Merge tag 'v3.6-rc7' into next

Linux 3.6-rc7

Requested by David Howells so he can merge his key susbsystem work into
my tree with requisite -linus changesets.

24 files changed, 90 insertions, 86 deletions

diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
index 318c5ba5360f..dfb744381c42 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/arizona.c b/sound/soc/codecs/arizona.c
index 5c9cacaf2d52..1cf7a32d1b21 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 8f726c063f42..115a40301810 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -659,7 +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,
@@ -670,7 +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,
@@ -692,14 +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/wm5102.c b/sound/soc/codecs/wm5102.c
index 6537f16d383e..e33d327396ad 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 8033f7065189..01ebbcc5c6a4 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 0013afe48e66..dc4262eea4b7 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 aa9ce9dd7d8a..ce6720073798 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3733,21 +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 04ef03175c51..6c9eeca85b95 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -4038,6 +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 f16fb361a4eb..c6d2076a796b 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"},
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 95441bfc8190..ce5e5cd254dd 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 fb21b17f17f5..199408ec4261 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 28dd76c7cb1c..81d7728cf67f 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 99a997f19bb9..b6fa77678d97 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/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
index 009533ab8d18..df65f98211ec 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 34835e8a9160..d33c48baaf71 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/samsung/dma.c b/sound/soc/samsung/dma.c
index f3ebc38c10fe..b70964ea448c 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 b7b2a1f91425..89b064650f14 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 f81c5976b961..c501af6d8dbe 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;
 	}
@@ -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 dd7c49fafd75..f90139b5f50d 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 7f8b3b7428bb..0c172938b82a 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 97c2cac8e92c..8c7f23729446 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 e463529b38bb..76cb1b363b71 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[] = {
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 5658bcec1931..8d6900c1ee47 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/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 5c472f335a64..eb85113d472a 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -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");