From 1d3c16a818e992c199844954d95c17fd7ce6cbba Mon Sep 17 00:00:00 2001
From: Jon Mason <jon.mason@exar.com>
Date: Tue, 30 Nov 2010 17:43:26 -0600
Subject: PCI: make pci_restore_state return void

pci_restore_state only ever returns 0, thus there is no benefit in
having it return any value.  Also, a large majority of the callers do
not check the return code of pci_restore_state.  Make the
pci_restore_state a void return and avoid the overhead.

Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Signed-off-by: Jon Mason <jon.mason@exar.com>
Signed-off-by: Jesse Barnes <jbarnes@virtuousgeek.org>
---
 sound/pci/cs5535audio/cs5535audio_pm.c | 7 +------
 1 file changed, 1 insertion(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index a3301cc4ab8..185b0008832 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -90,12 +90,7 @@ int snd_cs5535audio_resume(struct pci_dev *pci)
 	int i;
 
 	pci_set_power_state(pci, PCI_D0);
-	if (pci_restore_state(pci) < 0) {
-		printk(KERN_ERR "cs5535audio: pci_restore_state failed, "
-		       "disabling device\n");
-		snd_card_disconnect(card);
-		return -EIO;
-	}
+	pci_restore_state(pci);
 	if (pci_enable_device(pci) < 0) {
 		printk(KERN_ERR "cs5535audio: pci_enable_device failed, "
 		       "disabling device\n");
-- 
cgit v1.2.3-70-g09d2


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')

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')

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')

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')

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')

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')

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 f8fe80e4383bf5f542beb80bf2abe9fc1505c366 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Fri, 14 Jan 2011 08:07:50 +0100
Subject: ALSA: oxygen: Xonar DG: fix CS4245 register writes

Accidentally exchanging register addresses and register values leads to
many strange errors ...

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/xonar_dg.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e4de0b8d087..e1fa602eba7 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -75,7 +75,7 @@ static void cs4245_write(struct oxygen *chip, unsigned int reg, u8 value)
 			 OXYGEN_SPI_CEN_LATCH_CLOCK_HI,
 			 CS4245_SPI_ADDRESS |
 			 CS4245_SPI_WRITE |
-			 (value << 8) | reg);
+			 (reg << 8) | value);
 	data->cs4245_regs[reg] = value;
 }
 
-- 
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')

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')

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')

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 228dd545147a07c5e81e4732ad0e829d19ce5daa Mon Sep 17 00:00:00 2001
From: "Matti J. Aaltonen" <matti.j.aaltonen@nokia.com>
Date: Thu, 13 Jan 2011 15:22:45 +0200
Subject: ASoC: WL1273 FM radio: Fix breakage with MFD API changes

These changes are needed to keep up with the changes in the
MFD core and V4L2 parts of the wl1273 FM radio driver.

Use function pointers instead of exported functions for I2C IO.
Also move all preprocessor constants from the wl1273.h to
include/linux/mfd/wl1273-core.h.

Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/Kconfig  |  2 +-
 sound/soc/codecs/wl1273.c | 29 ++++++++-----------
 sound/soc/codecs/wl1273.h | 71 -----------------------------------------------
 3 files changed, 13 insertions(+), 89 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 883a312bb29..c48b23c1d4f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -44,7 +44,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TWL6040 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
-	select SND_SOC_WL1273 if WL1273_CORE
+	select SND_SOC_WL1273 if RADIO_WL1273
 	select SND_SOC_WM2000 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index d3ffa2f0122..861b28f543d 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -42,7 +42,7 @@ struct wl1273_priv {
 static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core,
 				      int rate, int width)
 {
-	struct device *dev = &core->i2c_dev->dev;
+	struct device *dev = &core->client->dev;
 	int r = 0;
 	u16 mode;
 
@@ -123,13 +123,13 @@ static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core,
 	dev_dbg(dev, "mode: 0x%04x\n", mode);
 
 	if (core->i2s_mode != mode) {
-		r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode);
+		r = core->write(core, WL1273_I2S_MODE_CONFIG_SET, mode);
 		if (r)
 			goto out;
 
 		core->i2s_mode = mode;
-		r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE,
-					WL1273_AUDIO_ENABLE_I2S);
+		r = core->write(core, WL1273_AUDIO_ENABLE,
+				WL1273_AUDIO_ENABLE_I2S);
 		if (r)
 			goto out;
 	}
@@ -142,8 +142,7 @@ out:
 static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core,
 					    int channel_number)
 {
-	struct i2c_client *client = core->i2c_dev;
-	struct device *dev = &client->dev;
+	struct device *dev = &core->client->dev;
 	int r = 0;
 
 	dev_dbg(dev, "%s\n", __func__);
@@ -154,17 +153,13 @@ static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core,
 		goto out;
 
 	if (channel_number == 1 && core->mode == WL1273_MODE_RX)
-		r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
-					WL1273_RX_MONO);
+		r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_MONO);
 	else if (channel_number == 1 && core->mode == WL1273_MODE_TX)
-		r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
-					WL1273_TX_MONO);
+		r = core->write(core, WL1273_MONO_SET, WL1273_TX_MONO);
 	else if (channel_number == 2 && core->mode == WL1273_MODE_RX)
-		r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
-					WL1273_RX_STEREO);
+		r = core->write(core, WL1273_MOST_MODE_SET, WL1273_RX_STEREO);
 	else if (channel_number == 2 && core->mode == WL1273_MODE_TX)
-		r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
-					WL1273_TX_STEREO);
+		r = core->write(core, WL1273_MONO_SET, WL1273_TX_STEREO);
 	else
 		r = -EINVAL;
 out:
@@ -237,7 +232,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
 	if (wl1273->core->audio_mode == val)
 		return 0;
 
-	r = wl1273_fm_set_audio(wl1273->core, val);
+	r = wl1273->core->set_audio(wl1273->core, val);
 	if (r < 0)
 		return r;
 
@@ -272,8 +267,8 @@ static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol,
 
 	dev_dbg(codec->dev, "%s: enter.\n", __func__);
 
-	r = wl1273_fm_set_volume(wl1273->core,
-				 ucontrol->value.integer.value[0]);
+	r = wl1273->core->set_volume(wl1273->core,
+				     ucontrol->value.integer.value[0]);
 	if (r)
 		return r;
 
diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h
index 14ed027fdcf..43ec7e668c5 100644
--- a/sound/soc/codecs/wl1273.h
+++ b/sound/soc/codecs/wl1273.h
@@ -25,77 +25,6 @@
 #ifndef __WL1273_CODEC_H__
 #define __WL1273_CODEC_H__
 
-/* I2S protocol, left channel first, data width 16 bits */
-#define WL1273_PCM_DEF_MODE		0x00
-
-/* Rx */
-#define WL1273_AUDIO_ENABLE_I2S		(1 << 0)
-#define WL1273_AUDIO_ENABLE_ANALOG	(1 << 1)
-
-/* Tx */
-#define WL1273_AUDIO_IO_SET_ANALOG	0
-#define WL1273_AUDIO_IO_SET_I2S		1
-
-#define WL1273_POWER_SET_OFF		0
-#define WL1273_POWER_SET_FM		(1 << 0)
-#define WL1273_POWER_SET_RDS		(1 << 1)
-#define WL1273_POWER_SET_RETENTION	(1 << 4)
-
-#define WL1273_PUPD_SET_OFF		0x00
-#define WL1273_PUPD_SET_ON		0x01
-#define WL1273_PUPD_SET_RETENTION	0x10
-
-/* I2S mode */
-#define WL1273_IS2_WIDTH_32	0x0
-#define WL1273_IS2_WIDTH_40	0x1
-#define WL1273_IS2_WIDTH_22_23	0x2
-#define WL1273_IS2_WIDTH_23_22	0x3
-#define WL1273_IS2_WIDTH_48	0x4
-#define WL1273_IS2_WIDTH_50	0x5
-#define WL1273_IS2_WIDTH_60	0x6
-#define WL1273_IS2_WIDTH_64	0x7
-#define WL1273_IS2_WIDTH_80	0x8
-#define WL1273_IS2_WIDTH_96	0x9
-#define WL1273_IS2_WIDTH_128	0xa
-#define WL1273_IS2_WIDTH	0xf
-
-#define WL1273_IS2_FORMAT_STD	(0x0 << 4)
-#define WL1273_IS2_FORMAT_LEFT	(0x1 << 4)
-#define WL1273_IS2_FORMAT_RIGHT	(0x2 << 4)
-#define WL1273_IS2_FORMAT_USER	(0x3 << 4)
-
-#define WL1273_IS2_MASTER	(0x0 << 6)
-#define WL1273_IS2_SLAVEW	(0x1 << 6)
-
-#define WL1273_IS2_TRI_AFTER_SENDING	(0x0 << 7)
-#define WL1273_IS2_TRI_ALWAYS_ACTIVE	(0x1 << 7)
-
-#define WL1273_IS2_SDOWS_RR	(0x0 << 8)
-#define WL1273_IS2_SDOWS_RF	(0x1 << 8)
-#define WL1273_IS2_SDOWS_FR	(0x2 << 8)
-#define WL1273_IS2_SDOWS_FF	(0x3 << 8)
-
-#define WL1273_IS2_TRI_OPT	(0x0 << 10)
-#define WL1273_IS2_TRI_ALWAYS	(0x1 << 10)
-
-#define WL1273_IS2_RATE_48K	(0x0 << 12)
-#define WL1273_IS2_RATE_44_1K	(0x1 << 12)
-#define WL1273_IS2_RATE_32K	(0x2 << 12)
-#define WL1273_IS2_RATE_22_05K	(0x4 << 12)
-#define WL1273_IS2_RATE_16K	(0x5 << 12)
-#define WL1273_IS2_RATE_12K	(0x8 << 12)
-#define WL1273_IS2_RATE_11_025	(0x9 << 12)
-#define WL1273_IS2_RATE_8K	(0xa << 12)
-#define WL1273_IS2_RATE		(0xf << 12)
-
-#define WL1273_I2S_DEF_MODE	(WL1273_IS2_WIDTH_32 | \
-				 WL1273_IS2_FORMAT_STD | \
-				 WL1273_IS2_MASTER | \
-				 WL1273_IS2_TRI_AFTER_SENDING | \
-				 WL1273_IS2_SDOWS_RR | \
-				 WL1273_IS2_TRI_OPT | \
-				 WL1273_IS2_RATE_48K)
-
 int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt);
 
 #endif	/* End of __WL1273_CODEC_H__ */
-- 
cgit v1.2.3-70-g09d2


From 3e8b3b90fecedcf20d895c4e6ad01a379fe252bf Mon Sep 17 00:00:00 2001
From: Hanno Boeck <hanno@hboeck.de>
Date: Fri, 14 Jan 2011 19:14:47 +0100
Subject: ALSA: constify functions in ac97

Signed-off-by: Hanno Boeck <hanno@hboeck.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/ac97_codec.h  |  2 +-
 sound/pci/ac97/ac97_codec.c |  2 +-
 sound/pci/ac97/ac97_patch.c | 62 ++++++++++++++++++++++-----------------------
 3 files changed, 33 insertions(+), 33 deletions(-)

(limited to 'sound')

diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 49400459b47..b602f475cdb 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -477,7 +477,7 @@ struct snd_ac97_template {
 
 struct snd_ac97 {
 	/* -- lowlevel (hardware) driver specific -- */
-	struct snd_ac97_build_ops * build_ops;
+	const struct snd_ac97_build_ops *build_ops;
 	void *private_data;
 	void (*private_free) (struct snd_ac97 *ac97);
 	/* --- */
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 0fc614ce16c..cb62d178b3e 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1961,7 +1961,7 @@ static int snd_ac97_dev_disconnect(struct snd_device *device)
 }
 
 /* build_ops to do nothing */
-static struct snd_ac97_build_ops null_build_ops;
+static const struct snd_ac97_build_ops null_build_ops;
 
 #ifdef CONFIG_SND_AC97_POWER_SAVE
 static void do_update_power(struct work_struct *work)
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index e68c98ef404..bf47574ca1f 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -371,7 +371,7 @@ static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 *ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = {
+static const struct snd_ac97_build_ops patch_yamaha_ymf743_ops = {
 	.build_spdif	= patch_yamaha_ymf743_build_spdif,
 	.build_3d	= patch_yamaha_ymf7x3_3d,
 };
@@ -455,7 +455,7 @@ static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = {
+static const struct snd_ac97_build_ops patch_yamaha_ymf753_ops = {
 	.build_3d	= patch_yamaha_ymf7x3_3d,
 	.build_post_spdif = patch_yamaha_ymf753_post_spdif
 };
@@ -502,7 +502,7 @@ static int patch_wolfson_wm9703_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_wolfson_wm9703_ops = {
+static const struct snd_ac97_build_ops patch_wolfson_wm9703_ops = {
 	.build_specific = patch_wolfson_wm9703_specific,
 };
 
@@ -533,7 +533,7 @@ static int patch_wolfson_wm9704_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_wolfson_wm9704_ops = {
+static const struct snd_ac97_build_ops patch_wolfson_wm9704_ops = {
 	.build_specific = patch_wolfson_wm9704_specific,
 };
 
@@ -677,7 +677,7 @@ static int patch_wolfson_wm9711_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_wolfson_wm9711_ops = {
+static const struct snd_ac97_build_ops patch_wolfson_wm9711_ops = {
 	.build_specific = patch_wolfson_wm9711_specific,
 };
 
@@ -871,7 +871,7 @@ static void patch_wolfson_wm9713_resume (struct snd_ac97 * ac97)
 }
 #endif
 
-static struct snd_ac97_build_ops patch_wolfson_wm9713_ops = {
+static const struct snd_ac97_build_ops patch_wolfson_wm9713_ops = {
 	.build_specific = patch_wolfson_wm9713_specific,
 	.build_3d = patch_wolfson_wm9713_3d,
 #ifdef CONFIG_PM	
@@ -976,7 +976,7 @@ static int patch_sigmatel_stac97xx_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = {
+static const struct snd_ac97_build_ops patch_sigmatel_stac9700_ops = {
 	.build_3d	= patch_sigmatel_stac9700_3d,
 	.build_specific	= patch_sigmatel_stac97xx_specific
 };
@@ -1023,7 +1023,7 @@ static int patch_sigmatel_stac9708_specific(struct snd_ac97 *ac97)
 	return patch_sigmatel_stac97xx_specific(ac97);
 }
 
-static struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = {
+static const struct snd_ac97_build_ops patch_sigmatel_stac9708_ops = {
 	.build_3d	= patch_sigmatel_stac9708_3d,
 	.build_specific	= patch_sigmatel_stac9708_specific
 };
@@ -1252,7 +1252,7 @@ static int patch_sigmatel_stac9758_specific(struct snd_ac97 *ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = {
+static const struct snd_ac97_build_ops patch_sigmatel_stac9758_ops = {
 	.build_3d	= patch_sigmatel_stac9700_3d,
 	.build_specific	= patch_sigmatel_stac9758_specific
 };
@@ -1327,7 +1327,7 @@ static int patch_cirrus_build_spdif(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_cirrus_ops = {
+static const struct snd_ac97_build_ops patch_cirrus_ops = {
 	.build_spdif = patch_cirrus_build_spdif
 };
 
@@ -1384,7 +1384,7 @@ static int patch_conexant_build_spdif(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_conexant_ops = {
+static const struct snd_ac97_build_ops patch_conexant_ops = {
 	.build_spdif = patch_conexant_build_spdif
 };
 
@@ -1560,7 +1560,7 @@ static void patch_ad1881_chained(struct snd_ac97 * ac97, int unchained_idx, int
 	}
 }
 
-static struct snd_ac97_build_ops patch_ad1881_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1881_build_ops = {
 #ifdef CONFIG_PM
 	.resume = ad18xx_resume
 #endif
@@ -1647,7 +1647,7 @@ static int patch_ad1885_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_ad1885_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1885_build_ops = {
 	.build_specific = &patch_ad1885_specific,
 #ifdef CONFIG_PM
 	.resume = ad18xx_resume
@@ -1674,7 +1674,7 @@ static int patch_ad1886_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_ad1886_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1886_build_ops = {
 	.build_specific = &patch_ad1886_specific,
 #ifdef CONFIG_PM
 	.resume = ad18xx_resume
@@ -1881,7 +1881,7 @@ static int patch_ad1981a_specific(struct snd_ac97 * ac97)
 				    ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
 }
 
-static struct snd_ac97_build_ops patch_ad1981a_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1981a_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1981a_specific,
 #ifdef CONFIG_PM
@@ -1936,7 +1936,7 @@ static int patch_ad1981b_specific(struct snd_ac97 *ac97)
 				    ARRAY_SIZE(snd_ac97_ad1981x_jack_sense));
 }
 
-static struct snd_ac97_build_ops patch_ad1981b_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1981b_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1981b_specific,
 #ifdef CONFIG_PM
@@ -2075,7 +2075,7 @@ static int patch_ad1888_specific(struct snd_ac97 *ac97)
 	return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls));
 }
 
-static struct snd_ac97_build_ops patch_ad1888_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1888_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1888_specific,
 #ifdef CONFIG_PM
@@ -2124,7 +2124,7 @@ static int patch_ad1980_specific(struct snd_ac97 *ac97)
 	return patch_build_controls(ac97, &snd_ac97_ad198x_2cmic, 1);
 }
 
-static struct snd_ac97_build_ops patch_ad1980_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1980_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1980_specific,
 #ifdef CONFIG_PM
@@ -2239,7 +2239,7 @@ static int patch_ad1985_specific(struct snd_ac97 *ac97)
 				    ARRAY_SIZE(snd_ac97_ad1985_controls));
 }
 
-static struct snd_ac97_build_ops patch_ad1985_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1985_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1985_specific,
 #ifdef CONFIG_PM
@@ -2531,7 +2531,7 @@ static int patch_ad1986_specific(struct snd_ac97 *ac97)
 				    ARRAY_SIZE(snd_ac97_ad1985_controls));
 }
 
-static struct snd_ac97_build_ops patch_ad1986_build_ops = {
+static const struct snd_ac97_build_ops patch_ad1986_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1986_specific,
 #ifdef CONFIG_PM
@@ -2636,7 +2636,7 @@ static int patch_alc650_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_alc650_ops = {
+static const struct snd_ac97_build_ops patch_alc650_ops = {
 	.build_specific	= patch_alc650_specific,
 	.update_jacks = alc650_update_jacks
 };
@@ -2788,7 +2788,7 @@ static int patch_alc655_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_alc655_ops = {
+static const struct snd_ac97_build_ops patch_alc655_ops = {
 	.build_specific	= patch_alc655_specific,
 	.update_jacks = alc655_update_jacks
 };
@@ -2900,7 +2900,7 @@ static int patch_alc850_specific(struct snd_ac97 *ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_alc850_ops = {
+static const struct snd_ac97_build_ops patch_alc850_ops = {
 	.build_specific	= patch_alc850_specific,
 	.update_jacks = alc850_update_jacks
 };
@@ -2962,7 +2962,7 @@ static int patch_cm9738_specific(struct snd_ac97 * ac97)
 	return patch_build_controls(ac97, snd_ac97_cm9738_controls, ARRAY_SIZE(snd_ac97_cm9738_controls));
 }
 
-static struct snd_ac97_build_ops patch_cm9738_ops = {
+static const struct snd_ac97_build_ops patch_cm9738_ops = {
 	.build_specific	= patch_cm9738_specific,
 	.update_jacks = cm9738_update_jacks
 };
@@ -3053,7 +3053,7 @@ static int patch_cm9739_post_spdif(struct snd_ac97 * ac97)
 	return patch_build_controls(ac97, snd_ac97_cm9739_controls_spdif, ARRAY_SIZE(snd_ac97_cm9739_controls_spdif));
 }
 
-static struct snd_ac97_build_ops patch_cm9739_ops = {
+static const struct snd_ac97_build_ops patch_cm9739_ops = {
 	.build_specific	= patch_cm9739_specific,
 	.build_post_spdif = patch_cm9739_post_spdif,
 	.update_jacks = cm9739_update_jacks
@@ -3227,7 +3227,7 @@ static int patch_cm9761_specific(struct snd_ac97 * ac97)
 	return patch_build_controls(ac97, snd_ac97_cm9761_controls, ARRAY_SIZE(snd_ac97_cm9761_controls));
 }
 
-static struct snd_ac97_build_ops patch_cm9761_ops = {
+static const struct snd_ac97_build_ops patch_cm9761_ops = {
 	.build_specific	= patch_cm9761_specific,
 	.build_post_spdif = patch_cm9761_post_spdif,
 	.update_jacks = cm9761_update_jacks
@@ -3323,7 +3323,7 @@ static int patch_cm9780_specific(struct snd_ac97 *ac97)
 	return patch_build_controls(ac97, cm9780_controls, ARRAY_SIZE(cm9780_controls));
 }
 
-static struct snd_ac97_build_ops patch_cm9780_ops = {
+static const struct snd_ac97_build_ops patch_cm9780_ops = {
 	.build_specific	= patch_cm9780_specific,
 	.build_post_spdif = patch_cm9761_post_spdif	/* identical with CM9761 */
 };
@@ -3443,7 +3443,7 @@ static int patch_vt1616_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_vt1616_ops = {
+static const struct snd_ac97_build_ops patch_vt1616_ops = {
 	.build_specific	= patch_vt1616_specific
 };
 
@@ -3797,7 +3797,7 @@ static int patch_it2646_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_it2646_ops = {
+static const struct snd_ac97_build_ops patch_it2646_ops = {
 	.build_specific	= patch_it2646_specific,
 	.update_jacks = it2646_update_jacks
 };
@@ -3831,7 +3831,7 @@ static int patch_si3036_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_si3036_ops = {
+static const struct snd_ac97_build_ops patch_si3036_ops = {
 	.build_specific	= patch_si3036_specific,
 };
 
@@ -3898,7 +3898,7 @@ static int patch_ucb1400_specific(struct snd_ac97 * ac97)
 	return 0;
 }
 
-static struct snd_ac97_build_ops patch_ucb1400_ops = {
+static const struct snd_ac97_build_ops patch_ucb1400_ops = {
 	.build_specific	= patch_ucb1400_specific,
 };
 
-- 
cgit v1.2.3-70-g09d2


From d9ab344336f74c012f6643ed3d1ad8ca0136de3b Mon Sep 17 00:00:00 2001
From: Raymond Yau <superquad.vortex2@gmail.com>
Date: Sun, 16 Jan 2011 10:55:54 +0800
Subject: ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu

Fix playback/capture channels patch to change supported playback
channels of au8830 to 1,2,4 and capture channels to 1,2.
This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to
set 3 Channels

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0_pcm.c | 24 ++++++++++++++++++++----
 1 file changed, 20 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index b9d2f202cf9..5439d662d10 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -42,11 +42,7 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = {
 	.rate_min = 5000,
 	.rate_max = 48000,
 	.channels_min = 1,
-#ifdef CHIP_AU8830
-	.channels_max = 4,
-#else
 	.channels_max = 2,
-#endif
 	.buffer_bytes_max = 0x10000,
 	.period_bytes_min = 0x1,
 	.period_bytes_max = 0x1000,
@@ -115,6 +111,17 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_wt = {
 	.periods_max = 64,
 };
 #endif
+#ifdef CHIP_AU8830
+static unsigned int au8830_channels[3] = {
+	1, 2, 4,
+};
+
+static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = {
+	.count = ARRAY_SIZE(au8830_channels),
+	.list = au8830_channels,
+	.mask = 0,
+};
+#endif
 /* open callback */
 static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
 {
@@ -156,6 +163,15 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
 		if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB
 		    || VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_I2S)
 			runtime->hw = snd_vortex_playback_hw_adb;
+#ifdef CHIP_AU8830
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+			VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) {
+			runtime->hw.channels_max = 4;
+			snd_pcm_hw_constraint_list(runtime, 0,
+				SNDRV_PCM_HW_PARAM_CHANNELS,
+				&hw_constraints_au8830_channels);
+		}
+#endif
 		substream->runtime->private_data = NULL;
 	}
 #ifndef CHIP_AU8810
-- 
cgit v1.2.3-70-g09d2


From 7ebcf5d6021a696680ee77d9162a2edec2d671dd Mon Sep 17 00:00:00 2001
From: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Date: Fri, 14 Jan 2011 15:59:13 +0000
Subject: ASoC: WM8990: msleep() takes milliseconds not jiffies

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8990.c | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 5c87a634fc0..100aeee5ba9 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1183,7 +1183,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 				     WM8990_VMIDTOG);
 
 			/* Delay to allow output caps to discharge */
-			msleep(msecs_to_jiffies(300));
+			msleep(300);
 
 			/* Disable VMIDTOG */
 			snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
@@ -1195,17 +1195,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 			/* Enable outputs */
 			snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00);
 
-			msleep(msecs_to_jiffies(50));
+			msleep(50);
 
 			/* Enable VMID at 2x50k */
 			snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02);
 
-			msleep(msecs_to_jiffies(100));
+			msleep(100);
 
 			/* Enable VREF */
 			snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
 
-			msleep(msecs_to_jiffies(600));
+			msleep(600);
 
 			/* Enable BUFIOEN */
 			snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
@@ -1250,7 +1250,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 		/* Disable VMID */
 		snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01);
 
-		msleep(msecs_to_jiffies(300));
+		msleep(300);
 
 		/* Enable all output discharge bits */
 		snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
-- 
cgit v1.2.3-70-g09d2


From 7322ce21cde92777a9b11e17429d61d1cda6d2c2 Mon Sep 17 00:00:00 2001
From: Alexander Sverdlin <subaparts@yandex.ru>
Date: Sun, 16 Jan 2011 15:48:05 +0300
Subject: ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code

Changelog:
1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is
hardware limitation and that's the way original Cirrus's driver worked.
This will fix distorted sound playback and make capture actually work in
present ep93xx drivers.

I've found, that author of code, on which modern ep93xx-i2s.c and
ep93xx-pcm.c are based, had faced this problem also in 2007:
http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3

Now SoC code uses his developments, but not overcomes the hardware
issues. Some details from EP93xx users guide:

Both I2S transmitter and receiver have similar 16x32bit FIFO, where they
store 8 samples for both left and right channels. The FIFO is always
32bit wide and should be properly aligned if you use samples of other
width. Transmitter and receiver have configuration registers for
selection of I2S word length (16, 24, 32). They are I2STXWrdLen and
I2SRXWrdLen.

Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for
transfers to and from peripherals is selected by particular module
configuration. Lucky AC97 module has such configuration: AC97RXCRx
registers, bit CM (Compact mode enable) switches between 16 and 32 bit
samples. AC97TXCRx registers have the same bits for transmitters.
ep93xx-ac97.c enables this compact mode and so has all the rights to use
S16_LE format.
No one has found such a configuration in I2S module until now in any
Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit
samples consecutively for left and right channels. You cannot use 32-bit
DMA transfers to transfer two 16-bit samples.

So we can use two formats for AC97, but should remove all but S32_LE for
I2S. Always using 32 bit chunks is not a problem for I2S, the codec I
use uses less bits too (24), it's permitted by I2S standard.

In proposed patch formats list shortened to just S32_LE, this makes all
the DMA transactions right, while ALSA will do all sample format
translation for us.

2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c
masks the first problem.

DMA takes two 16 bit samples instead of one, overall sound speed seems
to be normal, but you get actually 4000 sampling rate instead of
requested 8000 and therefore some noise... This is also the reason why
the capture function not worked at all in this driver...

If we take a look into I2S specification, we will figure that LRCLK MUST
be equal to sample rate, if we are talking about stereo (in mono too,
but it's not our case at all).

In proposed patch SCLK and LRCLK rates are corrected, assuming we always
send 32 bits * 2 channels to codec.

Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/ep93xx/ep93xx-i2s.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 9ac93f6b4f8..fff579a1c13 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -267,14 +267,16 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
 		ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len);
 
 	/*
-	 * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values.
-	 * If the lrclk is pulse length is larger than the word size, then the
-	 * bit clock will be gated for the unused bits.
+	 * EP93xx I2S module can be setup so SCLK / LRCLK value can be
+	 * 32, 64, 128. MCLK / SCLK value can be 2 and 4.
+	 * We set LRCLK equal to `rate' and minimum SCLK / LRCLK 
+	 * value is 64, because our sample size is 32 bit * 2 channels.
+	 * I2S standard permits us to transmit more bits than
+	 * the codec uses.
 	 */
-	div = (clk_get_rate(info->mclk) / params_rate(params)) *
-		params_channels(params);
+	div = clk_get_rate(info->mclk) / params_rate(params);
 	for (sdiv = 2; sdiv <= 4; sdiv += 2)
-		for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1)
+		for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1)
 			if (sdiv * lrdiv == div) {
 				found = 1;
 				goto out;
@@ -341,9 +343,7 @@ static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = {
 	.set_fmt	= ep93xx_i2s_set_dai_fmt,
 };
 
-#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
-			    SNDRV_PCM_FMTBIT_S24_LE | \
-			    SNDRV_PCM_FMTBIT_S32_LE)
+#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S32_LE)
 
 static struct snd_soc_dai_driver ep93xx_i2s_dai = {
 	.symmetric_rates= 1,
-- 
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')

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')

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')

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')

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 b8b1a4cb6842fb33769be1ad636f062d31d588c3 Mon Sep 17 00:00:00 2001
From: Brian Bloniarz <brian.bloniarz@gmail.com>
Date: Mon, 17 Jan 2011 23:20:03 -0800
Subject: ALSA: ice1712 delta - initialize SPI clock

The driver was using an initial value for the clock on the SPI bus
which was read from ICE1712 EEPROM,
ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02)

It appears some cards have it default high, some cards
have it default low. On my Delta 66 rev. E:
$ cat /proc/asound/M66/ice1712 | grep 'GPIO state'
  GPIO state       : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */
On my Audiophile 2496:
$ cat /proc/asound/M2496/ice1712 | grep 'GPIO state'
  GPIO state       : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */

It must be raised before the first SPI write happens, or the write will
fail, leading to:

[   23.248721] invalid CS8427 signature 0x0: let me try again...

I theorize that 4eb4550ab37d351ab0973ccec921a5a2d8560ec7
is no longer needed, it was a different way to workaround
the problem.

[fixed variable decleration by tiwai]

Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ice1712/delta.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index 7b62de089fe..20c6b079d0d 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -580,6 +580,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
 {
 	int err;
 	struct snd_akm4xxx *ak;
+	unsigned char tmp;
 
 	if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DELTA1010 &&
 	    ice->eeprom.gpiodir == 0x7b)
@@ -622,6 +623,12 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice)
 		break;
 	}
 
+	/* initialize the SPI clock to high */
+	tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA);
+	tmp |= ICE1712_DELTA_AP_CCLK;
+	snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp);
+	udelay(5);
+
 	/* initialize spdif */
 	switch (ice->eeprom.subvendor) {
 	case ICE1712_SUBDEVICE_AUDIOPHILE:
-- 
cgit v1.2.3-70-g09d2


From 15d2e22b820bad62854d6ad99d8af8320adf4a91 Mon Sep 17 00:00:00 2001
From: Mike Frysinger <vapier@gentoo.org>
Date: Tue, 11 Jan 2011 23:08:19 -0500
Subject: ASoC: Blackfin TDM: fix missed snd_soc_dai_get_drvdata update

One spot was missed in this driver when converting from
snd_soc_dai.private_data to snd_soc_dai_get_drvdata.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/blackfin/bf5xx-tdm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 125123929f1..b2cf239f20b 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -210,7 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
 #ifdef CONFIG_PM
 static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
 {
-	struct sport_device *sport = dai->private_data;
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
 		return 0;
-- 
cgit v1.2.3-70-g09d2


From e9c2048915048d605fd76539ddd96f00d593e1eb Mon Sep 17 00:00:00 2001
From: Mike Frysinger <vapier@gentoo.org>
Date: Tue, 11 Jan 2011 19:57:33 -0500
Subject: ASoC: Blackfin AC97: fix build error after multi-component update

We need to tweak how we query the active capture/playback state after
the recent overhauls of common code.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/blackfin/bf5xx-ac97.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index c5f856ec27c..ffbac26b9bc 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -260,9 +260,9 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 	pr_debug("%s : sport %d\n", __func__, dai->id);
 	if (!dai->active)
 		return 0;
-	if (dai->capture.active)
+	if (dai->capture_active)
 		sport_rx_stop(sport);
-	if (dai->playback.active)
+	if (dai->playback_active)
 		sport_tx_stop(sport);
 	return 0;
 }
-- 
cgit v1.2.3-70-g09d2


From 950a95d4e2e2c3a9fb0daceaaf55b969e4710ce7 Mon Sep 17 00:00:00 2001
From: Barry Song <barry.song@analog.com>
Date: Wed, 12 Jan 2011 02:59:55 -0500
Subject: ASoC: Blackfin TDM: use external frame syncs

We don't want to use internal frame syncs otherwise we sometimes
get out of sync, so don't enable them when setting up the SPORT.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/bf5xx-tdm.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index b2cf239f20b..5515ac9e05c 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -235,13 +235,13 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai)
 		ret = -EBUSY;
 	}
 
-	ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0);
+	ret = sport_config_rx(sport, 0, 0x1F, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		ret = -EBUSY;
 	}
 
-	ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0);
+	ret = sport_config_tx(sport, 0, 0x1F, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		ret = -EBUSY;
@@ -303,14 +303,14 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev)
 		goto sport_config_err;
 	}
 
-	ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0);
+	ret = sport_config_rx(sport_handle, 0, 0x1F, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		ret = -EBUSY;
 		goto sport_config_err;
 	}
 
-	ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0);
+	ret = sport_config_tx(sport_handle, 0, 0x1F, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		ret = -EBUSY;
-- 
cgit v1.2.3-70-g09d2


From 91056acbcb6f58265698a091a1a211f994fdb579 Mon Sep 17 00:00:00 2001
From: Barry Song <barry.song@analog.com>
Date: Tue, 11 Jan 2011 20:04:28 -0500
Subject: ASoC: Blackfin: fix DAI/SPORT config dependency issues

While I2S/TDM/AC97 DAI is built-in, others are compiled as modules,
SND_BF5XX_SOC_SPORT will be module, then DAI can't get some symbols.
Except that, SND_BF5XX_AC97 depends on SND_BF5XX_SOC_AC97 too.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/blackfin/Kconfig | 11 ++++++-----
 1 file changed, 6 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 3abeeddc67d..ae403597fd3 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,6 +1,7 @@
 config SND_BF5XX_I2S
 	tristate "SoC I2S Audio for the ADI BF5xx chip"
 	depends on BLACKFIN
+	select SND_BF5XX_SOC_SPORT
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -35,6 +36,7 @@ config SND_BFIN_AD73311_SE
 config SND_BF5XX_TDM
 	tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
 	depends on (BLACKFIN && SND_SOC)
+	select SND_BF5XX_SOC_SPORT
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in TDM
@@ -61,6 +63,10 @@ config SND_BF5XX_SOC_AD193X
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
 	depends on BLACKFIN
+	select AC97_BUS
+	select SND_SOC_AC97_BUS
+	select SND_BF5XX_SOC_SPORT
+	select SND_BF5XX_SOC_AC97
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in slot 16
@@ -122,17 +128,12 @@ config SND_BF5XX_SOC_SPORT
 
 config SND_BF5XX_SOC_I2S
 	tristate
-	select SND_BF5XX_SOC_SPORT
 
 config SND_BF5XX_SOC_TDM
 	tristate
-	select SND_BF5XX_SOC_SPORT
 
 config SND_BF5XX_SOC_AC97
 	tristate
-	select AC97_BUS
-	select SND_SOC_AC97_BUS
-	select SND_BF5XX_SOC_SPORT
 
 config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
-- 
cgit v1.2.3-70-g09d2


From 7cbf70041db039532d6b8972e88164ed45ae6460 Mon Sep 17 00:00:00 2001
From: Vasily Khoruzhick <anarsoul@gmail.com>
Date: Tue, 18 Jan 2011 16:54:24 +0200
Subject: ASoC: PXA: Fix jack detection on Zipit Z2

Fix jack detection on Zipit Z2, otherwise it
disables headphones output when jack is connected

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/z2.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 2d4f896d7fe..ca0d7e34240 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -104,6 +104,7 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
 		.name		= "hsdet-gpio",
 		.report		= SND_JACK_HEADSET,
 		.debounce_time	= 200,
+		.invert		= 1,
 	},
 };
 
-- 
cgit v1.2.3-70-g09d2


From c88c2823e87dd6f8214b8b8cdc36d45f205a8077 Mon Sep 17 00:00:00 2001
From: Vasily Khoruzhick <anarsoul@gmail.com>
Date: Tue, 18 Jan 2011 16:54:25 +0200
Subject: ASoC: PXA: Fix codec address on Zipit Z2

WM8750 address is 0x1b, not 0x1a. Without this fix ALSA detects no sound
cards on Zipit

Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/pxa/z2.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index ca0d7e34240..3ceaef68e01 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -193,7 +193,7 @@ static struct snd_soc_dai_link z2_dai = {
 	.cpu_dai_name	= "pxa2xx-i2s",
 	.codec_dai_name	= "wm8750-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name	= "wm8750-codec.0-001a",
+	.codec_name	= "wm8750-codec.0-001b",
 	.init		= z2_wm8750_init,
 	.ops		= &z2_ops,
 };
-- 
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')

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 a28287925555c93984115d37a1a25315ea369764 Mon Sep 17 00:00:00 2001
From: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Date: Wed, 19 Jan 2011 12:55:28 +0000
Subject: ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()

In the wm8995_set_tristate() function when updating the register
bits use the value and not the register index as the value argument.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8995.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 6045cbde492..608c84c5aa8 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1223,7 +1223,7 @@ static int wm8995_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	else
 		val = 0;
 
-	return snd_soc_update_bits(codec, reg, mask, reg);
+	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 /* The size in bits of the FLL divide multiplied by 10
-- 
cgit v1.2.3-70-g09d2


From 78b3fb46753872fc79bffecc8d50355a8622b39b Mon Sep 17 00:00:00 2001
From: Qiao Zhou <zhouqiao@marvell.com>
Date: Wed, 19 Jan 2011 19:10:47 +0800
Subject: ASoC: WM8994: fix wrong value in tristate function

fix wrong value in wm8994_set_tristate func. when updating reg bits,
it should use "value", not "reg".

Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8994.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 247a6a99feb..3351f77607b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2386,7 +2386,7 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	else
 		val = 0;
 
-	return snd_soc_update_bits(codec, reg, mask, reg);
+	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
 #define WM8994_RATES SNDRV_PCM_RATE_8000_96000
-- 
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')

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')

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')

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')

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 dc5a460a1bfa44273653700e33d4e7051194cbfd Mon Sep 17 00:00:00 2001
From: "Rajashekhara, Sudhakar" <sudhakar.raj@ti.com>
Date: Fri, 21 Jan 2011 20:10:01 +0530
Subject: ASoC: da8xx/omap-l1xx: match codec_name with i2c ids

The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c
is not matching with the i2c ids in the board file. Without this fix the
soundcard does not get detected on da850/omap-l138/am18x evm.

Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Tested-by: Dan Sharon <dansharon@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org (for 2.6.37)
---
 sound/soc/davinci/davinci-evm.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 0c2d6bacc68..b36f0b39b09 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
 	.stream_name = "AIC3X",
 	.cpu_dai_name= "davinci-mcasp.0",
 	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-0018",
 	.platform_name = "davinci-pcm-audio",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
-- 
cgit v1.2.3-70-g09d2


From 20a4e7fc7e213365ea3771d7bf1e10a6bab853be Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 21 Jan 2011 12:47:33 +0000
Subject: ASoC: Handle low measured DC offsets for wm_hubs devices

The DC servo codes are actually signed numbers so need to be treated as
such.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm_hubs.c | 15 +++++++--------
 1 file changed, 7 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index c466982eed2..613df5db0b3 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -91,6 +91,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
 static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
+	s8 offset;
 	u16 reg, reg_l, reg_r, dcs_cfg;
 
 	/* If we're using a digital only path and have a previously
@@ -149,16 +150,14 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 			hubs->dcs_codes);
 
 		/* HPOUT1L */
-		if (reg_l + hubs->dcs_codes > 0 &&
-		    reg_l + hubs->dcs_codes < 0xff)
-			reg_l += hubs->dcs_codes;
-		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		offset = reg_l;
+		offset += hubs->dcs_codes;
+		dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
 		/* HPOUT1R */
-		if (reg_r + hubs->dcs_codes > 0 &&
-		    reg_r + hubs->dcs_codes < 0xff)
-			reg_r += hubs->dcs_codes;
-		dcs_cfg |= reg_r;
+		offset = reg_r;
+		offset += hubs->dcs_codes;
+		dcs_cfg |= (u8)offset;
 
 		dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
 
-- 
cgit v1.2.3-70-g09d2


From 233d84c46c2253d13e10b42d88c14748fbb67a98 Mon Sep 17 00:00:00 2001
From: Jesper Juhl <jj@chaosbits.net>
Date: Thu, 20 Jan 2011 22:37:43 +0100
Subject: ALSA: Xonar, CS43xx: Don't overrun static array

'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of
8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers()

    		for (i = 2; i <= 8; ++i)
	  			snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);

will overrun the array when 'i == 8'.

I guess that what's needed to fix it is the trivial patch below, but I
must admit that I have no idea about this code, so I may very well be
wrong. Additionally, I have no way to actually test this, so all I know is
that the below compiles. Someone who actually knows this code should take
a look before anything is comitted - consider the below (not much more
than) a bug report.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/xonar_cs43xx.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c
index 9f72d424969..252719101c4 100644
--- a/sound/pci/oxygen/xonar_cs43xx.c
+++ b/sound/pci/oxygen/xonar_cs43xx.c
@@ -392,7 +392,7 @@ static void dump_d1_registers(struct oxygen *chip,
 	unsigned int i;
 
 	snd_iprintf(buffer, "\nCS4398: 7?");
-	for (i = 2; i <= 8; ++i)
+	for (i = 2; i < 8; ++i)
 		snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
 	snd_iprintf(buffer, "\n");
 	dump_cs4362a_registers(data, buffer);
-- 
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')

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')

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')

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

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

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

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

(limited to 'sound')

diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 6117595fc07..573594bf322 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -979,31 +979,25 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec,
 
 	snd_azf3328_dbgcallenter();
 	switch (bitrate) {
-#define AZF_FMT_XLATE(in_freq, out_bits) \
-	do { \
-		case AZF_FREQ_ ## in_freq: \
-			freq = SOUNDFORMAT_FREQ_ ## out_bits; \
-			break; \
-	} while (0);
-	AZF_FMT_XLATE(4000, SUSPECTED_4000)
-	AZF_FMT_XLATE(4800, SUSPECTED_4800)
-	/* the AZF3328 names it "5510" for some strange reason: */
-	AZF_FMT_XLATE(5512, 5510)
-	AZF_FMT_XLATE(6620, 6620)
-	AZF_FMT_XLATE(8000, 8000)
-	AZF_FMT_XLATE(9600, 9600)
-	AZF_FMT_XLATE(11025, 11025)
-	AZF_FMT_XLATE(13240, SUSPECTED_13240)
-	AZF_FMT_XLATE(16000, 16000)
-	AZF_FMT_XLATE(22050, 22050)
-	AZF_FMT_XLATE(32000, 32000)
+	case AZF_FREQ_4000:  freq = SOUNDFORMAT_FREQ_SUSPECTED_4000; break;
+	case AZF_FREQ_4800:  freq = SOUNDFORMAT_FREQ_SUSPECTED_4800; break;
+	case AZF_FREQ_5512:
+		/* the AZF3328 names it "5510" for some strange reason */
+			     freq = SOUNDFORMAT_FREQ_5510; break;
+	case AZF_FREQ_6620:  freq = SOUNDFORMAT_FREQ_6620; break;
+	case AZF_FREQ_8000:  freq = SOUNDFORMAT_FREQ_8000; break;
+	case AZF_FREQ_9600:  freq = SOUNDFORMAT_FREQ_9600; break;
+	case AZF_FREQ_11025: freq = SOUNDFORMAT_FREQ_11025; break;
+	case AZF_FREQ_13240: freq = SOUNDFORMAT_FREQ_SUSPECTED_13240; break;
+	case AZF_FREQ_16000: freq = SOUNDFORMAT_FREQ_16000; break;
+	case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break;
+	case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break;
 	default:
 		snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate);
 		/* fall-through */
-	AZF_FMT_XLATE(44100, 44100)
-	AZF_FMT_XLATE(48000, 48000)
-	AZF_FMT_XLATE(66200, SUSPECTED_66200)
-#undef AZF_FMT_XLATE
+	case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break;
+	case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break;
+	case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break;
 	}
 	/* val = 0xff07; 3m27.993s (65301Hz; -> 64000Hz???) hmm, 66120, 65967, 66123 */
 	/* val = 0xff09; 17m15.098s (13123,478Hz; -> 12000Hz???) hmm, 13237.2Hz? */
-- 
cgit v1.2.3-70-g09d2


From 81d7da5404aad930a4e4f6111e4f16b752183018 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 24 Jan 2011 22:09:22 +0100
Subject: ASoC: Fix codec device id format used by some dai_links

The id part of an I2C device name is created with the "%d-%04x" format string.

So for example for an I2C device which is connected to the adapter with the id 0
and has its address set to 0x1a the id part of the devices name would be
"0-001a".

Currently some sound board drivers have the id part the codec_name field of
their dai_link structures set as if it had been created by a "%d-0x%x" format
string. For example "0-0x1a" instead of "0-001a".

As a result there is no match between the codec device and the dai_link and no
sound card is instantiated.

This patch fixes it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/atmel/snd-soc-afeb9260.c             | 2 +-
 sound/soc/blackfin/bf5xx-ssm2602.c             | 2 +-
 sound/soc/samsung/neo1973_gta02_wm8753.c       | 4 ++--
 sound/soc/samsung/neo1973_wm8753.c             | 4 ++--
 sound/soc/samsung/s3c24xx_simtec_hermes.c      | 2 +-
 sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +-
 6 files changed, 8 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index da2208e06b0..5e4d499d843 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -129,7 +129,7 @@ static struct snd_soc_dai_link afeb9260_dai = {
 	.cpu_dai_name = "atmel-ssc-dai.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.platform_name = "atmel_pcm-audio",
-	.codec_name = "tlv320aic23-codec.0-0x1a",
+	.codec_name = "tlv320aic23-codec.0-001a",
 	.init = afeb9260_tlv320aic23_init,
 	.ops = &afeb9260_ops,
 };
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e902b24c185..ad28663f5bb 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = {
 	.cpu_dai_name = "bf5xx-i2s",
 	.codec_dai_name = "ssm2602-hifi",
 	.platform_name = "bf5xx-pcm-audio",
-	.codec_name = "ssm2602-codec.0-0x1b",
+	.codec_name = "ssm2602-codec.0-001b",
 	.ops = &bf5xx_ssm2602_ops,
 };
 
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 3eec610c10f..9e05e10b86a 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -401,7 +401,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
 	.codec_dai_name = "wm8753-hifi",
 	.init = neo1973_gta02_wm8753_init,
 	.platform_name = "samsung-audio",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.ops = &neo1973_gta02_hifi_ops,
 },
 { /* Voice via BT */
@@ -410,7 +410,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
 	.cpu_dai_name = "bluetooth-dai",
 	.codec_dai_name = "wm8753-voice",
 	.ops = &neo1973_gta02_voice_ops,
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.platform_name = "samsung-audio",
 },
 };
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index c7a24514beb..cf69e146890 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -561,7 +561,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.platform_name = "samsung-audio",
 	.cpu_dai_name = "s3c24xx-i2s",
 	.codec_dai_name = "wm8753-hifi",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.init = neo1973_wm8753_init,
 	.ops = &neo1973_hifi_ops,
 },
@@ -571,7 +571,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.platform_name = "samsung-audio",
 	.cpu_dai_name = "bluetooth-dai",
 	.codec_dai_name = "wm8753-voice",
-	.codec_name = "wm8753-codec.0-0x1a",
+	.codec_name = "wm8753-codec.0-001a",
 	.ops = &neo1973_voice_ops,
 },
 };
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index bb4292e3596..287a97164ab 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -94,7 +94,7 @@ static int simtec_hermes_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic33 = {
 	.name		= "tlv320aic33",
 	.stream_name	= "TLV320AIC33",
-	.codec_name	= "tlv320aic3x-codec.0-0x1a",
+	.codec_name	= "tlv320aic3x-codec.0-001a",
 	.cpu_dai_name	= "s3c24xx-i2s",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index fbba4e36772..d2b14ba6f9e 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -85,7 +85,7 @@ static int simtec_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link simtec_dai_aic23 = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
-	.codec_name	= "tlv320aic3x-codec.0-0x1a",
+	.codec_name	= "tlv320aic3x-codec.0-001a",
 	.cpu_dai_name	= "s3c24xx-i2s",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
-- 
cgit v1.2.3-70-g09d2


From 518aa59f6e45b3c90b849187ae1d56757d074b92 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Mon, 24 Jan 2011 22:12:42 +0100
Subject: ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s

During the multi-component patch the s3c24xx i2s driver was renamed from
"s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not
updated to reflect this change as well.

As a result there is no match between the dai_link and the i2s driver and no
sound card is instantiated.

This patch fixes the problem by updating the sound board drivers to use
"s3c24xx-iis" for the cpu_dai_name.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/samsung/neo1973_gta02_wm8753.c       | 2 +-
 sound/soc/samsung/neo1973_wm8753.c             | 2 +-
 sound/soc/samsung/s3c24xx_simtec_hermes.c      | 2 +-
 sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 2 +-
 sound/soc/samsung/s3c24xx_uda134x.c            | 2 +-
 5 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
index 9e05e10b86a..0d0ae2b9eef 100644
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ b/sound/soc/samsung/neo1973_gta02_wm8753.c
@@ -397,7 +397,7 @@ static struct snd_soc_dai_link neo1973_gta02_dai[] = {
 { /* Hifi Playback - for similatious use with voice below */
 	.name = "WM8753",
 	.stream_name = "WM8753 HiFi",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.codec_dai_name = "wm8753-hifi",
 	.init = neo1973_gta02_wm8753_init,
 	.platform_name = "samsung-audio",
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index cf69e146890..d20815d5ab2 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -559,7 +559,7 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 	.name = "WM8753",
 	.stream_name = "WM8753 HiFi",
 	.platform_name = "samsung-audio",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.codec_dai_name = "wm8753-hifi",
 	.codec_name = "wm8753-codec.0-001a",
 	.init = neo1973_wm8753_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c
index 287a97164ab..08fcaaa6690 100644
--- a/sound/soc/samsung/s3c24xx_simtec_hermes.c
+++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c
@@ -95,7 +95,7 @@ static struct snd_soc_dai_link simtec_dai_aic33 = {
 	.name		= "tlv320aic33",
 	.stream_name	= "TLV320AIC33",
 	.codec_name	= "tlv320aic3x-codec.0-001a",
-	.cpu_dai_name	= "s3c24xx-i2s",
+	.cpu_dai_name	= "s3c24xx-iis",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
 	.init		= simtec_hermes_init,
diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
index d2b14ba6f9e..116e3e67016 100644
--- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c
@@ -86,7 +86,7 @@ static struct snd_soc_dai_link simtec_dai_aic23 = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
 	.codec_name	= "tlv320aic3x-codec.0-001a",
-	.cpu_dai_name	= "s3c24xx-i2s",
+	.cpu_dai_name	= "s3c24xx-iis",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.platform_name	= "samsung-audio",
 	.init		= simtec_tlv320aic23_init,
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index cdc8ecbcb8e..2c09e93dd56 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -228,7 +228,7 @@ static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
 	.stream_name = "UDA134X",
 	.codec_name = "uda134x-hifi",
 	.codec_dai_name = "uda134x-hifi",
-	.cpu_dai_name = "s3c24xx-i2s",
+	.cpu_dai_name = "s3c24xx-iis",
 	.ops = &s3c24xx_uda134x_ops,
 	.platform_name	= "samsung-audio",
 };
-- 
cgit v1.2.3-70-g09d2


From a3adfa00e8089aa72826c6ba04bcb18cfceaf0a9 Mon Sep 17 00:00:00 2001
From: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Date: Fri, 21 Jan 2011 22:14:17 +0300
Subject: ASoC: correct link specifications for corgi, poodle and spitz

ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms
contained incorrect names for cpu_dai and codec, which effectievly disabled sound
on theese platforms. Fix that errors.

Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
---
 sound/soc/pxa/corgi.c  | 4 ++--
 sound/soc/pxa/poodle.c | 2 +-
 sound/soc/pxa/spitz.c  | 4 ++--
 3 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fc592f0d5fc..784cff5f67e 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -307,10 +307,10 @@ static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link corgi_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731",
-	.cpu_dai_name = "pxa-is2-dai",
+	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec-0.001a",
+	.codec_name = "wm8731-codec-0.001b",
 	.init = corgi_wm8731_init,
 	.ops = &corgi_ops,
 };
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6298ee115e2..a7d4999f9b2 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = {
 	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec.0-001a",
+	.codec_name = "wm8731-codec.0-001b",
 	.init = poodle_wm8731_init,
 	.ops = &poodle_ops,
 };
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index c2acb69b957..8e157135063 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -315,10 +315,10 @@ static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 static struct snd_soc_dai_link spitz_dai = {
 	.name = "wm8750",
 	.stream_name = "WM8750",
-	.cpu_dai_name = "pxa-is2",
+	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8750-codec.0-001a",
+	.codec_name = "wm8750-codec.0-001b",
 	.init = spitz_wm8750_init,
 	.ops = &spitz_ops,
 };
-- 
cgit v1.2.3-70-g09d2


From fd76804f3f5484b35e6a51214c91e916ebba05aa Mon Sep 17 00:00:00 2001
From: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Date: Mon, 24 Jan 2011 16:09:56 +0100
Subject: ALSA: fix invalid hardware.h include in ac97c for AVR32 architecture

This patch fixes the non-compiling AC97C driver for AVR32 architecture by
include mach/hardware.h only for AT91 architecture. The AVR32 architecture does
not supply the hardware.h include file.

Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
CC: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/atmel/ac97c.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 10c3a871a12..b310702c646 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -33,9 +33,12 @@
 #include <linux/dw_dmac.h>
 
 #include <mach/cpu.h>
-#include <mach/hardware.h>
 #include <mach/gpio.h>
 
+#ifdef CONFIG_ARCH_AT91
+#include <mach/hardware.h>
+#endif
+
 #include "ac97c.h"
 
 enum {
-- 
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')

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 69058cd6d10423126ab6aeb568f4af2bd34c49fe Mon Sep 17 00:00:00 2001
From: Russell King <rmk+kernel@arm.linux.org.uk>
Date: Wed, 12 Jan 2011 23:17:24 +0000
Subject: ALSA: AACI: fix timeout condition checking

Ensure that a timeout coincident with the condition being waited for
results in success rather than failure.  This helps avoid timeout
conditions being inappropriately flagged.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
---
 sound/arm/aaci.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 91acc9a243e..21ff6296d16 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -88,7 +88,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 		v = readl(aaci->base + AACI_SLFR);
 	} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
 
-	if (!timeout)
+	if (v & (SLFR_1TXB|SLFR_2TXB))
 		dev_err(&aaci->dev->dev,
 			"timeout waiting for write to complete\n");
 
@@ -124,7 +124,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 		v = readl(aaci->base + AACI_SLFR);
 	} while ((v & SLFR_1TXB) && --timeout);
 
-	if (!timeout) {
+	if (v & SLFR_1TXB) {
 		dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
 		v = ~0;
 		goto out;
@@ -145,7 +145,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 		v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
 	} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
 
-	if (!timeout) {
+	if (v != (SLFR_1RXV|SLFR_2RXV)) {
 		dev_err(&aaci->dev->dev, "timeout on RX valid\n");
 		v = ~0;
 		goto out;
-- 
cgit v1.2.3-70-g09d2


From 250c7a61c35a258e2422b3d55c61bfbad33326be Mon Sep 17 00:00:00 2001
From: Russell King <rmk+kernel@arm.linux.org.uk>
Date: Wed, 12 Jan 2011 23:42:57 +0000
Subject: ALSA: AACI: fix timeout duration

Relying on the access time of peripherals is unreliable - it depends
on the speed of the CPU and the bus.  On Versatile Express, these
timeouts were expiring, causing the driver to fail.

Add udelay(1) to ensure that they don't expire early, and adjust
timeouts to give a reasonable margin over the response times.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
---
 sound/arm/aaci.c | 42 +++++++++++++++++++++++-------------------
 1 file changed, 23 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 21ff6296d16..24d3013c023 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -30,6 +30,8 @@
 
 #define DRIVER_NAME	"aaci-pl041"
 
+#define FRAME_PERIOD_US	21
+
 /*
  * PM support is not complete.  Turn it off.
  */
@@ -64,8 +66,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 			    unsigned short val)
 {
 	struct aaci *aaci = ac97->private_data;
+	int timeout;
 	u32 v;
-	int timeout = 5000;
 
 	if (ac97->num >= 4)
 		return;
@@ -81,10 +83,13 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	writel(val << 4, aaci->base + AACI_SL2TX);
 	writel(reg << 12, aaci->base + AACI_SL1TX);
 
-	/*
-	 * Wait for the transmission of both slots to complete.
-	 */
+	/* Initially, wait one frame period */
+	udelay(FRAME_PERIOD_US);
+
+	/* And then wait an additional eight frame periods for it to be sent */
+	timeout = FRAME_PERIOD_US * 8;
 	do {
+		udelay(1);
 		v = readl(aaci->base + AACI_SLFR);
 	} while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
 
@@ -101,9 +106,8 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 {
 	struct aaci *aaci = ac97->private_data;
+	int timeout, retries = 10;
 	u32 v;
-	int timeout = 5000;
-	int retries = 10;
 
 	if (ac97->num >= 4)
 		return ~0;
@@ -117,10 +121,13 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 	 */
 	writel((reg << 12) | (1 << 19), aaci->base + AACI_SL1TX);
 
-	/*
-	 * Wait for the transmission to complete.
-	 */
+	/* Initially, wait one frame period */
+	udelay(FRAME_PERIOD_US);
+
+	/* And then wait an additional eight frame periods for it to be sent */
+	timeout = FRAME_PERIOD_US * 8;
 	do {
+		udelay(1);
 		v = readl(aaci->base + AACI_SLFR);
 	} while ((v & SLFR_1TXB) && --timeout);
 
@@ -130,17 +137,13 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 		goto out;
 	}
 
-	/*
-	 * Give the AC'97 codec more than enough time
-	 * to respond. (42us = ~2 frames at 48kHz.)
-	 */
-	udelay(42);
+	/* Now wait for the response frame */
+	udelay(FRAME_PERIOD_US);
 
-	/*
-	 * Wait for slot 2 to indicate data.
-	 */
-	timeout = 5000;
+	/* And then wait an additional eight frame periods for data */
+	timeout = FRAME_PERIOD_US * 8;
 	do {
+		udelay(1);
 		cond_resched();
 		v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
 	} while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
@@ -179,6 +182,7 @@ aaci_chan_wait_ready(struct aaci_runtime *aacirun, unsigned long mask)
 	int timeout = 5000;
 
 	do {
+		udelay(1);
 		val = readl(aacirun->base + AACI_SR);
 	} while (val & mask && timeout--);
 }
@@ -874,7 +878,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci)
 	 * Give the AC'97 codec more than enough time
 	 * to wake up. (42us = ~2 frames at 48kHz.)
 	 */
-	udelay(42);
+	udelay(FRAME_PERIOD_US * 2);
 
 	ret = snd_ac97_bus(aaci->card, 0, &aaci_bus_ops, aaci, &ac97_bus);
 	if (ret)
-- 
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')

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 c73e0c83f512012e7c357e516a0d7c0a832bfa34 Mon Sep 17 00:00:00 2001
From: Jarkko Nikula <jhnikula@gmail.com>
Date: Wed, 26 Jan 2011 16:39:37 +0200
Subject: ASoC: Fix module refcount for auxiliary devices

Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers"
moved codec driver refcount increments from soc_bind_dai_link into
soc_probe_codec.

However, the commit didn't remove try_module_get from soc_probe_aux_dev so
the auxiliary device reference counts are incremented twice as the
soc_probe_codec is called from soc_probe_aux_dev too.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index bac7291b6ff..c4b60610beb 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1664,9 +1664,6 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num)
 	goto out;
 
 found:
-	if (!try_module_get(codec->dev->driver->owner))
-		return -ENODEV;
-
 	ret = soc_probe_codec(card, codec);
 	if (ret < 0)
 		return ret;
-- 
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')

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 0fa63b69284c9bbedf876c677a9e650243cc40be Mon Sep 17 00:00:00 2001
From: "Manjunathappa, Prakash" <prakash.pm@ti.com>
Date: Thu, 27 Jan 2011 19:17:43 +0530
Subject: ASoC: DaVinci: fix kernel panic due to uninitialized platform_data

This patch fixes the Kernel panic issue on accessing davinci_vc in
cq93vc_probe function. struct davinci_vc is part of platform device's
private driver data(codec->dev->p->driver_data) and this is populated
by DaVinci Voice Codec MFD driver.

Signed-off-by: Manjunathappa, Prakash <prakash.pm@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cq93vc.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 46dbfd067f7..347a567b01e 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,7 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-	struct davinci_vc *davinci_vc = codec->dev->platform_data;
+	struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
 
 	davinci_vc->cq93vc.codec = codec;
 	codec->control_data = davinci_vc;
-- 
cgit v1.2.3-70-g09d2


From e9cf7049330cd44c8af43b0c5c7bef25a086c5b7 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Thu, 27 Jan 2011 14:54:05 -0700
Subject: ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw()

snd_soc_dapm_put_volsw() has variables for both the unshifted and
shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in
the middle of DAPM sequences) got confused between the two of these.
Since there's no need to keep a copy of the unshifted mask fix this and
simplify the code by using only one mask variable.

[Completely rewrote the changelog to describe the issue -- broonie.]

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 499730ab563..8194f150bab 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1742,7 +1742,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
-	unsigned int val, val_mask;
+	unsigned int val;
 	int connect, change;
 	struct snd_soc_dapm_update update;
 
@@ -1750,13 +1750,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
 
 	if (invert)
 		val = max - val;
-	val_mask = mask << shift;
+	mask = mask << shift;
 	val = val << shift;
 
 	mutex_lock(&widget->codec->mutex);
 	widget->value = val;
 
-	change = snd_soc_test_bits(widget->codec, reg, val_mask, val);
+	change = snd_soc_test_bits(widget->codec, reg, mask, val);
 	if (change) {
 		if (val)
 			/* new connection */
-- 
cgit v1.2.3-70-g09d2


From fdbc5d1b32e195b7775e103abd6263370f11af11 Mon Sep 17 00:00:00 2001
From: Amerigo Wang <amwang@redhat.com>
Date: Fri, 28 Jan 2011 16:52:00 +0800
Subject: sound: silent echo'ed messages in Makefile

Silent these echo's, please.

Signed-off-by: WANG Cong <amwang@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/oss/Makefile | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/oss/Makefile b/sound/oss/Makefile
index 96f14dcd0cd..90ffb99c6b1 100644
--- a/sound/oss/Makefile
+++ b/sound/oss/Makefile
@@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
 	$(obj)/bin2hex pss_synth < $< > $@
 else
     $(obj)/pss_boot.h:
-	(							\
+	$(Q)(							\
 	    echo 'static unsigned char * pss_synth = NULL;';	\
 	    echo 'static int pss_synthLen = 0;';		\
 	) > $@
@@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
 	$(obj)/hex2hex -i trix_boot < $< > $@
 else
     $(obj)/trix_boot.h:
-	(							\
+	$(Q)(							\
 	    echo 'static unsigned char * trix_boot = NULL;';	\
 	    echo 'static int trix_boot_len = 0;';		\
 	) > $@
-- 
cgit v1.2.3-70-g09d2


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

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

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

(limited to 'sound')

diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index c2ae63d17cd..f53897a708b 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -92,6 +92,8 @@ struct oxygen_model {
 	void (*update_dac_volume)(struct oxygen *chip);
 	void (*update_dac_mute)(struct oxygen *chip);
 	void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed);
+	unsigned int (*adjust_dac_routing)(struct oxygen *chip,
+					   unsigned int play_routing);
 	void (*gpio_changed)(struct oxygen *chip);
 	void (*uart_input)(struct oxygen *chip);
 	void (*ac97_switch)(struct oxygen *chip,
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 9bff14d5895..26c7e8bcb22 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -180,6 +180,8 @@ void oxygen_update_dac_routing(struct oxygen *chip)
 			    (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) |
 			    (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) |
 			    (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT);
+	if (chip->model.adjust_dac_routing)
+		reg_value = chip->model.adjust_dac_routing(chip, reg_value);
 	oxygen_write16_masked(chip, OXYGEN_PLAY_ROUTING, reg_value,
 			      OXYGEN_PLAY_DAC0_SOURCE_MASK |
 			      OXYGEN_PLAY_DAC1_SOURCE_MASK |
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index e1fa602eba7..bc6eb58be38 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -24,6 +24,11 @@
  *
  *   SPI 0 -> CS4245
  *
+ *   I²S 1 -> CS4245
+ *   I²S 2 -> CS4361 (center/LFE)
+ *   I²S 3 -> CS4361 (surround)
+ *   I²S 4 -> CS4361 (front)
+ *
  *   GPIO 3 <- ?
  *   GPIO 4 <- headphone detect
  *   GPIO 5 -> route input jack to line-in (0) or mic-in (1)
@@ -36,6 +41,7 @@
  *   input 1 <- aux
  *   input 2 <- front mic
  *   input 4 <- line/mic
+ *   DAC out -> headphones
  *   aux out -> front panel headphones
  */
 
@@ -207,6 +213,35 @@ static void set_cs4245_adc_params(struct oxygen *chip,
 	cs4245_write_cached(chip, CS4245_ADC_CTRL, value);
 }
 
+static inline unsigned int shift_bits(unsigned int value,
+				      unsigned int shift_from,
+				      unsigned int shift_to,
+				      unsigned int mask)
+{
+	if (shift_from < shift_to)
+		return (value << (shift_to - shift_from)) & mask;
+	else
+		return (value >> (shift_from - shift_to)) & mask;
+}
+
+static unsigned int adjust_dg_dac_routing(struct oxygen *chip,
+					  unsigned int play_routing)
+{
+	return (play_routing & OXYGEN_PLAY_DAC0_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC1_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC1_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC2_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC2_SOURCE_MASK) |
+	       shift_bits(play_routing,
+			  OXYGEN_PLAY_DAC0_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC3_SOURCE_SHIFT,
+			  OXYGEN_PLAY_DAC3_SOURCE_MASK);
+}
+
 static int output_switch_info(struct snd_kcontrol *ctl,
 			      struct snd_ctl_elem_info *info)
 {
@@ -557,6 +592,7 @@ struct oxygen_model model_xonar_dg = {
 	.resume = dg_resume,
 	.set_dac_params = set_cs4245_dac_params,
 	.set_adc_params = set_cs4245_adc_params,
+	.adjust_dac_routing = adjust_dg_dac_routing,
 	.dump_registers = dump_cs4245_registers,
 	.model_data_size = sizeof(struct dg),
 	.device_config = PLAYBACK_0_TO_I2S |
-- 
cgit v1.2.3-70-g09d2


From acd62276773b46810a3292af0c915c9782138ff2 Mon Sep 17 00:00:00 2001
From: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Date: Tue, 1 Feb 2011 11:11:55 +0100
Subject: ASoC: Amstrad Delta: fix const related build error

The Amstrad Delta ASoC driver used to override the digital_mute()
callback, expected to be not provided by the on-board CX20442 CODEC
driver, with its own implementation. While this is still posssible when
substituting the whole empty snd_soc_dai_driver.ops member (the CX20442
case), replacing snd_soc_dai_ops.digital_mute only is no longer correct
after the snd_soc_dai_driver.ops member has been constified, and results
in build error.

Drop this actually not used code path in hope the CX20442 driver never
provides its own snd_soc_dai_ops structure.

Created and tested against linux-2.6.38-rc2

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/omap/ams-delta.c | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 2101bdcee21..3167be68962 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -507,8 +507,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
 	/* Set up digital mute if not provided by the codec */
 	if (!codec_dai->driver->ops) {
 		codec_dai->driver->ops = &ams_delta_dai_ops;
-	} else if (!codec_dai->driver->ops->digital_mute) {
-		codec_dai->driver->ops->digital_mute = ams_delta_digital_mute;
 	} else {
 		ams_delta_ops.startup = ams_delta_startup;
 		ams_delta_ops.shutdown = ams_delta_shutdown;
-- 
cgit v1.2.3-70-g09d2


From f019ee5feb344ff0b22b58df4568676295aae14f Mon Sep 17 00:00:00 2001
From: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Date: Tue, 1 Feb 2011 13:01:17 +0100
Subject: ASoC: CX20442: fix NULL pointer dereference

The CX20442 codec driver never provided the snd_soc_codec_driver's
.reg_cache_default member. With the latest ASoC framework changes, it
seems to be referred unconditionally, resulting in a NULL pointer
dereference if missing. Provide it.

Created and tested on Amstrad Delta against linux-2.6.38-rc2

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/codecs/cx20442.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 03d1e860d22..bb4bf65b9e7 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,9 +367,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+
 static struct snd_soc_codec_driver cx20442_codec_dev = {
 	.probe = 	cx20442_codec_probe,
 	.remove = 	cx20442_codec_remove,
+	.reg_cache_default = &cx20442_reg,
 	.reg_cache_size = 1,
 	.reg_word_size = sizeof(u8),
 	.read = cx20442_read_reg_cache,
-- 
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')

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 ddfb319926462fd9670b7c1678a1f6a14a68e421 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Feb 2011 17:49:53 +0100
Subject: ALSA: use linux/io.h to fix compile warnings

For helping to reduce Greert's regression list...
  src/sound/drivers/mtpav.c: error: implicit declaration of function 'inb'
  src/sound/drivers/mtpav.c: error: implicit declaration of function 'outb'
  ...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/drivers/mtpav.c              | 3 +--
 sound/pcmcia/pdaudiocf/pdaudiocf.h | 2 +-
 sound/pcmcia/vx/vxp_ops.c          | 2 +-
 3 files changed, 3 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index da03597fc89..5c426df8767 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -55,14 +55,13 @@
 #include <linux/err.h>
 #include <linux/platform_device.h>
 #include <linux/ioport.h>
+#include <linux/io.h>
 #include <linux/moduleparam.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/rawmidi.h>
 #include <linux/delay.h>
 
-#include <asm/io.h>
-
 /*
  *      globals
  */
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index bd26e092aea..6ce9ad70029 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -22,7 +22,7 @@
 #define __PDAUDIOCF_H
 
 #include <sound/pcm.h>
-#include <asm/io.h>
+#include <linux/io.h>
 #include <linux/interrupt.h>
 #include <pcmcia/cistpl.h>
 #include <pcmcia/ds.h>
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 989e04abb52..fe33e122e37 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -23,8 +23,8 @@
 #include <linux/delay.h>
 #include <linux/device.h>
 #include <linux/firmware.h>
+#include <linux/io.h>
 #include <sound/core.h>
-#include <asm/io.h>
 #include "vxpocket.h"
 
 
-- 
cgit v1.2.3-70-g09d2


From 0962bb217ac74c4b8fae34c5367ebc63131c962c Mon Sep 17 00:00:00 2001
From: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Date: Wed, 2 Feb 2011 21:11:41 +0100
Subject: ASoC: fill in snd_soc_pcm_runtime.card before calling
 snd_soc_dai_link.init()

The .card member of the snd_soc_pcm_runtime structure pointed to by the
snd_soc_dai_link.init() argument used to be initialized before the
function being called. This has changed, probably unintentionally,
after recent refactorings. Since the function implementations are free
to make use of this pointer, move its assignment back before the
function is called to avoid NULL pointer dereferences.

Created and tested on Amstrad Delta againts linux-2.6.38-rc2

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-core.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c4b60610beb..c3f6f1e7279 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
 		rtd = &card->rtd_aux[num];
 		name = aux_dev->name;
 	}
+	rtd->card = card;
 
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
 	/* register the rtd device */
 	rtd->codec = codec;
-	rtd->card = card;
 	rtd->dev.parent = card->dev;
 	rtd->dev.release = rtd_release;
 	rtd->dev.init_name = name;
-- 
cgit v1.2.3-70-g09d2


From f9eb9dd14c2ca2a1f8d979637fb651512d16ad22 Mon Sep 17 00:00:00 2001
From: Vaibhav Bedia <vaibhav.bedia@ti.com>
Date: Thu, 3 Feb 2011 16:42:25 +0530
Subject: asoc: davinci: da830/omap-l137: correct cpu_dai_name

McASP1 is used on the DA830/OMAP-L137 platform for the codec.
This is different from the DA850/OMAP-L138 platform which uses McASP0.

This is fixed by adding a new snd_soc_dai_link struct.

Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++---
 1 file changed, 15 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index b36f0b39b09..fe7984221eb 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.ops = &evm_spdif_ops,
 	},
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcasp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-0018",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
 	.cpu_dai_name= "davinci-mcasp.0",
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
 
 static struct snd_soc_card da830_snd_soc_card = {
 	.name = "DA830/OMAP-L137 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da830_evm_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
 	.name = "DA850/OMAP-L138 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da850_evm_dai,
 	.num_links = 1,
 };
 
-- 
cgit v1.2.3-70-g09d2


From 7f94de483f4e37e14d646ad6e85a3c82f66fb487 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 3 Feb 2011 16:27:34 +0000
Subject: ASoC: Create an AIF1ADCDAT signal widget to match AIF2

Due to the different routing for AIF1 and AIF2 we weren't using a
single widget to represent the ADCDAT signal. For consistency add
one.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8994.c | 14 ++++++++++----
 1 file changed, 10 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3351f77607b..3e308ad97dd 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
 	{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
 	/* AIF3 output */
-- 
cgit v1.2.3-70-g09d2


From 6ed8f1485fc82d44ac464bc84a7dcdddd1fa096f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 3 Feb 2011 16:27:35 +0000
Subject: ASoC: Improve WM8994 digital power sequencing

On WM8994 revision D and earlier ensure optimal sequencing with
simultaneous usage of AIF1 and AIF2 by tying the signals together
so if paths through both are connected the streams are started
simultaneously.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
---
 sound/soc/codecs/wm8994.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3e308ad97dd..37b8aa8a680 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1584,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+	{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
+	{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
+	{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+	{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
 	{ "AIF2DACL", NULL, "AIF2DAC Mux" },
 	{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -3135,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	case WM8994:
 		snd_soc_dapm_add_routes(dapm, wm8994_intercon,
 					ARRAY_SIZE(wm8994_intercon));
+
+		if (wm8994->revision < 4)
+			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+						ARRAY_SIZE(wm8994_revd_intercon));
+			
 		break;
 	case WM8958:
 		snd_soc_dapm_add_routes(dapm, wm8958_intercon,
-- 
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')

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 7c289385b84d136089b8a1149321ebffa5193595 Mon Sep 17 00:00:00 2001
From: Russell King <rmk+kernel@arm.linux.org.uk>
Date: Sat, 5 Feb 2011 10:41:55 +0000
Subject: ALSA: AACI: allow writes to MAINCR to take effect

The AACI TRM requires the MAINCR enable bit to be held zero for two
bitclk cycles plus three apb_pclk cycles.  Use a delay of 1us to
ensure this.

Ensure that writes to MAINCR to change the addressed codec only happen
when required, and that they take effect in a similar manner to the
above, otherwise we seem to occasionally have stuck slot busy bits.

Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
---
 sound/arm/aaci.c | 8 +++++++-
 1 file changed, 7 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 24d3013c023..7c1fc64cb53 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -50,7 +50,11 @@ static void aaci_ac97_select_codec(struct aaci *aaci, struct snd_ac97 *ac97)
 	if (v & SLFR_1RXV)
 		readl(aaci->base + AACI_SL1RX);
 
-	writel(maincr, aaci->base + AACI_MAINCR);
+	if (maincr != readl(aaci->base + AACI_MAINCR)) {
+		writel(maincr, aaci->base + AACI_MAINCR);
+		readl(aaci->base + AACI_MAINCR);
+		udelay(1);
+	}
 }
 
 /*
@@ -993,6 +997,8 @@ static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
 	 * disabling the channel doesn't clear the FIFO.
 	 */
 	writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR);
+	readl(aaci->base + AACI_MAINCR);
+	udelay(1);
 	writel(aaci->maincr, aaci->base + AACI_MAINCR);
 
 	/*
-- 
cgit v1.2.3-70-g09d2


From 1cdfa9f34acb9780e0fe7b8a41fb1a885ab94735 Mon Sep 17 00:00:00 2001
From: Joseph Teichman <josteich@gmail.com>
Date: Tue, 8 Feb 2011 01:22:36 -0500
Subject: ALSA: usbaudio - Enable the E-MU 0204 USB

Signed-off-by: Joseph Teichman <josteich@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/usb/mixer.c        | 4 ++--
 sound/usb/quirks-table.h | 7 +++++++
 sound/usb/quirks.c       | 3 ++-
 3 files changed, 11 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 7df89b3d7de..85af6051b52 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -95,7 +95,7 @@ enum {
 };
 
 
-/*E-mu 0202(0404) eXtension Unit(XU) control*/
+/*E-mu 0202/0404/0204 eXtension Unit(XU) control*/
 enum {
 	USB_XU_CLOCK_RATE 		= 0xe301,
 	USB_XU_CLOCK_SOURCE		= 0xe302,
@@ -1566,7 +1566,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
 			cval->initialized = 1;
 		} else {
 			if (type == USB_XU_CLOCK_RATE) {
-				/* E-Mu USB 0404/0202/TrackerPre
+				/* E-Mu USB 0404/0202/TrackerPre/0204
 				 * samplerate control quirk
 				 */
 				cval->min = 0;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 35999874d30..921a86fd988 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -79,6 +79,13 @@
 	.idProduct = 0x3f0a,
 	.bInterfaceClass = USB_CLASS_AUDIO,
 },
+{
+	/* E-Mu 0204 USB */
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
+	.idVendor = 0x041e,
+	.idProduct = 0x3f19,
+	.bInterfaceClass = USB_CLASS_AUDIO,
+},
 
 /*
  * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index cf8bf088394..e314cdb8500 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -532,7 +532,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat
 }
 
 /*
- * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device,
+ * For E-Mu 0404USB/0202USB/TrackerPre/0204 sample rate should be set for device,
  * not for interface.
  */
 
@@ -589,6 +589,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
 	case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
 	case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
 	case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
+	case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
 		set_format_emu_quirk(subs, fmt);
 		break;
 	}
-- 
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')

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')

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')

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 b1d4f7f4bdcf9915c41ff8cfc4425c84dabb1fde Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 10 Feb 2011 16:15:44 +0100
Subject: ALSA: hrtimer: handle delayed timer interrupts

If a timer interrupt was delayed too much, hrtimer_forward_now() will
forward the timer expiry more than once.  When this happens, the
additional number of elapsed ALSA timer ticks must be passed to
snd_timer_interrupt() to prevent the ALSA timer from falling behind.

This mostly fixes MIDI slowdown problems on highly-loaded systems with
badly behaved interrupt handlers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/core/hrtimer.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 7730575bfad..07efa29dfd4 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -45,12 +45,13 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
 {
 	struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt);
 	struct snd_timer *t = stime->timer;
+	unsigned long oruns;
 
 	if (!atomic_read(&stime->running))
 		return HRTIMER_NORESTART;
 
-	hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
-	snd_timer_interrupt(stime->timer, t->sticks);
+	oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution));
+	snd_timer_interrupt(stime->timer, t->sticks * oruns);
 
 	if (!atomic_read(&stime->running))
 		return HRTIMER_NORESTART;
-- 
cgit v1.2.3-70-g09d2


From 2243c4d0727ad85aff3f54be9d178632cc9234b2 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Thu, 10 Feb 2011 16:16:32 +0100
Subject: ALSA: hrtimer: remove superfluous tasklet invocation

Commit bb758e9637e5ddc removed snd_hrtimer_callback() from the hardware
interrupt handler, thus moving it into a tasklet, but did not tell the
ALSA timer framework about this, so the timer handling would now be done
in the ALSA timer tasklet scheduled from another tasklet.

To fix this, add the flag to tell the ALSA timer framework that the
timer handler is already being invoked in a tasklet.

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

(limited to 'sound')

diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index 07efa29dfd4..b8b31c433d6 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -105,7 +105,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
 }
 
 static struct snd_timer_hardware hrtimer_hw = {
-	.flags =	SNDRV_TIMER_HW_AUTO,
+	.flags =	SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
 	.open =		snd_hrtimer_open,
 	.close =	snd_hrtimer_close,
 	.start =	snd_hrtimer_start,
-- 
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')

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