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Date:	Thu, 17 Sep 2009 21:14:54 +0200
From:	Takashi Iwai <tiwai@...e.de>
To:	Linus Torvalds <torvalds@...ux-foundation.org>
Cc:	Andrew Morton <akpm@...ux-foundation.org>,
	linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 2.6.32-rc1

Linus,

please pull ALSA fixes for v2.6.32-rc1 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

including the following a few fixes for ASoC and HD-audio.


Thanks!

Takashi

===

Huang Weiyi (1):
      ASoC: remove unused #include <linux/version.h>

Jassi (1):
      ASoC: S3C lrsync function made to work with IRQs disabled.

Julia Lawall (1):
      ASoC: Clean up error handling in MPC5200 DMA setup

Mark Brown (1):
      ASoC: Fix display of stream name in DAPM debugfs

Takashi Iwai (8):
      ALSA: hda - Add missing model=auto entry for ALC269
      ALSA: hda - Set default GPIO for STAC/IDT codecs
      ALSA: hda - Set default GPIO for IDT92HD71bxx
      ALSA: hda - Fix HP/line-out initialization with IDT/STAC codecs
      ALSA: hda - Add support for HP dv6
      ALSA: hda - Fix IDT92HD83* codec setup
      ALSA: hda - Fix Dell S14 pin setup
      ALSA: hda - Fix MSI GX620 mixer

---
 sound/pci/hda/patch_realtek.c  |   21 ++++++++++--
 sound/pci/hda/patch_sigmatel.c |   65 +++++++++++++++++++++-------------------
 sound/soc/codecs/ad1836.c      |    1 -
 sound/soc/codecs/ad1938.c      |    1 -
 sound/soc/codecs/wm8974.c      |    1 -
 sound/soc/fsl/mpc5200_dma.c    |   33 ++++++++++++--------
 sound/soc/s3c24xx/s3c-i2s-v2.c |   16 ++++++----
 sound/soc/soc-dapm.c           |    7 ++--
 8 files changed, 85 insertions(+), 60 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7ed47f6..1296058 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7927,8 +7927,9 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 
 static struct snd_kcontrol_new alc883_targa_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	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", 0x0e, 1, 0x0, HDA_OUTPUT),
@@ -7947,8 +7948,9 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = {
 
 static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -7960,6 +7962,15 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = {
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -9167,7 +9178,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc882_targa_automute,
 	},
 	[ALC883_TARGA_8ch_DIG] = {
-		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
+		.mixers = { alc883_targa_mixer, alc883_targa_8ch_mixer,
+			    alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
 				alc883_targa_verbs },
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
@@ -13370,7 +13382,8 @@ static const char *alc269_models[ALC269_MODEL_LAST] = {
 	[ALC269_ASUS_EEEPC_P703]	= "eeepc-p703",
 	[ALC269_ASUS_EEEPC_P901]	= "eeepc-p901",
 	[ALC269_FUJITSU]		= "fujitsu",
-	[ALC269_LIFEBOOK]		= "lifebook"
+	[ALC269_LIFEBOOK]		= "lifebook",
+	[ALC269_AUTO]			= "auto",
 };
 
 static struct snd_pci_quirk alc269_cfg_tbl[] = {
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index e31e53d..826137e 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -864,10 +864,6 @@ static struct hda_verb stac92hd73xx_core_init[] = {
 };
 
 static struct hda_verb stac92hd83xxx_core_init[] = {
-	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x0},
-
 	/* power state controls amps */
 	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
 	{}
@@ -1590,8 +1586,8 @@ static unsigned int ref92hd83xxx_pin_configs[10] = {
 };
 
 static unsigned int dell_s14_pin_configs[10] = {
-	0x02214030, 0x02211010, 0x02a19020, 0x01014050,
-	0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160,
+	0x0221403f, 0x0221101f, 0x02a19020, 0x90170110,
+	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a60160,
 	0x40f000f0, 0x40f000f0,
 };
 
@@ -1690,6 +1686,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP mini 1000", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
 		      "HP HDX", STAC_HP_HDX),  /* HDX16 */
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620,
+		      "HP dv6", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
 		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
@@ -4166,7 +4164,10 @@ static int stac92xx_init(struct hda_codec *codec)
 		stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
 				AC_PINCTL_OUT_EN);
 		/* fake event to set up pins */
-		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+		if (cfg->hp_pins[0])
+			stac_issue_unsol_event(codec, cfg->hp_pins[0]);
+		else if (cfg->line_out_pins[0])
+			stac_issue_unsol_event(codec, cfg->line_out_pins[0]);
 	} else {
 		stac92xx_auto_init_multi_out(codec);
 		stac92xx_auto_init_hp_out(codec);
@@ -4688,8 +4689,13 @@ static int stac92xx_resume(struct hda_codec *codec)
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
 	/* fake event to set up pins again to override cached values */
-	if (spec->hp_detect)
-		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+	if (spec->hp_detect) {
+		if (spec->autocfg.hp_pins[0])
+			stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
+		else if (spec->autocfg.line_out_pins[0])
+			stac_issue_unsol_event(codec,
+					       spec->autocfg.line_out_pins[0]);
+	}
 	return 0;
 }
 
@@ -5016,7 +5022,7 @@ again:
 		spec->eapd_switch = 1;
 		break;
 	}
-	if (spec->board_config > STAC_92HD73XX_REF) {
+	if (spec->board_config != STAC_92HD73XX_REF) {
 		/* GPIO0 High = Enable EAPD */
 		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1;
 		spec->gpio_data = 0x01;
@@ -5066,7 +5072,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 
 	codec->spec = spec;
 	codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs;
-	spec->mono_nid = 0x19;
 	spec->digbeep_nid = 0x21;
 	spec->mux_nids = stac92hd83xxx_mux_nids;
 	spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids);
@@ -5242,7 +5247,7 @@ again:
 		stac92xx_set_config_regs(codec,
 				stac92hd71bxx_brd_tbl[spec->board_config]);
 
-	if (spec->board_config > STAC_92HD71BXX_REF) {
+	if (spec->board_config != STAC_92HD71BXX_REF) {
 		/* GPIO0 = EAPD */
 		spec->gpio_mask = 0x01;
 		spec->gpio_dir = 0x01;
@@ -5375,6 +5380,11 @@ again:
 	case STAC_HP_DV5:
 		snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
 		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
+		/* HP dv6 gives the headphone pin as a line-out.  Thus we
+		 * need to set hp_detect flag here to force to enable HP
+		 * detection.
+		 */
+		spec->hp_detect = 1;
 		break;
 	case STAC_HP_HDX:
 		spec->num_dmics = 1;
@@ -5557,14 +5567,17 @@ static int patch_stac927x(struct hda_codec *codec)
 	spec->dac_list = stac927x_dac_nids;
 	spec->multiout.dac_nids = spec->dac_nids;
 
+	if (spec->board_config != STAC_D965_REF) {
+		/* GPIO0 High = Enable EAPD */
+		spec->eapd_mask = spec->gpio_mask = 0x01;
+		spec->gpio_dir = spec->gpio_data = 0x01;
+	}
+
 	switch (spec->board_config) {
 	case STAC_D965_3ST:
 	case STAC_D965_5ST:
 		/* GPIO0 High = Enable EAPD */
-		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x01;
-		spec->gpio_data = 0x01;
 		spec->num_dmics = 0;
-
 		spec->init = d965_core_init;
 		break;
 	case STAC_DELL_BIOS:
@@ -5583,16 +5596,11 @@ static int patch_stac927x(struct hda_codec *codec)
 		snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130);
 		/* fallthru */
 	case STAC_DELL_3ST:
-		/* GPIO2 High = Enable EAPD */
-		spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x04;
-		spec->gpio_data = 0x04;
-		switch (codec->subsystem_id) {
-		case 0x1028022f:
-			/* correct EAPD to be GPIO0 */
-			spec->eapd_mask = spec->gpio_mask = 0x01;
-			spec->gpio_dir = spec->gpio_data = 0x01;
-			break;
-		};
+		if (codec->subsystem_id != 0x1028022f) {
+			/* GPIO2 High = Enable EAPD */
+			spec->eapd_mask = spec->gpio_mask = 0x04;
+			spec->gpio_dir = spec->gpio_data = 0x04;
+		}
 		spec->dmic_nids = stac927x_dmic_nids;
 		spec->num_dmics = STAC927X_NUM_DMICS;
 
@@ -5601,14 +5609,9 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
 	default:
-		if (spec->board_config > STAC_D965_REF) {
-			/* GPIO0 High = Enable EAPD */
-			spec->eapd_mask = spec->gpio_mask = 0x01;
-			spec->gpio_dir = spec->gpio_data = 0x01;
-		}
 		spec->num_dmics = 0;
-
 		spec->init = stac927x_core_init;
+		break;
 	}
 
 	spec->num_caps = STAC927X_NUM_CAPS;
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 3612bb9..01343dc 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -18,7 +18,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index e62b277..9a049a1 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -28,7 +28,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index d8a013a..98d663a 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -12,7 +12,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 9ff62e3..6096d22 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	int size, irq, rc;
 	const __be32 *prop;
 	void __iomem *regs;
+	int ret;
 
 	/* Fetch the registers and IRQ of the PSC */
 	irq = irq_of_parse_and_map(op->node, 0);
@@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	/* Allocate and initialize the driver private data */
 	psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
 	if (!psc_dma) {
-		iounmap(regs);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto out_unmap;
 	}
 
 	/* Get the PSC ID */
 	prop = of_get_property(op->node, "cell-index", &size);
-	if (!prop || size < sizeof *prop)
-		return -ENODEV;
+	if (!prop || size < sizeof *prop) {
+		ret = -ENODEV;
+		goto out_free;
+	}
 
 	spin_lock_init(&psc_dma->lock);
 	mutex_init(&psc_dma->mutex);
@@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	if (!psc_dma->capture.bcom_task ||
 	    !psc_dma->playback.bcom_task) {
 		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
-		iounmap(regs);
-		kfree(psc_dma);
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out_free;
 	}
 
 	/* Disable all interrupts and reset the PSC */
@@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 			  &psc_dma_bcom_irq_tx, IRQF_SHARED,
 			  "psc-dma-playback", &psc_dma->playback);
 	if (rc) {
-		free_irq(psc_dma->irq, psc_dma);
-		free_irq(psc_dma->capture.irq,
-			 &psc_dma->capture);
-		free_irq(psc_dma->playback.irq,
-			 &psc_dma->playback);
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out_irq;
 	}
 
 	/* Save what we've done so it can be found again later */
@@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op)
 
 	/* Tell the ASoC OF helpers about it */
 	return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+out_irq:
+	free_irq(psc_dma->irq, psc_dma);
+	free_irq(psc_dma->capture.irq, &psc_dma->capture);
+	free_irq(psc_dma->playback.irq, &psc_dma->playback);
+out_free:
+	kfree(psc_dma);
+out_unmap:
+	iounmap(regs);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
 
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index aa7af0b..9bc4aa3 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -230,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
 
+#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
+
 /*
  * Wait for the LR signal to allow synchronisation to the L/R clock
  * from the codec. May only be needed for slave mode.
@@ -237,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
 {
 	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+	unsigned long loops = msecs_to_loops(5);
 
 	pr_debug("Entered %s\n", __func__);
 
-	while (1) {
+	while (--loops) {
 		iiscon = readl(i2s->regs + S3C2412_IISCON);
 		if (iiscon & S3C2412_IISCON_LRINDEX)
 			break;
 
-		if (timeout < jiffies) {
-			printk(KERN_ERR "%s: timeout\n", __func__);
-			return -ETIMEDOUT;
-		}
+		cpu_relax();
+	}
+
+	if (!loops) {
+		printk(KERN_ERR "%s: timeout\n", __func__);
+		return -ETIMEDOUT;
 	}
 
 	return 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0d8b08e..f79711b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1131,9 +1131,10 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 	ret = snprintf(buf, PAGE_SIZE, "%s: %s  in %d out %d\n",
 		       w->name, w->power ? "On" : "Off", in, out);
 
-	if (w->active && w->sname)
-		ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n",
-				w->sname);
+	if (w->sname)
+		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+				w->sname,
+				w->active ? "active" : "inactive");
 
 	list_for_each_entry(p, &w->sources, list_sink) {
 		if (p->connect)
--
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