Merge branch 'linus' into sched/core

Conflicts:
	fs/proc/array.c

Merge reason: resolve conflict and queue up dependent patch.

Signed-off-by: Ingo Molnar <mingo@elte.hu>

23 files changed, 167 insertions, 184 deletions

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ac927ffdc96..97f1a251e44 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,15 +7,6 @@ config SND_BF5XX_I2S
 	  mode (supports single stereo In/Out).
 	  You will also need to select the audio interfaces to support below.
 
-config SND_BF5XX_TDM
-	tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
-	depends on (BLACKFIN && SND_SOC)
-	help
-	  Say Y or M if you want to add support for codecs attached to
-	  the Blackfin SPORT (synchronous serial ports) interface in TDM
-	  mode.
-	  You will also need to select the audio interfaces to support below.
-
 config SND_BF5XX_SOC_SSM2602
 	tristate "SoC SSM2602 Audio support for BF52x ezkit"
 	depends on SND_BF5XX_I2S
@@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
 	  Enter the GPIO used to control AD73311's SE pin. Acceptable
 	  values are 0 to 7
 
+config SND_BF5XX_TDM
+	tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+	depends on (BLACKFIN && SND_SOC)
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the Blackfin SPORT (synchronous serial ports) interface in TDM
+	  mode.
+	  You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_AD1836
+	tristate "SoC AD1836 Audio support for BF5xx"
+	depends on SND_BF5XX_TDM
+	select SND_BF5XX_SOC_TDM
+	select SND_SOC_AD1836
+	help
+	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SOC_AD1938
+	tristate "SoC AD1938 Audio support for Blackfin"
+	depends on SND_BF5XX_TDM
+	select SND_BF5XX_SOC_TDM
+	select SND_SOC_AD1938
+	help
+	  Say Y if you want to add support for AD1938 codec on Blackfin.
+
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
 	depends on BLACKFIN
@@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
 	  Say y if you want AC97 driver to support up to 5.1 channel audio.
 	  this mode will consume much more memory for DMA.
 
+config SND_BF5XX_HAVE_COLD_RESET
+	bool "BOARD has COLD Reset GPIO"
+	depends on SND_BF5XX_AC97
+	default y if BFIN548_EZKIT
+	default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+	int "Set a GPIO for cold reset"
+	depends on SND_BF5XX_HAVE_COLD_RESET
+	range 0 159
+	default 19 if BFIN548_EZKIT
+	default 5 if BFIN537_STAMP
+	default 0
+	help
+	  Set the correct GPIO for RESET the sound chip.
+
+config SND_BF5XX_SOC_AD1980
+	tristate "SoC AD1980/1 Audio support for BF5xx"
+	depends on SND_BF5XX_AC97
+	select SND_BF5XX_SOC_AC97
+	select SND_SOC_AD1980
+	help
+	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
 config SND_BF5XX_SOC_SPORT
 	tristate
 
@@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
 	select SND_SOC_AC97_BUS
 	select SND_BF5XX_SOC_SPORT
 
-config SND_BF5XX_SOC_AD1836
-	tristate "SoC AD1836 Audio support for BF5xx"
-	depends on SND_BF5XX_TDM
-	select SND_BF5XX_SOC_TDM
-	select SND_SOC_AD1836
-	help
-	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1980
-	tristate "SoC AD1980/1 Audio support for BF5xx"
-	depends on SND_BF5XX_AC97
-	select SND_BF5XX_SOC_AC97
-	select SND_SOC_AD1980
-	help
-	  Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1938
-        tristate "SoC AD1938 Audio support for Blackfin"
-        depends on SND_BF5XX_TDM
-        select SND_BF5XX_SOC_TDM
-        select SND_SOC_AD1938
-        help
-          Say Y if you want to add support for AD1938 codec on Blackfin.
-
 config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
 	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
@@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
 	default 0
 	help
 	  Set the correct SPORT for sound chip.
-
-config SND_BF5XX_HAVE_COLD_RESET
-	bool "BOARD has COLD Reset GPIO"
-	depends on SND_BF5XX_AC97
-	default y if BFIN548_EZKIT
-	default n if !BFIN548_EZKIT
-
-config SND_BF5XX_RESET_GPIO_NUM
-	int "Set a GPIO for cold reset"
-	depends on SND_BF5XX_HAVE_COLD_RESET
-	range 0 159
-	default 19 if BFIN548_EZKIT
-	default 5 if BFIN537_STAMP
-	default 0
-	help
-	  Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 2758b9017a7..e6932297873 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -277,7 +277,11 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 	if (!dai->active)
 		return 0;
 
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	ret = sport_set_multichannel(sport, 16, 0x3FF, 1);
+#else
 	ret = sport_set_multichannel(sport, 16, 0x1F, 1);
+#endif
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		return -EBUSY;
@@ -334,7 +338,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
 		goto sport_err;
 	}
 	/*SPORT works in TDM mode to simulate AC97 transfers*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1);
+#else
 	ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
+#endif
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		ret = -EBUSY;
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f2a911fe0c..a1f97dd809d 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -1,5 +1,5 @@
 /*
- * linux/sound/arm/bf5xx-ac97.h
+ * sound/soc/blackfin/bf5xx-ac97.h
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 876abade27e..084b68884ad 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
  * TFS. When Port G is selected and EMAC then there is a conflict between
  * the PHY interrupt line and TFS.  Current settings prevent the conflict
  * by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
  */
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
 #define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
 #endif
 
 static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
@@ -227,7 +227,8 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct platform_device *pdev,
+			struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	peripheral_free_list(&sport_req[sport_num][0]);
@@ -236,36 +237,31 @@ static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
 #ifdef CONFIG_PM
 static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
 {
-	struct sport_device *sport =
-		(struct sport_device *)dai->private_data;
 
 	pr_debug("%s : sport %d\n", __func__, dai->id);
-	if (!dai->active)
-		return 0;
+
 	if (dai->capture.active)
-		sport_rx_stop(sport);
+		sport_rx_stop(sport_handle);
 	if (dai->playback.active)
-		sport_tx_stop(sport);
+		sport_tx_stop(sport_handle);
 	return 0;
 }
 
 static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	int ret;
-	struct sport_device *sport =
-		(struct sport_device *)dai->private_data;
 
 	pr_debug("%s : sport %d\n", __func__, dai->id);
-	if (!dai->active)
-		return 0;
 
-	ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0);
+	ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1,
+				      bf5xx_i2s.rcr2, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		return -EBUSY;
 	}
 
-	ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0);
+	ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1,
+				      bf5xx_i2s.tcr2, 0, 0);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
 		return -EBUSY;
diff --git a/sound/soc/blackfin/bf5xx-i2s.h b/sound/soc/blackfin/bf5xx-i2s.h
index 7107d1a0b06..264ecdcba35 100644
--- a/sound/soc/blackfin/bf5xx-i2s.h
+++ b/sound/soc/blackfin/bf5xx-i2s.h
@@ -1,5 +1,5 @@
 /*
- * linux/sound/arm/bf5xx-i2s.h
+ * sound/soc/blackfin/bf5xx-i2s.h
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index 469ce7fab20..99051ff0954 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -326,7 +326,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
 
 int sport_tx_start(struct sport_device *sport)
 {
-	unsigned flags;
+	unsigned long flags;
 	pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__,
 			sport->tx_run, sport->rx_run);
 	if (sport->tx_run)
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 3096badf09a..ff546e91a22 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
  * TFS. When Port G is selected and EMAC then there is a conflict between
  * the PHY interrupt line and TFS.  Current settings prevent the conflict
  * by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
  */
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
 #define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
 #endif
 
 static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 01343dc984f..c48485f2c55 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -251,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi)
 
 static struct spi_driver ad1836_spi_driver = {
 	.driver = {
-		.name	= "ad1836-spi",
-		.bus	= &spi_bus_type,
+		.name	= "ad1836",
 		.owner	= THIS_MODULE,
 	},
 	.probe		= ad1836_spi_probe,
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index 9a049a1995a..34b30efc3cb 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -456,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi)
 static struct spi_driver ad1938_spi_driver = {
 	.driver = {
 		.name	= "ad1938",
-		.bus	= &spi_bus_type,
 		.owner	= THIS_MODULE,
 	},
 	.probe		= ad1938_spi_probe,
@@ -515,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938)
 	codec->num_dai = 1;
 	codec->write = ad1938_write_reg;
 	codec->read = ad1938_read_reg_cache;
+	codec->set_bias_level = ad1938_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373dff8..593d5b9c9f0 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
 	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
 			    WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
 	SOC_DAPM_SINGLE("PGA Capture Switch",
-			WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+			WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
 };
 
 /* Right Input Mixer */
@@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
 	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
 			    WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
 	SOC_DAPM_SINGLE("PGA Capture Switch",
-			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
 };
 
 /* Left Mic Mixer */
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d80d414cfbb..5ad677ce80d 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* Mono Capture mixer-mux */
 	{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
+	{"Capture Left Mixer", "Stereo", "Capture Left Mux"},
 	{"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
 	{"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
 	{"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index da97aae475a..1ef2454c520 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
 	codec->reg_cache = &wm8940->reg_cache;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
-	if (ret == 0) {
+	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
 	}
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index c64e55aa63b..686e5aa9720 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1027,7 +1027,7 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
 		       - wm9081->fs);
 	for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
 		cur_val = abs((wm9081->sysclk_rate /
-			       clk_sys_rates[i].ratio) - wm9081->fs);;
+			       clk_sys_rates[i].ratio) - wm9081->fs);
 		if (cur_val < best_val) {
 			best = i;
 			best_val = cur_val;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 12a6c549ee6..4ae70704802 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,22 +97,19 @@ enum {
 	DAVINCI_MCBSP_WORD_32,
 };
 
-static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
-	.name = "I2S PCM Stereo out",
-};
-
-static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
-	.name = "I2S PCM Stereo in",
-};
-
 struct davinci_mcbsp_dev {
+	/*
+	 * dma_params must be first because rtd->dai->cpu_dai->private_data
+	 * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+	 * davinci_pcm_open.
+	 */
+	struct davinci_pcm_dma_params	dma_params[2];
 	void __iomem			*base;
 #define MOD_DSP_A	0
 #define MOD_DSP_B	1
 	int				mode;
 	u32				pcr;
 	struct clk			*clk;
-	struct davinci_pcm_dma_params	*dma_params[2];
 };
 
 static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
 	toggle_clock(dev, playback);
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *cpu_dai)
-{
-	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
-	cpu_dai->dma_data = dev->dma_params[substream->stream];
-	return 0;
-}
-
 #define DEFAULT_BITPERSAMPLE	16
 
 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct davinci_pcm_dma_params *dma_params = dai->dma_data;
 	struct davinci_mcbsp_dev *dev = dai->private_data;
+	struct davinci_pcm_dma_params *dma_params =
+					&dev->dma_params[substream->stream];
 	struct snd_interval *i = NULL;
 	int mcbsp_word_length;
 	unsigned int rcr, xcr, srgr;
@@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
 static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
-	.startup 	= davinci_i2s_startup,
 	.shutdown	= davinci_i2s_shutdown,
 	.prepare	= davinci_i2s_prepare,
 	.trigger	= davinci_i2s_trigger,
@@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 
 	dev->base = (void __iomem *)IO_ADDRESS(mem->start);
 
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
 	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
 
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
 	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
 
 	/* first TX, then RX */
@@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 		ret = -ENXIO;
 		goto err_free_mem;
 	}
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
@@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 		ret = -ENXIO;
 		goto err_free_mem;
 	}
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 
 	davinci_i2s_dai.private_data = dev;
 	ret = snd_soc_register_dai(&davinci_i2s_dai);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index eca22d7829d..5d1f98a4c97 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
 		printk(KERN_ERR "GBLCTL write error\n");
 }
 
-static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
-						struct snd_soc_dai *cpu_dai)
-{
-	struct davinci_audio_dev *dev = cpu_dai->private_data;
-	cpu_dai->dma_data = dev->dma_params[substream->stream];
-	return 0;
-}
-
 static void mcasp_start_rx(struct davinci_audio_dev *dev)
 {
 	mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
@@ -386,17 +378,17 @@ 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 (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (dev->txnumevt)	/* enable FIFO */
+			mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_start_tx(dev);
-	else
+	} else {
+		if (dev->rxnumevt)	/* enable FIFO */
+			mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_start_rx(dev);
-
-	/* enable FIFO */
-	if (dev->txnumevt)
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
-	if (dev->rxnumevt)
-		mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+	}
 }
 
 static void mcasp_stop_rx(struct davinci_audio_dev *dev)
@@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
 
 static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
 {
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (dev->txnumevt)	/* disable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_stop_tx(dev);
-	else
+	} else {
+		if (dev->rxnumevt)	/* disable FIFO */
+			mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+								FIFO_ENABLE);
 		mcasp_stop_rx(dev);
-
-	/* disable FIFO */
-	if (dev->txnumevt)
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
-	if (dev->rxnumevt)
-		mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+	}
 }
 
 static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -512,34 +504,49 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 				       int channel_size)
 {
 	u32 fmt = 0;
+	u32 mask, rotate;
 
 	switch (channel_size) {
 	case DAVINCI_AUDIO_WORD_8:
 		fmt = 0x03;
+		rotate = 6;
+		mask = 0x000000ff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_12:
 		fmt = 0x05;
+		rotate = 5;
+		mask = 0x00000fff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_16:
 		fmt = 0x07;
+		rotate = 4;
+		mask = 0x0000ffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_20:
 		fmt = 0x09;
+		rotate = 3;
+		mask = 0x000fffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_24:
 		fmt = 0x0B;
+		rotate = 2;
+		mask = 0x00ffffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_28:
 		fmt = 0x0D;
+		rotate = 1;
+		mask = 0x0fffffff;
 		break;
 
 	case DAVINCI_AUDIO_WORD_32:
 		fmt = 0x0F;
+		rotate = 0;
+		mask = 0xffffffff;
 		break;
 
 	default:
@@ -550,6 +557,13 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 					RXSSZ(fmt), RXSSZ(0x0F));
 	mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
 					TXSSZ(fmt), TXSSZ(0x0F));
+	mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXROT(rotate),
+							TXROT(7));
+	mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXROT(rotate),
+							RXROT(7));
+	mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, mask);
+	mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask);
+
 	return 0;
 }
 
@@ -638,7 +652,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
 			printk(KERN_ERR "playback tdm slot %d not supported\n",
 				dev->tdm_slots);
 
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
 	} else {
 		/* bit stream is MSB first with no delay */
@@ -655,7 +668,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
 			printk(KERN_ERR "capture tdm slot %d not supported\n",
 				dev->tdm_slots);
 
-		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF);
 		mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
 	}
 }
@@ -700,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 {
 	struct davinci_audio_dev *dev = cpu_dai->private_data;
 	struct davinci_pcm_dma_params *dma_params =
-					dev->dma_params[substream->stream];
+					&dev->dma_params[substream->stream];
 	int word_length;
 	u8 numevt;
 
@@ -778,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 }
 
 static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
-	.startup 	= davinci_mcasp_startup,
 	.trigger	= davinci_mcasp_trigger,
 	.hw_params	= davinci_mcasp_hw_params,
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
@@ -829,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	struct resource *mem, *ioarea, *res;
 	struct snd_platform_data *pdata;
 	struct davinci_audio_dev *dev;
-	int count = 0;
 	int ret = 0;
 
 	dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
 	if (!dev)
 		return	-ENOMEM;
 
-	dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
-								GFP_KERNEL);
-	if (!dma_data) {
-		ret = -ENOMEM;
-		goto err_release_dev;
-	}
-
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!mem) {
 		dev_err(&pdev->dev, "no mem resource?\n");
@@ -877,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dev->txnumevt = pdata->txnumevt;
 	dev->rxnumevt = pdata->rxnumevt;
 
-	dma_data[count].name = "I2S PCM Stereo out";
-	dma_data[count].eventq_no = pdata->eventq_no;
-	dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+	dma_data->eventq_no = pdata->eventq_no;
+	dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
 							io_v2p(dev->base));
-	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
 
 	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -890,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		goto err_release_region;
 	}
 
-	dma_data[count].channel = res->start;
-	count++;
-	dma_data[count].name = "I2S PCM Stereo in";
-	dma_data[count].eventq_no = pdata->eventq_no;
-	dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+	dma_data->channel = res->start;
+
+	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+	dma_data->eventq_no = pdata->eventq_no;
+	dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
 							io_v2p(dev->base));
-	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
@@ -904,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 		goto err_release_region;
 	}
 
-	dma_data[count].channel = res->start;
+	dma_data->channel = res->start;
 	davinci_mcasp_dai[pdata->op_mode].private_data = dev;
 	davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
 	ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
@@ -916,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 err_release_region:
 	release_mem_region(mem->start, (mem->end - mem->start) + 1);
 err_release_data:
-	kfree(dma_data);
-err_release_dev:
 	kfree(dev);
 
 	return ret;
@@ -926,7 +925,6 @@ err_release_dev:
 static int davinci_mcasp_remove(struct platform_device *pdev)
 {
 	struct snd_platform_data *pdata = pdev->dev.platform_data;
-	struct davinci_pcm_dma_params *dma_data;
 	struct davinci_audio_dev *dev;
 	struct resource *mem;
 
@@ -939,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	release_mem_region(mem->start, (mem->end - mem->start) + 1);
 
-	dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
-	kfree(dma_data);
 	kfree(dev);
 
 	return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 554354c1cc2..9d179cc88f7 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,10 +39,15 @@ enum {
 };
 
 struct davinci_audio_dev {
+	/*
+	 * dma_params must be first because rtd->dai->cpu_dai->private_data
+	 * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+	 * davinci_pcm_open.
+	 */
+	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
 	int sample_rate;
 	struct clk *clk;
-	struct davinci_pcm_dma_params *dma_params[2];
 	unsigned int codec_fmt;
 
 	/* McASP specific data */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 091dacb78b4..c73a915f233 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 {
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
 	struct edmacc_param p_ram;
 	int ret;
 
-	if (!dma_data)
-		return -ENODEV;
-
-	prtd->params = dma_data;
-
 	/* Request master DMA channel */
 	ret = edma_alloc_channel(prtd->params->channel,
 				  davinci_pcm_dma_irq, substream,
@@ -145,7 +138,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	prtd->master_lch = ret;
 
 	/* Request parameter RAM reload slot */
-	ret = edma_alloc_slot(EDMA_SLOT_ANY);
+	ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY);
 	if (ret < 0) {
 		edma_free_channel(prtd->master_lch);
 		return ret;
@@ -162,8 +155,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	 * so davinci_pcm_enqueue_dma() takes less time in IRQ.
 	 */
 	edma_read_slot(prtd->slave_lch, &p_ram);
-	p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
-	p_ram.link_bcntrld = prtd->slave_lch << 5;
+	p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch));
+	p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5;
 	edma_write_slot(prtd->slave_lch, &p_ram);
 
 	return 0;
@@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct davinci_runtime_data *prtd;
 	int ret = 0;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
+	struct davinci_pcm_dma_params *params = &pa[substream->stream];
+	if (!params)
+		return -ENODEV;
 
 	snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
 	/* ensure that buffer size is a multiple of period size */
@@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 		return -ENOMEM;
 
 	spin_lock_init(&prtd->lock);
+	prtd->params = params;
 
 	runtime->private_data = prtd;
 
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 63d96253c73..8746606efc8 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -17,7 +17,6 @@
 
 
 struct davinci_pcm_dma_params {
-	char *name;			/* stream identifier */
 	int channel;			/* sync dma channel ID */
 	unsigned short acnt;
 	dma_addr_t dma_addr;		/* device physical address for DMA */
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2c0cd..ccdefe60e75 100644
--- a/sound/soc/imx/mxc-ssi.c
+++ b/sound/soc/imx/mxc-ssi.c
@@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		break;
 	}
 
-	/* sync */
-	if (!(fmt & SND_SOC_DAIFMT_ASYNC))
-		scr |= SSI_SCR_SYN;
-
-	/* tdm - only for stereo atm */
-	if (fmt & SND_SOC_DAIFMT_TDM)
-		scr |= SSI_SCR_NET;
-
 	if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
 		SSI1_STCR = stcr;
 		SSI1_SRCR = srcr;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4ea525..dcb3181bb34 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
 
 config SND_PXA2XX_SOC_IMOTE2
        tristate "SoC Audio support for IMote 2"
-       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
        select SND_PXA2XX_SOC_I2S
        select SND_SOC_WM8940
        help
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 5b9ed646478..d11a6d7e384 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -351,7 +351,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
 			do_div(tmp, freq_out);
 			val = tmp;
 
-			val = (val << 16) | 64;;
+			val = (val << 16) | 64;
 			ssp_write_reg(ssp, SSACDD, val);
 
 			ssacd |= (0x6 << 4);
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index 8e79a416db5..c215d32d632 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -67,7 +67,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
 #ifdef ENFORCE_RATES
-	struct snd_pcm_runtime *runtime = substream->runtime;;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 #endif
 
 	mutex_lock(&clk_lock);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f79711b9fa5..8de6f9dec4a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
 
 		/* connected jack or spk ? */
 		if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
-			widget->id == snd_soc_dapm_line)
+		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
 			return 1;
 	}
 
@@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 			return 1;
 
 		/* connected jack ? */
-		if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+		if (widget->id == snd_soc_dapm_mic ||
+		    (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
 			return 1;
 	}