXUP_SKU_IGNORE] = {
-		.sku = ALC_FIXUP_SKU_IGNORE,
+		.type = ALC_FIXUP_SKU,
+		.v.sku = ALC_FIXUP_SKU_IGNORE,
 	},
 	[ALC269_FIXUP_ASUS_G73JW] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x17, 0x99130111 }, /* subwoofer */
 			{ }
 		}
 	},
 	[ALC269_FIXUP_LENOVO_EAPD] = {
-		.verbs = (const struct hda_verb[]) {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
 			{0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
 			{}
 		}
 	},
 	[ALC275_FIXUP_SONY_HWEQ] = {
-		.func = alc269_fixup_hweq,
-		.verbs = (const struct hda_verb[]) {
-			{0x01, AC_VERB_SET_GPIO_MASK, 0x04},
-			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04},
-			{0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-			{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD},
-			{ }
-		}
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_hweq,
+		.chained = true,
+		.chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2
 	}
 };
 
@@ -15174,8 +15234,10 @@ static int patch_alc269(struct hda_codec *codec)
 		board_config = ALC269_AUTO;
 	}
 
-	if (board_config == ALC269_AUTO)
-		alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1);
+	if (board_config == ALC269_AUTO) {
+		alc_pick_fixup(codec, NULL, alc269_fixup_tbl, alc269_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	}
 
 	if (board_config == ALC269_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -15236,8 +15298,7 @@ static int patch_alc269(struct hda_codec *codec)
 	if (has_cdefine_beep(codec))
 		set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
-	if (board_config == ALC269_AUTO)
-		alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	spec->vmaster_nid = 0x02;
 
@@ -16296,7 +16357,8 @@ enum {
 
 static const struct alc_fixup alc861_fixups[] = {
 	[PINFIX_FSC_AMILO_PI1505] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x0b, 0x0221101f }, /* HP */
 			{ 0x0f, 0x90170310 }, /* speaker */
 			{ }
@@ -16331,8 +16393,10 @@ static int patch_alc861(struct hda_codec *codec)
 		board_config = ALC861_AUTO;
 	}
 
-	if (board_config == ALC861_AUTO)
-		alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1);
+	if (board_config == ALC861_AUTO) {
+		alc_pick_fixup(codec, NULL, alc861_fixup_tbl, alc861_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	}
 
 	if (board_config == ALC861_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -16369,8 +16433,7 @@ static int patch_alc861(struct hda_codec *codec)
 
 	spec->vmaster_nid = 0x03;
 
-	if (board_config == ALC861_AUTO)
-		alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC861_AUTO) {
@@ -17252,7 +17315,8 @@ enum {
 /* reset GPIO1 */
 static const struct alc_fixup alc861vd_fixups[] = {
 	[ALC660VD_FIX_ASUS_GPIO1] = {
-		.verbs = (const struct hda_verb[]) {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
 			{0x01, AC_VERB_SET_GPIO_MASK, 0x03},
 			{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
 			{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
@@ -17287,8 +17351,10 @@ static int patch_alc861vd(struct hda_codec *codec)
 		board_config = ALC861VD_AUTO;
 	}
 
-	if (board_config == ALC861VD_AUTO)
-		alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1);
+	if (board_config == ALC861VD_AUTO) {
+		alc_pick_fixup(codec, NULL, alc861vd_fixup_tbl, alc861vd_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+	}
 
 	if (board_config == ALC861VD_AUTO) {
 		/* automatic parse from the BIOS config */
@@ -17336,8 +17402,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 
 	spec->vmaster_nid = 0x02;
 
-	if (board_config == ALC861VD_AUTO)
-		alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
 
 	codec->patch_ops = alc_patch_ops;
 
@@ -19368,9 +19433,9 @@ static void alc662_auto_init(struct hda_codec *codec)
 }
 
 static void alc272_fixup_mario(struct hda_codec *codec,
-			       const struct alc_fixup *fix, int pre_init)
+			       const struct alc_fixup *fix, int action)
 {
-	if (!pre_init)
+	if (action != ALC_FIXUP_ACT_PROBE)
 		return;
 	if (snd_hda_override_amp_caps(codec, 0x2, HDA_OUTPUT,
 				      (0x3b << AC_AMPCAP_OFFSET_SHIFT) |
@@ -19389,19 +19454,22 @@ enum {
 
 static const struct alc_fixup alc662_fixups[] = {
 	[ALC662_FIXUP_ASPIRE] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x15, 0x99130112 }, /* subwoofer */
 			{ }
 		}
 	},
 	[ALC662_FIXUP_IDEAPAD] = {
-		.pins = (const struct alc_pincfg[]) {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
 			{ 0x17, 0x99130112 }, /* subwoofer */
 			{ }
 		}
 	},
 	[ALC272_FIXUP_MARIO] = {
-		.func = alc272_fixup_mario,
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc272_fixup_mario,
 	}
 };
 
@@ -19455,7 +19523,9 @@ static int patch_alc662(struct hda_codec *codec)
 	}
 
 	if (board_config == ALC662_AUTO) {
-		alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 1);
+		alc_pick_fixup(codec, alc662_fixup_models,
+			       alc662_fixup_tbl, alc662_fixups);
+		alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 		/* automatic parse from the BIOS config */
 		err = alc662_parse_auto_config(codec);
 		if (err < 0) {
@@ -19513,12 +19583,11 @@ static int patch_alc662(struct hda_codec *codec)
 	}
 	spec->vmaster_nid = 0x02;
 
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC662_AUTO) {
+	if (board_config == ALC662_AUTO)
 		spec->init_hook = alc662_auto_init;
-		alc_pick_fixup_model(codec, alc662_fixup_models,
-				     alc662_fixup_tbl, alc662_fixups, 0);
-	}
 
 	alc_init_jacks(codec);
 
-- 
cgit v1.2.3-70-g09d2


From 5870112021fb38e73b25dad3baec4ca0819c594a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Jan 2011 15:41:45 +0100
Subject: ALSA: hda - Add fixup-call in init callback

In some cases, the fix-up is required in the init callback to be called
both at the first initialization and at the resume.  The new action type
ALC_FIXUP_ACT_INIT is used for this case.

So far, only ALC275_FIXUP_SONY_HWEQ uses this.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a06c9437cde..b445ae98942 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1714,6 +1714,7 @@ enum {
 enum {
 	ALC_FIXUP_ACT_PRE_PROBE,
 	ALC_FIXUP_ACT_PROBE,
+	ALC_FIXUP_ACT_INIT,
 };
 
 static void alc_apply_fixup(struct hda_codec *codec, int action)
@@ -3910,6 +3911,8 @@ static int alc_init(struct hda_codec *codec)
 	if (spec->init_hook)
 		spec->init_hook(codec);
 
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT);
+
 	hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
@@ -14871,7 +14874,7 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
 {
 	int coef;
 
-	if (action != ALC_FIXUP_ACT_PROBE)
+	if (action != ALC_FIXUP_ACT_INIT)
 		return;
 	coef = alc_read_coef_idx(codec, 0x1e);
 	alc_write_coef_idx(codec, 0x1e, coef | 0x80);
-- 
cgit v1.2.3-70-g09d2


From ad09fc9d2156f3d37537b34418a6b79309013d33 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 14 Jan 2011 09:42:27 +0100
Subject: ALSA: hda - Suppress the odd number of channels for HDMI

It looks like that HDMI codecs don't support the odd number of channels
although HD-audio spec doesn't have the restriction.  Add the
hw_constraint to limit to only the even number of channels.

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index f29b97b5de8..2d288793ceb 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1238,6 +1238,9 @@ static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo,
 		snd_pcm_hw_constraint_list(substream->runtime, 0,
 				SNDRV_PCM_HW_PARAM_CHANNELS,
 				hw_constraints_channels);
+	} else {
+		snd_pcm_hw_constraint_step(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 	}
 
 	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-- 
cgit v1.2.3-70-g09d2


From 361fe6e90888af83d5bfdfc152d737018cbede43 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 14 Jan 2011 09:55:32 +0100
Subject: ALSA: hda - Rearrange fixup struct in patch_realtek.c

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b445ae98942..69554061c16 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1691,6 +1691,8 @@ struct alc_model_fixup {
 
 struct alc_fixup {
 	int type;
+	bool chained;
+	int chain_id;
 	union {
 		unsigned int sku;
 		const struct alc_pincfg *pins;
@@ -1699,8 +1701,6 @@ struct alc_fixup {
 			     const struct alc_fixup *fix,
 			     int action);
 	} v;
-	bool chained;
-	int chain_id;
 };
 
 enum {
-- 
cgit v1.2.3-70-g09d2


From 639cef0eb6df05d5516520aa89b0c9fe62ee2d3b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 14 Jan 2011 10:30:46 +0100
Subject: ALSA: hda - Store PCM parameters properly in HDMI open callback

In hdmi_pcm_open(), the evaluated PCM hw parameters are stored in
hinfo, but these aren't properly set back to the current runtime
record since these have been set beforehand in azx_pcm_open().
This patch fixes the behavior.

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d288793ceb..5980552f597 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -817,6 +817,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	struct hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld;
 	struct hda_pcm_stream *codec_pars;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	unsigned int idx;
 
 	for (idx = 0; idx < spec->num_cvts; idx++)
@@ -844,6 +845,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 		hinfo->formats = codec_pars->formats;
 		hinfo->maxbps = codec_pars->maxbps;
 	}
+	/* store the updated parameters */
+	runtime->hw.channels_min = hinfo->channels_min;
+	runtime->hw.channels_max = hinfo->channels_max;
+	runtime->hw.formats = hinfo->formats;
+	runtime->hw.rates = hinfo->rates;
 	return 0;
 }
 
-- 
cgit v1.2.3-70-g09d2


From 4fe2ca14678174d9df254ae3ba2b79bacc19e2ee Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 14 Jan 2011 10:33:26 +0100
Subject: ALSA: hda - More coverage for odd-number channels elimination for
 HDMI

The commit ad09fc9d2156f3d37537b34418a6b79309013d33 didn't cover the
case for Intel and Nvidia HDMIs, where hdmi_pcm_open() is called.
Put the hw_constraint there, too.

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 5980552f597..2d5b83fa8d2 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -850,6 +850,9 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo,
 	runtime->hw.channels_max = hinfo->channels_max;
 	runtime->hw.formats = hinfo->formats;
 	runtime->hw.rates = hinfo->rates;
+
+	snd_pcm_hw_constraint_step(substream->runtime, 0,
+				   SNDRV_PCM_HW_PARAM_CHANNELS, 2);
 	return 0;
 }
 
-- 
cgit v1.2.3-70-g09d2


From c66ddf32dda0d5bcf9db7b4cc42ef5da7baadd3e Mon Sep 17 00:00:00 2001
From: Raymond Yau <superquad.vortex2@gmail.com>
Date: Mon, 17 Jan 2011 11:19:03 +0100
Subject: ALSA: hda - Add add multi-streaming playback for AD1988

Attached a patch which add a new model to support multi-streaming
playback for ad1988.

playback another stereo stream through the front panel headphone on
device 2 while playback through the speakers connected to rear panel
on device 0 at the same time.

Tested with ad1988a rev2 codec on asus P5B-V motherboard.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 182 ++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 178 insertions(+), 4 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 46780670162..34ee1169f2e 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -46,6 +46,9 @@ struct ad198x_spec {
 	unsigned int cur_eapd;
 	unsigned int need_dac_fix;
 
+	hda_nid_t *alt_dac_nid;
+	struct hda_pcm_stream *stream_analog_alt_playback;
+
 	/* capture */
 	unsigned int num_adc_nids;
 	hda_nid_t *adc_nids;
@@ -156,6 +159,25 @@ static const char *ad_slave_sws[] = {
 	NULL
 };
 
+static const char *ad1988_6stack_fp_slave_vols[] = {
+	"Front Playback Volume",
+	"Surround Playback Volume",
+	"Center Playback Volume",
+	"LFE Playback Volume",
+	"Side Playback Volume",
+	"IEC958 Playback Volume",
+	NULL
+};
+
+static const char *ad1988_6stack_fp_slave_sws[] = {
+	"Front Playback Switch",
+	"Surround Playback Switch",
+	"Center Playback Switch",
+	"LFE Playback Switch",
+	"Side Playback Switch",
+	"IEC958 Playback Switch",
+	NULL
+};
 static void ad198x_free_kctls(struct hda_codec *codec);
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -309,6 +331,38 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
+static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+				struct hda_codec *codec,
+				unsigned int stream_tag,
+				unsigned int format,
+				struct snd_pcm_substream *substream)
+{
+	struct ad198x_spec *spec = codec->spec;
+	snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag,
+					0, format);
+	return 0;
+}
+
+static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				struct hda_codec *codec,
+				struct snd_pcm_substream *substream)
+{
+	struct ad198x_spec *spec = codec->spec;
+	snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]);
+	return 0;
+}
+
+static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in ad198x_build_pcms */
+	.ops = {
+		.prepare = ad198x_alt_playback_pcm_prepare,
+		.cleanup = ad198x_alt_playback_pcm_cleanup
+	},
+};
+
 /*
  * Digital out
  */
@@ -446,6 +500,17 @@ static int ad198x_build_pcms(struct hda_codec *codec)
 		}
 	}
 
+	if (spec->alt_dac_nid && spec->stream_analog_alt_playback) {
+		codec->num_pcms++;
+		info = spec->pcm_rec + 2;
+		info->name = "AD198x Headphone";
+		info->pcm_type = HDA_PCM_TYPE_AUDIO;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+			*spec->stream_analog_alt_playback;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+			spec->alt_dac_nid[0];
+	}
+
 	return 0;
 }
 
@@ -2015,6 +2080,7 @@ static int patch_ad1981(struct hda_codec *codec)
 enum {
 	AD1988_6STACK,
 	AD1988_6STACK_DIG,
+	AD1988_6STACK_DIG_FP,
 	AD1988_3STACK,
 	AD1988_3STACK_DIG,
 	AD1988_LAPTOP,
@@ -2047,6 +2113,10 @@ static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
 	0x04, 0x05, 0x0a, 0x06
 };
 
+static hda_nid_t ad1988_alt_dac_nid[1] = {
+	0x03
+};
+
 static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
 	0x04, 0x0a, 0x06
 };
@@ -2166,6 +2236,35 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = {
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+
+	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
+	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
+	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
+
+	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
+
+	{ } /* end */
+};
+
 /* 3-stack mode */
 static struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
@@ -2445,6 +2544,68 @@ static struct hda_verb ad1988_6stack_init_verbs[] = {
 	{ }
 };
 
+static struct hda_verb ad1988_6stack_fp_init_verbs[] = {
+	/* Front, Surround, CLFE, side DAC; unmute as default */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Headphone; unmute as default */
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Port-A front headphon path */
+	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Port-D line-out path */
+	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* Port-F surround path */
+	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* Port-G CLFE path */
+	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* Port-H side path */
+	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	/* Mono out path */
+	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
+	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
+	/* Port-B front mic-in path */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	/* Port-C line-in path */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Port-E mic-in path */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Analog CD Input */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	/* Analog Mix output amp */
+	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
+
+	{ }
+};
+
 static struct hda_verb ad1988_capture_init_verbs[] = {
 	/* mute analog mix */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -3074,13 +3235,13 @@ static int ad1988_auto_init(struct hda_codec *codec)
 	return 0;
 }
 
-
 /*
  */
 
 static const char *ad1988_models[AD1988_MODEL_LAST] = {
 	[AD1988_6STACK]		= "6stack",
 	[AD1988_6STACK_DIG]	= "6stack-dig",
+	[AD1988_6STACK_DIG_FP]	= "6stack-dig-fp",
 	[AD1988_3STACK]		= "3stack",
 	[AD1988_3STACK_DIG]	= "3stack-dig",
 	[AD1988_LAPTOP]		= "laptop",
@@ -3140,6 +3301,7 @@ static int patch_ad1988(struct hda_codec *codec)
 	switch (board_config) {
 	case AD1988_6STACK:
 	case AD1988_6STACK_DIG:
+	case AD1988_6STACK_DIG_FP:
 		spec->multiout.max_channels = 8;
 		spec->multiout.num_dacs = 4;
 		if (is_rev2(codec))
@@ -3152,10 +3314,22 @@ static int patch_ad1988(struct hda_codec *codec)
 			spec->mixers[0] = ad1988_6stack_mixers1_rev2;
 		else
 			spec->mixers[0] = ad1988_6stack_mixers1;
-		spec->mixers[1] = ad1988_6stack_mixers2;
+		if (board_config == AD1988_6STACK_DIG_FP) {
+			spec->mixers[1] = ad1988_6stack_fp_mixers;
+			spec->slave_vols = ad1988_6stack_fp_slave_vols;
+			spec->slave_sws = ad1988_6stack_fp_slave_sws;
+			spec->alt_dac_nid = ad1988_alt_dac_nid;
+			spec->stream_analog_alt_playback =
+				&ad198x_pcm_analog_alt_playback;
+		} else
+			spec->mixers[1] = ad1988_6stack_mixers2;
 		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_6stack_init_verbs;
-		if (board_config == AD1988_6STACK_DIG) {
+		if (board_config == AD1988_6STACK_DIG_FP)
+			spec->init_verbs[0] = ad1988_6stack_fp_init_verbs;
+		else
+			spec->init_verbs[0] = ad1988_6stack_init_verbs;
+		if ((board_config == AD1988_6STACK_DIG) ||
+			(board_config == AD1988_6STACK_DIG_FP)) {
 			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
 			spec->dig_in_nid = AD1988_SPDIF_IN;
 		}
-- 
cgit v1.2.3-70-g09d2


From ea73496324c1d990504e27f551e159388f891a4c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 17 Jan 2011 11:29:34 +0100
Subject: ALSA: hda - consitify string arrays

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 10 +++++-----
 sound/pci/hda/hda_generic.c    |  7 ++++---
 sound/pci/hda/hda_local.h      |  6 +++---
 sound/pci/hda/hda_proc.c       |  2 +-
 sound/pci/hda/patch_analog.c   | 30 ++++++++++++++++--------------
 sound/pci/hda/patch_cirrus.c   |  4 ++--
 sound/pci/hda/patch_cmedia.c   |  2 +-
 sound/pci/hda/patch_conexant.c | 14 +++++++-------
 sound/pci/hda/patch_realtek.c  | 34 ++++++++++++++++++----------------
 sound/pci/hda/patch_sigmatel.c | 36 ++++++++++++++++++------------------
 sound/pci/hda/patch_via.c      | 28 ++++++++++++++++++++--------
 11 files changed, 95 insertions(+), 78 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 05e5ec88c2d..ae5c5d5e4b7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2134,10 +2134,10 @@ int snd_hda_codec_reset(struct hda_codec *codec)
  * This function returns zero if successful or a negative error code.
  */
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
-			unsigned int *tlv, const char **slaves)
+			unsigned int *tlv, const char * const *slaves)
 {
 	struct snd_kcontrol *kctl;
-	const char **s;
+	const char * const *s;
 	int err;
 
 	for (s = slaves; *s && !snd_hda_find_mixer_ctl(codec, *s); s++)
@@ -3689,7 +3689,7 @@ EXPORT_SYMBOL_HDA(snd_hda_build_pcms);
  * If no entries are matching, the function returns a negative value.
  */
 int snd_hda_check_board_config(struct hda_codec *codec,
-			       int num_configs, const char **models,
+			       int num_configs, const char * const *models,
 			       const struct snd_pci_quirk *tbl)
 {
 	if (codec->modelname && models) {
@@ -3753,7 +3753,7 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config);
  * If no entries are matching, the function returns a negative value.
  */
 int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
-			       int num_configs, const char **models,
+			       int num_configs, const char * const *models,
 			       const struct snd_pci_quirk *tbl)
 {
 	const struct snd_pci_quirk *q;
@@ -4690,7 +4690,7 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin,
 					int check_location)
 {
 	unsigned int def_conf;
-	static const char *mic_names[] = {
+	static const char * const mic_names[] = {
 		"Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic",
 	};
 	int attr;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index fb0582f8d72..a63c54d9d76 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -762,7 +762,8 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con
 /*
  * build output mixer controls
  */
-static int create_output_mixers(struct hda_codec *codec, const char **names)
+static int create_output_mixers(struct hda_codec *codec,
+				const char * const *names)
 {
 	struct hda_gspec *spec = codec->spec;
 	int i, err;
@@ -780,8 +781,8 @@ static int create_output_mixers(struct hda_codec *codec, const char **names)
 static int build_output_controls(struct hda_codec *codec)
 {
 	struct hda_gspec *spec = codec->spec;
-	static const char *types_speaker[] = { "Speaker", "Headphone" };
-	static const char *types_line[] = { "Front", "Headphone" };
+	static const char * const types_speaker[] = { "Speaker", "Headphone" };
+	static const char * const types_line[] = { "Front", "Headphone" };
 
 	switch (spec->pcm_vol_nodes) {
 	case 1:
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 46bbefe2e4a..3ab5e7a303d 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -140,7 +140,7 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
 struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 					    const char *name);
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
-			unsigned int *tlv, const char **slaves);
+			unsigned int *tlv, const char * const *slaves);
 int snd_hda_codec_reset(struct hda_codec *codec);
 
 /* amp value bits */
@@ -341,10 +341,10 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen);
  * Misc
  */
 int snd_hda_check_board_config(struct hda_codec *codec, int num_configs,
-			       const char **modelnames,
+			       const char * const *modelnames,
 			       const struct snd_pci_quirk *pci_list);
 int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
-                               int num_configs, const char **models,
+                               int num_configs, const char * const *models,
                                const struct snd_pci_quirk *tbl);
 int snd_hda_add_new_ctls(struct hda_codec *codec,
 			 struct snd_kcontrol_new *knew);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index f025200f2a6..bfe74c2fb07 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -418,7 +418,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 
 static const char *get_pwr_state(u32 state)
 {
-	static const char *buf[4] = {
+	static const char * const buf[4] = {
 		"D0", "D1", "D2", "D3"
 	};
 	if (state < 4)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 34ee1169f2e..8dabab79868 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -84,8 +84,8 @@ struct ad198x_spec {
 #endif
 	/* for virtual master */
 	hda_nid_t vmaster_nid;
-	const char **slave_vols;
-	const char **slave_sws;
+	const char * const *slave_vols;
+	const char * const *slave_sws;
 };
 
 /*
@@ -133,7 +133,7 @@ static int ad198x_init(struct hda_codec *codec)
 	return 0;
 }
 
-static const char *ad_slave_vols[] = {
+static const char * const ad_slave_vols[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
 	"Center Playback Volume",
@@ -146,7 +146,7 @@ static const char *ad_slave_vols[] = {
 	NULL
 };
 
-static const char *ad_slave_sws[] = {
+static const char * const ad_slave_sws[] = {
 	"Front Playback Switch",
 	"Surround Playback Switch",
 	"Center Playback Switch",
@@ -159,7 +159,7 @@ static const char *ad_slave_sws[] = {
 	NULL
 };
 
-static const char *ad1988_6stack_fp_slave_vols[] = {
+static const char * const ad1988_6stack_fp_slave_vols[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
 	"Center Playback Volume",
@@ -169,7 +169,7 @@ static const char *ad1988_6stack_fp_slave_vols[] = {
 	NULL
 };
 
-static const char *ad1988_6stack_fp_slave_sws[] = {
+static const char * const ad1988_6stack_fp_slave_sws[] = {
 	"Front Playback Switch",
 	"Surround Playback Switch",
 	"Center Playback Switch",
@@ -1134,7 +1134,7 @@ enum {
 	AD1986A_MODELS
 };
 
-static const char *ad1986a_models[AD1986A_MODELS] = {
+static const char * const ad1986a_models[AD1986A_MODELS] = {
 	[AD1986A_6STACK]	= "6stack",
 	[AD1986A_3STACK]	= "3stack",
 	[AD1986A_LAPTOP]	= "laptop",
@@ -1878,7 +1878,7 @@ enum {
 	AD1981_MODELS
 };
 
-static const char *ad1981_models[AD1981_MODELS] = {
+static const char * const ad1981_models[AD1981_MODELS] = {
 	[AD1981_HP]		= "hp",
 	[AD1981_THINKPAD]	= "thinkpad",
 	[AD1981_BASIC]		= "basic",
@@ -2953,7 +2953,9 @@ static int ad1988_auto_create_multi_out_ctls(struct ad198x_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
+	static const char * const chname[4] = {
+		"Front", "Surround", NULL /*CLFE*/, "Side"
+	};
 	hda_nid_t nid;
 	int i, err;
 
@@ -3238,7 +3240,7 @@ static int ad1988_auto_init(struct hda_codec *codec)
 /*
  */
 
-static const char *ad1988_models[AD1988_MODEL_LAST] = {
+static const char * const ad1988_models[AD1988_MODEL_LAST] = {
 	[AD1988_6STACK]		= "6stack",
 	[AD1988_6STACK_DIG]	= "6stack-dig",
 	[AD1988_6STACK_DIG_FP]	= "6stack-dig-fp",
@@ -3573,7 +3575,7 @@ static struct hda_amp_list ad1884_loopbacks[] = {
 };
 #endif
 
-static const char *ad1884_slave_vols[] = {
+static const char * const ad1884_slave_vols[] = {
 	"PCM Playback Volume",
 	"Mic Playback Volume",
 	"Mono Playback Volume",
@@ -3811,7 +3813,7 @@ enum {
 	AD1984_MODELS
 };
 
-static const char *ad1984_models[AD1984_MODELS] = {
+static const char * const ad1984_models[AD1984_MODELS] = {
 	[AD1984_BASIC]		= "basic",
 	[AD1984_THINKPAD]	= "thinkpad",
 	[AD1984_DELL_DESKTOP]	= "dell_desktop",
@@ -4482,7 +4484,7 @@ enum {
 	AD1884A_MODELS
 };
 
-static const char *ad1884a_models[AD1884A_MODELS] = {
+static const char * const ad1884a_models[AD1884A_MODELS] = {
 	[AD1884A_DESKTOP]	= "desktop",
 	[AD1884A_LAPTOP]	= "laptop",
 	[AD1884A_MOBILE]	= "mobile",
@@ -4870,7 +4872,7 @@ enum {
 	AD1882_MODELS
 };
 
-static const char *ad1882_models[AD1986A_MODELS] = {
+static const char * const ad1882_models[AD1986A_MODELS] = {
 	[AD1882_3STACK]		= "3stack",
 	[AD1882_6STACK]		= "6stack",
 };
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 18af38ebf75..a07b031090d 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -490,7 +490,7 @@ static int parse_digital_input(struct hda_codec *codec)
  * create mixer controls
  */
 
-static const char *dir_sfx[2] = { "Playback", "Capture" };
+static const char * const dir_sfx[2] = { "Playback", "Capture" };
 
 static int add_mute(struct hda_codec *codec, const char *name, int index,
 		    unsigned int pval, int dir, struct snd_kcontrol **kctlp)
@@ -1156,7 +1156,7 @@ static int cs_parse_auto_config(struct hda_codec *codec)
 	return 0;
 }
 
-static const char *cs420x_models[CS420X_MODELS] = {
+static const char * const cs420x_models[CS420X_MODELS] = {
 	[CS420X_MBP53] = "mbp53",
 	[CS420X_MBP55] = "mbp55",
 	[CS420X_IMAC27] = "imac27",
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index ff60908f455..1f8bbcd0f80 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -608,7 +608,7 @@ static void cmi9880_free(struct hda_codec *codec)
 /*
  */
 
-static const char *cmi9880_models[CMI_MODELS] = {
+static const char * const cmi9880_models[CMI_MODELS] = {
 	[CMI_MINIMAL]	= "minimal",
 	[CMI_MIN_FP]	= "min_fp",
 	[CMI_FULL]	= "full",
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e96581fcdbd..9bb030a469c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -537,13 +537,13 @@ static struct snd_kcontrol_new cxt_beep_mixer[] = {
 };
 #endif
 
-static const char *slave_vols[] = {
+static const char * const slave_vols[] = {
 	"Headphone Playback Volume",
 	"Speaker Playback Volume",
 	NULL
 };
 
-static const char *slave_sws[] = {
+static const char * const slave_sws[] = {
 	"Headphone Playback Switch",
 	"Speaker Playback Switch",
 	NULL
@@ -1134,7 +1134,7 @@ enum {
 	CXT5045_MODELS
 };
 
-static const char *cxt5045_models[CXT5045_MODELS] = {
+static const char * const cxt5045_models[CXT5045_MODELS] = {
 	[CXT5045_LAPTOP_HPSENSE]	= "laptop-hpsense",
 	[CXT5045_LAPTOP_MICSENSE]	= "laptop-micsense",
 	[CXT5045_LAPTOP_HPMICSENSE]	= "laptop-hpmicsense",
@@ -1579,7 +1579,7 @@ enum {
 	CXT5047_MODELS
 };
 
-static const char *cxt5047_models[CXT5047_MODELS] = {
+static const char * const cxt5047_models[CXT5047_MODELS] = {
 	[CXT5047_LAPTOP]	= "laptop",
 	[CXT5047_LAPTOP_HP]	= "laptop-hp",
 	[CXT5047_LAPTOP_EAPD]	= "laptop-eapd",
@@ -1995,7 +1995,7 @@ enum {
 	CXT5051_MODELS
 };
 
-static const char *cxt5051_models[CXT5051_MODELS] = {
+static const char *const cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_LAPTOP]	= "laptop",
 	[CXT5051_HP]		= "hp",
 	[CXT5051_HP_DV6736]	= "hp-dv6736",
@@ -3084,7 +3084,7 @@ enum {
 	CXT5066_MODELS
 };
 
-static const char *cxt5066_models[CXT5066_MODELS] = {
+static const char * const cxt5066_models[CXT5066_MODELS] = {
 	[CXT5066_LAPTOP]	= "laptop",
 	[CXT5066_DELL_LAPTOP]	= "dell-laptop",
 	[CXT5066_OLPC_XO_1_5]	= "olpc-xo-1_5",
@@ -3746,7 +3746,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	int i, err;
 	int num_line = 0, num_hp = 0, num_spk = 0;
-	static const char *texts[3] = { "Front", "Surround", "CLFE" };
+	static const char * const texts[3] = { "Front", "Surround", "CLFE" };
 
 	if (spec->dac_info_filled == 1)
 		return cx_auto_add_pb_volume(codec, spec->dac_info[0].dac,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 69554061c16..4f006eedd7e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2916,7 +2916,7 @@ static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
 /*
  * slave controls for virtual master
  */
-static const char *alc_slave_vols[] = {
+static const char * const alc_slave_vols[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
 	"Center Playback Volume",
@@ -2930,7 +2930,7 @@ static const char *alc_slave_vols[] = {
 	NULL,
 };
 
-static const char *alc_slave_sws[] = {
+static const char * const alc_slave_sws[] = {
 	"Front Playback Switch",
 	"Surround Playback Switch",
 	"Center Playback Switch",
@@ -4611,7 +4611,7 @@ static struct hda_verb alc880_test_init_verbs[] = {
 /*
  */
 
-static const char *alc880_models[ALC880_MODEL_LAST] = {
+static const char * const alc880_models[ALC880_MODEL_LAST] = {
 	[ALC880_3ST]		= "3stack",
 	[ALC880_TCL_S700]	= "tcl",
 	[ALC880_3ST_DIG]	= "3stack-digout",
@@ -5144,7 +5144,7 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg,
 static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
-	static const char *chname[4] = {
+	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
 	const char *pfx = alc_get_line_out_pfx(cfg, false);
@@ -7158,7 +7158,7 @@ static struct snd_pci_quirk alc260_fixup_tbl[] = {
 /*
  * ALC260 configurations
  */
-static const char *alc260_models[ALC260_MODEL_LAST] = {
+static const char * const alc260_models[ALC260_MODEL_LAST] = {
 	[ALC260_BASIC]		= "basic",
 	[ALC260_HP]		= "hp",
 	[ALC260_HP_3013]	= "hp-3013",
@@ -9781,7 +9781,7 @@ static hda_nid_t alc1200_slave_dig_outs[] = {
 /*
  * configuration and preset
  */
-static const char *alc882_models[ALC882_MODEL_LAST] = {
+static const char * const alc882_models[ALC882_MODEL_LAST] = {
 	[ALC882_3ST_DIG]	= "3stack-dig",
 	[ALC882_6ST_DIG]	= "6stack-dig",
 	[ALC882_ARIMA]		= "arima",
@@ -12601,7 +12601,7 @@ static void alc262_auto_init(struct hda_codec *codec)
 /*
  * configuration and preset
  */
-static const char *alc262_models[ALC262_MODEL_LAST] = {
+static const char * const alc262_models[ALC262_MODEL_LAST] = {
 	[ALC262_BASIC]		= "basic",
 	[ALC262_HIPPO]		= "hippo",
 	[ALC262_HIPPO_1]	= "hippo_1",
@@ -13789,7 +13789,7 @@ static void alc268_auto_init(struct hda_codec *codec)
 /*
  * configuration and preset
  */
-static const char *alc268_models[ALC268_MODEL_LAST] = {
+static const char * const alc268_models[ALC268_MODEL_LAST] = {
 	[ALC267_QUANTA_IL1]	= "quanta-il1",
 	[ALC268_3ST]		= "3stack",
 	[ALC268_TOSHIBA]	= "toshiba",
@@ -14961,7 +14961,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
 /*
  * configuration and preset
  */
-static const char *alc269_models[ALC269_MODEL_LAST] = {
+static const char * const alc269_models[ALC269_MODEL_LAST] = {
 	[ALC269_BASIC]			= "basic",
 	[ALC269_QUANTA_FL1]		= "quanta",
 	[ALC269_AMIC]			= "laptop-amic",
@@ -16004,7 +16004,7 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
 					     const struct auto_pin_cfg *cfg)
 {
 	struct alc_spec *spec = codec->spec;
-	static const char *chname[4] = {
+	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
 	const char *pfx = alc_get_line_out_pfx(cfg, true);
@@ -16210,7 +16210,7 @@ static struct hda_amp_list alc861_loopbacks[] = {
 /*
  * configuration and preset
  */
-static const char *alc861_models[ALC861_MODEL_LAST] = {
+static const char * const alc861_models[ALC861_MODEL_LAST] = {
 	[ALC861_3ST]		= "3stack",
 	[ALC660_3ST]		= "3stack-660",
 	[ALC861_3ST_DIG]	= "3stack-dig",
@@ -16913,7 +16913,7 @@ static void alc861vd_dallas_setup(struct hda_codec *codec)
 /*
  * configuration and preset
  */
-static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
+static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = {
 	[ALC660VD_3ST]		= "3stack-660",
 	[ALC660VD_3ST_DIG]	= "3stack-660-digout",
 	[ALC660VD_ASUS_V1S]	= "asus-v1s",
@@ -17133,7 +17133,9 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
-	static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"};
+	static const char * const chname[4] = {
+		"Front", "Surround", "CLFE", "Side"
+	};
 	const char *pfx = alc_get_line_out_pfx(cfg, true);
 	hda_nid_t nid_v, nid_s;
 	int i, err;
@@ -18688,7 +18690,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = {
 /*
  * configuration and preset
  */
-static const char *alc662_models[ALC662_MODEL_LAST] = {
+static const char * const alc662_models[ALC662_MODEL_LAST] = {
 	[ALC662_3ST_2ch_DIG]	= "3stack-dig",
 	[ALC662_3ST_6ch_DIG]	= "3stack-6ch-dig",
 	[ALC662_3ST_6ch]	= "3stack-6ch",
@@ -19203,7 +19205,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
 					     const struct auto_pin_cfg *cfg)
 {
 	struct alc_spec *spec = codec->spec;
-	static const char *chname[4] = {
+	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
 	const char *pfx = alc_get_line_out_pfx(cfg, true);
@@ -19978,7 +19980,7 @@ static void alc680_auto_init(struct hda_codec *codec)
 /*
  * configuration and preset
  */
-static const char *alc680_models[ALC680_MODEL_LAST] = {
+static const char * const alc680_models[ALC680_MODEL_LAST] = {
 	[ALC680_BASE]		= "base",
 	[ALC680_AUTO]		= "auto",
 };
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4ab019d0924..056f52df68c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -266,7 +266,7 @@ struct sigmatel_spec {
 	struct sigmatel_mic_route int_mic;
 	struct sigmatel_mic_route dock_mic;
 
-	const char **spdif_labels;
+	const char * const *spdif_labels;
 
 	hda_nid_t dig_in_nid;
 	hda_nid_t mono_nid;
@@ -524,7 +524,7 @@ static unsigned long stac927x_capsws[] = {
 	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
 };
 
-static const char *stac927x_spdif_labels[5] = {
+static const char * const stac927x_spdif_labels[5] = {
 	"Digital Playback", "ADAT", "Analog Mux 1",
 	"Analog Mux 2", "Analog Mux 3"
 };
@@ -1062,7 +1062,7 @@ static struct snd_kcontrol_new stac_smux_mixer = {
 	.put = stac92xx_smux_enum_put,
 };
 
-static const char *slave_vols[] = {
+static const char * const slave_vols[] = {
 	"Front Playback Volume",
 	"Surround Playback Volume",
 	"Center Playback Volume",
@@ -1073,7 +1073,7 @@ static const char *slave_vols[] = {
 	NULL
 };
 
-static const char *slave_sws[] = {
+static const char * const slave_sws[] = {
 	"Front Playback Switch",
 	"Surround Playback Switch",
 	"Center Playback Switch",
@@ -1354,7 +1354,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
 	[STAC_9200_PANASONIC] = ref9200_pin_configs,
 };
 
-static const char *stac9200_models[STAC_9200_MODELS] = {
+static const char * const stac9200_models[STAC_9200_MODELS] = {
 	[STAC_AUTO] = "auto",
 	[STAC_REF] = "ref",
 	[STAC_9200_OQO] = "oqo",
@@ -1500,7 +1500,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = {
 	[STAC_M6] = stac925xM6_pin_configs,
 };
 
-static const char *stac925x_models[STAC_925x_MODELS] = {
+static const char * const stac925x_models[STAC_925x_MODELS] = {
 	[STAC_925x_AUTO] = "auto",
 	[STAC_REF] = "ref",
 	[STAC_M1] = "m1",
@@ -1574,7 +1574,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_INTEL]	= intel_dg45id_pin_configs,
 };
 
-static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
+static const char * const stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_AUTO] = "auto",
 	[STAC_92HD73XX_NO_JD] = "no-jd",
 	[STAC_92HD73XX_REF] = "ref",
@@ -1660,7 +1660,7 @@ static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
 };
 
-static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
+static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_AUTO] = "auto",
 	[STAC_92HD83XXX_REF] = "ref",
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
@@ -1722,7 +1722,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 	[STAC_HP_DV4_1222NR]	= NULL,
 };
 
-static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
+static const char * const stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
 	[STAC_92HD71BXX_AUTO] = "auto",
 	[STAC_92HD71BXX_REF] = "ref",
 	[STAC_DELL_M4_1] = "dell-m4-1",
@@ -1915,7 +1915,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
 	[STAC_922X_DELL_M82] = dell_922x_m82_pin_configs,	
 };
 
-static const char *stac922x_models[STAC_922X_MODELS] = {
+static const char * const stac922x_models[STAC_922X_MODELS] = {
 	[STAC_922X_AUTO] = "auto",
 	[STAC_D945_REF]	= "ref",
 	[STAC_D945GTP5]	= "5stack",
@@ -2077,7 +2077,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 	[STAC_927X_VOLKNOB] = NULL,
 };
 
-static const char *stac927x_models[STAC_927X_MODELS] = {
+static const char * const stac927x_models[STAC_927X_MODELS] = {
 	[STAC_927X_AUTO]	= "auto",
 	[STAC_D965_REF_NO_JD]	= "ref-no-jd",
 	[STAC_D965_REF]		= "ref",
@@ -2180,7 +2180,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
 	[STAC_9205_EAPD] = NULL,
 };
 
-static const char *stac9205_models[STAC_9205_MODELS] = {
+static const char * const stac9205_models[STAC_9205_MODELS] = {
 	[STAC_9205_AUTO] = "auto",
 	[STAC_9205_REF] = "ref",
 	[STAC_9205_DELL_M42] = "dell-m42",
@@ -3123,7 +3123,7 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 				 int type)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	static const char *chname[4] = {
+	static const char * const chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
 	hda_nid_t nid;
@@ -3256,7 +3256,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 }
 
 /* labels for mono mux outputs */
-static const char *stac92xx_mono_labels[4] = {
+static const char * const stac92xx_mono_labels[4] = {
 	"DAC0", "DAC1", "Mixer", "DAC2"
 };
 
@@ -3380,7 +3380,7 @@ static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
 	return 0;
 };
 
-static const char *stac92xx_spdif_labels[3] = {
+static const char * const stac92xx_spdif_labels[3] = {
 	"Digital Playback", "Analog Mux 1", "Analog Mux 2",
 };
 
@@ -3388,7 +3388,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	struct hda_input_mux *spdif_mux = &spec->private_smux;
-	const char **labels = spec->spdif_labels;
+	const char * const *labels = spec->spdif_labels;
 	int i, num_cons;
 	hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
 
@@ -3409,7 +3409,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec)
 }
 
 /* labels for dmic mux inputs */
-static const char *stac92xx_dmic_labels[5] = {
+static const char * const stac92xx_dmic_labels[5] = {
 	"Analog Inputs", "Digital Mic 1", "Digital Mic 2",
 	"Digital Mic 3", "Digital Mic 4"
 };
@@ -6270,7 +6270,7 @@ static unsigned int stac9872_vaio_pin_configs[9] = {
 	0x90a7013e
 };
 
-static const char *stac9872_models[STAC_9872_MODELS] = {
+static const char * const stac9872_models[STAC_9872_MODELS] = {
 	[STAC_9872_AUTO] = "auto",
 	[STAC_9872_VAIO] = "vaio",
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index d1c3f8defc4..71f78456d68 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -2282,7 +2282,9 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+	static const char * const chname[4] = {
+		"Front", "Surround", "C/LFE", "Side"
+	};
 	hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b};
 	int i, err;
 
@@ -2371,7 +2373,7 @@ static void create_hp_imux(struct via_spec *spec)
 {
 	int i;
 	struct hda_input_mux *imux = &spec->private_imux[1];
-	static const char *texts[] = { "OFF", "ON", NULL};
+	static const char * const texts[] = { "OFF", "ON", NULL};
 
 	/* for hp mode select */
 	for (i = 0; texts[i]; i++)
@@ -2891,7 +2893,9 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+	static const char * const chname[4] = {
+		"Front", "Surround", "C/LFE", "Side"
+	};
 	hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29};
 	int i, err;
 
@@ -3434,7 +3438,9 @@ static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+	static const char * const chname[4] = {
+		"Front", "Surround", "C/LFE", "Side"
+	};
 	hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27};
 	hda_nid_t nid, nid_vol = 0;
 	int i, err;
@@ -3862,7 +3868,9 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+	static const char * const chname[4] = {
+		"Front", "Surround", "C/LFE", "Side"
+	};
 	hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25};
 	hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27};
 	hda_nid_t nid, nid_vol, nid_mute;
@@ -4305,7 +4313,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin)
 {
 	int err, i;
 	struct hda_input_mux *imux;
-	static const char *texts[] = { "ON", "OFF", NULL};
+	static const char * const texts[] = { "ON", "OFF", NULL};
 	if (!pin)
 		return 0;
 	spec->multiout.hp_nid = 0x1D;
@@ -4616,7 +4624,9 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" };
+	static const char * const chname[4] = {
+		"Front", "Surround", "C/LFE", "Side"
+	};
 	hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb};
 	hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27};
 	hda_nid_t nid, nid_vol, nid_mute = 0;
@@ -5065,7 +5075,9 @@ static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec,
 					      const struct auto_pin_cfg *cfg)
 {
 	char name[32];
-	static const char *chname[3] = { "Front", "Surround", "C/LFE" };
+	static const char * const chname[3] = {
+		"Front", "Surround", "C/LFE"
+	};
 	hda_nid_t nid_vols[] = {0x10, 0x11, 0x25};
 	hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27};
 	hda_nid_t nid, nid_vol, nid_mute;
-- 
cgit v1.2.3-70-g09d2


From 0f0714c5ed0a98fdeaa2287d3b159989bbe6d842 Mon Sep 17 00:00:00 2001
From: Bankim Bhavsar <bbhavsar@vmware.com>
Date: Mon, 17 Jan 2011 15:23:21 +0100
Subject: ALSA: hda - Add support for VMware controller

Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio
Controller.

[changed to use AZX_DRIVER_GENERIC by tiwai]

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a1c4008af89..07c522fd2b1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2806,6 +2806,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 #endif
 	/* Vortex86MX */
 	{ PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC },
+	/* VMware HDAudio */
+	{ PCI_DEVICE(0x15ad, 0x1977), .driver_data = AZX_DRIVER_GENERIC },
 	/* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */
 	{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
-- 
cgit v1.2.3-70-g09d2


From cbbf50b22f9693218f9f0d460432266b04fc960d Mon Sep 17 00:00:00 2001
From: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Date: Fri, 14 Jan 2011 17:21:13 -0600
Subject: ALSA: hda - Fix initialization for HP 2011 notebooks

Fixes for HP 2011 notebooks: enable dock ports and disable BTL
initialization in the driver.

Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 53 ++++++++----------------------------------
 1 file changed, 10 insertions(+), 43 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 056f52df68c..9ea48b425d0 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5333,7 +5333,7 @@ again:
 	return 0;
 }
 
-static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec)
+static int hp_bnb2011_with_dock(struct hda_codec *codec)
 {
 	if (codec->vendor_id != 0x111d7605 &&
 	    codec->vendor_id != 0x111d76d1)
@@ -5348,10 +5348,6 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec)
 	case 0x103c161d:
 	case 0x103c161e:
 	case 0x103c161f:
-	case 0x103c1620:
-	case 0x103c1621:
-	case 0x103c1622:
-	case 0x103c1623:
 
 	case 0x103c162a:
 	case 0x103c162b:
@@ -5360,41 +5356,9 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec)
 	case 0x103c1631:
 
 	case 0x103c1633:
-
+	case 0x103c1634:
 	case 0x103c1635:
 
-	case 0x103c164f:
-
-	case 0x103c1676:
-	case 0x103c1677:
-	case 0x103c1678:
-	case 0x103c1679:
-	case 0x103c167a:
-	case 0x103c167b:
-	case 0x103c167c:
-	case 0x103c167d:
-	case 0x103c167e:
-	case 0x103c167f:
-	case 0x103c1680:
-	case 0x103c1681:
-	case 0x103c1682:
-	case 0x103c1683:
-	case 0x103c1684:
-	case 0x103c1685:
-	case 0x103c1686:
-	case 0x103c1687:
-	case 0x103c1688:
-	case 0x103c1689:
-	case 0x103c168a:
-	case 0x103c168b:
-	case 0x103c168c:
-	case 0x103c168d:
-	case 0x103c168e:
-	case 0x103c168f:
-	case 0x103c1690:
-	case 0x103c1691:
-	case 0x103c1692:
-
 	case 0x103c3587:
 	case 0x103c3588:
 	case 0x103c3589:
@@ -5402,9 +5366,9 @@ static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec)
 
 	case 0x103c3667:
 	case 0x103c3668:
-		/* set BTL amp level to 13.43dB for louder speaker output */
-		return snd_hda_codec_write_cache(codec, codec->afg, 0,
-						 0x7F4, 0x14);
+	case 0x103c3669:
+
+		return 1;
 	}
 	return 0;
 }
@@ -5420,6 +5384,11 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	if (spec == NULL)
 		return -ENOMEM;
 
+	if (hp_bnb2011_with_dock(codec)) {
+		snd_hda_codec_set_pincfg(codec, 0xa, 0x2101201f);
+		snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e);
+	}
+
 	/* reset pin power-down; Windows may leave these bits after reboot */
 	snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7EC, 0);
 	snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0);
@@ -5546,8 +5515,6 @@ again:
 			AC_VERB_SET_CONNECT_SEL, num_dacs);
 	}
 
-	stac92hd83xxx_set_system_btl_amp(codec);
-
 	codec->proc_widget_hook = stac92hd_proc_hook;
 
 	return 0;
-- 
cgit v1.2.3-70-g09d2


From 569ed348ecef309fae5a71b86015951680ea3415 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Jan 2011 10:14:46 +0100
Subject: Revert "ALSA: HDA: Create mixers on ALC887"

This reverts commit 03b7a1ab557efe34e8f79b78660e514bd7374248.

This commit was mistakenly re-introduced.  While the change is harmless
(as ALC887 uses patch_alc888() now), we should get rid of any wrong code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 10 ++--------
 1 file changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 269dbff70b9..4f006eedd7e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10930,9 +10930,6 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
 	return 0;
 }
 
-static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
-					     const struct auto_pin_cfg *cfg);
-
 /* almost identical with ALC880 parser... */
 static int alc882_parse_auto_config(struct hda_codec *codec)
 {
@@ -10950,10 +10947,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 	err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	if (codec->vendor_id == 0x10ec0887)
-		err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg);
-	else
-		err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
 	err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
@@ -17134,7 +17128,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 #define alc861vd_idx_to_mixer_switch(nid)	((nid) + 0x0c)
 
 /* add playback controls from the parsed DAC table */
-/* Based on ALC880 version. But ALC861VD and ALC887 have separate,
+/* Based on ALC880 version. But ALC861VD has separate,
  * different NIDs for mute/unmute switch and volume control */
 static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
 					     const struct auto_pin_cfg *cfg)
-- 
cgit v1.2.3-70-g09d2


From 5734a07cbb8d4600a74a374c839620ddc62b2cf2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Jan 2011 17:07:12 +0100
Subject: ALSA: hda - Add quirk for HP Z-series workstation

It seems working well with model=hp-bpc.

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4f006eedd7e..7874023db0f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12629,6 +12629,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
 			   ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
+			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
-- 
cgit v1.2.3-70-g09d2


From aa1d0c5261f17d48636bf6d10bde0f38045511c0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Jan 2011 17:27:58 +0100
Subject: ALSA: hda - Fix "unused variable" compile warning
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

  sound/pci/hda/patch_realtek.c: In function ‘alc_apply_fixup’:
  sound/pci/hda/patch_realtek.c:1724:14: warning: unused variable ‘modelname’

snd_printdd() is evaluated only when CONFIG_SND_DEBUG_VERBOSE=y.

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

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7874023db0f..2b055e2780b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1721,7 +1721,9 @@ static void alc_apply_fixup(struct hda_codec *codec, int action)
 {
 	struct alc_spec *spec = codec->spec;
 	int id = spec->fixup_id;
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 	const char *modelname = spec->fixup_name;
+#endif
 	int depth = 0;
 
 	if (!spec->fixup_list)
-- 
cgit v1.2.3-70-g09d2


From fb228af7060d02a81a7bcc2ce329ba3ab1af0c7f Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 19 Jan 2011 11:59:01 +0100
Subject: ALSA: HDA: Add SKU ignore for another Thinkpad Edge 14

BugLink: http://bugs.launchpad.net/bugs/705323

Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute.

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/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2b055e2780b..5ea60c6d24a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -14955,6 +14955,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
-- 
cgit v1.2.3-70-g09d2


From d2ebd4798744c401faf3fdc6493383912ccd0b80 Mon Sep 17 00:00:00 2001
From: Anisse Astier <anisse@astier.eu>
Date: Thu, 20 Jan 2011 12:36:21 +0100
Subject: ALSA: hda - Fix EAPD to low on CZC P10T tablet computer with ALC662

Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 ++++++++++-
 1 file changed, 10 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5ea60c6d24a..be4df4c6fd5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -19460,6 +19460,7 @@ enum {
 	ALC662_FIXUP_ASPIRE,
 	ALC662_FIXUP_IDEAPAD,
 	ALC272_FIXUP_MARIO,
+	ALC662_FIXUP_CZC_P10T,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -19480,7 +19481,14 @@ static const struct alc_fixup alc662_fixups[] = {
 	[ALC272_FIXUP_MARIO] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc272_fixup_mario,
-	}
+	},
+	[ALC662_FIXUP_CZC_P10T] = {
+		.type = ALC_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{0x14, AC_VERB_SET_EAPD_BTLENABLE, 0},
+			{}
+		}
+	},
 };
 
 static struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -19488,6 +19496,7 @@ static struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
+	SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
 	{}
 };
 
-- 
cgit v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9bb030a469c..7cd59b9f0e9 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 v1.2.3-70-g09d2


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/hda')

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 16ae4300c74..0caf77e59be 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 7cd59b9f0e9..19f0daf6497 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 19f0daf6497..9867afc7895 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 4a663471dad..74b0560289c 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index be4df4c6fd5..2fa9ed99c32 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9867afc7895..7e1ca43bd66 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 7e1ca43bd66..fbe97d32140 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2fa9ed99c32..2571d977df2 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 2d5b83fa8d2..a5876773672 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2e91a991eb1..0baffcdee8f 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2571d977df2..089a7de2439 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 v1.2.3-70-g09d2


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/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 089a7de2439..3328a259a24 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 v1.2.3-70-g09d2