Merge branch 'linus' into timers/core

Pick up the latest fixes and refresh the branch.

Signed-off-by: Ingo Molnar <mingo@kernel.org>

20 files changed, 145 insertions, 94 deletions

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 8697cedccd21..1ead3c977a51 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 
 	dma_params = ssc_p->dma_params[dir];
 
-	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
 	ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error);
 
 	pr_debug("%s enabled SSC_SR=0x%08x\n",
@@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int atmel_ssc_trigger(struct snd_pcm_substream *substream,
+			     int cmd, struct snd_soc_dai *dai)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+		break;
+	default:
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+		break;
+	}
+
+	return 0;
+}
 
 #ifdef CONFIG_PM
 static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
@@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = {
 	.startup	= atmel_ssc_startup,
 	.shutdown	= atmel_ssc_shutdown,
 	.prepare	= atmel_ssc_prepare,
+	.trigger	= atmel_ssc_trigger,
 	.hw_params	= atmel_ssc_hw_params,
 	.set_fmt	= atmel_ssc_set_dai_fmt,
 	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c
index 992ae38d5a15..7d6a9055874b 100644
--- a/sound/soc/atmel/sam9x5_wm8731.c
+++ b/sound/soc/atmel/sam9x5_wm8731.c
@@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 		goto out;
 	}
 
+	snd_soc_card_set_drvdata(card, priv);
+
 	card->dev = &pdev->dev;
 	card->owner = THIS_MODULE;
 	card->dai_link = dai;
@@ -107,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev)
 	dai->stream_name = "WM8731 PCM";
 	dai->codec_dai_name = "wm8731-hifi";
 	dai->init = sam9x5_wm8731_init;
-	dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+	dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF
 		| SND_SOC_DAIFMT_CBM_CFM;
 
 	ret = snd_soc_of_parse_card_name(card, "atmel,model");
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index c3c7396a6181..0ab2dc296474 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
 
-SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
-	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
-	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
-SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
-	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
-SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
-	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
-	   ARIZONA_OUT5_OSR_SHIFT, 1, 0),
-SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
-	   ARIZONA_OUT6_OSR_SHIFT, 1, 0),
-
 SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
 		 ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
 		 0xbf, 0, digital_tlv),
 
-SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
-		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
-		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
-		       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
-		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
-		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
-		       0x34, 0x40, 0, ana_tlv),
-SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
-		       ARIZONA_OUTPUT_PATH_CONFIG_3R,
-		       ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
-
 SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
 	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
 SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
@@ -1037,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
 	{ "AEC Loopback", "HPOUT3L", "OUT3L" },
 	{ "AEC Loopback", "HPOUT3R", "OUT3R" },
 	{ "HPOUT3L", NULL, "OUT3L" },
-	{ "HPOUT3R", NULL, "OUT3L" },
+	{ "HPOUT3R", NULL, "OUT3R" },
 
 	{ "AEC Loopback", "SPKOUTL", "OUT4L" },
 	{ "SPKOUTLN", NULL, "OUT4L" },
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 456bb8c6d759..bc7472c968e3 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface |= 0x0001;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
-		iface |= 0x0003;
+		iface |= 0x0013;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x0013;
+		iface |= 0x0003;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3938fb1c203e..53bbfac6a83a 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
-		aif1 |= WM8904_AIF_LRCLK_INV;
+		aif1 |= 0x3 | WM8904_AIF_LRCLK_INV;
 	case SND_SOC_DAIFMT_DSP_A:
 		aif1 |= 0x3;
 		break;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 543c5c2631b6..0f17ed3e29f4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8962_CLOCKING_4,
 			    WM8962_SYSCLK_RATE_MASK, clocking4);
 
+	/* DSPCLK_DIV can be only generated correctly after enabling SYSCLK.
+	 * So we here provisionally enable it and then disable it afterward
+	 * if current bias_level hasn't reached SND_SOC_BIAS_ON.
+	 */
+	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, WM8962_CLOCKING2,
+				WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA);
+
 	dspclk = snd_soc_read(codec, WM8962_CLOCKING1);
+
+	if (codec->dapm.bias_level != SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, WM8962_CLOCKING2,
+				WM8962_SYSCLK_ENA_MASK, 0);
+
 	if (dspclk < 0) {
 		dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk);
 		return;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 253c88bb7a4c..4f05fb88bddf 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
 		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 		snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+		codec->cache_sync = 1;
 		break;
 	}
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 46ec0e9744d4..4fbcab63e61f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp)
 		return ret;
 
 	/* Wait for the RAM to start, should be near instantaneous */
-	count = 0;
-	do {
+	for (count = 0; count < 10; ++count) {
 		ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1,
 				  &val);
 		if (ret != 0)
 			return ret;
-	} while (!(val & ADSP2_RAM_RDY) && ++count < 10);
+
+		if (val & ADSP2_RAM_RDY)
+			break;
+
+		msleep(1);
+	}
 
 	if (!(val & ADSP2_RAM_RDY)) {
 		adsp_err(dsp, "Failed to start DSP RAM\n");
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 61e48852b9e8..3fd76bc391de 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 		break;
 	}
 
-	dapm->bias_level = level;
-
 	return 0;
 }
 
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index eb4373840bb6..3665f612819d 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op)
 		return -ENOMEM;
 
 	card->dev = &op->dev;
-	platform_set_drvdata(op, pdata);
 
 	pdata->card = card;
 
@@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op)
 	if (ret)
 		dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
 
+	platform_set_drvdata(op, pdata);
+
 	return ret;
 }
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index d34d91743e3f..3920a5e8125f 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -33,6 +33,10 @@
 	 SNDRV_PCM_FMTBIT_S24_LE | \
 	 SNDRV_PCM_FMTBIT_S32_LE)
 
+#define KIRKWOOD_SPDIF_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | \
+	 SNDRV_PCM_FMTBIT_S24_LE)
+
 static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
 {
@@ -244,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
 				   ctl);
 	}
 
-	if (dai->id == 0)
-		ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;	/* i2s */
-	else
-		ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;	/* spdif */
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		/* configure */
 		ctl = priv->ctl_play;
+		if (dai->id == 0)
+			ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN;	/* i2s */
+		else
+			ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN;	/* spdif */
+
 		value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK;
 		writel(value, priv->io + KIRKWOOD_PLAYCTL);
 
@@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = {
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
 				SNDRV_PCM_RATE_96000,
-		.formats = KIRKWOOD_I2S_FORMATS,
+		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
 				SNDRV_PCM_RATE_96000,
-		.formats = KIRKWOOD_I2S_FORMATS,
+		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
     },
@@ -469,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
 		.formats = KIRKWOOD_I2S_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
@@ -490,18 +494,18 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = {
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
-		.formats = KIRKWOOD_I2S_FORMATS,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
+		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.capture = {
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000_192000 |
-			 SNDRV_PCM_RATE_CONTINUOUS |
-			 SNDRV_PCM_RATE_KNOT,
-		.formats = KIRKWOOD_I2S_FORMATS,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.rate_min = 5512,
+		.rate_max = 192000,
+		.formats = KIRKWOOD_SPDIF_FORMATS,
 	},
 	.ops = &kirkwood_i2s_dai_ops,
     },
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6d216cb6c19b..3fde9e402710 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream)
 				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
 
 	n810_ext_control(&codec->dapm);
-	return clk_enable(sys_clkout2);
+	return clk_prepare_enable(sys_clkout2);
 }
 
 static void n810_shutdown(struct snd_pcm_substream *substream)
 {
-	clk_disable(sys_clkout2);
+	clk_disable_unprepare(sys_clkout2);
 }
 
 static int n810_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 14011d90d70a..ff60e11ecb56 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU
 config SND_SOC_RCAR
 	tristate "R-Car series SRU/SCU/SSIU/SSI support"
 	select SND_SIMPLE_CARD
+	select REGMAP
 	help
 	  This option enables R-Car SUR/SCU/SSIU/SSI sound support
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4e53d87e881d..a66783e13a9c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
 			break;
 		case 2:
 			((u16 *)(&ucontrol->value.bytes.data))[0]
-				&= ~params->mask;
+				&= cpu_to_be16(~params->mask);
 			break;
 		case 4:
 			((u32 *)(&ucontrol->value.bytes.data))[0]
-				&= ~params->mask;
+				&= cpu_to_be32(~params->mask);
 			break;
 		default:
 			return -EINVAL;
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index b1d732255c02..3449c1e909ae 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res)
  */
 int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 {
-	struct device **ptr;
+	struct snd_soc_card **ptr;
 	int ret;
 
 	ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL);
@@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card)
 
 	ret = snd_soc_register_card(card);
 	if (ret == 0) {
-		*ptr = dev;
+		*ptr = card;
 		devres_add(dev, ptr);
 	} else {
 		devres_free(ptr);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index cbc9c96ce1f4..41949af3baae 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 	}
 }
 
+static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
+{
+	unsigned int i;
+
+	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
+	     i++) {
+		if (!pcm->chan[i])
+			continue;
+		dma_release_channel(pcm->chan[i]);
+		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
+			break;
+	}
+}
+
 /**
  * snd_dmaengine_pcm_register - Register a dmaengine based PCM device
  * @dev: The parent device for the PCM device
@@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	const struct snd_dmaengine_pcm_config *config, unsigned int flags)
 {
 	struct dmaengine_pcm *pcm;
+	int ret;
 
 	pcm = kzalloc(sizeof(*pcm), GFP_KERNEL);
 	if (!pcm)
@@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev,
 	dmaengine_pcm_request_chan_of(pcm, dev);
 
 	if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE)
-		return snd_soc_add_platform(dev, &pcm->platform,
+		ret = snd_soc_add_platform(dev, &pcm->platform,
 				&dmaengine_no_residue_pcm_platform);
 	else
-		return snd_soc_add_platform(dev, &pcm->platform,
+		ret = snd_soc_add_platform(dev, &pcm->platform,
 				&dmaengine_pcm_platform);
+	if (ret)
+		goto err_free_dma;
+
+	return 0;
+
+err_free_dma:
+	dmaengine_pcm_release_chan(pcm);
+	kfree(pcm);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register);
 
@@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 {
 	struct snd_soc_platform *platform;
 	struct dmaengine_pcm *pcm;
-	unsigned int i;
 
 	platform = snd_soc_lookup_platform(dev);
 	if (!platform)
@@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
 
 	pcm = soc_platform_to_pcm(platform);
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
-		if (pcm->chan[i]) {
-			dma_release_channel(pcm->chan[i]);
-			if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
-				break;
-		}
-	}
-
 	snd_soc_remove_platform(platform);
+	dmaengine_pcm_release_chan(pcm);
 	kfree(pcm);
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 42782c01e413..891b9a9bcbf8 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream,
 	}
 }
 
-static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
+static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
 	struct snd_soc_pcm_stream *codec_stream,
 	struct snd_soc_pcm_stream *cpu_stream)
 {
-	hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min);
-	hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max);
+	struct snd_pcm_hardware *hw = &runtime->hw;
+
 	hw->channels_min = max(codec_stream->channels_min,
 		cpu_stream->channels_min);
 	hw->channels_max = min(codec_stream->channels_max,
@@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw,
 	if (cpu_stream->rates
 		& (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))
 		hw->rates |= codec_stream->rates;
+
+	snd_pcm_limit_hw_rates(runtime);
+
+	hw->rate_min = max(hw->rate_min, cpu_stream->rate_min);
+	hw->rate_min = max(hw->rate_min, codec_stream->rate_min);
+	hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max);
+	hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max);
 }
 
 /*
@@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 
 	/* Check that the codec and cpu DAIs are compatible */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback,
+		soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback,
 			&cpu_dai_drv->playback);
 	} else {
-		soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture,
+		soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture,
 			&cpu_dai_drv->capture);
 	}
 
 	ret = -EINVAL;
-	snd_pcm_limit_hw_rates(runtime);
 	if (!runtime->hw.rates) {
 		printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n",
 			codec_dai->name, cpu_dai->name);
@@ -594,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
 	/* apply codec digital mute */
-	if (!codec->active)
+	if ((playback && codec_dai->playback_active == 1) ||
+	    (!playback && codec_dai->capture_active == 1))
 		snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
 
 	/* free any machine hw params */
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 364bf6a907e1..8c819f811470 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+	mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
+		val |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 08bc6931c7c7..8c7c1028e579 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream,
 {
 	struct device *dev = dai->dev;
 	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 	int ret, spdifclock;
 
-	mask = TEGRA20_SPDIF_CTRL_PACK |
-	       TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+	mask |= TEGRA20_SPDIF_CTRL_PACK |
+		TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		val = TEGRA20_SPDIF_CTRL_PACK |
-		      TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+		val |= TEGRA20_SPDIF_CTRL_PACK |
+		       TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 231a785b3921..02247fee1cf7 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 				unsigned int fmt)
 {
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	unsigned int mask, val;
+	unsigned int mask = 0, val = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai,
 		return -EINVAL;
 	}
 
-	mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+	mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
+		val |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;