Merge commit 'v2.6.34-rc1' into for-2.6.35

25 files changed, 1313 insertions, 219 deletions

diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 410a893..4b67140 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97
 ##
 ## Boards
 ##
-config SND_SOC_SAMPLE_PSC_AC97
-	tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+config SND_SOC_DB1200
+	tristate "DB1200 AC97+I2S audio support"
 	depends on SND_SOC_AU1XPSC
 	select SND_SOC_AU1XPSC_AC97
 	select SND_SOC_AC97_CODEC
+	select SND_SOC_AU1XPSC_I2S
+	select SND_SOC_WM8731
 	help
-	  This is a sample AC97 sound machine for use in Au12x0/Au1550
-	  based systems which have audio on PSC1 (e.g. Db1200 demoboard).
+	  Select this option to enable audio (AC97 or I2S) on the
+	  Alchemy/AMD/RMI DB1200 demoboard.
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 6c6950b..1687307 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
 obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
 
 # Boards
-snd-soc-sample-ac97-objs := sample-ac97.o
+snd-soc-db1200-objs := db1200.o
 
-obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o
+obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
new file mode 100644
index 0000000..cdf7be1
--- /dev/null
+++ b/sound/soc/au1x/db1200.c
@@ -0,0 +1,141 @@
+/*
+ * DB1200 ASoC audio fabric support code.
+ *
+ * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "../codecs/ac97.h"
+#include "../codecs/wm8731.h"
+#include "psc.h"
+
+/*-------------------------  AC97 PART  ---------------------------*/
+
+static struct snd_soc_dai_link db1200_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.cpu_dai	= &au1xpsc_ac97_dai,
+	.codec_dai	= &ac97_dai,
+};
+
+static struct snd_soc_card db1200_ac97_machine = {
+	.name		= "DB1200_AC97",
+	.dai_link	= &db1200_ac97_dai,
+	.num_links	= 1,
+	.platform	= &au1xpsc_soc_platform,
+};
+
+static struct snd_soc_device db1200_ac97_devdata = {
+	.card		= &db1200_ac97_machine,
+	.codec_dev	= &soc_codec_dev_ac97,
+};
+
+/*-------------------------  I2S PART  ---------------------------*/
+
+static int db1200_i2s_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* WM8731 has its own 12MHz crystal */
+	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+				12000000, SND_SOC_CLOCK_IN);
+
+	/* codec is bitclock and lrclk master */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		goto out;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		goto out;
+
+	ret = 0;
+out:
+	return ret;
+}
+
+static struct snd_soc_ops db1200_i2s_wm8731_ops = {
+	.startup	= db1200_i2s_startup,
+};
+
+static struct snd_soc_dai_link db1200_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.cpu_dai	= &au1xpsc_i2s_dai,
+	.codec_dai	= &wm8731_dai,
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1200_i2s_machine = {
+	.name		= "DB1200_I2S",
+	.dai_link	= &db1200_i2s_dai,
+	.num_links	= 1,
+	.platform	= &au1xpsc_soc_platform,
+};
+
+static struct snd_soc_device db1200_i2s_devdata = {
+	.card		= &db1200_i2s_machine,
+	.codec_dev	= &soc_codec_dev_wm8731,
+};
+
+/*-------------------------  COMMON PART  ---------------------------*/
+
+static struct platform_device *db1200_asoc_dev;
+
+static int __init db1200_audio_load(void)
+{
+	int ret;
+
+	ret = -ENOMEM;
+	db1200_asoc_dev = platform_device_alloc("soc-audio", -1);
+	if (!db1200_asoc_dev)
+		goto out;
+
+	/* DB1200 board setup set PSC1MUX to preferred audio device */
+	if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
+		platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata);
+	else
+		platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata);
+
+	db1200_ac97_devdata.dev = &db1200_asoc_dev->dev;
+	db1200_i2s_devdata.dev = &db1200_asoc_dev->dev;
+	ret = platform_device_add(db1200_asoc_dev);
+
+	if (ret) {
+		platform_device_put(db1200_asoc_dev);
+		db1200_asoc_dev = NULL;
+	}
+out:
+	return ret;
+}
+
+static void __exit db1200_audio_unload(void)
+{
+	platform_device_unregister(db1200_asoc_dev);
+}
+
+module_init(db1200_audio_load);
+module_exit(db1200_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1200 ASoC audio support");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 19e4d37..6d9f4c6 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -51,8 +51,8 @@ struct au1xpsc_audio_dmadata {
 	struct snd_pcm_substream *substream;
 	unsigned long curr_period;	/* current segment DDMA is working on */
 	unsigned long q_period;		/* queue period(s) */
-	unsigned long dma_area;		/* address of queued DMA area */
-	unsigned long dma_area_s;	/* start address of DMA area */
+	dma_addr_t dma_area;		/* address of queued DMA area */
+	dma_addr_t dma_area_s;		/* start address of DMA area */
 	unsigned long pos;		/* current byte position being played */
 	unsigned long periods;		/* number of SG segments in total */
 	unsigned long period_bytes;	/* size in bytes of one SG segment */
@@ -94,8 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 
 static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
 {
-	au1xxx_dbdma_put_source_flags(cd->ddma_chan,
-				(void *)phys_to_virt(cd->dma_area),
+	au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area,
 				cd->period_bytes, DDMA_FLAGS_IE);
 
 	/* update next-to-queue period */
@@ -109,9 +108,8 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
 
 static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
 {
-	au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
-				(void *)phys_to_virt(cd->dma_area),
-				cd->period_bytes, DDMA_FLAGS_IE);
+	au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area,
+			      cd->period_bytes, DDMA_FLAGS_IE);
 
 	/* update next-to-queue period */
 	++cd->q_period;
@@ -233,7 +231,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
 	pcd->substream = substream;
 	pcd->period_bytes = params_period_bytes(params);
 	pcd->periods = params_periods(params);
-	pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
+	pcd->dma_area_s = pcd->dma_area = runtime->dma_addr;
 	pcd->q_period = 0;
 	pcd->curr_period = 0;
 	pcd->pos = 0;
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
deleted file mode 100644
index 27683eb..0000000
--- a/sound/soc/au1x/sample-ac97.c
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Sample Au12x0/Au1550 PSC AC97 sound machine.
- *
- * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
- *
- *  This program is free software; you can redistribute it and/or modify
- *  it under the terms outlined in the file COPYING at the root of this
- *  source archive.
- *
- * This is a very generic AC97 sound machine driver for boards which
- * have (AC97) audio at PSC1 (e.g. DB1200 demoboards).
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-#include <asm/mach-au1x00/au1000.h>
-#include <asm/mach-au1x00/au1xxx_psc.h>
-#include <asm/mach-au1x00/au1xxx_dbdma.h>
-
-#include "../codecs/ac97.h"
-#include "psc.h"
-
-static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_sync(codec);
-	return 0;
-}
-
-static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
-	.name		= "AC97",
-	.stream_name	= "AC97 HiFi",
-	.cpu_dai	= &au1xpsc_ac97_dai,	/* see psc-ac97.c */
-	.codec_dai	= &ac97_dai,		/* see codecs/ac97.c */
-	.init		= au1xpsc_sample_ac97_init,
-	.ops		= NULL,
-};
-
-static struct snd_soc_card au1xpsc_sample_ac97_machine = {
-	.name		= "Au1xxx PSC AC97 Audio",
-	.dai_link	= &au1xpsc_sample_ac97_dai,
-	.num_links	= 1,
-};
-
-static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
-	.card		= &au1xpsc_sample_ac97_machine,
-	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
-	.codec_dev	= &soc_codec_dev_ac97,
-};
-
-static struct resource au1xpsc_psc1_res[] = {
-	[0] = {
-		.start	= CPHYSADDR(PSC1_BASE_ADDR),
-		.end	= CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff,
-		.flags	= IORESOURCE_MEM,
-	},
-	[1] = {
-#ifdef CONFIG_SOC_AU1200
-		.start	= AU1200_PSC1_INT,
-		.end	= AU1200_PSC1_INT,
-#elif defined(CONFIG_SOC_AU1550)
-		.start	= AU1550_PSC1_INT,
-		.end	= AU1550_PSC1_INT,
-#endif
-		.flags	= IORESOURCE_IRQ,
-	},
-	[2] = {
-		.start	= DSCR_CMD0_PSC1_TX,
-		.end	= DSCR_CMD0_PSC1_TX,
-		.flags	= IORESOURCE_DMA,
-	},
-	[3] = {
-		.start	= DSCR_CMD0_PSC1_RX,
-		.end	= DSCR_CMD0_PSC1_RX,
-		.flags	= IORESOURCE_DMA,
-	},
-};
-
-static struct platform_device *au1xpsc_sample_ac97_dev;
-
-static int __init au1xpsc_sample_ac97_load(void)
-{
-	int ret;
-
-#ifdef CONFIG_SOC_AU1200
-	unsigned long io;
-
-	/* modify sys_pinfunc for AC97 on PSC1 */
-	io = au_readl(SYS_PINFUNC);
-	io |= SYS_PINFUNC_P1C;
-	io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B);
-	au_writel(io, SYS_PINFUNC);
-	au_sync();
-#endif
-
-	ret = -ENOMEM;
-
-	/* setup PSC clock source for AC97 part: external clock provided
-	 * by codec.  The psc-ac97.c driver depends on this setting!
-	 */
-	au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET);
-	au_sync();
-
-	au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1);
-	if (!au1xpsc_sample_ac97_dev)
-		goto out;
-
-	au1xpsc_sample_ac97_dev->resource =
-		kmemdup(au1xpsc_psc1_res, sizeof(struct resource) *
-			ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL);
-	au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res);
-	au1xpsc_sample_ac97_dev->id = 1;
-
-	platform_set_drvdata(au1xpsc_sample_ac97_dev,
-			     &au1xpsc_sample_ac97_devdata);
-	au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
-	ret = platform_device_add(au1xpsc_sample_ac97_dev);
-
-	if (ret) {
-		platform_device_put(au1xpsc_sample_ac97_dev);
-		au1xpsc_sample_ac97_dev = NULL;
-	}
-
-out:
-	return ret;
-}
-
-static void __exit au1xpsc_sample_ac97_exit(void)
-{
-	platform_device_unregister(au1xpsc_sample_ac97_dev);
-}
-
-module_init(au1xpsc_sample_ac97_load);
-module_exit(au1xpsc_sample_ac97_exit);
-
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index b9ef7e4..b68d99f 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -90,12 +90,10 @@ static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (reg >= codec->reg_cache_size)
 		return -EINVAL;
 
-	reg &= AK4104_REG_MASK;
-	reg |= AK4104_WRITE;
-
 	/* only write to the hardware if value has changed */
 	if (cache[reg] != value) {
-		u8 tmp[2] = { reg, value };
+		u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value };
+
 		if (spi_write(spi, tmp, sizeof(tmp))) {
 			dev_err(&spi->dev, "SPI write failed\n");
 			return -EIO;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a2763c2..9cd0a66 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -137,7 +137,7 @@ static void uda1380_flush_work(struct work_struct *work)
 {
 	int bit, reg;
 
-	for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+	for_each_set_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
 		reg = 0x10 + bit;
 		pr_debug("uda1380: flush reg %x val %x:\n", reg,
 				uda1380_read_reg_cache(uda1380_codec, reg));
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 718ef91..df2c6d9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1349,7 +1349,7 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data)
 	int mask;
 	struct wm8350_jack_data *jack = NULL;
 
-	switch (irq) {
+	switch (irq - wm8350->irq_base) {
 	case WM8350_IRQ_CODEC_JCK_DET_L:
 		jack = &priv->hpl;
 		mask = WM8350_JACK_L_LVL;
@@ -1424,7 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
 	wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
 
 	/* Sync status */
-	wm8350_hp_jack_handler(irq, priv);
+	wm8350_hp_jack_handler(irq + wm8350->irq_base, priv);
 
 	return 0;
 }
@@ -1521,8 +1521,8 @@ static int wm8350_remove(struct platform_device *pdev)
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
 
-	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
-	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv);
 
 	priv->hpl.jack = NULL;
 	priv->hpr.jack = NULL;
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 3326e2a..1a5b8e0 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -55,7 +55,7 @@ static __init int efika_fabric_init(void)
 	struct platform_device *pdev;
 	int rc;
 
-	if (!machine_is_compatible("bplan,efika"))
+	if (!of_machine_is_compatible("bplan,efika"))
 		return -ENODEV;
 
 	card.platform = &mpc5200_audio_dma_platform;
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index b928ef7..6644cba 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -55,7 +55,7 @@ static __init int pcm030_fabric_init(void)
 	struct platform_device *pdev;
 	int rc;
 
-	if (!machine_is_compatible("phytec,pcm030"))
+	if (!of_machine_is_compatible("phytec,pcm030"))
 		return -ENODEV;
 
 	card.platform = &mpc5200_audio_dma_platform;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 5532579..d9cb984 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -35,22 +35,25 @@
 struct imx_pcm_runtime_data {
 	int period;
 	int periods;
-	unsigned long dma_addr;
-	int dma;
 	unsigned long offset;
+	unsigned long last_offset;
 	unsigned long size;
-	unsigned long period_cnt;
-	void *buf;
 	struct timer_list timer;
-	int period_time;
+	int poll_time;
 };
 
+static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd)
+{
+	iprtd->timer.expires = jiffies + iprtd->poll_time;
+}
+
 static void imx_ssi_timer_callback(unsigned long data)
 {
 	struct snd_pcm_substream *substream = (void *)data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	struct pt_regs regs;
+	unsigned long delta;
 
 	get_fiq_regs(&regs);
 
@@ -59,9 +62,25 @@ static void imx_ssi_timer_callback(unsigned long data)
 	else
 		iprtd->offset = regs.ARM_r9 & 0xffff;
 
-	iprtd->timer.expires = jiffies + iprtd->period_time;
+	/* How much data have we transferred since the last period report? */
+	if (iprtd->offset >= iprtd->last_offset)
+		delta = iprtd->offset - iprtd->last_offset;
+	else
+		delta = runtime->buffer_size + iprtd->offset
+			- iprtd->last_offset;
+
+	/* If we've transferred at least a period then report it and
+	 * reset our poll time */
+	if (delta >= runtime->period_size) {
+		snd_pcm_period_elapsed(substream);
+		iprtd->last_offset = iprtd->offset;
+
+		imx_ssi_set_next_poll(iprtd);
+	}
+
+	/* Restart the timer; if we didn't report we'll run on the next tick */
 	add_timer(&iprtd->timer);
-	snd_pcm_period_elapsed(substream);
+
 }
 
 static struct fiq_handler fh = {
@@ -76,9 +95,10 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 	iprtd->size = params_buffer_bytes(params);
 	iprtd->periods = params_periods(params);
-	iprtd->period = params_period_bytes(params);
+	iprtd->period = params_period_bytes(params) ;
 	iprtd->offset = 0;
-	iprtd->period_time = HZ / (params_rate(params) / params_period_size(params));
+	iprtd->last_offset = 0;
+	iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params));
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 
@@ -114,7 +134,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		iprtd->timer.expires = jiffies + iprtd->period_time;
+		imx_ssi_set_next_poll(iprtd);
 		add_timer(&iprtd->timer);
 		if (++fiq_enable == 1)
 			enable_fiq(imx_pcm_fiq);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 18ebdc7..f11963c 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP
 	tristate
 	select OMAP_MCBSP
 
+config SND_OMAP_SOC_MCPDM
+	tristate
+
 config SND_OMAP_SOC_N810
 	tristate "SoC Audio support for Nokia N810"
 	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 19283e5..0bc00ca 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -1,9 +1,11 @@
 # OMAP Platform Support
 snd-soc-omap-objs := omap-pcm.o
 snd-soc-omap-mcbsp-objs := omap-mcbsp.o
+snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o
 
 obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
 obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c
new file mode 100644
index 0000000..ad8df6c
--- /dev/null
+++ b/sound/soc/omap/mcpdm.c
@@ -0,0 +1,484 @@
+/*
+ * mcpdm.c  --  McPDM interface driver
+ *
+ * Author: Jorge Eduardo Candelaria <x0107209@ti.com>
+ * Copyright (C) 2009 - Texas Instruments, Inc.
+ *
+ * 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 published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/wait.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/irq.h>
+
+#include "mcpdm.h"
+
+static struct omap_mcpdm *mcpdm;
+
+static inline void omap_mcpdm_write(u16 reg, u32 val)
+{
+       __raw_writel(val, mcpdm->io_base + reg);
+}
+
+static inline int omap_mcpdm_read(u16 reg)
+{
+       return __raw_readl(mcpdm->io_base + reg);
+}
+
+static void omap_mcpdm_reg_dump(void)
+{
+       dev_dbg(mcpdm->dev, "***********************\n");
+       dev_dbg(mcpdm->dev, "IRQSTATUS_RAW:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_IRQSTATUS_RAW));
+       dev_dbg(mcpdm->dev, "IRQSTATUS:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_IRQSTATUS));
+       dev_dbg(mcpdm->dev, "IRQENABLE_SET:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_IRQENABLE_SET));
+       dev_dbg(mcpdm->dev, "IRQENABLE_CLR:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_IRQENABLE_CLR));
+       dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
+                       omap_mcpdm_read(MCPDM_IRQWAKE_EN));
+       dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
+                       omap_mcpdm_read(MCPDM_DMAENABLE_SET));
+       dev_dbg(mcpdm->dev, "DMAENABLE_CLR:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_DMAENABLE_CLR));
+       dev_dbg(mcpdm->dev, "DMAWAKEEN:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_DMAWAKEEN));
+       dev_dbg(mcpdm->dev, "CTRL:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_CTRL));
+       dev_dbg(mcpdm->dev, "DN_DATA:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_DN_DATA));
+       dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
+                       omap_mcpdm_read(MCPDM_UP_DATA));
+       dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
+                       omap_mcpdm_read(MCPDM_FIFO_CTRL_DN));
+       dev_dbg(mcpdm->dev, "FIFO_CTRL_UP:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_FIFO_CTRL_UP));
+       dev_dbg(mcpdm->dev, "DN_OFFSET:  0x%04x\n",
+                       omap_mcpdm_read(MCPDM_DN_OFFSET));
+       dev_dbg(mcpdm->dev, "***********************\n");
+}
+
+/*
+ * Takes the McPDM module in and out of reset state.
+ * Uplink and downlink can be reset individually.
+ */
+static void omap_mcpdm_reset_capture(int reset)
+{
+       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+       if (reset)
+               ctrl |= SW_UP_RST;
+       else
+               ctrl &= ~SW_UP_RST;
+
+       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+static void omap_mcpdm_reset_playback(int reset)
+{
+       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+       if (reset)
+               ctrl |= SW_DN_RST;
+       else
+               ctrl &= ~SW_DN_RST;
+
+       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Enables the transfer through the PDM interface to/from the Phoenix
+ * codec by enabling the corresponding UP or DN channels.
+ */
+void omap_mcpdm_start(int stream)
+{
+       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+       if (stream)
+               ctrl |= mcpdm->up_channels;
+       else
+               ctrl |= mcpdm->dn_channels;
+
+       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Disables the transfer through the PDM interface to/from the Phoenix
+ * codec by disabling the corresponding UP or DN channels.
+ */
+void omap_mcpdm_stop(int stream)
+{
+       int ctrl = omap_mcpdm_read(MCPDM_CTRL);
+
+       if (stream)
+               ctrl &= ~mcpdm->up_channels;
+       else
+               ctrl &= ~mcpdm->dn_channels;
+
+       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+}
+
+/*
+ * Configures McPDM uplink for audio recording.
+ * This function should be called before omap_mcpdm_start.
+ */
+int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink)
+{
+       int irq_mask = 0;
+       int ctrl;
+
+       if (!uplink)
+               return -EINVAL;
+
+       mcpdm->uplink = uplink;
+
+       /* Enable irq request generation */
+       irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+       omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+
+       /* Configure uplink threshold */
+       if (uplink->threshold > UP_THRES_MAX)
+               uplink->threshold = UP_THRES_MAX;
+
+       omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold);
+
+       /* Configure DMA controller */
+       omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE);
+
+       /* Set pdm out format */
+       ctrl = omap_mcpdm_read(MCPDM_CTRL);
+       ctrl &= ~PDMOUTFORMAT;
+       ctrl |= uplink->format & PDMOUTFORMAT;
+
+       /* Uplink channels */
+       mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK);
+
+       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+
+       return 0;
+}
+
+/*
+ * Configures McPDM downlink for audio playback.
+ * This function should be called before omap_mcpdm_start.
+ */
+int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink)
+{
+       int irq_mask = 0;
+       int ctrl;
+
+       if (!downlink)
+               return -EINVAL;
+
+       mcpdm->downlink = downlink;
+
+       /* Enable irq request generation */
+       irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+       omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask);
+
+       /* Configure uplink threshold */
+       if (downlink->threshold > DN_THRES_MAX)
+               downlink->threshold = DN_THRES_MAX;
+
+       omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold);
+
+       /* Enable DMA request generation */
+       omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE);
+
+       /* Set pdm out format */
+       ctrl = omap_mcpdm_read(MCPDM_CTRL);
+       ctrl &= ~PDMOUTFORMAT;
+       ctrl |= downlink->format & PDMOUTFORMAT;
+
+       /* Downlink channels */
+       mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK);
+
+       omap_mcpdm_write(MCPDM_CTRL, ctrl);
+
+       return 0;
+}
+
+/*
+ * Cleans McPDM uplink configuration.
+ * This function should be called when the stream is closed.
+ */
+int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink)
+{
+       int irq_mask = 0;
+
+       if (!uplink)
+               return -EINVAL;
+
+       /* Disable irq request generation */
+       irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK;
+       omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+
+       /* Disable DMA request generation */
+       omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE);
+
+       /* Clear Downlink channels */
+       mcpdm->up_channels = 0;
+
+       mcpdm->uplink = NULL;
+
+       return 0;
+}
+
+/*
+ * Cleans McPDM downlink configuration.
+ * This function should be called when the stream is closed.
+ */
+int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink)
+{
+       int irq_mask = 0;
+
+       if (!downlink)
+               return -EINVAL;
+
+       /* Disable irq request generation */
+       irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK;
+       omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask);
+
+       /* Disable DMA request generation */
+       omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE);
+
+       /* clear Downlink channels */
+       mcpdm->dn_channels = 0;
+
+       mcpdm->downlink = NULL;
+
+       return 0;
+}
+
+static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id)
+{
+       struct omap_mcpdm *mcpdm_irq = dev_id;
+       int irq_status;
+
+       irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS);
+
+       /* Acknowledge irq event */
+       omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status);
+
+       if (irq & MCPDM_DN_IRQ_FULL) {
+               dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+               omap_mcpdm_reset_playback(1);
+               omap_mcpdm_playback_open(mcpdm_irq->downlink);
+               omap_mcpdm_reset_playback(0);
+       }
+
+       if (irq & MCPDM_DN_IRQ_EMPTY) {
+               dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status);
+               omap_mcpdm_reset_playback(1);
+               omap_mcpdm_playback_open(mcpdm_irq->downlink);
+               omap_mcpdm_reset_playback(0);
+       }
+
+       if (irq & MCPDM_DN_IRQ) {
+               dev_dbg(mcpdm_irq->dev, "DN write request\n");
+       }
+
+       if (irq & MCPDM_UP_IRQ_FULL) {
+               dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+               omap_mcpdm_reset_capture(1);
+               omap_mcpdm_capture_open(mcpdm_irq->uplink);
+               omap_mcpdm_reset_capture(0);
+       }
+
+       if (irq & MCPDM_UP_IRQ_EMPTY) {
+               dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status);
+               omap_mcpdm_reset_capture(1);
+               omap_mcpdm_capture_open(mcpdm_irq->uplink);
+               omap_mcpdm_reset_capture(0);
+       }
+
+       if (irq & MCPDM_UP_IRQ) {
+               dev_dbg(mcpdm_irq->dev, "UP write request\n");
+       }
+
+       return IRQ_HANDLED;
+}
+
+int omap_mcpdm_request(void)
+{
+       int ret;
+
+       clk_enable(mcpdm->clk);
+
+       spin_lock(&mcpdm->lock);
+
+       if (!mcpdm->free) {
+               dev_err(mcpdm->dev, "McPDM interface is in use\n");
+               spin_unlock(&mcpdm->lock);
+               ret = -EBUSY;
+               goto err;
+       }
+       mcpdm->free = 0;
+
+       spin_unlock(&mcpdm->lock);
+
+       /* Disable lines while request is ongoing */
+       omap_mcpdm_write(MCPDM_CTRL, 0x00);
+
+       ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+                               0, "McPDM", (void *)mcpdm);
+       if (ret) {
+               dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n");
+               goto err;
+       }
+
+       return 0;
+
+err:
+       clk_disable(mcpdm->clk);
+       return ret;
+}
+
+void omap_mcpdm_free(void)
+{
+       spin_lock(&mcpdm->lock);
+       if (mcpdm->free) {
+               dev_err(mcpdm->dev, "McPDM interface is already free\n");
+               spin_unlock(&mcpdm->lock);
+               return;
+       }
+       mcpdm->free = 1;
+       spin_unlock(&mcpdm->lock);
+
+       clk_disable(mcpdm->clk);
+
+       free_irq(mcpdm->irq, (void *)mcpdm);
+}
+
+/* Enable/disable DC offset cancelation for the analog
+ * headset path (PDM channels 1 and 2).
+ */
+int omap_mcpdm_set_offset(int offset1, int offset2)
+{
+       int offset;
+
+       if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX))
+               return -EINVAL;
+
+       offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2);
+
+       /* offset cancellation for channel 1 */
+       if (offset1)
+               offset |= DN_OFST_RX1_EN;
+       else
+               offset &= ~DN_OFST_RX1_EN;
+
+       /* offset cancellation for channel 2 */
+       if (offset2)
+               offset |= DN_OFST_RX2_EN;
+       else
+               offset &= ~DN_OFST_RX2_EN;
+
+       omap_mcpdm_write(MCPDM_DN_OFFSET, offset);
+
+       return 0;
+}
+
+static int __devinit omap_mcpdm_probe(struct platform_device *pdev)
+{
+       struct resource *res;
+       int ret = 0;
+
+       mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL);
+       if (!mcpdm) {
+               ret = -ENOMEM;
+               goto exit;
+       }
+
+       res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+       if (res == NULL) {
+               dev_err(&pdev->dev, "no resource\n");
+               goto err_resource;
+       }
+
+       spin_lock_init(&mcpdm->lock);
+       mcpdm->free = 1;
+       mcpdm->io_base = ioremap(res->start, resource_size(res));
+       if (!mcpdm->io_base) {
+               ret = -ENOMEM;
+               goto err_resource;
+       }
+
+       mcpdm->irq = platform_get_irq(pdev, 0);
+
+       mcpdm->clk = clk_get(&pdev->dev, "pdm_ck");
+       if (IS_ERR(mcpdm->clk)) {
+               ret = PTR_ERR(mcpdm->clk);
+               dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret);
+               goto err_clk;
+       }
+
+       mcpdm->dev = &pdev->dev;
+       platform_set_drvdata(pdev, mcpdm);
+
+       return 0;
+
+err_clk:
+       iounmap(mcpdm->io_base);
+err_resource:
+       kfree(mcpdm);
+exit:
+       return ret;
+}
+
+static int __devexit omap_mcpdm_remove(struct platform_device *pdev)
+{
+       struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev);
+
+       platform_set_drvdata(pdev, NULL);
+
+       clk_put(mcpdm_ptr->clk);
+
+       iounmap(mcpdm_ptr->io_base);
+
+       mcpdm_ptr->clk = NULL;
+       mcpdm_ptr->free = 0;
+       mcpdm_ptr->dev = NULL;
+
+       kfree(mcpdm_ptr);
+
+       return 0;
+}
+
+static struct platform_driver omap_mcpdm_driver = {
+       .probe = omap_mcpdm_probe,
+       .remove = __devexit_p(omap_mcpdm_remove),
+       .driver = {
+               .name = "omap-mcpdm",
+       },
+};
+
+static struct platform_device *omap_mcpdm_device;
+
+static int __init omap_mcpdm_init(void)
+{
+       return platform_driver_register(&omap_mcpdm_driver);
+}
+arch_initcall(omap_mcpdm_init);
diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h
new file mode 100644
index 0000000..7bb326e
--- /dev/null
+++ b/sound/soc/omap/mcpdm.h
@@ -0,0 +1,151 @@
+/*
+ * mcpdm.h -- Defines for McPDM driver
+ *
+ * Author: Jorge Eduardo Candelaria <x0107209@ti.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 published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+/* McPDM registers */
+
+#define MCPDM_REVISION         0x00
+#define MCPDM_SYSCONFIG                0x10
+#define MCPDM_IRQSTATUS_RAW    0x24
+#define MCPDM_IRQSTATUS                0x28
+#define MCPDM_IRQENABLE_SET    0x2C
+#define MCPDM_IRQENABLE_CLR    0x30
+#define MCPDM_IRQWAKE_EN       0x34
+#define MCPDM_DMAENABLE_SET    0x38
+#define MCPDM_DMAENABLE_CLR    0x3C
+#define MCPDM_DMAWAKEEN                0x40
+#define MCPDM_CTRL             0x44
+#define MCPDM_DN_DATA          0x48
+#define MCPDM_UP_DATA          0x4C
+#define MCPDM_FIFO_CTRL_DN     0x50
+#define MCPDM_FIFO_CTRL_UP     0x54
+#define MCPDM_DN_OFFSET                0x58
+
+/*
+ * MCPDM_IRQ bit fields
+ * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR
+ */
+
+#define MCPDM_DN_IRQ                   (1 << 0)
+#define MCPDM_DN_IRQ_EMPTY             (1 << 1)
+#define MCPDM_DN_IRQ_ALMST_EMPTY       (1 << 2)
+#define MCPDM_DN_IRQ_FULL              (1 << 3)
+
+#define MCPDM_UP_IRQ                   (1 << 8)
+#define MCPDM_UP_IRQ_EMPTY             (1 << 9)
+#define MCPDM_UP_IRQ_ALMST_FULL                (1 << 10)
+#define MCPDM_UP_IRQ_FULL              (1 << 11)
+
+#define MCPDM_DOWNLINK_IRQ_MASK                0x00F
+#define MCPDM_UPLINK_IRQ_MASK          0xF00
+
+/*
+ * MCPDM_DMAENABLE bit fields
+ */
+
+#define DMA_DN_ENABLE          0x1
+#define DMA_UP_ENABLE          0x2
+
+/*
+ * MCPDM_CTRL bit fields
+ */
+
+#define PDM_UP1_EN             0x0001
+#define PDM_UP2_EN             0x0002
+#define PDM_UP3_EN             0x0004
+#define PDM_DN1_EN             0x0008
+#define PDM_DN2_EN             0x0010
+#define PDM_DN3_EN             0x0020
+#define PDM_DN4_EN             0x0040
+#define PDM_DN5_EN             0x0080
+#define PDMOUTFORMAT           0x0100
+#define CMD_INT                        0x0200
+#define STATUS_INT             0x0400
+#define SW_UP_RST              0x0800
+#define SW_DN_RST              0x1000
+#define PDM_UP_MASK            0x007
+#define PDM_DN_MASK            0x0F8
+#define PDM_CMD_MASK           0x200
+#define PDM_STATUS_MASK                0x400
+
+
+#define PDMOUTFORMAT_LJUST     (0 << 8)
+#define PDMOUTFORMAT_RJUST     (1 << 8)
+
+/*
+ * MCPDM_FIFO_CTRL bit fields
+ */
+
+#define UP_THRES_MAX           0xF
+#define DN_THRES_MAX           0xF
+
+/*
+ * MCPDM_DN_OFFSET bit fields
+ */
+
+#define DN_OFST_RX1_EN         0x0001
+#define DN_OFST_RX2_EN         0x0100
+
+#define DN_OFST_RX1            1
+#define DN_OFST_RX2            9
+#define DN_OFST_MAX            0x1F
+
+#define MCPDM_UPLINK           1
+#define MCPDM_DOWNLINK         2
+
+struct omap_mcpdm_link {
+       int irq_mask;
+       int threshold;
+       int format;
+       int channels;
+};
+
+struct omap_mcpdm_platform_data {
+       unsigned long phys_base;
+       u16 irq;
+};
+
+struct omap_mcpdm {
+       struct device *dev;
+       unsigned long phys_base;
+       void __iomem *io_base;
+       u8 free;
+       int irq;
+
+       spinlock_t lock;
+       struct omap_mcpdm_platform_data *pdata;
+       struct clk *clk;
+       struct omap_mcpdm_link *downlink;
+       struct omap_mcpdm_link *uplink;
+       struct completion irq_completion;
+
+       int dn_channels;
+       int up_channels;
+};
+
+extern void omap_mcpdm_start(int stream);
+extern void omap_mcpdm_stop(int stream);
+extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink);
+extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink);
+extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink);
+extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink);
+extern int omap_mcpdm_request(void);
+extern void omap_mcpdm_free(void);
+extern int omap_mcpdm_set_offset(int offset1, int offset2);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6bbbd2a..e814a95 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -39,6 +39,14 @@
 
 #define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_8000_96000)
 
+#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \
+	xhandler_get, xhandler_put) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = omap_mcbsp_st_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long) &(struct soc_mixer_control) \
+	{.min = xmin, .max = xmax} }
+
 struct omap_mcbsp_data {
 	unsigned int			bus_id;
 	struct omap_mcbsp_reg_cfg	regs;
@@ -82,11 +90,11 @@ static const int omap1_dma_reqs[][2] = {};
 static const unsigned long omap1_mcbsp_port[][2] = {};
 #endif
 
-#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP2) || defined(CONFIG_ARCH_OMAP3)
 static const int omap24xx_dma_reqs[][2] = {
 	{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
 	{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
 	{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
 	{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
 	{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
@@ -124,7 +132,7 @@ static const unsigned long omap2430_mcbsp_port[][2] = {
 static const unsigned long omap2430_mcbsp_port[][2] = {};
 #endif
 
-#if defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP3)
 static const unsigned long omap34xx_mcbsp_port[][2] = {
 	{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
 	  OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
@@ -287,6 +295,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
 	omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
 	omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
+	omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
+							OMAP_DMA_DATA_TYPE_S16;
 	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
 
 	if (mcbsp_data->configured) {
@@ -637,6 +647,136 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
 
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
+int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int max = mc->max;
+	int min = mc->min;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = min;
+	uinfo->value.integer.max = max;
+	return 0;
+}
+
+#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel)			\
+static int								\
+omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc,	\
+					struct snd_ctl_elem_value *uc)	\
+{									\
+	struct soc_mixer_control *mc =					\
+		(struct soc_mixer_control *)kc->private_value;		\
+	int max = mc->max;						\
+	int min = mc->min;						\
+	int val = uc->value.integer.value[0];				\
+									\
+	if (val < min || val > max)					\
+		return -EINVAL;						\
+									\
+	/* OMAP McBSP implementation uses index values 0..4 */		\
+	return omap_st_set_chgain((id)-1, channel, val);		\
+}
+
+#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel)			\
+static int								\
+omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc,	\
+					struct snd_ctl_elem_value *uc)	\
+{									\
+	s16 chgain;							\
+									\
+	if (omap_st_get_chgain((id)-1, channel, &chgain))		\
+		return -EAGAIN;						\
+									\
+	uc->value.integer.value[0] = chgain;				\
+	return 0;							\
+}
+
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0)
+OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0)
+OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1)
+
+static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u8 value = ucontrol->value.integer.value[0];
+
+	if (value == omap_st_is_enabled(mc->reg))
+		return 0;
+
+	if (value)
+		omap_st_enable(mc->reg);
+	else
+		omap_st_disable(mc->reg);
+
+	return 1;
+}
+
+static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg);
+	return 0;
+}
+
+static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = {
+	SOC_SINGLE_EXT("McBSP2 Sidetone Switch", 1, 0, 1, 0,
+			omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume",
+				      -32768, 32767,
+				      omap_mcbsp2_get_st_ch0_volume,
+				      omap_mcbsp2_set_st_ch0_volume),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume",
+				      -32768, 32767,
+				      omap_mcbsp2_get_st_ch1_volume,
+				      omap_mcbsp2_set_st_ch1_volume),
+};
+
+static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = {
+	SOC_SINGLE_EXT("McBSP3 Sidetone Switch", 2, 0, 1, 0,
+			omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume",
+				      -32768, 32767,
+				      omap_mcbsp3_get_st_ch0_volume,
+				      omap_mcbsp3_set_st_ch0_volume),
+	OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume",
+				      -32768, 32767,
+				      omap_mcbsp3_get_st_ch1_volume,
+				      omap_mcbsp3_set_st_ch1_volume),
+};
+
+int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id)
+{
+	if (!cpu_is_omap34xx())
+		return -ENODEV;
+
+	switch (mcbsp_id) {
+	case 1: /* McBSP 2 */
+		return snd_soc_add_controls(codec, omap_mcbsp2_st_controls,
+					ARRAY_SIZE(omap_mcbsp2_st_controls));
+	case 2: /* McBSP 3 */
+		return snd_soc_add_controls(codec, omap_mcbsp3_st_controls,
+					ARRAY_SIZE(omap_mcbsp3_st_controls));
+	default:
+		break;
+	}
+
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls);
+
 static int __init snd_omap_mcbsp_init(void)
 {
 	return snd_soc_register_dais(omap_mcbsp_dai,
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 647d2f9..6c363e5 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -50,11 +50,13 @@ enum omap_mcbsp_div {
 #undef  NUM_LINKS
 #define NUM_LINKS	3
 #endif
-#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
 #undef  NUM_LINKS
 #define NUM_LINKS	5
 #endif
 
 extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
 
+int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id);
+
 #endif
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
new file mode 100644
index 0000000..25f19e4
--- /dev/null
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -0,0 +1,251 @@
+/*
+ * omap-mcpdm.c  --  OMAP ALSA SoC DAI driver using McPDM port
+ *
+ * Copyright (C) 2009 Texas Instruments
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ * Contact: Jorge Eduardo Candelaria <x0107209@ti.com>
+ *          Margarita Olaya <magi.olaya@ti.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 published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/control.h>
+#include <plat/dma.h>
+#include <plat/mcbsp.h>
+#include "mcpdm.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+
+struct omap_mcpdm_data {
+	struct omap_mcpdm_link *links;
+	int active;
+};
+
+static struct omap_mcpdm_link omap_mcpdm_links[] = {
+	/* downlink */
+	{
+		.irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL,
+		.threshold = 1,
+		.format = PDMOUTFORMAT_LJUST,
+	},
+	/* uplink */
+	{
+		.irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL,
+		.threshold = 1,
+		.format = PDMOUTFORMAT_LJUST,
+	},
+};
+
+static struct omap_mcpdm_data mcpdm_data = {
+	.links = omap_mcpdm_links,
+	.active = 0,
+};
+
+/*
+ * Stream DMA parameters
+ */
+static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = {
+	{
+		.name = "Audio playback",
+		.dma_req = OMAP44XX_DMA_MCPDM_DL,
+		.data_type = OMAP_DMA_DATA_TYPE_S32,
+		.sync_mode = OMAP_DMA_SYNC_PACKET,
+		.packet_size = 16,
+		.port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA,
+	},
+	{
+		.name = "Audio capture",
+		.dma_req = OMAP44XX_DMA_MCPDM_UP,
+		.data_type = OMAP_DMA_DATA_TYPE_S32,
+		.sync_mode = OMAP_DMA_SYNC_PACKET,
+		.packet_size = 16,
+		.port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA,
+	},
+};
+
+static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int err = 0;
+
+	if (!cpu_dai->active)
+		err = omap_mcpdm_request();
+
+	return err;
+}
+
+static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	if (!cpu_dai->active)
+		omap_mcpdm_free();
+}
+
+static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data;
+	int stream = substream->stream;
+	int err = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (!mcpdm_priv->active++)
+			omap_mcpdm_start(stream);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (!--mcpdm_priv->active)
+			omap_mcpdm_stop(stream);
+		break;
+	default:
+		err = -EINVAL;
+	}
+
+	return err;
+}
+
+static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data;
+	struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links;
+	int stream = substream->stream;
+	int channels, err, link_mask = 0;
+
+	cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream];
+
+	channels = params_channels(params);
+	switch (channels) {
+	case 4:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 2 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 3;
+	case 3:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 2 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 2;
+	case 2:
+		link_mask |= 1 << 1;
+	case 1:
+		link_mask |= 1 << 0;
+		break;
+	default:
+		/* unsupported number of channels */
+		return -EINVAL;
+	}
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mcpdm_links[stream].channels = link_mask << 3;
+		err = omap_mcpdm_playback_open(&mcpdm_links[stream]);
+	} else {
+		mcpdm_links[stream].channels = link_mask << 0;
+		err = omap_mcpdm_capture_open(&mcpdm_links[stream]);
+	}
+
+	return err;
+}
+
+static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data;
+	struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links;
+	int stream = substream->stream;
+	int err;
+
+	if (substream->stream ==  SNDRV_PCM_STREAM_PLAYBACK)
+		err = omap_mcpdm_playback_close(&mcpdm_links[stream]);
+	else
+		err = omap_mcpdm_capture_close(&mcpdm_links[stream]);
+
+	return err;
+}
+
+static struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
+	.startup	= omap_mcpdm_dai_startup,
+	.shutdown	= omap_mcpdm_dai_shutdown,
+	.trigger	= omap_mcpdm_dai_trigger,
+	.hw_params	= omap_mcpdm_dai_hw_params,
+	.hw_free	= omap_mcpdm_dai_hw_free,
+};
+
+#define OMAP_MCPDM_RATES	(SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+#define OMAP_MCPDM_FORMATS	(SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai omap_mcpdm_dai = {
+	.name = "omap-mcpdm",
+	.id = -1,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 4,
+		.rates = OMAP_MCPDM_RATES,
+		.formats = OMAP_MCPDM_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = OMAP_MCPDM_RATES,
+		.formats = OMAP_MCPDM_FORMATS,
+	},
+	.ops = &omap_mcpdm_dai_ops,
+	.private_data = &mcpdm_data,
+};
+EXPORT_SYMBOL_GPL(omap_mcpdm_dai);
+
+static int __init snd_omap_mcpdm_init(void)
+{
+	return snd_soc_register_dai(&omap_mcpdm_dai);
+}
+module_init(snd_omap_mcpdm_init);
+
+static void __exit snd_omap_mcpdm_exit(void)
+{
+	snd_soc_unregister_dai(&omap_mcpdm_dai);
+}
+module_exit(snd_omap_mcpdm_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("OMAP PDM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h
new file mode 100644
index 0000000..73b80d5
--- /dev/null
+++ b/sound/soc/omap/omap-mcpdm.h
@@ -0,0 +1,29 @@
+/*
+ * omap-mcpdm.h
+ *
+ * Copyright (C) 2009 Texas Instruments
+ *
+ * Contact: Misael Lopez Cruz <x0052729@ti.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 published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_MCPDM_H__
+#define __OMAP_MCPDM_H__
+
+extern struct snd_soc_dai omap_mcpdm_dai;
+
+#endif	/* End of __OMAP_MCPDM_H__ */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 9db2770..825db38 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_PAUSE |
 				  SNDRV_PCM_INFO_RESUME,
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 64 * 1024,
 	.periods_min		= 2,
@@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	struct omap_runtime_data *prtd = runtime->private_data;
 	struct omap_pcm_dma_data *dma_data = prtd->dma_data;
 	struct omap_dma_channel_params dma_params;
+	int bytes;
 
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
@@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		return 0;
 
 	memset(&dma_params, 0, sizeof(dma_params));
-	/*
-	 * Note: Regardless of interface data formats supported by OMAP McBSP
-	 * or EAC blocks, internal representation is always fixed 16-bit/sample
-	 */
-	dma_params.data_type			= OMAP_DMA_DATA_TYPE_S16;
+	dma_params.data_type			= dma_data->data_type;
 	dma_params.trigger			= dma_data->dma_req;
 	dma_params.sync_mode			= dma_data->sync_mode;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		dma_params.src_start		= runtime->dma_addr;
 		dma_params.dst_start		= dma_data->port_addr;
 		dma_params.dst_port		= OMAP_DMA_PORT_MPUI;
+		dma_params.dst_fi		= dma_data->packet_size;
 	} else {
 		dma_params.src_amode		= OMAP_DMA_AMODE_CONSTANT;
 		dma_params.dst_amode		= OMAP_DMA_AMODE_POST_INC;
@@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		dma_params.src_start		= dma_data->port_addr;
 		dma_params.dst_start		= runtime->dma_addr;
 		dma_params.src_port		= OMAP_DMA_PORT_MPUI;
+		dma_params.src_fi		= dma_data->packet_size;
 	}
 	/*
 	 * Set DMA transfer frame size equal to ALSA period size and frame
@@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	 * we can transfer the whole ALSA buffer with single DMA transfer but
 	 * still can get an interrupt at each period bounary
 	 */
-	dma_params.elem_count	= snd_pcm_lib_period_bytes(substream) / 2;
+	bytes = snd_pcm_lib_period_bytes(substream);
+	dma_params.elem_count	= bytes >> dma_data->data_type;
 	dma_params.frame_count	= runtime->periods;
 	omap_set_dma_params(prtd->dma_ch, &dma_params);
 
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 38a821d..b19975d 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,8 +29,10 @@ struct omap_pcm_dma_data {
 	char		*name;		/* stream identifier */
 	int		dma_req;	/* DMA request line */
 	unsigned long	port_addr;	/* transmit/receive register */
-	int		sync_mode;	/* DMA sync mode */
 	void (*set_threshold)(struct snd_pcm_substream *substream);
+	int		data_type;	/* data type 8,16,32 */
+	int		sync_mode;	/* DMA sync mode */
+	int		packet_size;	/* packet size only in PACKET mode */
 };
 
 extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3c36d24..993abb7 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -388,7 +388,7 @@ static void fsi_soft_all_reset(struct fsi_master *master)
 }
 
 /* playback interrupt */
-static int fsi_data_push(struct fsi_priv *fsi)
+static int fsi_data_push(struct fsi_priv *fsi, int startup)
 {
 	struct snd_pcm_runtime *runtime;
 	struct snd_pcm_substream *substream = NULL;
@@ -397,7 +397,7 @@ static int fsi_data_push(struct fsi_priv *fsi)
 	int fifo_free;
 	int width;
 	u8 *start;
-	int i, ret, over_period;
+	int i, over_period;
 
 	if (!fsi			||
 	    !fsi->substream		||
@@ -453,24 +453,26 @@ static int fsi_data_push(struct fsi_priv *fsi)
 
 	fsi->byte_offset += send * width;
 
-	ret = 0;
 	status = fsi_reg_read(fsi, DOFF_ST);
-	if (status & ERR_OVER) {
+	if (!startup) {
 		struct snd_soc_dai *dai = fsi_get_dai(substream);
-		dev_err(dai->dev, "over run error\n");
-		fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR);
-		ret = -EIO;
+
+		if (status & ERR_OVER)
+			dev_err(dai->dev, "over run\n");
+		if (status & ERR_UNDER)
+			dev_err(dai->dev, "under run\n");
 	}
+	fsi_reg_write(fsi, DOFF_ST, 0);
 
 	fsi_irq_enable(fsi, 1);
 
 	if (over_period)
 		snd_pcm_period_elapsed(substream);
 
-	return ret;
+	return 0;
 }
 
-static int fsi_data_pop(struct fsi_priv *fsi)
+static int fsi_data_pop(struct fsi_priv *fsi, int startup)
 {
 	struct snd_pcm_runtime *runtime;
 	struct snd_pcm_substream *substream = NULL;
@@ -479,7 +481,7 @@ static int fsi_data_pop(struct fsi_priv *fsi)
 	int fifo_fill;
 	int width;
 	u8 *start;
-	int i, ret, over_period;
+	int i, over_period;
 
 	if (!fsi			||
 	    !fsi->substream		||
@@ -534,21 +536,23 @@ static int fsi_data_pop(struct fsi_priv *fsi)
 
 	fsi->byte_offset += fifo_fill * width;
 
-	ret = 0;
 	status = fsi_reg_read(fsi, DIFF_ST);
-	if (status & ERR_UNDER) {
+	if (!startup) {
 		struct snd_soc_dai *dai = fsi_get_dai(substream);
-		dev_err(dai->dev, "under run error\n");
-		fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR);
-		ret = -EIO;
+
+		if (status & ERR_OVER)
+			dev_err(dai->dev, "over run\n");
+		if (status & ERR_UNDER)
+			dev_err(dai->dev, "under run\n");
 	}
+	fsi_reg_write(fsi, DIFF_ST, 0);
 
 	fsi_irq_enable(fsi, 0);
 
 	if (over_period)
 		snd_pcm_period_elapsed(substream);
 
-	return ret;
+	return 0;
 }
 
 static irqreturn_t fsi_interrupt(int irq, void *data)
@@ -562,13 +566,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data)
 	fsi_master_write(master, SOFT_RST, status | 0x00000010);
 
 	if (int_st & INT_A_OUT)
-		fsi_data_push(&master->fsia);
+		fsi_data_push(&master->fsia, 0);
 	if (int_st & INT_B_OUT)
-		fsi_data_push(&master->fsib);
+		fsi_data_push(&master->fsib, 0);
 	if (int_st & INT_A_IN)
-		fsi_data_pop(&master->fsia);
+		fsi_data_pop(&master->fsia, 0);
 	if (int_st & INT_B_IN)
-		fsi_data_pop(&master->fsib);
+		fsi_data_pop(&master->fsib, 0);
 
 	fsi_master_write(master, INT_ST, 0x0000000);
 
@@ -726,7 +730,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 		fsi_stream_push(fsi, substream,
 				frames_to_bytes(runtime, runtime->buffer_size),
 				frames_to_bytes(runtime, runtime->period_size));
-		ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi);
+		ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		fsi_irq_disable(fsi, is_play);
diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h
index 9cc04ab..c0bfab8 100644
--- a/sound/soc/sh/siu.h
+++ b/sound/soc/sh/siu.h
@@ -72,7 +72,7 @@ struct siu_firmware {
 #include <linux/interrupt.h>
 #include <linux/io.h>
 
-#include <asm/dma-sh.h>
+#include <asm/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index c5efc30..ba7f8d0 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -32,7 +32,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc-dai.h>
 
-#include <asm/dma-sh.h>
+#include <asm/dmaengine.h>
 #include <asm/siu.h>
 
 #include "siu.h"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4011ad3..06c38d1 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -427,24 +427,24 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	if (!runtime->hw.rates) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
 			codec_dai->name, cpu_dai->name);
-		goto machine_err;
+		goto config_err;
 	}
 	if (!runtime->hw.formats) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
 			codec_dai->name, cpu_dai->name);
-		goto machine_err;
+		goto config_err;
 	}
 	if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
 			codec_dai->name, cpu_dai->name);
-		goto machine_err;
+		goto config_err;
 	}
 
 	/* Symmetry only applies if we've already got an active stream. */
 	if (cpu_dai->active || codec_dai->active) {
 		ret = soc_pcm_apply_symmetry(substream);
 		if (ret != 0)
-			goto machine_err;
+			goto config_err;
 	}
 
 	pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
@@ -467,10 +467,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	mutex_unlock(&pcm_mutex);
 	return 0;
 
-machine_err:
+config_err:
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
 
+machine_err:
+	if (codec_dai->ops->shutdown)
+		codec_dai->ops->shutdown(substream, codec_dai);
+
 codec_dai_err:
 	if (platform->pcm_ops->close)
 		platform->pcm_ops->close(substream);
@@ -1002,6 +1006,12 @@ static int soc_resume(struct device *dev)
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
 
+	/* If the initialization of this soc device failed, there is no codec
+	 * associated with it. Just bail out in this case.
+	 */
+	if (!card->codec)
+		return 0;
+
 	/* AC97 devices might have other drivers hanging off them so
 	 * need to resume immediately.  Other drivers don't have that
 	 * problem and may take a substantial amount of time to resume