Merge branch 'staging/for_v3.7' into v4l_for_linus

* staging/for_v3.7: (2891 commits) em28xx: regression fix: use DRX-K sync firmware requests on em28xx drxk: allow loading firmware synchrousnously em28xx: Make all em28xx extensions to be initialized asynchronously [media] tda18271: properly report read errors in tda18271_get_id [media] tda18271: delay IR & RF calibration until init() if delay_cal is set [media] MAINTAINERS: add Michael Krufky as tda827x maintainer [media] MAINTAINERS: add Michael Krufky as tda8290 maintainer [media] MAINTAINERS: add Michael Krufky as cxusb maintainer [media] MAINTAINERS: add Michael Krufky as lg2160 maintainer [media] MAINTAINERS: add Michael Krufky as lgdt3305 maintainer [media] MAINTAINERS: add Michael Krufky as mxl111sf maintainer [media] MAINTAINERS: add Michael Krufky as mxl5007t maintainer [media] MAINTAINERS: add Michael Krufky as tda18271 maintainer [media] s5p-tv: Report only multi-plane capabilities in vidioc_querycap [media] s5p-mfc: Fix misplaced return statement in s5p_mfc_suspend() [media] exynos-gsc: Add missing static storage class specifiers [media] exynos-gsc: Remove <linux/version.h> header file inclusion [media] s5p-fimc: Fix incorrect condition in fimc_lite_reqbufs() [media] s5p-tv: Fix potential NULL pointer dereference error [media] s5k6aa: Fix possible NULL pointer dereference ... Conflicts: drivers/media/platform/s5p-fimc/fimc-capture.c drivers/media/platform/s5p-fimc/fimc-lite.c
author: Mauro Carvalho Chehab <mchehab@redhat.com> 2012-10-05 09:36:26 -0300
committer: Mauro Carvalho Chehab <mchehab@redhat.com> 2012-10-05 09:36:26 -0300
commit: bd0d10498826ed150da5e4c45baf8b9c7088fb71 (patch)
tree: cdee4371121a355d627a655c4eef5c0047b0462a /sound
parent: 89de0f2cda8b784e51ebd6655fff7339e4ac552b (diff)
parent: 2425bb3d4016ed95ce83a90b53bd92c7f31091e4 (diff)
76 files changed, 696 insertions, 395 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 0d7b25e8164..4e1fda75c1c 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -106,7 +106,7 @@ static struct pxa2xx_pcm_client pxa2xx_ac97_pcm_client = {
 	.prepare		= pxa2xx_ac97_pcm_prepare,
 };
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 
 static int pxa2xx_ac97_do_suspend(struct snd_card *card)
 {
@@ -243,7 +243,7 @@ static struct platform_driver pxa2xx_ac97_driver = {
 	.driver		= {
 		.name	= "pxa2xx-ac97",
 		.owner	= THIS_MODULE,
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 		.pm	= &pxa2xx_ac97_pm_ops,
 #endif
 	},
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index eb4ceb71123..277ebce23a4 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -452,6 +452,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 	dac->regs = ioremap(regs->start, resource_size(regs));
 	if (!dac->regs) {
 		dev_dbg(&pdev->dev, "could not remap register memory\n");
+		retval = -ENOMEM;
 		goto out_free_card;
 	}
 
@@ -534,7 +535,7 @@ out_put_pclk:
 	return retval;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int atmel_abdac_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index bf47025bdf4..9052aff37f6 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -278,14 +278,9 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream,
 	if (retval < 0)
 		return retval;
 	/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
-	if (cpu_is_at32ap7000()) {
-		if (retval < 0)
-			return retval;
-		/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
-		if (retval == 1)
-			if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
-				dw_dma_cyclic_free(chip->dma.rx_chan);
-	}
+	if (cpu_is_at32ap7000() && retval == 1)
+		if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
+			dw_dma_cyclic_free(chip->dma.rx_chan);
 
 	/* Set restrictions to params. */
 	mutex_lock(&opened_mutex);
@@ -980,6 +975,7 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
 
 	if (!chip->regs) {
 		dev_dbg(&pdev->dev, "could not remap register memory\n");
+		retval = -ENOMEM;
 		goto err_ioremap;
 	}
 
@@ -1134,7 +1130,7 @@ err_snd_card_new:
 	return retval;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int atmel_ac97c_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index ec2118d0e27..eb60cb8dbb8 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f)
 	int maj = imajor(inode);
 	int ret;
 
-	if (f->f_flags & O_WRONLY)
+	if ((f->f_flags & O_ACCMODE) == O_WRONLY)
 		dirn = SND_COMPRESS_PLAYBACK;
-	else if (f->f_flags & O_RDONLY)
+	else if ((f->f_flags & O_ACCMODE) == O_RDONLY)
 		dirn = SND_COMPRESS_CAPTURE;
-	else {
-		pr_err("invalid direction\n");
+	else
 		return -EINVAL;
-	}
 
 	if (maj == snd_major)
 		compr = snd_lookup_minor_data(iminor(inode),
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index 4e7ec2b4987..d0f00356fc1 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -101,7 +101,7 @@ void *snd_malloc_sgbuf_pages(struct device *device,
 		if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, device,
 						 chunk, &tmpb) < 0) {
 			if (!sgbuf->pages)
-				return NULL;
+				goto _failed;
 			if (!res_size)
 				goto _failed;
 			size = sgbuf->pages * PAGE_SIZE;
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 1128b35b2b0..5a34355e78e 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1176,7 +1176,7 @@ static int __devexit loopback_remove(struct platform_device *devptr)
 	return 0;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int loopback_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index f7d3bfc6bca..54bb6644a59 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1064,7 +1064,7 @@ static int __devexit snd_dummy_remove(struct platform_device *devptr)
 	return 0;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int snd_dummy_suspend(struct device *pdev)
 {
 	struct snd_card *card = dev_get_drvdata(pdev);
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 1cff331a228..4608c2ca43f 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -554,6 +554,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
 	spin_lock_init(&mpu->output_lock);
 	spin_lock_init(&mpu->timer_lock);
 	mpu->hardware = hardware;
+	mpu->irq = -1;
 	if (! (info_flags & MPU401_INFO_INTEGRATED)) {
 		int res_size = hardware == MPU401_HW_PC98II ? 4 : 2;
 		mpu->res = request_region(port, res_size, "MPU401 UART");
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 6ca59fc6dcb..ef171295f6d 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -199,7 +199,7 @@ static void pcsp_stop_beep(struct snd_pcsp *chip)
 	pcspkr_stop_sound();
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int pcsp_suspend(struct device *dev)
 {
 	struct snd_pcsp *chip = dev_get_drvdata(dev);
@@ -212,7 +212,7 @@ static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL);
 #define PCSP_PM_OPS	&pcsp_pm
 #else
 #define PCSP_PM_OPS	NULL
-#endif	/* CONFIG_PM */
+#endif	/* CONFIG_PM_SLEEP */
 
 static void pcsp_shutdown(struct platform_device *dev)
 {
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index d14edb7d648..3c6c1e3226f 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -37,9 +37,6 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
 MODULE_LICENSE("GPL");
 
-#define FREQ_LO		((tea->tea5759 ? 760 :  875) * 1600U)
-#define FREQ_HI		((tea->tea5759 ? 910 : 1080) * 1600U)
-
 /*
  * definitions
  */
@@ -50,8 +47,8 @@ MODULE_LICENSE("GPL");
 #define TEA575X_BIT_BAND_MASK	(3<<20)
 #define TEA575X_BIT_BAND_FM	(0<<20)
 #define TEA575X_BIT_BAND_MW	(1<<20)
-#define TEA575X_BIT_BAND_LW	(1<<21)
-#define TEA575X_BIT_BAND_SW	(1<<22)
+#define TEA575X_BIT_BAND_LW	(2<<20)
+#define TEA575X_BIT_BAND_SW	(3<<20)
 #define TEA575X_BIT_PORT_0	(1<<19)		/* user bit */
 #define TEA575X_BIT_PORT_1	(1<<18)		/* user bit */
 #define TEA575X_BIT_SEARCH_MASK	(3<<16)		/* search level */
@@ -62,6 +59,37 @@ MODULE_LICENSE("GPL");
 #define TEA575X_BIT_DUMMY	(1<<15)		/* buffer */
 #define TEA575X_BIT_FREQ_MASK	0x7fff
 
+enum { BAND_FM, BAND_FM_JAPAN, BAND_AM };
+
+static const struct v4l2_frequency_band bands[] = {
+	{
+		.type = V4L2_TUNER_RADIO,
+		.index = 0,
+		.capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO |
+			      V4L2_TUNER_CAP_FREQ_BANDS,
+		.rangelow   =  87500 * 16,
+		.rangehigh  = 108000 * 16,
+		.modulation = V4L2_BAND_MODULATION_FM,
+	},
+	{
+		.type = V4L2_TUNER_RADIO,
+		.index = 0,
+		.capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO |
+			      V4L2_TUNER_CAP_FREQ_BANDS,
+		.rangelow   = 76000 * 16,
+		.rangehigh  = 91000 * 16,
+		.modulation = V4L2_BAND_MODULATION_FM,
+	},
+	{
+		.type = V4L2_TUNER_RADIO,
+		.index = 1,
+		.capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_FREQ_BANDS,
+		.rangelow   =  530 * 16,
+		.rangehigh  = 1710 * 16,
+		.modulation = V4L2_BAND_MODULATION_AM,
+	},
+};
+
 /*
  * lowlevel part
  */
@@ -133,16 +161,29 @@ static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val)
 	if (freq == 0)
 		return freq;
 
-	/* freq *= 12.5 */
-	freq *= 125;
-	freq /= 10;
-	/* crystal fixup */
-	if (tea->tea5759)
-		freq += TEA575X_FMIF;
-	else
+	switch (tea->band) {
+	case BAND_FM:
+		/* freq *= 12.5 */
+		freq *= 125;
+		freq /= 10;
+		/* crystal fixup */
 		freq -= TEA575X_FMIF;
+		break;
+	case BAND_FM_JAPAN:
+		/* freq *= 12.5 */
+		freq *= 125;
+		freq /= 10;
+		/* crystal fixup */
+		freq += TEA575X_FMIF;
+		break;
+	case BAND_AM:
+		/* crystal fixup */
+		freq -= TEA575X_AMIF;
+		break;
+	}
 
-	return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */
+	return clamp(freq * 16, bands[tea->band].rangelow,
+				bands[tea->band].rangehigh); /* from kHz */
 }
 
 static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
@@ -150,21 +191,37 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
 	return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea));
 }
 
-static void snd_tea575x_set_freq(struct snd_tea575x *tea)
+void snd_tea575x_set_freq(struct snd_tea575x *tea)
 {
-	u32 freq = tea->freq;
+	u32 freq = tea->freq / 16;	/* to kHz */
+	u32 band = 0;
 
-	freq /= 16;		/* to kHz */
-	/* crystal fixup */
-	if (tea->tea5759)
-		freq -= TEA575X_FMIF;
-	else
+	switch (tea->band) {
+	case BAND_FM:
+		band = TEA575X_BIT_BAND_FM;
+		/* crystal fixup */
 		freq += TEA575X_FMIF;
-	/* freq /= 12.5 */
-	freq *= 10;
-	freq /= 125;
+		/* freq /= 12.5 */
+		freq *= 10;
+		freq /= 125;
+		break;
+	case BAND_FM_JAPAN:
+		band = TEA575X_BIT_BAND_FM;
+		/* crystal fixup */
+		freq -= TEA575X_FMIF;
+		/* freq /= 12.5 */
+		freq *= 10;
+		freq /= 125;
+		break;
+	case BAND_AM:
+		band = TEA575X_BIT_BAND_MW;
+		/* crystal fixup */
+		freq += TEA575X_AMIF;
+		break;
+	}
 
-	tea->val &= ~TEA575X_BIT_FREQ_MASK;
+	tea->val &= ~(TEA575X_BIT_FREQ_MASK | TEA575X_BIT_BAND_MASK);
+	tea->val |= band;
 	tea->val |= freq & TEA575X_BIT_FREQ_MASK;
 	snd_tea575x_write(tea, tea->val);
 	tea->freq = snd_tea575x_val_to_freq(tea, tea->val);
@@ -190,23 +247,57 @@ static int vidioc_querycap(struct file *file, void  *priv,
 	return 0;
 }
 
+static int vidioc_enum_freq_bands(struct file *file, void *priv,
+					 struct v4l2_frequency_band *band)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+	int index;
+
+	if (band->tuner != 0)
+		return -EINVAL;
+
+	switch (band->index) {
+	case 0:
+		if (tea->tea5759)
+			index = BAND_FM_JAPAN;
+		else
+			index = BAND_FM;
+		break;
+	case 1:
+		if (tea->has_am) {
+			index = BAND_AM;
+			break;
+		}
+		/* Fall through */
+	default:
+		return -EINVAL;
+	}
+
+	*band = bands[index];
+	if (!tea->cannot_read_data)
+		band->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED;
+
+	return 0;
+}
+
 static int vidioc_g_tuner(struct file *file, void *priv,
 					struct v4l2_tuner *v)
 {
 	struct snd_tea575x *tea = video_drvdata(file);
+	struct v4l2_frequency_band band_fm = { 0, };
 
 	if (v->index > 0)
 		return -EINVAL;
 
 	snd_tea575x_read(tea);
+	vidioc_enum_freq_bands(file, priv, &band_fm);
 
-	strcpy(v->name, "FM");
+	memset(v, 0, sizeof(*v));
+	strlcpy(v->name, tea->has_am ? "FM/AM" : "FM", sizeof(v->name));
 	v->type = V4L2_TUNER_RADIO;
-	v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO;
-	if (!tea->cannot_read_data)
-		v->capability |= V4L2_TUNER_CAP_HWSEEK_BOUNDED;
-	v->rangelow = FREQ_LO;
-	v->rangehigh = FREQ_HI;
+	v->capability = band_fm.capability;
+	v->rangelow = tea->has_am ? bands[BAND_AM].rangelow : band_fm.rangelow;
+	v->rangehigh = band_fm.rangehigh;
 	v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO;
 	v->audmode = (tea->val & TEA575X_BIT_MONO) ?
 		V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO;
@@ -218,13 +309,17 @@ static int vidioc_s_tuner(struct file *file, void *priv,
 					struct v4l2_tuner *v)
 {
 	struct snd_tea575x *tea = video_drvdata(file);
+	u32 orig_val = tea->val;
 
 	if (v->index)
 		return -EINVAL;
 	tea->val &= ~TEA575X_BIT_MONO;
 	if (v->audmode == V4L2_TUNER_MODE_MONO)
 		tea->val |= TEA575X_BIT_MONO;
-	snd_tea575x_write(tea, tea->val);
+	/* Only apply changes if currently tuning FM */
+	if (tea->band != BAND_AM && tea->val != orig_val)
+		snd_tea575x_set_freq(tea);
+
 	return 0;
 }
 
@@ -248,24 +343,56 @@ static int vidioc_s_frequency(struct file *file, void *priv,
 	if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO)
 		return -EINVAL;
 
-	tea->val &= ~TEA575X_BIT_SEARCH;
-	tea->freq = clamp(f->frequency, FREQ_LO, FREQ_HI);
+	if (tea->has_am && f->frequency < (20000 * 16))
+		tea->band = BAND_AM;
+	else if (tea->tea5759)
+		tea->band = BAND_FM_JAPAN;
+	else
+		tea->band = BAND_FM;
+
+	tea->freq = clamp(f->frequency, bands[tea->band].rangelow,
+					bands[tea->band].rangehigh);
 	snd_tea575x_set_freq(tea);
 	return 0;
 }
 
 static int vidioc_s_hw_freq_seek(struct file *file, void *fh,
-					struct v4l2_hw_freq_seek *a)
+					const struct v4l2_hw_freq_seek *a)
 {
 	struct snd_tea575x *tea = video_drvdata(file);
 	unsigned long timeout;
-	int i;
+	int i, spacing;
 
 	if (tea->cannot_read_data)
 		return -ENOTTY;
 	if (a->tuner || a->wrap_around)
 		return -EINVAL;
 
+	if (file->f_flags & O_NONBLOCK)
+		return -EWOULDBLOCK;
+
+	if (a->rangelow || a->rangehigh) {
+		for (i = 0; i < ARRAY_SIZE(bands); i++) {
+			if ((i == BAND_FM && tea->tea5759) ||
+			    (i == BAND_FM_JAPAN && !tea->tea5759) ||
+			    (i == BAND_AM && !tea->has_am))
+				continue;
+			if (bands[i].rangelow  == a->rangelow &&
+			    bands[i].rangehigh == a->rangehigh)
+				break;
+		}
+		if (i == ARRAY_SIZE(bands))
+			return -EINVAL; /* No matching band found */
+		if (i != tea->band) {
+			tea->band = i;
+			tea->freq = clamp(tea->freq, bands[i].rangelow,
+						     bands[i].rangehigh);
+			snd_tea575x_set_freq(tea);
+		}
+	}
+
+	spacing = (tea->band == BAND_AM) ? 5 : 50; /* kHz */
+
 	/* clear the frequency, HW will fill it in */
 	tea->val &= ~TEA575X_BIT_FREQ_MASK;
 	tea->val |= TEA575X_BIT_SEARCH;
@@ -297,10 +424,10 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh,
 			if (freq == 0) /* shouldn't happen */
 				break;
 			/*
-			 * if we moved by less than 50 kHz, or in the wrong
-			 * direction, continue seeking
+			 * if we moved by less than the spacing, or in the
+			 * wrong direction, continue seeking
 			 */
-			if (abs(tea->freq - freq) < 16 * 50 ||
+			if (abs(tea->freq - freq) < 16 * spacing ||
 					(a->seek_upward && freq < tea->freq) ||
 					(!a->seek_upward && freq > tea->freq)) {
 				snd_tea575x_write(tea, tea->val);
@@ -344,6 +471,7 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = {
 	.vidioc_g_frequency = vidioc_g_frequency,
 	.vidioc_s_frequency = vidioc_s_frequency,
 	.vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek,
+	.vidioc_enum_freq_bands = vidioc_enum_freq_bands,
 	.vidioc_log_status  = v4l2_ctrl_log_status,
 	.vidioc_subscribe_event = v4l2_ctrl_subscribe_event,
 	.vidioc_unsubscribe_event = v4l2_event_unsubscribe,
@@ -446,3 +574,4 @@ module_exit(alsa_tea575x_module_exit)
 
 EXPORT_SYMBOL(snd_tea575x_init);
 EXPORT_SYMBOL(snd_tea575x_exit);
+EXPORT_SYMBOL(snd_tea575x_set_freq);
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 2d67c78c9f4..f7cdaf51512 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev,
 			irq[dev], dma8[dev], dma16[dev]);
 	}
 
-	if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
+	if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) {
 		snd_card_free(card);
 		return error;
 	}
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 1d47be8170b..b3b4f15e45b 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -612,10 +612,10 @@ static int snd_es1688_capture_close(struct snd_pcm_substream *substream)
 
 static int snd_es1688_free(struct snd_es1688 *chip)
 {
-	if (chip->res_port) {
+	if (chip->hardware != ES1688_HW_UNDEF)
 		snd_es1688_init(chip, 0);
+	if (chip->res_port)
 		release_and_free_resource(chip->res_port);
-	}
 	if (chip->irq >= 0)
 		free_irq(chip->irq, (void *) chip);
 	if (chip->dma8 >= 0) {
@@ -657,19 +657,27 @@ int snd_es1688_create(struct snd_card *card,
 		return -ENOMEM;
 	chip->irq = -1;
 	chip->dma8 = -1;
+	chip->hardware = ES1688_HW_UNDEF;
 	
-	if ((chip->res_port = request_region(port + 4, 12, "ES1688")) == NULL) {
+	chip->res_port = request_region(port + 4, 12, "ES1688");
+	if (chip->res_port == NULL) {
 		snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
-		return -EBUSY;
+		err = -EBUSY;
+		goto exit;
 	}
-	if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) {
+
+	err = request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip);
+	if (err < 0) {
 		snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
-		return -EBUSY;
+		goto exit;
 	}
+
 	chip->irq = irq;
-	if (request_dma(dma8, "ES1688")) {
+	err = request_dma(dma8, "ES1688");
+
+	if (err < 0) {
 		snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8);
-		return -EBUSY;
+		goto exit;
 	}
 	chip->dma8 = dma8;
 
@@ -685,14 +693,18 @@ int snd_es1688_create(struct snd_card *card,
 
 	err = snd_es1688_probe(chip);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	err = snd_es1688_init(chip, 1);
 	if (err < 0)
-		return err;
+		goto exit;
 
 	/* Register device */
-	return snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+exit:
+	if (err)
+		snd_es1688_free(chip);
+	return err;
 }
 
 static struct snd_pcm_ops snd_es1688_playback_ops = {
diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c
index 733b014ec7d..b2b3c014221 100644
--- a/sound/oss/sb_audio.c
+++ b/sound/oss/sb_audio.c
@@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed)
 	if (speed > 0)
 	{
 		int tmp;
-		int s = speed * devc->channels;
+		int s;
 
 		if (speed < 5000)
 			speed = 5000;
 		if (speed > 44100)
 			speed = 44100;
 
+		s = speed * devc->channels;
+
 		devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff;
 
 		tmp = 256 - devc->tconst;
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index f75f5ffdfdf..a71d1c14a0f 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
 
 	if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX &&
 		       codec_index != CS46XX_SECONDARY_CODEC_INDEX))
-		return -EINVAL;
+		return 0xffff;
 
 	chip->active_ctrl(chip, 1);
 
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 8e40262d411..2f6e9c762d3 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1725,8 +1725,10 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
 	atc_connect_resources(atc);
 
 	atc->timer = ct_timer_new(atc);
-	if (!atc->timer)
+	if (!atc->timer) {
+		err = -ENOMEM;
 		goto error1;
+	}
 
 	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops);
 	if (err < 0)
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index 4f502a2bdc3..0a436626182 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -326,7 +326,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst
 	for (page = blk->first_page; page <= blk->last_page; page++, idx++) {
 		unsigned long ofs = idx << PAGE_SHIFT;
 		dma_addr_t addr;
-		addr = snd_pcm_sgbuf_get_addr(substream, ofs);
+		if (ofs >= runtime->dma_bytes)
+			addr = emu->silent_page.addr;
+		else
+			addr = snd_pcm_sgbuf_get_addr(substream, ofs);
 		if (! is_valid_page(emu, addr)) {
 			printk(KERN_ERR "emu: failure page = %d\n", idx);
 			mutex_unlock(&hdr->block_mutex);
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 647218d69f6..4f7d2dfcef7 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -332,13 +332,12 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	if (cfg->dig_outs)
 		snd_printd("   dig-out=0x%x/0x%x\n",
 			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
-	snd_printd("   inputs:");
+	snd_printd("   inputs:\n");
 	for (i = 0; i < cfg->num_inputs; i++) {
-		snd_printd(" %s=0x%x",
+		snd_printd("     %s=0x%x\n",
 			    hda_get_autocfg_input_label(codec, cfg, i),
 			    cfg->inputs[i].pin);
 	}
-	snd_printd("\n");
 	if (cfg->dig_in_pin)
 		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
 
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 0bc2315b181..0849aac449f 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -231,16 +231,22 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
 
+static bool ctl_has_mute(struct snd_kcontrol *kcontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	return query_amp_caps(codec, get_amp_nid(kcontrol),
+			      get_amp_direction(kcontrol)) & AC_AMPCAP_MUTE;
+}
+
 /* get/put callbacks for beep mute mixer switches */
 int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep) {
+	if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
 		ucontrol->value.integer.value[0] =
-			ucontrol->value.integer.value[1] =
-			beep->enabled;
+			ucontrol->value.integer.value[1] = beep->enabled;
 		return 0;
 	}
 	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -252,9 +258,20 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_beep *beep = codec->beep;
-	if (beep)
-		snd_hda_enable_beep_device(codec,
-					   *ucontrol->value.integer.value);
+	if (beep) {
+		u8 chs = get_amp_channels(kcontrol);
+		int enable = 0;
+		long *valp = ucontrol->value.integer.value;
+		if (chs & 1) {
+			enable |= *valp;
+			valp++;
+		}
+		if (chs & 2)
+			enable |= *valp;
+		snd_hda_enable_beep_device(codec, enable);
+	}
+	if (!ctl_has_mute(kcontrol))
+		return 0;
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 88a9c20eb7a..1c65cc5e3a3 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	kfree(codec);
 }
 
+static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec,
+				hda_nid_t fg, unsigned int power_state);
+
 static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state);
 
@@ -1317,6 +1320,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus,
 					   AC_VERB_GET_SUBSYSTEM_ID, 0);
 	}
 
+	codec->epss = snd_hda_codec_get_supported_ps(codec,
+					codec->afg ? codec->afg : codec->mfg,
+					AC_PWRST_EPSS);
+
 	/* power-up all before initialization */
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
@@ -1386,6 +1393,44 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
 
+/* update the stream-id if changed */
+static void update_pcm_stream_id(struct hda_codec *codec,
+				 struct hda_cvt_setup *p, hda_nid_t nid,
+				 u32 stream_tag, int channel_id)
+{
+	unsigned int oldval, newval;
+
+	if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
+		oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
+		newval = (stream_tag << 4) | channel_id;
+		if (oldval != newval)
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_CHANNEL_STREAMID,
+					    newval);
+		p->stream_tag = stream_tag;
+		p->channel_id = channel_id;
+	}
+}
+
+/* update the format-id if changed */
+static void update_pcm_format(struct hda_codec *codec, struct hda_cvt_setup *p,
+			      hda_nid_t nid, int format)
+{
+	unsigned int oldval;
+
+	if (p->format_id != format) {
+		oldval = snd_hda_codec_read(codec, nid, 0,
+					    AC_VERB_GET_STREAM_FORMAT, 0);
+		if (oldval != format) {
+			msleep(1);
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_STREAM_FORMAT,
+					    format);
+		}
+		p->format_id = format;
+	}
+}
+
 /**
  * snd_hda_codec_setup_stream - set up the codec for streaming
  * @codec: the CODEC to set up
@@ -1400,7 +1445,6 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 {
 	struct hda_codec *c;
 	struct hda_cvt_setup *p;
-	unsigned int oldval, newval;
 	int type;
 	int i;
 
@@ -1413,29 +1457,13 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
 	p = get_hda_cvt_setup(codec, nid);
 	if (!p)
 		return;
-	/* update the stream-id if changed */
-	if (p->stream_tag != stream_tag || p->channel_id != channel_id) {
-		oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
-		newval = (stream_tag << 4) | channel_id;
-		if (oldval != newval)
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_CHANNEL_STREAMID,
-					    newval);
-		p->stream_tag = stream_tag;
-		p->channel_id = channel_id;
-	}
-	/* update the format-id if changed */
-	if (p->format_id != format) {
-		oldval = snd_hda_codec_read(codec, nid, 0,
-					    AC_VERB_GET_STREAM_FORMAT, 0);
-		if (oldval != format) {
-			msleep(1);
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_STREAM_FORMAT,
-					    format);
-		}
-		p->format_id = format;
-	}
+
+	if (codec->pcm_format_first)
+		update_pcm_format(codec, p, nid, format);
+	update_pcm_stream_id(codec, p, nid, stream_tag, channel_id);
+	if (!codec->pcm_format_first)
+		update_pcm_format(codec, p, nid, format);
+
 	p->active = 1;
 	p->dirty = 0;
 
@@ -2325,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	}
 	if (codec->patch_ops.free)
 		codec->patch_ops.free(codec);
+	memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
 	snd_hda_jack_tbl_clear(codec);
 	codec->proc_widget_hook = NULL;
 	codec->spec = NULL;
@@ -2340,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
 	codec->preset = NULL;
-	memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
 	codec->slave_dig_outs = NULL;
 	codec->spdif_status_reset = 0;
 	module_put(codec->owner);
@@ -3497,7 +3525,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg
 {
 	int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
 
-	if (sup < 0)
+	if (sup == -1)
 		return false;
 	if (sup & power_state)
 		return true;
@@ -3522,8 +3550,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	/* this delay seems necessary to avoid click noise at power-down */
 	if (power_state == AC_PWRST_D3) {
 		/* transition time less than 10ms for power down */
-		bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
-		msleep(epss ? 10 : 100);
+		msleep(codec->epss ? 10 : 100);
 	}
 
 	/* repeat power states setting at most 10 times*/
@@ -4433,6 +4460,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 	 * then there is no need to go through power up here.
 	 */
 	if (codec->power_on) {
+		if (codec->power_transition < 0)
+			codec->power_transition = 0;
 		spin_unlock(&codec->power_lock);
 		return;
 	}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c422d330ca5..e5a7e19a807 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -861,6 +861,8 @@ struct hda_codec {
 	unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 	unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
 	unsigned int no_jack_detect:1;	/* Machine has no jack-detection */
+	unsigned int pcm_format_first:1; /* PCM format must be set first */
+	unsigned int epss:1;		/* supporting EPSS? */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	int power_transition;	/* power-state in transition */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8aced182fd..c4763c52eaf 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, CPT},"
 			 "{Intel, PPT},"
 			 "{Intel, LPT},"
+			 "{Intel, LPT_LP},"
 			 "{Intel, HPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
@@ -2700,6 +2701,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF),
+	SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
@@ -3270,6 +3273,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
 	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Lynx Point-LP */
+	{ PCI_DEVICE(0x8086, 0x9c20),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Lynx Point-LP */
+	{ PCI_DEVICE(0x8086, 0x9c21),
+	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* Haswell */
 	{ PCI_DEVICE(0x8086, 0x0c0c),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 7e46258fc70..6894ec66258 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -412,7 +412,7 @@ static void print_digital_conv(struct snd_info_buffer *buffer,
 	if (digi1 & AC_DIG1_EMPHASIS)
 		snd_iprintf(buffer, " Preemphasis");
 	if (digi1 & AC_DIG1_COPYRIGHT)
-		snd_iprintf(buffer, " Copyright");
+		snd_iprintf(buffer, " Non-Copyright");
 	if (digi1 & AC_DIG1_NONAUDIO)
 		snd_iprintf(buffer, " Non-Audio");
 	if (digi1 & AC_DIG1_PROFESSIONAL)
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index d0d3540e39e..49750a96d64 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -246,7 +246,7 @@ static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_OUT_UNMUTE);
 	}
-	if (dac)
+	if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
 		snd_hda_codec_write(codec, dac, 0,
 				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
 }
@@ -261,7 +261,7 @@ static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
 					    AC_VERB_SET_AMP_GAIN_MUTE,
 					    AMP_IN_UNMUTE(0));
 	}
-	if (adc)
+	if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP))
 		snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_IN_UNMUTE(0));
 }
@@ -275,6 +275,10 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+	if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_MUTE) == 0) {
+		snd_printdd("Skipping '%s %s Switch' (no mute on node 0x%x)\n", pfx, dirstr[dir], nid);
+		return 0;
+	}
 	sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
@@ -286,6 +290,10 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
 	int type = dir ? HDA_INPUT : HDA_OUTPUT;
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+	if ((query_amp_caps(codec, nid, type) & AC_AMPCAP_NUM_STEPS) == 0) {
+		snd_printdd("Skipping '%s %s Volume' (no amp on node 0x%x)\n", pfx, dirstr[dir], nid);
+		return 0;
+	}
 	sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
 	return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
 }
@@ -464,50 +472,17 @@ exit:
 }
 
 /*
- * PCM stuffs
+ * PCM callbacks
  */
-static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid,
-				 u32 stream_tag,
-				 int channel_id, int format)
+static int ca0132_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    struct snd_pcm_substream *substream)
 {
-	unsigned int oldval, newval;
-
-	if (!nid)
-		return;
-
-	snd_printdd("ca0132_setup_stream: "
-		"NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
-		nid, stream_tag, channel_id, format);
-
-	/* update the format-id if changed */
-	oldval = snd_hda_codec_read(codec, nid, 0,
-				    AC_VERB_GET_STREAM_FORMAT,
-				    0);
-	if (oldval != format) {
-		msleep(20);
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_STREAM_FORMAT,
-				    format);
-	}
-
-	oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0);
-	newval = (stream_tag << 4) | channel_id;
-	if (oldval != newval) {
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_CHANNEL_STREAMID,
-				    newval);
-	}
-}
-
-static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid)
-{
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0);
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+	struct ca0132_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+					     hinfo);
 }
 
-/*
- * PCM callbacks
- */
 static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			unsigned int stream_tag,
@@ -515,10 +490,8 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+						stream_tag, format, substream);
 }
 
 static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -526,92 +499,45 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dacs[0]);
-
-	return 0;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
 }
 
 /*
  * Digital out
  */
-static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			unsigned int stream_tag,
-			unsigned int format,
-			struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dig_out, stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
 }
 
-static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
-{
-	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dig_out);
-
-	return 0;
-}
-
-/*
- * Analog capture
- */
-static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_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 ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->adcs[substream->number],
-			     stream_tag, 0, format);
-
-	return 0;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+					     stream_tag, format, substream);
 }
 
-static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 			struct hda_codec *codec,
 			struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->adcs[substream->number]);
-
-	return 0;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
 }
 
-/*
- * Digital capture
- */
-static int ca0132_dig_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			unsigned int stream_tag,
-			unsigned int format,
-			struct snd_pcm_substream *substream)
+static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 struct snd_pcm_substream *substream)
 {
 	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_setup_stream(codec, spec->dig_in, stream_tag, 0, format);
-
-	return 0;
-}
-
-static int ca0132_dig_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-			struct hda_codec *codec,
-			struct snd_pcm_substream *substream)
-{
-	struct ca0132_spec *spec = codec->spec;
-
-	ca0132_cleanup_stream(codec, spec->dig_in);
-
-	return 0;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
 /*
@@ -621,6 +547,7 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.ops = {
+		.open = ca0132_playback_pcm_open,
 		.prepare = ca0132_playback_pcm_prepare,
 		.cleanup = ca0132_playback_pcm_cleanup
 	},
@@ -630,10 +557,6 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.ops = {
-		.prepare = ca0132_capture_pcm_prepare,
-		.cleanup = ca0132_capture_pcm_cleanup
-	},
 };
 
 static struct hda_pcm_stream ca0132_pcm_digital_playback = {
@@ -641,6 +564,8 @@ static struct hda_pcm_stream ca0132_pcm_digital_playback = {
 	.channels_min = 2,
 	.channels_max = 2,
 	.ops = {
+		.open = ca0132_dig_playback_pcm_open,
+		.close = ca0132_dig_playback_pcm_close,
 		.prepare = ca0132_dig_playback_pcm_prepare,
 		.cleanup = ca0132_dig_playback_pcm_cleanup
 	},
@@ -650,10 +575,6 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 2,
-	.ops = {
-		.prepare = ca0132_dig_capture_pcm_prepare,
-		.cleanup = ca0132_dig_capture_pcm_cleanup
-	},
 };
 
 static int ca0132_build_pcms(struct hda_codec *codec)
@@ -928,18 +849,16 @@ static int ca0132_build_controls(struct hda_codec *codec)
 						    spec->dig_out);
 		if (err < 0)
 			return err;
-		err = add_out_volume(codec, spec->dig_out, "IEC958");
+		err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
 		if (err < 0)
 			return err;
+		/* spec->multiout.share_spdif = 1; */
 	}
 
 	if (spec->dig_in) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
 		if (err < 0)
 			return err;
-		err = add_in_volume(codec, spec->dig_in, "IEC958");
-		if (err < 0)
-			return err;
 	}
 	return 0;
 }
@@ -961,6 +880,9 @@ static void ca0132_config(struct hda_codec *codec)
 	struct ca0132_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 
+	codec->pcm_format_first = 1;
+	codec->no_sticky_stream = 1;
+
 	/* line-outs */
 	cfg->line_outs = 1;
 	cfg->line_out_pins[0] = 0x0b; /* front */
@@ -988,14 +910,24 @@ static void ca0132_config(struct hda_codec *codec)
 
 	/* Mic-in */
 	spec->input_pins[0] = 0x12;
-	spec->input_labels[0] = "Mic-In";
+	spec->input_labels[0] = "Mic";
 	spec->adcs[0] = 0x07;
 
 	/* Line-In */
 	spec->input_pins[1] = 0x11;
-	spec->input_labels[1] = "Line-In";
+	spec->input_labels[1] = "Line";
 	spec->adcs[1] = 0x08;
 	spec->num_inputs = 2;
+
+	/* SPDIF I/O */
+	spec->dig_out = 0x05;
+	spec->multiout.dig_out_nid = spec->dig_out;
+	cfg->dig_out_pins[0] = 0x0c;
+	cfg->dig_outs = 1;
+	cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+	spec->dig_in = 0x09;
+	cfg->dig_in_pin = 0x0e;
+	cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
 }
 
 static void ca0132_init_chip(struct hda_codec *codec)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 14361184ae1..5e22a8f43d2 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2967,12 +2967,10 @@ static const char * const cxt5066_models[CXT5066_MODELS] = {
 };
 
 static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO),
 	SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
-	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
@@ -2988,14 +2986,10 @@ static const 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, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
-	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
-	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
 	SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
-	SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
-	SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 641408dc28c..8f23374fa64 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1165,14 +1165,20 @@ static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 					     struct hda_codec *codec,
 					     struct snd_pcm_substream *substream)
 {
+	snd_hda_codec_cleanup_stream(codec, hinfo->nid);
+	return 0;
+}
+
+static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
+			  struct hda_codec *codec,
+			  struct snd_pcm_substream *substream)
+{
 	struct hdmi_spec *spec = codec->spec;
 	int cvt_idx, pin_idx;
 	struct hdmi_spec_per_cvt *per_cvt;
 	struct hdmi_spec_per_pin *per_pin;
 	int pinctl;
 
-	snd_hda_codec_cleanup_stream(codec, hinfo->nid);
-
 	if (hinfo->nid) {
 		cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid);
 		if (snd_BUG_ON(cvt_idx < 0))
@@ -1195,12 +1201,12 @@ static int generic_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 				    pinctl & ~PIN_OUT);
 		snd_hda_spdif_ctls_unassign(codec, pin_idx);
 	}
-
 	return 0;
 }
 
 static const struct hda_pcm_ops generic_ops = {
 	.open = hdmi_pcm_open,
+	.close = hdmi_pcm_close,
 	.prepare = generic_hdmi_playback_pcm_prepare,
 	.cleanup = generic_hdmi_playback_pcm_cleanup,
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f141395dfee..4f81dd44c83 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -203,6 +203,7 @@ struct alc_spec {
 	unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
 	unsigned int inv_dmic_fixup:1; /* has inverted digital-mic workaround */
 	unsigned int inv_dmic_muted:1; /* R-ch of inv d-mic is muted? */
+	unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
 
 	/* auto-mute control */
 	int automute_mode;
@@ -4323,7 +4324,8 @@ static int alc_parse_auto_config(struct hda_codec *codec,
 		return 0; /* can't find valid BIOS pin config */
 	}
 
-	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
+	if (!spec->no_primary_hp &&
+	    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
 	    cfg->line_outs <= cfg->hp_outs) {
 		/* use HP as primary out */
 		cfg->speaker_outs = cfg->line_outs;
@@ -5050,6 +5052,7 @@ enum {
 	ALC889_FIXUP_MBP_VREF,
 	ALC889_FIXUP_IMAC91_VREF,
 	ALC882_FIXUP_INV_DMIC,
+	ALC882_FIXUP_NO_PRIMARY_HP,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -5171,6 +5174,17 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec,
 	spec->keep_vref_in_automute = 1;
 }
 
+/* Don't take HP output as primary
+ * strangely, the speaker output doesn't work on VAIO Z through DAC 0x05
+ */
+static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
+				       const struct alc_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action == ALC_FIXUP_ACT_PRE_PROBE)
+		spec->no_primary_hp = 1;
+}
+
 static const struct alc_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
 		.type = ALC_FIXUP_PINS,
@@ -5357,6 +5371,10 @@ static const struct alc_fixup alc882_fixups[] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc_fixup_inv_dmic_0x12,
 	},
+	[ALC882_FIXUP_NO_PRIMARY_HP] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc882_fixup_no_primary_hp,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -5391,6 +5409,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
 	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
+	SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
 
 	/* All Apple entries are in codec SSIDs */
 	SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF),
@@ -5432,6 +5451,7 @@ static const struct alc_model_fixup alc882_fixup_models[] = {
 	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
 	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
 	{.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"},
 	{}
 };
 
@@ -6079,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_PCM_44K] = {
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_pcm_44k,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_QUANTA_MUTE
 	},
 	[ALC269_FIXUP_STEREO_DMIC] = {
 		.type = ALC_FIXUP_FUNC,
@@ -6186,9 +6208,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	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(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
-	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
+	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 
 #if 0
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a1596a3b171..3d4722f0a1c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -101,6 +101,8 @@ enum {
 	STAC_92HD83XXX_HP_cNB11_INTQUAD,
 	STAC_HP_DV7_4000,
 	STAC_HP_ZEPHYR,
+	STAC_92HD83XXX_HP_LED,
+	STAC_92HD83XXX_HP_INV_LED,
 	STAC_92HD83XXX_MODELS
 };
 
@@ -1073,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = {
 
 static const char * const slave_pfxs[] = {
 	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Speaker", "IEC958",
+	"Headphone", "Speaker", "IEC958", "PCM",
 	NULL
 };
 
@@ -1675,6 +1677,8 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
 	[STAC_HP_DV7_4000] = "hp-dv7-4000",
 	[STAC_HP_ZEPHYR] = "hp-zephyr",
+	[STAC_92HD83XXX_HP_LED] = "hp-led",
+	[STAC_92HD83XXX_HP_INV_LED] = "hp-inv-led",
 };
 
 static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1729,6 +1733,8 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
 			  "HP", STAC_HP_ZEPHYR),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3660,
+			  "HP Mini", STAC_92HD83XXX_HP_LED),
 	{} /* terminator */
 };
 
@@ -4266,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec)
 	unsigned int gpio;
 	int i;
 
-	snd_hda_sequence_write(codec, spec->init);
+	if (spec->init)
+		snd_hda_sequence_write(codec, spec->init);
 
 	/* power down adcs initially */
 	if (spec->powerdown_adcs)
@@ -4414,7 +4421,12 @@ static int stac92xx_init(struct hda_codec *codec)
 	snd_hda_jack_report_sync(codec);
 
 	/* sync mute LED */
-	snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+	if (spec->gpio_led) {
+		if (spec->vmaster_mute.hook)
+			snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+		else /* the very first init call doesn't have vmaster yet */
+			stac92xx_update_led_status(codec, false);
+	}
 
 	/* sync the power-map */
 	if (spec->num_pwrs)
@@ -4531,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec,
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	int i;
 
+	if (cfg->speaker_outs == 0)
+		return;
+
 	for (i = 0; i < cfg->line_outs; i++) {
 		if (presence)
 			break;
@@ -5507,6 +5522,7 @@ static void stac92hd8x_fill_auto_spec(struct hda_codec *codec)
 static int patch_stac92hd83xxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	int default_polarity = -1; /* no default cfg */
 	int err;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -5518,6 +5534,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 		snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e);
 	}
 
+	codec->epss = 0; /* longer delay needed for D3 */
 	codec->no_trigger_sense = 1;
 	codec->spec = spec;
 
@@ -5555,9 +5572,15 @@ again:
 	case STAC_HP_ZEPHYR:
 		spec->init = stac92hd83xxx_hp_zephyr_init;
 		break;
+	case STAC_92HD83XXX_HP_LED:
+		default_polarity = 0;
+		break;
+	case STAC_92HD83XXX_HP_INV_LED:
+		default_polarity = 1;
+		break;
 	}
 
-	if (find_mute_led_cfg(codec, -1/*no default cfg*/))
+	if (find_mute_led_cfg(codec, default_polarity))
 		snd_printd("mute LED gpio %d polarity %d\n",
 				spec->gpio_led,
 				spec->gpio_led_polarity);
@@ -5730,7 +5753,6 @@ again:
 		/* fallthru */
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
-		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92xx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5755,7 +5777,6 @@ again:
 			spec->stream_delay = 40; /* 40 milliseconds */
 
 		/* disable VSW */
-		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
 		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
 		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5770,7 +5791,6 @@ again:
 
 		/* fallthru */
 	default:
-		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92xx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
@@ -5778,6 +5798,9 @@ again:
 		break;
 	}
 
+	if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB)
+		spec->init = stac92hd71bxx_core_init;
+
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
 		snd_hda_sequence_write_cache(codec, unmute_init);
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 90645560ed3..43077177691 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	vt1708_stop_hp_work(spec);
+
+	if (spec->codec_type == VT1802) {
+		/* Fix pop noise on headphones */
+		int i;
+		for (i = 0; i < spec->autocfg.hp_outs; i++)
+			snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0);
+	}
+
 	return 0;
 }
 #endif
@@ -3226,7 +3234,7 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	int imux_is_smixer;
-	unsigned int parm;
+	unsigned int parm, parm2;
 	/* MUX6 (1eh) = stereo mixer */
 	imux_is_smixer =
 	snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5;
@@ -3249,7 +3257,7 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
 	parm = AC_PWRST_D3;
 	set_pin_power_state(codec, 0x27, &parm);
 	update_power_state(codec, 0x1a, parm);
-	update_power_state(codec, 0xb, parm);
+	parm2 = parm; /* for pin 0x0b */
 
 	/* PW2 (26h), AOW2 (ah) */
 	parm = AC_PWRST_D3;
@@ -3264,6 +3272,9 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
 	if (!spec->hp_independent_mode) /* check for redirected HP */
 		set_pin_power_state(codec, 0x28, &parm);
 	update_power_state(codec, 0x8, parm);
+	if (!spec->hp_independent_mode && parm2 != AC_PWRST_D3)
+		parm = parm2;
+	update_power_state(codec, 0xb, parm);
 	/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
 	update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm);
 
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 764cc93dbca..075d5aa1fee 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 }
 
 static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
 
 static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
     {
@@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = {
 	.info = ak4396_dac_vol_info,
 	.get = ak4396_dac_vol_get,
 	.put = ak4396_dac_vol_put,
-	.tlv = { .p = db_scale_wm_dac },
+	.tlv = { .p = ak4396_db_scale },
     },
 };
 
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index d1ab4370673..5579b08bb35 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -851,6 +851,8 @@ static int __devinit lx_pcm_create(struct lx6464es *chip)
 	/* hardcoded device name & channel count */
 	err = snd_pcm_new(chip->card, (char *)card_name, 0,
 			  1, 1, &pcm);
+	if (err < 0)
+		return err;
 
 	pcm->private_data = chip;
 
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index b8ac8710f47..b12308b5ba2 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6585,7 +6585,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card,
 		snd_printk(KERN_ERR "HDSPM: "
 				"unable to kmalloc Mixer memory of %d Bytes\n",
 				(int)sizeof(struct hdspm_mixer));
-		return err;
+		return -ENOMEM;
 	}
 
 	hdspm->port_names_in = NULL;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 512434efcc3..805ab6e9a78 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1377,8 +1377,9 @@ static int __devinit sis_chip_create(struct snd_card *card,
 	if (rc)
 		goto error_out_cleanup;
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
-			sis)) {
+	rc = request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME,
+			 sis);
+	if (rc) {
 		dev_err(&pci->dev, "unable to allocate irq %d\n", sis->irq);
 		goto error_out_cleanup;
 	}
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index f5ceb6f282d..210cafe0489 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -143,7 +143,7 @@ static int __devexit snd_pmac_remove(struct platform_device *devptr)
 	return 0;
 }
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 static int snd_pmac_driver_suspend(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index 1aa52eff526..9b18b5243a5 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -1040,6 +1040,7 @@ static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
 				   GFP_KERNEL);
 	if (!the_card.null_buffer_start_vaddr) {
 		pr_info("%s: nullbuffer alloc failed\n", __func__);
+		ret = -ENOMEM;
 		goto clean_preallocate;
 	}
 	pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__,
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index 318c5ba5360..dfb744381c4 100644
--- a/sound/soc/blackfin/bf6xx-sport.c
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -413,7 +413,14 @@ EXPORT_SYMBOL(sport_create);
 
 void sport_delete(struct sport_device *sport)
 {
+	if (sport->tx_desc)
+		dma_free_coherent(NULL, sport->tx_desc_size,
+				sport->tx_desc, 0);
+	if (sport->rx_desc)
+		dma_free_coherent(NULL, sport->rx_desc_size,
+				sport->rx_desc, 0);
 	sport_free_resource(sport);
+	kfree(sport);
 }
 EXPORT_SYMBOL(sport_delete);
 
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 3c795921c5f..23b40186f9b 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 
 	/* Setup AB8500 according to board-settings */
 	pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
+
+	/* Inform SoC Core that we have our own I/O arrangements. */
+	codec->control_data = (void *)true;
+
 	status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
 	if (status < 0) {
 		pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 8c39dddd7d0..11b1b714b8b 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -186,6 +186,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec)
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
+	codec->control_data = codec;	/* we don't use regmap! */
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register AC97 codec\n");
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 5c9cacaf2d5..1cf7a32d1b2 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -426,7 +426,7 @@ static const int arizona_44k1_bclk_rates[] = {
 	940800,
 	1411200,
 	1881600,
-	2882400,
+	2822400,
 	3763200,
 	5644800,
 	7526400,
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 6276e352125..115a4030181 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -581,6 +581,8 @@ static int mc13783_probe(struct snd_soc_codec *codec)
 {
 	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
 
+	codec->control_data = priv->mc13xxx;
+
 	mc13xxx_lock(priv->mc13xxx);
 
 	/* these are the reset values */
@@ -657,7 +659,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = {
 		.id = MC13783_ID_STEREO_DAC,
 		.playback = {
 			.stream_name = "Playback",
-			.channels_min = 1,
+			.channels_min = 2,
 			.channels_max = 2,
 			.rates = SNDRV_PCM_RATE_8000_96000,
 			.formats = MC13783_FORMATS,
@@ -668,7 +670,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = {
 		.id = MC13783_ID_STEREO_CODEC,
 		.capture = {
 			.stream_name = "Capture",
-			.channels_min = 1,
+			.channels_min = 2,
 			.channels_max = 2,
 			.rates = MC13783_RATES_RECORD,
 			.formats = MC13783_FORMATS,
@@ -690,14 +692,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = {
 		.id = MC13783_ID_SYNC,
 		.playback = {
 			.stream_name = "Playback",
-			.channels_min = 1,
+			.channels_min = 2,
 			.channels_max = 2,
 			.rates = SNDRV_PCM_RATE_8000_96000,
 			.formats = MC13783_FORMATS,
 		},
 		.capture = {
 			.stream_name = "Capture",
-			.channels_min = 1,
+			.channels_min = 2,
 			.channels_max = 2,
 			.rates = MC13783_RATES_RECORD,
 			.formats = MC13783_FORMATS,
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 8af6a5245b1..df2f99d1d42 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -239,6 +239,7 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
 	{"Headphone Mux", "DAC", "DAC"},	/* dac --> hp_mux */
 	{"LO", NULL, "DAC"},			/* dac --> line_out */
 
+	{"LINE_IN", NULL, "VAG_POWER"},
 	{"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */
 	{"HP", NULL, "Headphone Mux"},		/* hp_mux --> hp */
 
@@ -1357,8 +1358,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 	if (ret)
 		goto err;
 
-	snd_soc_dapm_new_widgets(&codec->dapm);
-
 	return 0;
 
 err:
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 982e437799a..33c0f3d39c8 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -340,6 +340,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec)
 
 	printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
 
+	codec->control_data = codec;	/* we don't use regmap! */
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0)
 		goto codec_err;
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 3fd5b29dc93..a3acb7a85f6 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -702,7 +702,7 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
 }
 
 static const struct regmap_config wm2000_regmap = {
-	.reg_bits = 8,
+	.reg_bits = 16,
 	.val_bits = 8,
 
 	.max_register = WM2000_REG_IF_CTL,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 6537f16d383..e33d327396a 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
 
 ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
 
 SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
 		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
-SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
-		   ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
 
 ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
@@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
 
 ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
-ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
 
 ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
@@ -349,10 +343,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
 		 NULL, 0),
 SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
 		 NULL, 0),
-SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
-		 NULL, 0),
-SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
-		 NULL, 0),
 
 SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
 		 NULL, 0),
@@ -466,8 +456,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
 
 ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
 ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
-ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
-ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
 
 ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
 ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
@@ -553,8 +541,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
 	{ name, "EQ4", "EQ4" }, \
 	{ name, "DRC1L", "DRC1L" }, \
 	{ name, "DRC1R", "DRC1R" }, \
-	{ name, "DRC2L", "DRC2L" }, \
-	{ name, "DRC2R", "DRC2R" }, \
 	{ name, "LHPF1", "LHPF1" }, \
 	{ name, "LHPF2", "LHPF2" }, \
 	{ name, "LHPF3", "LHPF3" }, \
@@ -639,6 +625,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "SYSCLK" },
 	{ "AIF3 Capture", NULL, "SYSCLK" },
 
+	{ "IN1L PGA", NULL, "IN1L" },
+	{ "IN1R PGA", NULL, "IN1R" },
+
+	{ "IN2L PGA", NULL, "IN2L" },
+	{ "IN2R PGA", NULL, "IN2R" },
+
+	{ "IN3L PGA", NULL, "IN3L" },
+	{ "IN3R PGA", NULL, "IN3R" },
+
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
@@ -675,8 +670,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
 
 	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
 	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
-	ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
-	ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
 
 	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
 	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 8033f706518..01ebbcc5c6a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -681,6 +681,18 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "AIF2 Capture", NULL, "SYSCLK" },
 	{ "AIF3 Capture", NULL, "SYSCLK" },
 
+	{ "IN1L PGA", NULL, "IN1L" },
+	{ "IN1R PGA", NULL, "IN1R" },
+
+	{ "IN2L PGA", NULL, "IN2L" },
+	{ "IN2R PGA", NULL, "IN2R" },
+
+	{ "IN3L PGA", NULL, "IN3L" },
+	{ "IN3R PGA", NULL, "IN3R" },
+
+	{ "IN4L PGA", NULL, "IN4L" },
+	{ "IN4R PGA", NULL, "IN4R" },
+
 	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
 	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
 	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 0013afe48e6..dc4262eea4b 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = {
 	{ 14,  0x0000 },     /* R14  - Power Management 2 */
 	{ 15,  0x0000 },     /* R15  - Power Management 3 */
 	{ 18,  0x0000 },     /* R18  - Power Management 6 */
-	{ 19,  0x945E },     /* R20  - Clock Rates 0 */
+	{ 20,  0x945E },     /* R20  - Clock Rates 0 */
 	{ 21,  0x0C05 },     /* R21  - Clock Rates 1 */
 	{ 22,  0x0006 },     /* R22  - Clock Rates 2 */
 	{ 24,  0x0050 },     /* R24  - Audio Interface 0 */
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index eaf65863ec2..ce672007379 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2501,6 +2501,9 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
 		/* VMID 2*250k */
 		snd_soc_update_bits(codec, WM8962_PWR_MGMT_1,
 				    WM8962_VMID_SEL_MASK, 0x100);
+
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			msleep(100);
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -3730,21 +3733,6 @@ static int wm8962_runtime_resume(struct device *dev)
 
 	regcache_sync(wm8962->regmap);
 
-	regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
-			   WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA,
-			   WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA);
-
-	/* Bias enable at 2*50k for ramp */
-	regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
-			   WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA,
-			   WM8962_BIAS_ENA | 0x180);
-
-	msleep(5);
-
-	/* VMID back to 2x250k for standby */
-	regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1,
-			   WM8962_VMID_SEL_MASK, 0x100);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index bb62f4b3d56..6c9eeca85b9 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2649,7 +2649,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	bclk_rate = params_rate(params) * 2;
+	bclk_rate = params_rate(params) * 4;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		bclk_rate *= 16;
@@ -3253,10 +3253,13 @@ static void wm8994_mic_work(struct work_struct *work)
 	int ret;
 	int report;
 
+	pm_runtime_get_sync(dev);
+
 	ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, &reg);
 	if (ret < 0) {
 		dev_err(dev, "Failed to read microphone status: %d\n",
 			ret);
+		pm_runtime_put(dev);
 		return;
 	}
 
@@ -3299,6 +3302,8 @@ static void wm8994_mic_work(struct work_struct *work)
 
 	snd_soc_jack_report(priv->micdet[1].jack, report,
 			    SND_JACK_HEADSET | SND_JACK_BTN_0);
+
+	pm_runtime_put(dev);
 }
 
 static irqreturn_t wm8994_mic_irq(int irq, void *data)
@@ -3421,12 +3426,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 	int reg;
 	bool present;
 
+	pm_runtime_get_sync(codec->dev);
+
 	mutex_lock(&wm8994->accdet_lock);
 
 	reg = snd_soc_read(codec, WM1811_JACKDET_CTRL);
 	if (reg < 0) {
 		dev_err(codec->dev, "Failed to read jack status: %d\n", reg);
 		mutex_unlock(&wm8994->accdet_lock);
+		pm_runtime_put(codec->dev);
 		return IRQ_NONE;
 	}
 
@@ -3491,6 +3499,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
 				    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
 				    wm8994->btn_mask);
 
+	pm_runtime_put(codec->dev);
 	return IRQ_HANDLED;
 }
 
@@ -3602,6 +3611,8 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 	if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA))
 		return IRQ_HANDLED;
 
+	pm_runtime_get_sync(codec->dev);
+
 	/* We may occasionally read a detection without an impedence
 	 * range being provided - if that happens loop again.
 	 */
@@ -3612,6 +3623,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 			dev_err(codec->dev,
 				"Failed to read mic detect status: %d\n",
 				reg);
+			pm_runtime_put(codec->dev);
 			return IRQ_NONE;
 		}
 
@@ -3639,6 +3651,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 		dev_warn(codec->dev, "Accessory detection with no callback\n");
 
 out:
+	pm_runtime_put(codec->dev);
 	return IRQ_HANDLED;
 }
 
@@ -4025,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		break;
 	case WM8958:
 		if (wm8994->revision < 1) {
+			snd_soc_dapm_add_routes(dapm, wm8994_intercon,
+						ARRAY_SIZE(wm8994_intercon));
 			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
 						ARRAY_SIZE(wm8994_revd_intercon));
 			snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 099e6ec3212..c6d2076a796 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -148,7 +148,7 @@ SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
 
 SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
 SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
-SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
 SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
 
 SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
@@ -272,7 +272,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]);
 
 /* Mic select */
 static const struct snd_kcontrol_new wm9712_mic_src_controls =
-SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
 
 /* diff select */
 static const struct snd_kcontrol_new wm9712_diff_sel_controls =
@@ -291,7 +291,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
 	&wm9712_capture_selectl_controls),
 SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
 	&wm9712_capture_selectr_controls),
-SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
+	&wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
 	&wm9712_mic_src_controls),
 SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
 	&wm9712_diff_sel_controls),
@@ -319,6 +321,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
 SND_SOC_DAPM_OUTPUT("MONOOUT"),
 SND_SOC_DAPM_OUTPUT("HPOUTL"),
@@ -379,6 +382,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = {
 	{"Mic PGA", NULL, "MIC1"},
 	{"Mic PGA", NULL, "MIC2"},
 
+	/* microphones */
+	{"Differential Mic", NULL, "MIC1"},
+	{"Differential Mic", NULL, "MIC2"},
+	{"Left Mic Select Source", "Mic 1", "MIC1"},
+	{"Left Mic Select Source", "Mic 2", "MIC2"},
+	{"Left Mic Select Source", "Stereo", "MIC1"},
+	{"Left Mic Select Source", "Differential", "Differential Mic"},
+	{"Right Mic Select Source", "Mic 1", "MIC1"},
+	{"Right Mic Select Source", "Mic 2", "MIC2"},
+	{"Right Mic Select Source", "Stereo", "MIC2"},
+	{"Right Mic Select Source", "Differential", "Differential Mic"},
+
 	/* left capture selector */
 	{"Left Capture Select", "Mic", "MIC1"},
 	{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
@@ -619,6 +634,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
+	codec->control_data = codec;	/* we don't use regmap! */
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0) {
 		printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 3eb19fb71d1..d0b8a3287a8 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1196,6 +1196,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec)
 	if (wm9713 == NULL)
 		return -ENOMEM;
 	snd_soc_codec_set_drvdata(codec, wm9713);
+	codec->control_data = wm9713;	/* we don't use regmap! */
 
 	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
 	if (ret < 0)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 95441bfc819..ce5e5cd254d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -380,14 +380,20 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
 static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
 {
 	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (dev->txnumevt)	/* enable FIFO */
+		if (dev->txnumevt) {	/* enable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+								FIFO_ENABLE);
 			mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
 								FIFO_ENABLE);
+		}
 		mcasp_start_tx(dev);
 	} else {
-		if (dev->rxnumevt)	/* enable FIFO */
+		if (dev->rxnumevt) {	/* enable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+								FIFO_ENABLE);
 			mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
 								FIFO_ENABLE);
+		}
 		mcasp_start_rx(dev);
 	}
 }
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index fb21b17f17f..199408ec426 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "audmux internal port setup failed\n");
 		return ret;
 	}
-	imx_audmux_v2_configure_port(ext_port,
+	ret = imx_audmux_v2_configure_port(ext_port,
 			IMX_AUDMUX_V2_PTCR_SYN,
 			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
 	if (ret) {
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 28dd76c7cb1..81d7728cf67 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
 static struct snd_soc_dai_driver imx_ssi_dai = {
 	.probe = imx_ssi_dai_probe,
 	.playback = {
-		.channels_min = 1,
+		/* The SSI does not support monaural audio. */
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
-		.channels_min = 1,
+		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
index 99a997f19bb..b6fa77678d9 100644
--- a/sound/soc/mxs/Kconfig
+++ b/sound/soc/mxs/Kconfig
@@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC
 if SND_MXS_SOC
 
 config SND_SOC_MXS_SGTL5000
-	tristate "SoC Audio support for i.MX boards with sgtl5000"
+	tristate "SoC Audio support for MXS boards with sgtl5000"
 	depends on I2C
 	select SND_SOC_SGTL5000
 	help
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index aba71bfa33b..b3030718c22 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -394,9 +394,14 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *cpu_dai)
 {
 	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+	struct mxs_saif *master_saif;
 	u32 scr, stat;
 	int ret;
 
+	master_saif = mxs_saif_get_master(saif);
+	if (!master_saif)
+		return -EINVAL;
+
 	/* mclk should already be set */
 	if (!saif->mclk && saif->mclk_in_use) {
 		dev_err(cpu_dai->dev, "set mclk first\n");
@@ -420,6 +425,25 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	/* prepare clk in hw_param, enable in trigger */
+	clk_prepare(saif->clk);
+	if (saif != master_saif) {
+		/*
+		* Set an initial clock rate for the saif internal logic to work
+		* properly. This is important when working in EXTMASTER mode
+		* that uses the other saif's BITCLK&LRCLK but it still needs a
+		* basic clock which should be fast enough for the internal
+		* logic.
+		*/
+		clk_enable(saif->clk);
+		ret = clk_set_rate(saif->clk, 24000000);
+		clk_disable(saif->clk);
+		if (ret)
+			return ret;
+
+		clk_prepare(master_saif->clk);
+	}
+
 	scr = __raw_readl(saif->base + SAIF_CTRL);
 
 	scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 009533ab8d1..df65f98211e 100644
--- a/sound/soc/omap/am3517evm.c
+++ b/sound/soc/omap/am3517evm.c
@@ -59,7 +59,7 @@ static int am3517evm_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
-	snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
 				SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 34835e8a916..d33c48baaf7 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -745,7 +745,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux)
 {
 	const char *signal, *src;
 
-	if (mcbsp->pdata->mux_signal)
+	if (!mcbsp->pdata->mux_signal)
 		return -EINVAL;
 
 	switch (mux) {
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 1046083e90a..acdd3ef14e0 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -820,3 +820,4 @@ module_platform_driver(asoc_mcbsp_driver);
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-mcbsp");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5a649da9122..f0feb06615f 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -441,3 +441,4 @@ module_platform_driver(omap_pcm_driver);
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-pcm-audio");
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index f3ebc38c10f..b70964ea448 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				    SNDRV_PCM_INFO_MMAP |
-				    SNDRV_PCM_INFO_MMAP_VALID |
-				    SNDRV_PCM_INFO_PAUSE |
-				    SNDRV_PCM_INFO_RESUME,
+				    SNDRV_PCM_INFO_MMAP_VALID,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
 				    SNDRV_PCM_FMTBIT_U16_LE |
 				    SNDRV_PCM_FMTBIT_U8 |
@@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd)
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		prtd->state |= ST_RUNNING;
 		prtd->params->ops->trigger(prtd->params->ch);
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		prtd->state &= ~ST_RUNNING;
 		prtd->params->ops->stop(prtd->params->ch);
 		break;
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index b7b2a1f9142..89b064650f1 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -20,7 +20,7 @@
 #include <sound/pcm_params.h>
 
 #include <plat/audio.h>
-#include <plat/dma.h>
+#include <mach/dma.h>
 
 #include "dma.h"
 #include "pcm.h"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index f219b2f7ee6..c501af6d8db 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	}
 
 	if (!rtd->cpu_dai) {
-		dev_dbg(card->dev, "CPU DAI %s not registered\n",
+		dev_err(card->dev, "CPU DAI %s not registered\n",
 			dai_link->cpu_dai_name);
 		return -EPROBE_DEFER;
 	}
@@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 		}
 
 		if (!rtd->codec_dai) {
-			dev_dbg(card->dev, "CODEC DAI %s not registered\n",
+			dev_err(card->dev, "CODEC DAI %s not registered\n",
 				dai_link->codec_dai_name);
 			return -EPROBE_DEFER;
 		}
 	}
 
 	if (!rtd->codec) {
-		dev_dbg(card->dev, "CODEC %s not registered\n",
+		dev_err(card->dev, "CODEC %s not registered\n",
 			dai_link->codec_name);
 		return -EPROBE_DEFER;
 	}
@@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 		rtd->platform = platform;
 	}
 	if (!rtd->platform) {
-		dev_dbg(card->dev, "platform %s not registered\n",
+		dev_err(card->dev, "platform %s not registered\n",
 			dai_link->platform_name);
 		return -EPROBE_DEFER;
 	}
@@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card,
 	}
 
 	/* If the driver didn't set I/O up try regmap */
-	if (!codec->control_data)
+	if (!codec->write && dev_get_regmap(codec->dev, NULL))
 		snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
 
 	if (driver->controls)
@@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num)
 			return 0;
 	}
 
+	dev_err(card->dev, "%s not registered\n", aux_dev->codec_name);
+
 	return -EPROBE_DEFER;
 }
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dd7c49fafd7..f90139b5f50 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -291,8 +291,11 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 		if (dapm->codec->driver->set_bias_level)
 			ret = dapm->codec->driver->set_bias_level(dapm->codec,
 								  level);
-	} else
+		else
+			dapm->bias_level = level;
+	} else if (!card || dapm != &card->dapm) {
 		dapm->bias_level = level;
+	}
 
 	if (ret != 0)
 		goto out;
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7f8b3b7428b..0c172938b82 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -103,7 +103,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 	}
 
 	/* Report before the DAPM sync to help users updating micbias status */
-	blocking_notifier_call_chain(&jack->notifier, status, jack);
+	blocking_notifier_call_chain(&jack->notifier, jack->status, jack);
 
 	snd_soc_dapm_sync(dapm);
 
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 97c2cac8e92..8c7f2372944 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm)
 			continue;
 
 		buf = &substream->dma_buffer;
-		if (!buf && !buf->area)
+		if (!buf || !buf->area)
 			continue;
 
 		dma_free_writecombine(pcm->card->dev, buf->bytes,
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index d684df294c0..76cb1b363b7 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = {
 	.name = "Headset detection",
 	.report = SND_JACK_HEADSET,
 	.debounce_time = 150,
-	.invert = 1,
 };
 
 static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = {
@@ -177,7 +176,7 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 	}
 
 	alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
-	if (alc5632->gpio_hp_det == -ENODEV)
+	if (alc5632->gpio_hp_det == -EPROBE_DEFER)
 		return -EPROBE_DEFER;
 
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 5658bcec193..8d6900c1ee4 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
 		slave_config.dst_addr = dmap->addr;
-		slave_config.src_maxburst = 0;
+		slave_config.dst_maxburst = 4;
 	} else {
 		slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
 		slave_config.src_addr = dmap->addr;
-		slave_config.dst_maxburst = 0;
+		slave_config.src_maxburst = 4;
 	}
 	slave_config.slave_id = dmap->req_sel;
 
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0c5bb33d258..d4f14e49234 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -284,27 +284,27 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 	} else if (np) {
 		pdata->gpio_spkr_en = of_get_named_gpio(np,
 						"nvidia,spkr-en-gpios", 0);
-		if (pdata->gpio_spkr_en == -ENODEV)
+		if (pdata->gpio_spkr_en == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_hp_mute = of_get_named_gpio(np,
 						"nvidia,hp-mute-gpios", 0);
-		if (pdata->gpio_hp_mute == -ENODEV)
+		if (pdata->gpio_hp_mute == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_hp_det = of_get_named_gpio(np,
 						"nvidia,hp-det-gpios", 0);
-		if (pdata->gpio_hp_det == -ENODEV)
+		if (pdata->gpio_hp_det == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_int_mic_en = of_get_named_gpio(np,
 						"nvidia,int-mic-en-gpios", 0);
-		if (pdata->gpio_int_mic_en == -ENODEV)
+		if (pdata->gpio_int_mic_en == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 
 		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
 						"nvidia,ext-mic-en-gpios", 0);
-		if (pdata->gpio_ext_mic_en == -ENODEV)
+		if (pdata->gpio_ext_mic_en == -EPROBE_DEFER)
 			return -EPROBE_DEFER;
 	}
 
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 62ac0285bfa..057e28ef770 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -21,7 +21,7 @@
 #include <linux/mfd/dbx500-prcmu.h>
 
 #include <mach/hardware.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
 
 #include <sound/soc.h>
 #include <sound/soc-dai.h>
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index ee14d2dac2f..eb85113d472 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -19,7 +19,7 @@
 #include <linux/slab.h>
 
 #include <mach/hardware.h>
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
 
 #include <sound/soc.h>
 
@@ -663,7 +663,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
 			struct ux500_msp **msp_p,
 			struct msp_i2s_platform_data *platform_data)
 {
-	int ret = 0;
 	struct resource *res = NULL;
 	struct i2s_controller *i2s_cont;
 	struct ux500_msp *msp;
@@ -685,15 +684,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
 	if (res == NULL) {
 		dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n",
 			__func__);
-		ret = -ENOMEM;
-		goto err_res;
+		return -ENOMEM;
 	}
 
-	msp->registers = ioremap(res->start, (res->end - res->start + 1));
+	msp->registers = devm_ioremap(&pdev->dev, res->start,
+				      resource_size(res));
 	if (msp->registers == NULL) {
 		dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__);
-		ret = -ENOMEM;
-		goto err_res;
+		return -ENOMEM;
 	}
 
 	msp->msp_state = MSP_STATE_IDLE;
@@ -705,7 +703,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
 		dev_err(&pdev->dev,
 			"%s: ERROR: Failed to allocate I2S-controller!\n",
 			__func__);
-		goto err_i2s_cont;
+		return -ENOMEM;
 	}
 	i2s_cont->dev.parent = &pdev->dev;
 	i2s_cont->data = (void *)msp;
@@ -716,14 +714,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev,
 	msp->i2s_cont = i2s_cont;
 
 	return 0;
-
-err_i2s_cont:
-	iounmap(msp->registers);
-
-err_res:
-	devm_kfree(&pdev->dev, msp);
-
-	return ret;
 }
 
 void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
@@ -732,11 +722,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev,
 	dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id);
 
 	device_unregister(&msp->i2s_cont->dev);
-	devm_kfree(&pdev->dev, msp->i2s_cont);
-
-	iounmap(msp->registers);
-
-	devm_kfree(&pdev->dev, msp);
 }
 
 MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h
index 7f71b4a0d4b..2d9136da986 100644
--- a/sound/soc/ux500/ux500_msp_i2s.h
+++ b/sound/soc/ux500/ux500_msp_i2s.h
@@ -17,7 +17,7 @@
 
 #include <linux/platform_device.h>
 
-#include <mach/board-mop500-msp.h>
+#include <mach/msp.h>
 
 #define MSP_INPUT_FREQ_APB 48000000
 
diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c
index 7e96249536b..37711a5d0d6 100644
--- a/sound/sound_firmware.c
+++ b/sound/sound_firmware.c
@@ -23,14 +23,14 @@ static int do_mod_firmware_load(const char *fn, char **fp)
 	if (l <= 0 || l > 131072)
 	{
 		printk(KERN_INFO "Invalid firmware '%s'\n", fn);
-		filp_close(filp, current->files);
+		filp_close(filp, NULL);
 		return 0;
 	}
 	dp = vmalloc(l);
 	if (dp == NULL)
 	{
 		printk(KERN_INFO "Out of memory loading '%s'.\n", fn);
-		filp_close(filp, current->files);
+		filp_close(filp, NULL);
 		return 0;
 	}
 	pos = 0;
@@ -38,10 +38,10 @@ static int do_mod_firmware_load(const char *fn, char **fp)
 	{
 		printk(KERN_INFO "Failed to read '%s'.\n", fn);
 		vfree(dp);
-		filp_close(filp, current->files);
+		filp_close(filp, NULL);
 		return 0;
 	}
-	filp_close(filp, current->files);
+	filp_close(filp, NULL);
 	*fp = dp;
 	return (int) l;
 }
diff --git a/sound/usb/card.c b/sound/usb/card.c
index d5b5c3388e2..4a469f0cb6d 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
 				     struct snd_usb_audio *chip)
 {
 	struct snd_card *card;
-	struct list_head *p;
+	struct list_head *p, *n;
 
 	if (chip == (void *)-1L)
 		return;
@@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev,
 			snd_usb_stream_disconnect(p);
 		}
 		/* release the endpoint resources */
-		list_for_each(p, &chip->ep_list) {
+		list_for_each_safe(p, n, &chip->ep_list) {
 			snd_usb_endpoint_free(p);
 		}
 		/* release the midi resources */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 379baad3d5a..5e634a2eb28 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -111,7 +111,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
 		return 0;
 
 	/* If a clock source can't tell us whether it's valid, we assume it is */
-	if (!uac2_control_is_readable(cs_desc->bmControls, UAC2_CS_CONTROL_CLOCK_VALID))
+	if (!uac2_control_is_readable(cs_desc->bmControls,
+				      UAC2_CS_CONTROL_CLOCK_VALID - 1))
 		return 1;
 
 	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 0f647d22cb4..060dccb9ec7 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -141,7 +141,7 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep)
  *
  * For implicit feedback, next_packet_size() is unused.
  */
-static int next_packet_size(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
 {
 	unsigned long flags;
 	int ret;
@@ -177,15 +177,6 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep,
 		ep->retire_data_urb(ep->data_subs, urb);
 }
 
-static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep,
-				       struct snd_urb_ctx *ctx)
-{
-	int i;
-
-	for (i = 0; i < ctx->packets; ++i)
-		ctx->packet_size[i] = next_packet_size(ep);
-}
-
 /*
  * Prepare a PLAYBACK urb for submission to the bus.
  */
@@ -206,7 +197,13 @@ static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
 			/* no data provider, so send silence */
 			unsigned int offs = 0;
 			for (i = 0; i < ctx->packets; ++i) {
-				int counts = ctx->packet_size[i];
+				int counts;
+
+				if (ctx->packet_size[i])
+					counts = ctx->packet_size[i];
+				else
+					counts = snd_usb_endpoint_next_packet_size(ep);
+
 				urb->iso_frame_desc[i].offset = offs * ep->stride;
 				urb->iso_frame_desc[i].length = counts * ep->stride;
 				offs += counts;
@@ -370,7 +367,6 @@ static void snd_complete_urb(struct urb *urb)
 			goto exit_clear;
 		}
 
-		prepare_outbound_urb_sizes(ep, ctx);
 		prepare_outbound_urb(ep, ctx);
 	} else {
 		retire_inbound_urb(ep, ctx);
@@ -799,7 +795,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 /**
  * snd_usb_endpoint_start: start an snd_usb_endpoint
  *
- * @ep: the endpoint to start
+ * @ep:		the endpoint to start
+ * @can_sleep:	flag indicating whether the operation is executed in
+ * 		non-atomic context
  *
  * A call to this function will increment the use count of the endpoint.
  * In case it is not already running, the URBs for this endpoint will be
@@ -809,7 +807,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
  *
  * Returns an error if the URB submission failed, 0 in all other cases.
  */
-int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep)
 {
 	int err;
 	unsigned int i;
@@ -822,8 +820,9 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 		return 0;
 
 	/* just to be sure */
-	deactivate_urbs(ep, 0, 1);
-	wait_clear_urbs(ep);
+	deactivate_urbs(ep, 0, can_sleep);
+	if (can_sleep)
+		wait_clear_urbs(ep);
 
 	ep->active_mask = 0;
 	ep->unlink_mask = 0;
@@ -854,7 +853,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 			goto __error;
 
 		if (usb_pipeout(ep->pipe)) {
-			prepare_outbound_urb_sizes(ep, urb->context);
 			prepare_outbound_urb(ep, urb->context);
 		} else {
 			prepare_inbound_urb(ep, urb->context);
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index ee2723fb174..cbbbdf226d6 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -13,7 +13,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 				struct audioformat *fmt,
 				struct snd_usb_endpoint *sync_ep);
 
-int  snd_usb_endpoint_start(struct snd_usb_endpoint *ep);
+int  snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep);
 void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep,
 			   int force, int can_sleep, int wait);
 int  snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
@@ -21,6 +21,7 @@ int  snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_free(struct list_head *head);
 
 int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep);
 
 void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
 			     struct snd_usb_endpoint *sender,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a1298f37942..f782ce19bf5 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -212,7 +212,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
 	}
 }
 
-static int start_endpoints(struct snd_usb_substream *subs)
+static int start_endpoints(struct snd_usb_substream *subs, int can_sleep)
 {
 	int err;
 
@@ -225,7 +225,7 @@ static int start_endpoints(struct snd_usb_substream *subs)
 		snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep);
 
 		ep->data_subs = subs;
-		err = snd_usb_endpoint_start(ep);
+		err = snd_usb_endpoint_start(ep, can_sleep);
 		if (err < 0) {
 			clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
 			return err;
@@ -236,10 +236,25 @@ static int start_endpoints(struct snd_usb_substream *subs)
 	    !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
 		struct snd_usb_endpoint *ep = subs->sync_endpoint;
 
+		if (subs->data_endpoint->iface != subs->sync_endpoint->iface ||
+		    subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) {
+			err = usb_set_interface(subs->dev,
+						subs->sync_endpoint->iface,
+						subs->sync_endpoint->alt_idx);
+			if (err < 0) {
+				snd_printk(KERN_ERR
+					   "%d:%d:%d: cannot set interface (%d)\n",
+					   subs->dev->devnum,
+					   subs->sync_endpoint->iface,
+					   subs->sync_endpoint->alt_idx, err);
+				return -EIO;
+			}
+		}
+
 		snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep);
 
 		ep->sync_slave = subs->data_endpoint;
-		err = snd_usb_endpoint_start(ep);
+		err = snd_usb_endpoint_start(ep, can_sleep);
 		if (err < 0) {
 			clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
 			return err;
@@ -547,7 +562,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
 	/* for playback, submit the URBs now; otherwise, the first hwptr_done
 	 * updates for all URBs would happen at the same time when starting */
 	if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
-		return start_endpoints(subs);
+		return start_endpoints(subs, 1);
 
 	return 0;
 }
@@ -1029,6 +1044,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 				 struct urb *urb)
 {
 	struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
+	struct snd_usb_endpoint *ep = subs->data_endpoint;
 	struct snd_urb_ctx *ctx = urb->context;
 	unsigned int counts, frames, bytes;
 	int i, stride, period_elapsed = 0;
@@ -1040,7 +1056,11 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 	urb->number_of_packets = 0;
 	spin_lock_irqsave(&subs->lock, flags);
 	for (i = 0; i < ctx->packets; i++) {
-		counts = ctx->packet_size[i];
+		if (ctx->packet_size[i])
+			counts = ctx->packet_size[i];
+		else
+			counts = snd_usb_endpoint_next_packet_size(ep);
+
 		/* set up descriptor */
 		urb->iso_frame_desc[i].offset = frames * stride;
 		urb->iso_frame_desc[i].length = counts * stride;
@@ -1091,7 +1111,16 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 	subs->hwptr_done += bytes;
 	if (subs->hwptr_done >= runtime->buffer_size * stride)
 		subs->hwptr_done -= runtime->buffer_size * stride;
+
+	/* update delay with exact number of samples queued */
+	runtime->delay = subs->last_delay;
 	runtime->delay += frames;
+	subs->last_delay = runtime->delay;
+
+	/* realign last_frame_number */
+	subs->last_frame_number = usb_get_current_frame_number(subs->dev);
+	subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
+
 	spin_unlock_irqrestore(&subs->lock, flags);
 	urb->transfer_buffer_length = bytes;
 	if (period_elapsed)
@@ -1109,12 +1138,32 @@ static void retire_playback_urb(struct snd_usb_substream *subs,
 	struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
 	int stride = runtime->frame_bits >> 3;
 	int processed = urb->transfer_buffer_length / stride;
+	int est_delay;
+
+	/* ignore the delay accounting when procssed=0 is given, i.e.
+	 * silent payloads are procssed before handling the actual data
+	 */
+	if (!processed)
+		return;
 
 	spin_lock_irqsave(&subs->lock, flags);
-	if (processed > runtime->delay)
-		runtime->delay = 0;
+	est_delay = snd_usb_pcm_delay(subs, runtime->rate);
+	/* update delay with exact number of samples played */
+	if (processed > subs->last_delay)
+		subs->last_delay = 0;
 	else
-		runtime->delay -= processed;
+		subs->last_delay -= processed;
+	runtime->delay = subs->last_delay;
+
+	/*
+	 * Report when delay estimate is off by more than 2ms.
+	 * The error should be lower than 2ms since the estimate relies
+	 * on two reads of a counter updated every ms.
+	 */
+	if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
+		snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n",
+			est_delay, subs->last_delay);
+
 	spin_unlock_irqrestore(&subs->lock, flags);
 }
 
@@ -1172,7 +1221,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		err = start_endpoints(subs);
+		err = start_endpoints(subs, 0);
 		if (err < 0)
 			return err;
author	Mauro Carvalho Chehab <mchehab@redhat.com>	2012-10-05 09:36:26 -0300
committer	Mauro Carvalho Chehab <mchehab@redhat.com>	2012-10-05 09:36:26 -0300
commit	bd0d10498826ed150da5e4c45baf8b9c7088fb71 (patch)
tree	cdee4371121a355d627a655c4eef5c0047b0462a /sound
parent	89de0f2cda8b784e51ebd6655fff7339e4ac552b (diff)
parent	2425bb3d4016ed95ce83a90b53bd92c7f31091e4 (diff)