Merge branch 'for-2.6.32' into for-2.6.33

16 files changed, 174 insertions, 78 deletions

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0758a1b3ca44..475cb7ed6bec 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -205,7 +205,6 @@ struct snd_soc_jack_gpio;
 #endif
 
 typedef int (*hw_write_t)(void *,const char* ,int);
-typedef int (*hw_read_t)(void *,char* ,int);
 
 extern struct snd_ac97_bus_ops soc_ac97_ops;
 
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 479d7bdf1865..a521aa90ddee 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -1,8 +1,8 @@
 /*
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.com>
  *
  * 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
@@ -19,6 +19,7 @@
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/delay.h>
+#include <linux/mutex.h>
 #include <linux/suspend.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -29,6 +30,9 @@
 
 #include "psc.h"
 
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES	5
+
 #define AC97_DIR	\
 	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
 
@@ -45,6 +49,9 @@
 #define AC97PCR_CLRFIFO(stype)	\
 	((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
 
+#define AC97STAT_BUSY(stype)	\
+	((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+
 /* instance data. There can be only one, MacLeod!!!! */
 static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
 
@@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
-	unsigned short data, tmo;
+	unsigned short data, retry, tmo;
 
-	au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata));
+	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
 	au_sync();
 
-	tmo = 1000;
-	while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
-		udelay(2);
+	retry = AC97_RW_RETRIES;
+	do {
+		mutex_lock(&pscdata->lock);
+
+		au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg),
+			  AC97_CDC(pscdata));
+		au_sync();
+
+		tmo = 2000;
+		while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
+			&& --tmo)
+			udelay(2);
 
-	if (!tmo)
-		data = 0xffff;
-	else
 		data = au_readl(AC97_CDC(pscdata)) & 0xffff;
 
-	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
-	au_sync();
+		au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+		au_sync();
+
+		mutex_unlock(&pscdata->lock);
+	} while (--retry && !tmo);
 
-	return data;
+	return retry ? data : 0xffff;
 }
 
 /* AC97 controller writes to codec register */
@@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
-	unsigned int tmo;
+	unsigned int tmo, retry;
 
-	au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata));
+	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
 	au_sync();
-	tmo = 1000;
-	while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+
+	retry = AC97_RW_RETRIES;
+	do {
+		mutex_lock(&pscdata->lock);
+
+		au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff),
+			  AC97_CDC(pscdata));
 		au_sync();
 
-	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
-	au_sync();
+		tmo = 2000;
+		while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
+		       && --tmo)
+			udelay(2);
+
+		au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+		au_sync();
+
+		mutex_unlock(&pscdata->lock);
+	} while (--retry && !tmo);
 }
 
 /* AC97 controller asserts a warm reset */
@@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
 	au_sync();
 
 	/* wait for PSC to indicate it's ready */
-	i = 100000;
+	i = 1000;
 	while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
-		au_sync();
+		msleep(1);
 
 	if (i == 0) {
 		printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
@@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
 	au_sync();
 
 	/* wait for AC97 core to become ready */
-	i = 100000;
+	i = 1000;
 	while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
-		au_sync();
+		msleep(1);
 	if (i == 0)
 		printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
 }
@@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
-	unsigned long r, stat;
+	unsigned long r, ro, stat;
 	int chans, stype = SUBSTREAM_TYPE(substream);
 
 	chans = params_channels(params);
 
-	r = au_readl(AC97_CFG(pscdata));
+	r = ro = au_readl(AC97_CFG(pscdata));
 	stat = au_readl(AC97_STAT(pscdata));
 
 	/* already active? */
@@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 		    (pscdata->rate != params_rate(params)))
 			return -EINVAL;
 	} else {
-		/* disable AC97 device controller first */
-		au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
-		au_sync();
 
 		/* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
 		r &= ~PSC_AC97CFG_LEN_MASK;
@@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 			r |= PSC_AC97CFG_RXSLOT_ENA(4);
 		}
 
-		/* finally enable the AC97 controller again */
+		/* do we need to poke the hardware? */
+		if (!(r ^ ro))
+			goto out;
+
+		/* ac97 engine is about to be disabled */
+		mutex_lock(&pscdata->lock);
+
+		/* disable AC97 device controller first... */
+		au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+		au_sync();
+
+		/* ...wait for it... */
+		while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)
+			asm volatile ("nop");
+
+		/* ...write config... */
+		au_writel(r, AC97_CFG(pscdata));
+		au_sync();
+
+		/* ...enable the AC97 controller again... */
 		au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
 		au_sync();
 
+		/* ...and wait for ready bit */
+		while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR))
+			asm volatile ("nop");
+
+		mutex_unlock(&pscdata->lock);
+
 		pscdata->cfg = r;
 		pscdata->rate = params_rate(params);
 	}
 
+out:
 	return 0;
 }
 
@@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
+		au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+		au_sync();
 		au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
 		au_sync();
 		break;
@@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
 		au_sync();
+
+		while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype))
+			asm volatile ("nop");
+
+		au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+		au_sync();
+
 		break;
 	default:
 		ret = -EINVAL;
@@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev,
 	if (!au1xpsc_ac97_workdata)
 		return -ENOMEM;
 
+	mutex_init(&au1xpsc_ac97_workdata->lock);
+
 	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!r) {
 		ret = -ENODEV;
@@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev,
 		goto out1;
 
 	/* configuration: max dma trigger threshold, enable ac97 */
-	 au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
-				      PSC_AC97CFG_TT_FIFO8 |
-				      PSC_AC97CFG_DE_ENABLE;
+	au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
+				     PSC_AC97CFG_TT_FIFO8 |
+				     PSC_AC97CFG_DE_ENABLE;
 
 	/* preserve PSC clock source set up by platform (dev.platform_data
 	 * is already occupied by soc layer)
@@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index 8fdb1a04a07b..3f474e8ed4f6 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -29,6 +29,7 @@ struct au1xpsc_audio_data {
 
 	unsigned long pm[2];
 	struct resource *ioarea;
+	struct mutex lock;
 };
 
 #define PCM_TX	0
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 2758b9017a7f..e69322978739 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 3f2a911fe0cb..a1f97dd809d6 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 876abade27e1..1e9d161c76c4 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -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 7107d1a0b06b..264ecdcba35a 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 469ce7fab20c..99051ff0954e 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/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 3612bb92df90..01343dc984fd 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -18,7 +18,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index e62b27701a49..9a049a1995a3 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -28,7 +28,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 0ebd99b7493e..3f7e8a8b387e 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -612,7 +612,7 @@ SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
 
 /* Out4 Capture Mux */
 static const struct snd_kcontrol_new wm8350_out4_capture_controls =
-SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+SOC_DAPM_ENUM("Route", wm8350_enum[7]);
 
 static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
 
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index fa4d85bd048b..93d66e30f109 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -12,7 +12,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index eca22d7829d2..7a06c0a86665 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -512,34 +512,49 @@ 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 +565,13 @@ 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 +660,6 @@ 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 +676,6 @@ 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);
 	}
 }
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 9ff62e3a9b1d..6096d22283e6 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	int size, irq, rc;
 	const __be32 *prop;
 	void __iomem *regs;
+	int ret;
 
 	/* Fetch the registers and IRQ of the PSC */
 	irq = irq_of_parse_and_map(op->node, 0);
@@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	/* Allocate and initialize the driver private data */
 	psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
 	if (!psc_dma) {
-		iounmap(regs);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto out_unmap;
 	}
 
 	/* Get the PSC ID */
 	prop = of_get_property(op->node, "cell-index", &size);
-	if (!prop || size < sizeof *prop)
-		return -ENODEV;
+	if (!prop || size < sizeof *prop) {
+		ret = -ENODEV;
+		goto out_free;
+	}
 
 	spin_lock_init(&psc_dma->lock);
 	mutex_init(&psc_dma->mutex);
@@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	if (!psc_dma->capture.bcom_task ||
 	    !psc_dma->playback.bcom_task) {
 		dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
-		iounmap(regs);
-		kfree(psc_dma);
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out_free;
 	}
 
 	/* Disable all interrupts and reset the PSC */
@@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 			  &psc_dma_bcom_irq_tx, IRQF_SHARED,
 			  "psc-dma-playback", &psc_dma->playback);
 	if (rc) {
-		free_irq(psc_dma->irq, psc_dma);
-		free_irq(psc_dma->capture.irq,
-			 &psc_dma->capture);
-		free_irq(psc_dma->playback.irq,
-			 &psc_dma->playback);
-		return -ENODEV;
+		ret = -ENODEV;
+		goto out_irq;
 	}
 
 	/* Save what we've done so it can be found again later */
@@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op)
 
 	/* Tell the ASoC OF helpers about it */
 	return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+out_irq:
+	free_irq(psc_dma->irq, psc_dma);
+	free_irq(psc_dma->capture.irq, &psc_dma->capture);
+	free_irq(psc_dma->playback.irq, &psc_dma->playback);
+out_free:
+	kfree(psc_dma);
+out_unmap:
+	iounmap(regs);
+	return ret;
 }
 EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
 
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 819c3c086d69..11c45a37c631 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -230,6 +230,8 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
 
+#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
+
 /*
  * Wait for the LR signal to allow synchronisation to the L/R clock
  * from the codec. May only be needed for slave mode.
@@ -237,19 +239,21 @@ static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
 {
 	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+	unsigned long loops = msecs_to_loops(5);
 
 	pr_debug("Entered %s\n", __func__);
 
-	while (1) {
+	while (--loops) {
 		iiscon = readl(i2s->regs + S3C2412_IISCON);
 		if (iiscon & S3C2412_IISCON_LRINDEX)
 			break;
 
-		if (timeout < jiffies) {
-			printk(KERN_ERR "%s: timeout\n", __func__);
-			return -ETIMEDOUT;
-		}
+		cpu_relax();
+	}
+
+	if (!loops) {
+		printk(KERN_ERR "%s: timeout\n", __func__);
+		return -ETIMEDOUT;
 	}
 
 	return 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 37f7adeae323..9babda559c92 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1135,9 +1135,10 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 	ret = snprintf(buf, PAGE_SIZE, "%s: %s  in %d out %d\n",
 		       w->name, w->power ? "On" : "Off", in, out);
 
-	if (w->active && w->sname)
-		ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n",
-				w->sname);
+	if (w->sname)
+		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+				w->sname,
+				w->active ? "active" : "inactive");
 
 	list_for_each_entry(p, &w->sources, list_sink) {
 		if (p->connected && !p->connected(w, p->sink))