Add support for the i.MX53 QSB2.6.35.3-mx53-early-201103141049

This patch seems to have originated from the 11.01.00 release
from Freescale, which is no longer available except through the
gitweb interface from Freescale.

http://opensource.freescale.com/git?p=imx/linux-2.6-imx.git;a=commit;h=27fdf7bae11978d21e8aba09bb635f49b07edd4a

13 files changed, 2634 insertions, 0 deletions

diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
new file mode 100644
index 000000000000..213feb404a6b
--- /dev/null
+++ b/sound/soc/mxs/Kconfig
@@ -0,0 +1,51 @@
+config SND_MXS_SOC
+	tristate "SoC Audio for the MXS chips"
+	depends on ARCH_MXS && SND_SOC
+	select SND_PCM
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the MXS I2S or SSP interface.
+
+
+config SND_MXS_SOC_SPDIF_DAI
+	tristate
+
+config SND_MXS_SOC_EVK_DEVB
+	tristate "SoC Audio support for MXS-EVK SGTL5000"
+	depends on SND_MXS_SOC && ARCH_MX28
+	select SND_SOC_SGTL5000
+	help
+	  Say Y if you want to add support for SoC audio on MXS EVK development
+	  board with the sgtl5000 codec.
+
+config SND_MXS_SOC_EVK_ADC
+	tristate "SoC Audio support for MXS-EVK ADC/DAC"
+	depends on SND_MXS_SOC && ARCH_MX23
+	select SND_SOC_MXS_ADC_CODEC
+	help
+	  Say Y if you want to add support for SoC audio on MXS EVK development
+	  board with ADC/DAC audio interface.
+
+config SND_MXS_SOC_DAI
+	tristate "MXS Digital Audio Interface SAIF"
+	default y
+	depends on SND_MXS_SOC_EVK_DEVB
+	help
+	   Enable MXS Digital Audio Interface SAIF
+
+config SND_MXS_SOC_ADC
+	tristate "MXS ADC/DAC Audio Interface"
+	default y
+	depends on SND_MXS_SOC_EVK_ADC
+	help
+	  Enable MXS ADC/DAC Audio Interface
+
+config SND_MXS_SOC_EVK_DEVB_SPDIF
+	tristate "SoC SPDIF support for MXS EVK Development Board"
+	default n
+	depends on SND_MXS_SOC && ARCH_MXS
+	select SND_MXS_SOC_SPDIF_DAI
+	select SND_SOC_MXS_SPDIF
+	help
+	  Say Y if you want to add support for SoC audio on MXS EVK development
+	  board with the SPDIF transmitter.
diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile
new file mode 100644
index 000000000000..971c622a34fb
--- /dev/null
+++ b/sound/soc/mxs/Makefile
@@ -0,0 +1,16 @@
+# MXS platfrom support
+snd-soc-mxs-objs := mxs-pcm.o
+snd-soc-mxs-dai-objs := mxs-dai.o
+snd-soc-mxs-spdif-dai-objs := mxs-spdif-dai.o
+snd-soc-mxs-devb-objs := mxs-devb.o
+snd-soc-mxs-devb-spdif-objs := mxs-devb-spdif.o
+snd-soc-mxs-adc-objs := mxs-adc.o
+snd-soc-mxs-evk-adc-objs := mxs-evk-adc.o
+
+obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o
+obj-$(CONFIG_SND_MXS_SOC_DAI) += snd-soc-mxs-dai.o
+obj-$(CONFIG_SND_MXS_SOC_SPDIF_DAI) += snd-soc-mxs-spdif-dai.o
+obj-$(CONFIG_SND_MXS_SOC_EVK_DEVB) += snd-soc-mxs-devb.o
+obj-$(CONFIG_SND_MXS_SOC_EVK_DEVB_SPDIF) += snd-soc-mxs-devb-spdif.o
+obj-$(CONFIG_SND_MXS_SOC_ADC) += snd-soc-mxs-adc.o
+obj-$(CONFIG_SND_MXS_SOC_EVK_ADC) += snd-soc-mxs-evk-adc.o
diff --git a/sound/soc/mxs/mxs-adc.c b/sound/soc/mxs/mxs-adc.c
new file mode 100644
index 000000000000..6733652f76bc
--- /dev/null
+++ b/sound/soc/mxs/mxs-adc.c
@@ -0,0 +1,477 @@
+/*
+ * ASoC Audio Layer for Freescale MXS ADC/DAC
+ *
+ * Author: Vladislav Buzov <vbuzov@embeddedalley.com>
+ *
+ * Copyright 2008-2010 Freescale Semiconductor, Inc.
+ * Copyright 2008 Embedded Alley Solutions, Inc All Rights Reserved.
+ */
+
+/*
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/dma.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/regs-audioin.h>
+#include <mach/regs-audioout.h>
+#include <mach/dmaengine.h>
+
+#include "mxs-pcm.h"
+
+#define MXS_ADC_RATES	SNDRV_PCM_RATE_8000_192000
+#define MXS_ADC_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_S32_LE)
+#define ADC_VOLUME_MIN  0x37
+
+struct mxs_pcm_dma_params mxs_audio_in = {
+	.name = "mxs-audio-in",
+	.dma_ch	= MXS_DMA_CHANNEL_AHB_APBX_AUDIOADC,
+	.irq = IRQ_ADC_DMA,
+};
+
+struct mxs_pcm_dma_params mxs_audio_out = {
+	.name = "mxs-audio-out",
+	.dma_ch	= MXS_DMA_CHANNEL_AHB_APBX_AUDIODAC,
+	.irq = IRQ_DAC_DMA,
+};
+
+static struct delayed_work work;
+static struct delayed_work adc_ramp_work;
+static struct delayed_work dac_ramp_work;
+static bool adc_ramp_done = 1;
+static bool dac_ramp_done = 1;
+
+static void mxs_adc_schedule_work(struct delayed_work *work)
+{
+	schedule_delayed_work(work, HZ / 10);
+}
+static void mxs_adc_work(struct work_struct *work)
+{
+	/* disable irq */
+	disable_irq(IRQ_HEADPHONE_SHORT);
+
+	while (true) {
+		__raw_writel(BM_AUDIOOUT_PWRDN_HEADPHONE,
+		      REGS_AUDIOOUT_BASE + HW_AUDIOOUT_PWRDN_CLR);
+		msleep(10);
+		if ((__raw_readl(REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL)
+			& BM_AUDIOOUT_ANACTRL_SHORT_LR_STS) != 0) {
+			/* rearm the short protection */
+			__raw_writel(BM_AUDIOOUT_ANACTRL_SHORTMODE_LR,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_CLR);
+			__raw_writel(BM_AUDIOOUT_ANACTRL_SHORT_LR_STS,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_CLR);
+			__raw_writel(BF_AUDIOOUT_ANACTRL_SHORTMODE_LR(0x1),
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_SET);
+
+			__raw_writel(BM_AUDIOOUT_PWRDN_HEADPHONE,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_PWRDN_SET);
+			printk(KERN_WARNING "WARNING : Headphone LR short!\r\n");
+		} else {
+			printk(KERN_WARNING "INFO : Headphone LR no longer short!\r\n");
+			break;
+		}
+		msleep(1000);
+	}
+
+	/* power up the HEADPHONE and un-mute the HPVOL */
+	__raw_writel(BM_AUDIOOUT_HPVOL_MUTE,
+	      REGS_AUDIOOUT_BASE + HW_AUDIOOUT_HPVOL_CLR);
+	__raw_writel(BM_AUDIOOUT_PWRDN_HEADPHONE,
+		      REGS_AUDIOOUT_BASE + HW_AUDIOOUT_PWRDN_CLR);
+
+	/* enable irq for next short detect*/
+	enable_irq(IRQ_HEADPHONE_SHORT);
+}
+
+static void mxs_adc_schedule_ramp_work(struct delayed_work *work)
+{
+	schedule_delayed_work(work, msecs_to_jiffies(2));
+	adc_ramp_done = 0;
+}
+
+static void mxs_adc_ramp_work(struct work_struct *work)
+{
+	u32 reg = 0;
+	u32 reg1 = 0;
+	u32 reg2 = 0;
+	u32 l, r;
+	u32 ll, rr;
+	int i;
+
+	reg = __raw_readl(REGS_AUDIOIN_BASE + \
+		HW_AUDIOIN_ADCVOLUME);
+
+	reg1 = reg & ~BM_AUDIOIN_ADCVOLUME_VOLUME_LEFT;
+	reg1 = reg1 & ~BM_AUDIOIN_ADCVOLUME_VOLUME_RIGHT;
+	/* minimize adc volume */
+	reg2 = reg1 |
+	    BF_AUDIOIN_ADCVOLUME_VOLUME_LEFT(ADC_VOLUME_MIN) |
+	    BF_AUDIOIN_ADCVOLUME_VOLUME_RIGHT(ADC_VOLUME_MIN);
+	__raw_writel(reg2,
+		REGS_AUDIOIN_BASE + HW_AUDIOIN_ADCVOLUME);
+	msleep(1);
+
+	l = (reg & BM_AUDIOIN_ADCVOLUME_VOLUME_LEFT) >>
+		BP_AUDIOIN_ADCVOLUME_VOLUME_LEFT;
+	r = (reg & BM_AUDIOIN_ADCVOLUME_VOLUME_RIGHT) >>
+		BP_AUDIOIN_ADCVOLUME_VOLUME_RIGHT;
+
+	/* fade in adc vol */
+	for (i = ADC_VOLUME_MIN; (i < l) || (i < r);) {
+		i += 0x8;
+		ll = i < l ? i : l;
+		rr = i < r ? i : r;
+		reg2 = reg1 |
+		    BF_AUDIOIN_ADCVOLUME_VOLUME_LEFT(ll) |
+		    BF_AUDIOIN_ADCVOLUME_VOLUME_RIGHT(rr);
+		__raw_writel(reg2,
+		    REGS_AUDIOIN_BASE + HW_AUDIOIN_ADCVOLUME);
+		msleep(1);
+	}
+	adc_ramp_done = 1;
+}
+
+static void mxs_dac_schedule_ramp_work(struct delayed_work *work)
+{
+	schedule_delayed_work(work, msecs_to_jiffies(2));
+	dac_ramp_done = 0;
+}
+
+static void mxs_dac_ramp_work(struct work_struct *work)
+{
+	u32 reg = 0;
+	u32 reg1 = 0;
+	u32 l, r;
+	u32 ll, rr;
+	int i;
+
+	/* unmute hp and speaker */
+	__raw_writel(BM_AUDIOOUT_HPVOL_MUTE,
+		REGS_AUDIOOUT_BASE + HW_AUDIOOUT_HPVOL_CLR);
+	__raw_writel(BM_AUDIOOUT_SPEAKERCTRL_MUTE,
+		REGS_AUDIOOUT_BASE + HW_AUDIOOUT_SPEAKERCTRL_CLR);
+
+	reg = __raw_readl(REGS_AUDIOOUT_BASE + \
+			HW_AUDIOOUT_HPVOL);
+
+	reg1 = reg & ~BM_AUDIOOUT_HPVOL_VOL_LEFT;
+	reg1 = reg1 & ~BM_AUDIOOUT_HPVOL_VOL_RIGHT;
+
+	l = (reg & BM_AUDIOOUT_HPVOL_VOL_LEFT) >>
+		BP_AUDIOOUT_HPVOL_VOL_LEFT;
+	r = (reg & BM_AUDIOOUT_HPVOL_VOL_RIGHT) >>
+		BP_AUDIOOUT_HPVOL_VOL_RIGHT;
+	/* fade in hp vol */
+	for (i = 0x7f; i > 0 ;) {
+		i -= 0x8;
+		ll = i > (int)l ? i : l;
+		rr = i > (int)r ? i : r;
+		reg = reg1 | BF_AUDIOOUT_HPVOL_VOL_LEFT(ll)
+			| BF_AUDIOOUT_HPVOL_VOL_RIGHT(rr);
+		__raw_writel(reg,
+			REGS_AUDIOOUT_BASE + HW_AUDIOOUT_HPVOL);
+		msleep(1);
+	}
+	dac_ramp_done = 1;
+}
+
+static irqreturn_t mxs_short_irq(int irq, void *dev_id)
+{
+	__raw_writel(BM_AUDIOOUT_ANACTRL_SHORTMODE_LR,
+		REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_CLR);
+	__raw_writel(BM_AUDIOOUT_ANACTRL_SHORT_LR_STS,
+		REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_CLR);
+	__raw_writel(BF_AUDIOOUT_ANACTRL_SHORTMODE_LR(0x1),
+		REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_SET);
+
+	__raw_writel(BM_AUDIOOUT_HPVOL_MUTE,
+	      REGS_AUDIOOUT_BASE + HW_AUDIOOUT_HPVOL_SET);
+	__raw_writel(BM_AUDIOOUT_PWRDN_HEADPHONE,
+		      REGS_AUDIOOUT_BASE + HW_AUDIOOUT_PWRDN_SET);
+	__raw_writel(BM_AUDIOOUT_ANACTRL_HP_CLASSAB,
+		REGS_AUDIOOUT_BASE + HW_AUDIOOUT_ANACTRL_SET);
+
+	mxs_adc_schedule_work(&work);
+	return IRQ_HANDLED;
+}
+static irqreturn_t mxs_err_irq(int irq, void *dev_id)
+{
+	struct snd_pcm_substream *substream = dev_id;
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	u32 ctrl_reg;
+	u32 overflow_mask;
+	u32 underflow_mask;
+
+	if (playback) {
+		ctrl_reg = __raw_readl(REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL);
+		underflow_mask = BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ;
+		overflow_mask = BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ;
+	} else {
+		ctrl_reg = __raw_readl(REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL);
+		underflow_mask = BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ;
+		overflow_mask = BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ;
+	}
+
+	if (ctrl_reg & underflow_mask) {
+		printk(KERN_DEBUG "%s underflow detected\n",
+		       playback ? "DAC" : "ADC");
+
+		if (playback)
+			__raw_writel(
+				BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
+		else
+			__raw_writel(
+				BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ,
+				REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
+
+	} else if (ctrl_reg & overflow_mask) {
+		printk(KERN_DEBUG "%s overflow detected\n",
+		       playback ? "DAC" : "ADC");
+
+		if (playback)
+			__raw_writel(
+				BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
+		else
+			__raw_writel(BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ,
+				REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
+	} else
+		printk(KERN_WARNING "Unknown DAC error interrupt\n");
+
+	return IRQ_HANDLED;
+}
+
+static int mxs_adc_trigger(struct snd_pcm_substream *substream,
+				int cmd,
+				struct snd_soc_dai *dai)
+{
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	struct mxs_runtime_data *prtd = substream->runtime->private_data;
+	int ret = 0;
+	u32 xfer_count1 = 0;
+	u32 xfer_count2 = 0;
+	u32 cur_bar1 = 0;
+	u32 cur_bar2 = 0;
+	u32 reg;
+	struct mxs_dma_info dma_info;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+
+		if (playback) {
+			/* enable the fifo error interrupt */
+		    __raw_writel(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN,
+			REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_SET);
+			/* write a data to data reg to trigger the transfer */
+		    __raw_writel(0x0,
+			REGS_AUDIOOUT_BASE + HW_AUDIOOUT_DATA);
+		    mxs_dac_schedule_ramp_work(&dac_ramp_work);
+		} else {
+		    mxs_dma_get_info(prtd->dma_ch, &dma_info);
+		    cur_bar1 = dma_info.buf_addr;
+		    xfer_count1 = dma_info.xfer_count;
+
+		    __raw_writel(BM_AUDIOIN_CTRL_RUN,
+			REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_SET);
+		    udelay(100);
+
+		    mxs_dma_get_info(prtd->dma_ch, &dma_info);
+		    cur_bar2 = dma_info.buf_addr;
+		    xfer_count2 = dma_info.xfer_count;
+
+		    /* check if DMA getting stuck */
+		    if ((xfer_count1 == xfer_count2) && (cur_bar1 == cur_bar2))
+			/* read a data from data reg to trigger the receive */
+			reg = __raw_readl(REGS_AUDIOIN_BASE + HW_AUDIOIN_DATA);
+
+		    mxs_adc_schedule_ramp_work(&adc_ramp_work);
+		}
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+
+		if (playback) {
+			if (dac_ramp_done == 0) {
+				cancel_delayed_work(&dac_ramp_work);
+				dac_ramp_done = 1;
+			}
+			__raw_writel(BM_AUDIOOUT_HPVOL_MUTE,
+			  REGS_AUDIOOUT_BASE + HW_AUDIOOUT_HPVOL_SET);
+			__raw_writel(BM_AUDIOOUT_SPEAKERCTRL_MUTE,
+			  REGS_AUDIOOUT_BASE + HW_AUDIOOUT_SPEAKERCTRL_SET);
+			/* disable the fifo error interrupt */
+			__raw_writel(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
+			mdelay(50);
+		} else {
+			if (adc_ramp_done == 0) {
+				cancel_delayed_work(&adc_ramp_work);
+				adc_ramp_done = 1;
+			}
+			__raw_writel(BM_AUDIOIN_CTRL_RUN,
+				REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
+		}
+		break;
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int mxs_adc_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 playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	int irq;
+	int irq_short;
+	int ret;
+
+	INIT_DELAYED_WORK(&work, mxs_adc_work);
+	INIT_DELAYED_WORK(&adc_ramp_work, mxs_adc_ramp_work);
+	INIT_DELAYED_WORK(&dac_ramp_work, mxs_dac_ramp_work);
+
+	if (playback) {
+		irq = IRQ_DAC_ERROR;
+		snd_soc_dai_set_dma_data(dai, substream, &mxs_audio_out);
+	} else {
+		irq = IRQ_ADC_ERROR;
+		snd_soc_dai_set_dma_data(dai, substream, &mxs_audio_in);
+	}
+
+	ret = request_irq(irq, mxs_err_irq, 0, "MXS DAC and ADC Error",
+			  substream);
+	if (ret) {
+		printk(KERN_ERR "%s: Unable to request ADC/DAC error irq %d\n",
+		       __func__, IRQ_DAC_ERROR);
+		return ret;
+	}
+
+	irq_short = IRQ_HEADPHONE_SHORT;
+	ret = request_irq(irq_short, mxs_short_irq,
+		IRQF_DISABLED | IRQF_SHARED, "MXS DAC and ADC HP SHORT", substream);
+	if (ret) {
+		printk(KERN_ERR "%s: Unable to request ADC/DAC HP SHORT irq %d\n",
+		       __func__, IRQ_DAC_ERROR);
+		return ret;
+	}
+
+	/* Enable error interrupt */
+	if (playback) {
+		__raw_writel(BM_AUDIOOUT_CTRL_FIFO_OVERFLOW_IRQ,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
+		__raw_writel(BM_AUDIOOUT_CTRL_FIFO_UNDERFLOW_IRQ,
+				REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
+	} else {
+		__raw_writel(BM_AUDIOIN_CTRL_FIFO_OVERFLOW_IRQ,
+			REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
+		__raw_writel(BM_AUDIOIN_CTRL_FIFO_UNDERFLOW_IRQ,
+			REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
+		__raw_writel(BM_AUDIOIN_CTRL_FIFO_ERROR_IRQ_EN,
+			REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_SET);
+	}
+
+	return 0;
+}
+
+static void mxs_adc_shutdown(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+
+	/* Disable error interrupt */
+	if (playback) {
+		__raw_writel(BM_AUDIOOUT_CTRL_FIFO_ERROR_IRQ_EN,
+			REGS_AUDIOOUT_BASE + HW_AUDIOOUT_CTRL_CLR);
+		free_irq(IRQ_DAC_ERROR, substream);
+	} else {
+		__raw_writel(BM_AUDIOIN_CTRL_FIFO_ERROR_IRQ_EN,
+			REGS_AUDIOIN_BASE + HW_AUDIOIN_CTRL_CLR);
+		free_irq(IRQ_ADC_ERROR, substream);
+	}
+}
+
+#ifdef CONFIG_PM
+static int mxs_adc_suspend(struct snd_soc_dai *cpu_dai)
+{
+	return 0;
+}
+
+static int mxs_adc_resume(struct snd_soc_dai *cpu_dai)
+{
+	return 0;
+}
+#else
+#define mxs_adc_suspend	NULL
+#define mxs_adc_resume	NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_dai_ops mxs_adc_dai_ops = {
+	.startup = mxs_adc_startup,
+	.shutdown = mxs_adc_shutdown,
+	.trigger = mxs_adc_trigger,
+};
+
+struct snd_soc_dai mxs_adc_dai = {
+	.name = "mxs adc/dac",
+	.id = 0,
+	.suspend = mxs_adc_suspend,
+	.resume = mxs_adc_resume,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_ADC_RATES,
+		.formats = MXS_ADC_FORMATS,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_ADC_RATES,
+		.formats = MXS_ADC_FORMATS,
+	},
+	.ops = &mxs_adc_dai_ops,
+};
+EXPORT_SYMBOL_GPL(mxs_adc_dai);
+
+static int __init mxs_adc_dai_init(void)
+{
+	return snd_soc_register_dai(&mxs_adc_dai);
+}
+
+static void __exit mxs_adc_dai_exit(void)
+{
+	snd_soc_unregister_dai(&mxs_adc_dai);
+}
+module_init(mxs_adc_dai_init);
+module_exit(mxs_adc_dai_exit);
+
+MODULE_AUTHOR("Vladislav Buzov");
+MODULE_DESCRIPTION("MXS ADC/DAC DAI");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-adc.h b/sound/soc/mxs/mxs-adc.h
new file mode 100644
index 000000000000..b922c14cb894
--- /dev/null
+++ b/sound/soc/mxs/mxs-adc.h
@@ -0,0 +1,21 @@
+/*
+ * ASoC Audio Layer for Freescale MXS ADC/DAC
+ *
+ * Author: Vladislav Buzov <vbuzov@embeddedalley.com>
+ *
+ * Copyright 2008-2010 Freescale Semiconductor, Inc.
+ * Copyright 2008 Embedded Alley Solutions, Inc All Rights Reserved.
+ */
+
+/*
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+#ifndef _MXS_ADC_H
+#define _MXS_ADC_H
+extern struct snd_soc_dai mxs_adc_dai;
+#endif
diff --git a/sound/soc/mxs/mxs-dai.c b/sound/soc/mxs/mxs-dai.c
new file mode 100644
index 000000000000..e8c4642e660c
--- /dev/null
+++ b/sound/soc/mxs/mxs-dai.c
@@ -0,0 +1,689 @@
+/*
+ * Copyright (C) 2010 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/dma.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+
+#include <mach/mx28.h>
+
+#include "mxs-pcm.h"
+#include "mxs-dai.h"
+
+#define SAIF0_CTRL (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_CTRL)
+#define SAIF1_CTRL (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_CTRL)
+#define SAIF0_CTRL_SET (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_CTRL_SET)
+#define SAIF1_CTRL_SET (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_CTRL_SET)
+#define SAIF0_CTRL_CLR (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_CTRL_CLR)
+#define SAIF1_CTRL_CLR (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_CTRL_CLR)
+
+#define SAIF0_STAT (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_STAT)
+#define SAIF1_STAT (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_STAT)
+#define SAIF0_STAT_SET (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_STAT_SET)
+#define SAIF1_STAT_SET (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_STAT_SET)
+#define SAIF0_STAT_CLR (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_STAT_CLR)
+#define SAIF1_STAT_CLR (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_STAT_CLR)
+
+#define SAIF0_DATA (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_DATA)
+#define SAIF1_DATA (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_DATA)
+#define SAIF0_DATA_SET (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_DATA_SET)
+#define SAIF1_DATA_SET (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_DATA_SET)
+#define SAIF0_DATA_CLR (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_DATA_CLR)
+#define SAIF1_DATA_CLR (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_DATA_CLR)
+
+#define SAIF0_VERSION (IO_ADDRESS(SAIF0_PHYS_ADDR) + HW_SAIF_VERSION)
+#define SAIF1_VERSION (IO_ADDRESS(SAIF1_PHYS_ADDR) + HW_SAIF_VERSION)
+
+#define HW_SAIF_CTRL	(0x00000000)
+#define HW_SAIF_CTRL_SET	(0x00000004)
+#define HW_SAIF_CTRL_CLR	(0x00000008)
+#define HW_SAIF_CTRL_TOG	(0x0000000c)
+
+#define BM_SAIF_CTRL_SFTRST	0x80000000
+#define BM_SAIF_CTRL_CLKGATE	0x40000000
+#define BP_SAIF_CTRL_BITCLK_MULT_RATE	27
+#define BM_SAIF_CTRL_BITCLK_MULT_RATE	0x38000000
+#define BF_SAIF_CTRL_BITCLK_MULT_RATE(v)  \
+		(((v) << 27) & BM_SAIF_CTRL_BITCLK_MULT_RATE)
+#define BM_SAIF_CTRL_BITCLK_BASE_RATE	0x04000000
+#define BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN	0x02000000
+#define BM_SAIF_CTRL_FIFO_SERVICE_IRQ_EN	0x01000000
+#define BP_SAIF_CTRL_RSRVD2	21
+#define BM_SAIF_CTRL_RSRVD2	0x00E00000
+#define BF_SAIF_CTRL_RSRVD2(v)  \
+		(((v) << 21) & BM_SAIF_CTRL_RSRVD2)
+#define BP_SAIF_CTRL_DMAWAIT_COUNT	16
+#define BM_SAIF_CTRL_DMAWAIT_COUNT	0x001F0000
+#define BF_SAIF_CTRL_DMAWAIT_COUNT(v)  \
+		(((v) << 16) & BM_SAIF_CTRL_DMAWAIT_COUNT)
+#define BP_SAIF_CTRL_CHANNEL_NUM_SELECT	14
+#define BM_SAIF_CTRL_CHANNEL_NUM_SELECT	0x0000C000
+#define BF_SAIF_CTRL_CHANNEL_NUM_SELECT(v)  \
+		(((v) << 14) & BM_SAIF_CTRL_CHANNEL_NUM_SELECT)
+#define BM_SAIF_CTRL_LRCLK_PULSE	0x00002000
+#define BM_SAIF_CTRL_BIT_ORDER	0x00001000
+#define BM_SAIF_CTRL_DELAY	0x00000800
+#define BM_SAIF_CTRL_JUSTIFY	0x00000400
+#define BM_SAIF_CTRL_LRCLK_POLARITY	0x00000200
+#define BM_SAIF_CTRL_BITCLK_EDGE	0x00000100
+#define BP_SAIF_CTRL_WORD_LENGTH	4
+#define BM_SAIF_CTRL_WORD_LENGTH	0x000000F0
+#define BF_SAIF_CTRL_WORD_LENGTH(v)  \
+		(((v) << 4) & BM_SAIF_CTRL_WORD_LENGTH)
+#define BM_SAIF_CTRL_BITCLK_48XFS_ENABLE	0x00000008
+#define BM_SAIF_CTRL_SLAVE_MODE	0x00000004
+#define BM_SAIF_CTRL_READ_MODE	0x00000002
+#define BM_SAIF_CTRL_RUN	0x00000001
+
+#define HW_SAIF_STAT	(0x00000010)
+#define HW_SAIF_STAT_SET	(0x00000014)
+#define HW_SAIF_STAT_CLR	(0x00000018)
+#define HW_SAIF_STAT_TOG	(0x0000001c)
+
+#define BM_SAIF_STAT_PRESENT	0x80000000
+#define BP_SAIF_STAT_RSRVD2	17
+#define BM_SAIF_STAT_RSRVD2	0x7FFE0000
+#define BF_SAIF_STAT_RSRVD2(v)  \
+		(((v) << 17) & BM_SAIF_STAT_RSRVD2)
+#define BM_SAIF_STAT_DMA_PREQ	0x00010000
+#define BP_SAIF_STAT_RSRVD1	7
+#define BM_SAIF_STAT_RSRVD1	0x0000FF80
+#define BF_SAIF_STAT_RSRVD1(v)  \
+		(((v) << 7) & BM_SAIF_STAT_RSRVD1)
+#define BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ	0x00000040
+#define BM_SAIF_STAT_FIFO_OVERFLOW_IRQ	0x00000020
+#define BM_SAIF_STAT_FIFO_SERVICE_IRQ	0x00000010
+#define BP_SAIF_STAT_RSRVD0	1
+#define BM_SAIF_STAT_RSRVD0	0x0000000E
+#define BF_SAIF_STAT_RSRVD0(v)  \
+		(((v) << 1) & BM_SAIF_STAT_RSRVD0)
+#define BM_SAIF_STAT_BUSY	0x00000001
+
+#define HW_SAIF_DATA	(0x00000020)
+#define HW_SAIF_DATA_SET	(0x00000024)
+#define HW_SAIF_DATA_CLR	(0x00000028)
+#define HW_SAIF_DATA_TOG	(0x0000002c)
+
+#define BP_SAIF_DATA_PCM_RIGHT	16
+#define BM_SAIF_DATA_PCM_RIGHT	0xFFFF0000
+#define BF_SAIF_DATA_PCM_RIGHT(v) \
+		(((v) << 16) & BM_SAIF_DATA_PCM_RIGHT)
+#define BP_SAIF_DATA_PCM_LEFT	0
+#define BM_SAIF_DATA_PCM_LEFT	0x0000FFFF
+#define BF_SAIF_DATA_PCM_LEFT(v)  \
+		(((v) << 0) & BM_SAIF_DATA_PCM_LEFT)
+
+#define HW_SAIF_VERSION	(0x00000030)
+
+#define BP_SAIF_VERSION_MAJOR	24
+#define BM_SAIF_VERSION_MAJOR	0xFF000000
+#define BF_SAIF_VERSION_MAJOR(v) \
+		(((v) << 24) & BM_SAIF_VERSION_MAJOR)
+#define BP_SAIF_VERSION_MINOR	16
+#define BM_SAIF_VERSION_MINOR	0x00FF0000
+#define BF_SAIF_VERSION_MINOR(v)  \
+		(((v) << 16) & BM_SAIF_VERSION_MINOR)
+#define BP_SAIF_VERSION_STEP	0
+#define BM_SAIF_VERSION_STEP	0x0000FFFF
+#define BF_SAIF_VERSION_STEP(v)  \
+		(((v) << 0) & BM_SAIF_VERSION_STEP)
+/* debug */
+#define MXS_SAIF_DEBUG 0
+#if MXS_SAIF_DEBUG
+#define dbg(format, arg...) printk(format, ## arg)
+#else
+#define dbg(format, arg...)
+#endif
+
+#define MXS_SAIF_DUMP 0
+#if MXS_SAIF_DUMP
+#define SAIF_DUMP() \
+	do { \
+		printk(KERN_INFO "dump @ %s\n", __func__);\
+		printk(KERN_INFO "scr %x\t, %x\n", \
+			__raw_readl(SAIF0_CTRL), __raw_readl(SAIF1_CTRL));\
+		printk(KERN_INFO "stat %x\t, %x\n", \
+			__raw_readl(SAIF0_STAT), __raw_readl(SAIF1_STAT));\
+		printk(KERN_INFO "data %x\t, %x\n", \
+			__raw_readl(SAIF0_DATA), __raw_readl(SAIF1_DATA));\
+		printk(KERN_INFO "version %x\t, %x\n", \
+			__raw_readl(SAIF0_VERSION), __raw_readl(SAIF1_VERSION));
+	} while (0);
+#else
+#define SAIF_DUMP()
+#endif
+
+#define SAIF0_PORT	0
+#define SAIF1_PORT	1
+
+#define MXS_DAI_SAIF0 0
+#define MXS_DAI_SAIF1 1
+
+static struct mxs_saif mxs_saif_en;
+
+static int saif_active[2] = { 0, 0 };
+
+struct mxs_pcm_dma_params mxs_saif_0 = {
+	.name = "mxs-saif-0",
+	.dma_ch	= MXS_DMA_CHANNEL_AHB_APBX_SAIF0,
+	.irq = IRQ_SAIF0_DMA,
+};
+
+struct mxs_pcm_dma_params mxs_saif_1 = {
+	.name = "mxs-saif-1",
+	.dma_ch	= MXS_DMA_CHANNEL_AHB_APBX_SAIF1,
+	.irq = IRQ_SAIF1_DMA,
+};
+
+/*
+* SAIF system clock configuration.
+* Should only be called when port is inactive.
+*/
+static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct clk *saif_clk;
+	u32 scr;
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+
+	switch (clk_id) {
+	case IMX_SSP_SYS_CLK:
+		saif_clk = saif_select->saif_mclk;
+		if (IS_ERR(saif_clk)) {
+			pr_err("%s:failed to get sys_clk\n", __func__);
+			return -EINVAL;
+		}
+		clk_set_rate(saif_clk, freq);
+		clk_enable(saif_clk);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+/*
+ * SAIF Clock dividers
+ * Should only be called when port is inactive.
+ */
+static int mxs_saif_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+				  int div_id, int div)
+{
+	u32 scr;
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+
+	if (saif_select->saif_clk == SAIF0)
+		scr = __raw_readl(SAIF0_CTRL);
+	else
+		scr = __raw_readl(SAIF1_CTRL);
+
+	scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE;
+	scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+
+	switch (div_id) {
+	case IMX_SSP_SYS_MCLK:
+		switch (div) {
+		case 32:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4);
+			scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 64:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+			scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 128:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+			scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 256:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+			scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 512:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+			scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 48:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+			scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 96:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+			scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 192:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+			scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		case 384:
+			scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+			scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (saif_select->saif_clk == SAIF0)
+		__raw_writel(scr, SAIF0_CTRL);
+	else
+		__raw_writel(scr, SAIF1_CTRL);
+
+	return 0;
+}
+
+/*
+ * SAIF DAI format configuration.
+ * Should only be called when port is inactive.
+ * Note: We don't use the I2S modes but instead manually configure the
+ * SAIF for I2S.
+ */
+static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	u32 scr, stat;
+	u32 scr0, scr1;
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+
+	stat = (__raw_readl(SAIF0_STAT)) | (__raw_readl(SAIF1_STAT));
+	if (stat & BM_SAIF_STAT_BUSY)
+		return 0;
+
+	scr0 = __raw_readl(SAIF0_CTRL);
+	scr1 = __raw_readl(SAIF1_CTRL);
+	scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \
+		& ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY;
+	scr1 = scr1 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \
+		& ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY;
+	scr = 0;
+
+	/* DAI mode */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* data frame low 1clk before data */
+		scr |= BM_SAIF_CTRL_DELAY;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* data frame high with data */
+		scr &= ~BM_SAIF_CTRL_DELAY;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		scr &= ~BM_SAIF_CTRL_JUSTIFY;
+		break;
+	}
+
+	/* DAI clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_IF:
+		scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+		scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+		scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	}
+
+	/* DAI clock master masks */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		if (saif_select->saif_clk == SAIF0) {
+			scr &= ~BM_SAIF_CTRL_SLAVE_MODE;
+			__raw_writel(scr | scr0, SAIF0_CTRL);
+			scr |= BM_SAIF_CTRL_SLAVE_MODE;
+			__raw_writel(scr | scr1, SAIF1_CTRL);
+		} else {
+			scr &= ~BM_SAIF_CTRL_SLAVE_MODE;
+			__raw_writel(scr | scr1, SAIF1_CTRL);
+			scr |= BM_SAIF_CTRL_SLAVE_MODE;
+			__raw_writel(scr | scr0, SAIF0_CTRL);
+		}
+		break;
+	}
+
+	SAIF_DUMP();
+	return 0;
+}
+
+static int mxs_saif_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *cpu_dai)
+{
+	/* we cant really change any saif values after saif is enabled*/
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE)))
+		snd_soc_dai_set_dma_data(cpu_dai, substream, &mxs_saif_0);
+	else
+		snd_soc_dai_set_dma_data(cpu_dai, substream, &mxs_saif_1);
+
+	if (cpu_dai->playback.active && cpu_dai->capture.active)
+		return 0;
+
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE)))
+		if (saif_active[SAIF0_PORT]++)
+			return 0;
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_CAPTURE)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)))
+		if (saif_active[SAIF1_PORT]++)
+			return 0;
+	SAIF_DUMP();
+	return 0;
+}
+
+/*
+ * Should only be called when port is inactive.
+ * although can be called multiple times by upper layers.
+ */
+static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *cpu_dai)
+{
+	u32 scr, stat;
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE))) {
+		scr = __raw_readl(SAIF0_CTRL);
+		stat = __raw_readl(SAIF0_STAT);
+	} else {
+		scr = __raw_readl(SAIF1_CTRL);
+		stat = __raw_readl(SAIF1_STAT);
+	}
+	/* cant change any parameters when SAIF is running */
+	/* DAI data (word) size */
+	scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
+	scr &= ~BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		scr |= BF_SAIF_CTRL_WORD_LENGTH(0);
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		scr |= BF_SAIF_CTRL_WORD_LENGTH(4);
+		scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		scr |= BF_SAIF_CTRL_WORD_LENGTH(8);
+		scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+		break;
+	}
+	/* Tx/Rx config */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* enable TX mode */
+		scr &= ~BM_SAIF_CTRL_READ_MODE;
+	} else {
+		/* enable RX mode */
+		scr |= BM_SAIF_CTRL_READ_MODE;
+	}
+
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE)))
+		__raw_writel(scr, SAIF0_CTRL);
+	else
+		__raw_writel(scr, SAIF1_CTRL);
+	return 0;
+}
+
+static int mxs_saif_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE)))
+		__raw_writel(BM_SAIF_CTRL_CLKGATE, SAIF0_CTRL_CLR);
+	else
+		__raw_writel(BM_SAIF_CTRL_CLKGATE, SAIF1_CTRL_CLR);
+	SAIF_DUMP();
+	return 0;
+}
+
+static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *cpu_dai)
+{
+	void __iomem *reg;
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+
+	 if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE)))
+			reg = (void __iomem *)SAIF0_DATA;
+		else
+			reg = (void __iomem *)SAIF1_DATA;
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			/*write a data to saif data register to trigger
+				the transfer*/
+			__raw_writel(0, reg);
+		else
+			/*read a data from saif data register to trigger
+				the receive*/
+			__raw_readl(reg);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		break;
+	default:
+		return -EINVAL;
+	}
+	SAIF_DUMP();
+	return 0;
+}
+
+static void mxs_saif_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif_select = (struct mxs_saif *)cpu_dai->private_data;
+	/* shutdown SAIF if neither Tx or Rx is active */
+	if (cpu_dai->playback.active || cpu_dai->capture.active)
+		return;
+
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_CAPTURE)))
+		if (--saif_active[SAIF0_PORT] > 1)
+			return;
+
+	if (((saif_select->stream_mapping == PLAYBACK_SAIF0_CAPTURE_SAIF1) && \
+		(substream->stream == SNDRV_PCM_STREAM_CAPTURE)) || \
+		((saif_select->stream_mapping == PLAYBACK_SAIF1_CAPTURE_SAIF0) \
+		&& (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)))
+		if (--saif_active[SAIF1_PORT])
+			return;
+}
+
+#ifdef CONFIG_PM
+static int mxs_saif_suspend(struct snd_soc_dai *dai)
+{
+	if (!dai->active)
+		return 0;
+	/* do we need to disable any clocks? */
+	return 0;
+}
+
+static int mxs_saif_resume(struct snd_soc_dai *dai)
+{
+	if (!dai->active)
+		return 0;
+	/* do we need to enable any clocks? */
+	return 0;
+}
+#else
+#define mxs_saif_suspend	NULL
+#define mxs_saif_resume	NULL
+#endif
+
+static int fifo_err_counter;
+
+static irqreturn_t saif0_irq(int irq, void *dev_id)
+{
+	if (fifo_err_counter++ % 100 == 0)
+		printk(KERN_ERR "saif0_irq SAIF_STAT %x SAIF_CTRL %x fifo_errs=\
+			%d\n",
+		       __raw_readl(SAIF0_STAT),
+		       __raw_readl(SAIF0_CTRL),
+		       fifo_err_counter);
+	__raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ | \
+			BM_SAIF_STAT_FIFO_OVERFLOW_IRQ, SAIF0_STAT_CLR);
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t saif1_irq(int irq, void *dev_id)
+{
+	if (fifo_err_counter++ % 100 == 0)
+		printk(KERN_ERR "saif1_irq SAIF_STAT %x SAIF_CTRL %x \
+			fifo_errs=%d\n",
+		       __raw_readl(SAIF1_STAT),
+		       __raw_readl(SAIF1_CTRL),
+		       fifo_err_counter);
+	__raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ | \
+			BM_SAIF_STAT_FIFO_OVERFLOW_IRQ, SAIF1_STAT_CLR);
+	return IRQ_HANDLED;
+}
+
+static int mxs_saif_probe(struct platform_device *pdev, struct snd_soc_dai *dai)
+{
+	if (request_irq(IRQ_SAIF0, saif0_irq, 0, "saif0", dai)) {
+		printk(KERN_ERR "%s: failure requesting irq %s\n",
+		       __func__, "saif0");
+		return -EBUSY;
+	}
+
+	if (request_irq(IRQ_SAIF1, saif1_irq, 0, "saif1", dai)) {
+		printk(KERN_ERR "%s: failure requesting irq %s\n",
+		       __func__, "saif1");
+		return -EBUSY;
+	}
+	return 0;
+}
+
+static void mxs_saif_remove(struct platform_device *pdev,
+			   struct snd_soc_dai *dai)
+{
+	free_irq(IRQ_SAIF0, dai);
+	free_irq(IRQ_SAIF1, dai);
+}
+
+#define MXS_SAIF_RATES \
+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
+	SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+	SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
+	SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+	SNDRV_PCM_RATE_192000)
+
+#define MXS_SAIF_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mxs_saif_dai_ops = {
+	.startup = mxs_saif_startup,
+	.shutdown = mxs_saif_shutdown,
+	.trigger = mxs_saif_trigger,
+	.prepare = mxs_saif_prepare,
+	.hw_params = mxs_saif_hw_params,
+	.set_sysclk = mxs_saif_set_dai_sysclk,
+	.set_clkdiv = mxs_saif_set_dai_clkdiv,
+	.set_fmt = mxs_saif_set_dai_fmt,
+};
+
+struct snd_soc_dai mxs_saif_dai[] = {
+	{
+	.name = "mxs-saif",
+	.probe = mxs_saif_probe,
+	.remove = mxs_saif_remove,
+	.suspend = mxs_saif_suspend,
+	.resume = mxs_saif_resume,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_SAIF_RATES,
+		.formats = MXS_SAIF_FORMATS,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_SAIF_RATES,
+		.formats = MXS_SAIF_FORMATS,
+	},
+	.ops = &mxs_saif_dai_ops,
+	.private_data = &mxs_saif_en,
+	}
+};
+EXPORT_SYMBOL_GPL(mxs_saif_dai);
+
+static int __init mxs_saif_init(void)
+{
+	return snd_soc_register_dais(mxs_saif_dai, ARRAY_SIZE(mxs_saif_dai));
+}
+
+static void __exit mxs_saif_exit(void)
+{
+	snd_soc_unregister_dais(mxs_saif_dai, ARRAY_SIZE(mxs_saif_dai));
+}
+
+module_init(mxs_saif_init);
+module_exit(mxs_saif_exit);
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("i.MX28 ASoC I2S driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-dai.h b/sound/soc/mxs/mxs-dai.h
new file mode 100644
index 000000000000..2476feb832b3
--- /dev/null
+++ b/sound/soc/mxs/mxs-dai.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2009-2010 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef _MXS_SAIF_H
+#define _MXS_SAIF_H
+
+#include <mach/hardware.h>
+
+/* SAIF clock sources */
+#define IMX_SSP_SYS_CLK			0
+#define IMX_SSP_SYS_MCLK		1
+
+#define SAIF0 0
+#define SAIF1 1
+
+/*private info*/
+struct mxs_saif {
+	u8 saif_clk;
+#define PLAYBACK_SAIF0_CAPTURE_SAIF1 0
+#define PLAYBACK_SAIF1_CAPTURE_SAIF0 1
+	u16 stream_mapping;
+	struct clk *saif_mclk;
+};
+
+extern struct snd_soc_dai mxs_saif_dai[];
+
+#endif
diff --git a/sound/soc/mxs/mxs-devb-spdif.c b/sound/soc/mxs/mxs-devb-spdif.c
new file mode 100644
index 000000000000..9ed62110dff7
--- /dev/null
+++ b/sound/soc/mxs/mxs-devb-spdif.c
@@ -0,0 +1,90 @@
+/*
+ * ASoC driver for MXS Evk development board
+ *
+ * Copyright (C) 2008-2010 Freescale Semiconductor, Inc.
+ *
+ * based on stmp3780_devb_spdif.c
+ *
+ * Vladimir Barinov <vbarinov@embeddedalley.com>
+ *
+ * Copyright 2008 SigmaTel, Inc
+ * Copyright 2008 Embedded Alley Solutions, Inc
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program  is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+#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-types.h>
+#include <asm/dma.h>
+#include <mach/hardware.h>
+
+#include "mxs-spdif-dai.h"
+#include "../codecs/mxs_spdif.h"
+#include "mxs-pcm.h"
+
+/* mxs devb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link mxs_devb_dai = {
+	.name = "MXS SPDIF",
+	.stream_name = "MXS SPDIF",
+	.cpu_dai = &mxs_spdif_dai,
+	.codec_dai = &mxs_spdif_codec_dai,
+};
+
+/* mxs devb audio machine driver */
+static struct snd_soc_card snd_soc_machine_mxs_devb = {
+	.name = "mxs-evk",
+	.platform = &mxs_soc_platform,
+	.dai_link = &mxs_devb_dai,
+	.num_links = 1,
+};
+
+/* mxs devb audio subsystem */
+static struct snd_soc_device mxs_devb_snd_devdata = {
+	.card = &snd_soc_machine_mxs_devb,
+	.codec_dev = &soc_spdif_codec_dev_mxs,
+};
+
+static struct platform_device *mxs_devb_snd_device;
+
+static int __init mxs_devb_init(void)
+{
+	int ret = 0;
+
+	mxs_devb_snd_device = platform_device_alloc("soc-audio", 2);
+	if (!mxs_devb_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mxs_devb_snd_device,
+			     &mxs_devb_snd_devdata);
+	mxs_devb_snd_devdata.dev = &mxs_devb_snd_device->dev;
+	mxs_devb_snd_device->dev.platform_data =
+		&mxs_devb_snd_devdata;
+
+	ret = platform_device_add(mxs_devb_snd_device);
+	if (ret)
+		platform_device_put(mxs_devb_snd_device);
+
+	return ret;
+}
+
+static void __exit mxs_devb_exit(void)
+{
+	platform_device_unregister(mxs_devb_snd_device);
+}
+
+module_init(mxs_devb_init);
+module_exit(mxs_devb_exit);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("MXS EVK development board ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-devb.c b/sound/soc/mxs/mxs-devb.c
new file mode 100644
index 000000000000..6847c54a6485
--- /dev/null
+++ b/sound/soc/mxs/mxs-devb.c
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2010 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/err.h>
+#include <linux/irq.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <mach/dma.h>
+#include <mach/device.h>
+
+#include "mxs-dai.h"
+#include "mxs-pcm.h"
+#include "../codecs/sgtl5000.h"
+
+struct mxs_evk_priv {
+	int sysclk;
+	int hw;
+	struct platform_device *pdev;
+};
+
+static struct mxs_evk_priv card_priv;
+
+static int mxs_evk_audio_hw_params(struct snd_pcm_substream *substream,
+				      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
+	struct mxs_evk_priv *priv = &card_priv;
+	unsigned int rate = params_rate(params);
+	int ret = 0;
+
+	u32 dai_format;
+
+	/* only need to do this once as capture and playback are sync */
+	if (priv->hw)
+		return 0;
+	priv->hw = 1;
+	priv->sysclk = 512 * rate;
+
+	snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, (priv->sysclk)/2, 0);
+	snd_soc_dai_set_sysclk(codec_dai, SGTL5000_LRCLK, rate, 0);
+
+	snd_soc_dai_set_clkdiv(cpu_dai, IMX_SSP_SYS_MCLK, 256);
+	/* set codec to slave mode */
+	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+	    SND_SOC_DAIFMT_CBS_CFS;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
+	if (ret < 0)
+		return ret;
+	/* set cpu_dai to master mode for playback, slave mode for record */
+	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
+	if (ret < 0)
+		return ret;
+
+	/* set the SAIF system clock as output */
+	snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, priv->sysclk, \
+		SND_SOC_CLOCK_OUT);
+
+	return 0;
+}
+
+static int mxs_evk_startup(struct snd_pcm_substream *substream)
+{
+	return 0;
+}
+
+static void mxs_evk_shutdown(struct snd_pcm_substream *substream)
+{
+	struct mxs_evk_priv *priv = &card_priv;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+
+	if (cpu_dai->playback.active || cpu_dai->capture.active)
+		priv->hw = 1;
+	else
+		priv->hw = 0;
+}
+
+/*
+ * mxs_evk SGTL5000 audio DAI opserations.
+ */
+static struct snd_soc_ops mxs_evk_ops = {
+	.startup = mxs_evk_startup,
+	.shutdown = mxs_evk_shutdown,
+	.hw_params = mxs_evk_audio_hw_params,
+};
+
+/* mxs_evk machine connections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Mic Jack --> MIC_IN (with automatic bias) */
+	{"MIC_IN", NULL, "Mic Jack"},
+
+	/* Line in Jack --> LINE_IN */
+	{"LINE_IN", NULL, "Line In Jack"},
+
+	/* HP_OUT --> Headphone Jack */
+	{"Headphone Jack", NULL, "HP_OUT"},
+
+	/* LINE_OUT --> Ext Speaker */
+	{"Ext Spk", NULL, "LINE_OUT"},
+};
+
+static const char *jack_function[] = { "off", "on"};
+
+static const char *spk_function[] = { "off", "on" };
+
+static const char *line_in_function[] = { "off", "on" };
+
+static const struct soc_enum sgtl5000_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+	SOC_ENUM_SINGLE_EXT(2, line_in_function),
+};
+
+/* mxs_evk card dapm widgets */
+static const struct snd_soc_dapm_widget mxs_evk_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static int mxs_evk_sgtl5000_init(struct snd_soc_codec *codec)
+{
+	/* Add mxs_evk specific widgets */
+	snd_soc_dapm_new_controls(codec, mxs_evk_dapm_widgets,
+				  ARRAY_SIZE(mxs_evk_dapm_widgets));
+
+	/* Set up mxs_evk specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* mxs_evk digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link mxs_evk_dai = {
+	.name = "SGTL5000",
+	.stream_name = "SGTL5000",
+	.codec_dai = &sgtl5000_dai,
+	.init = mxs_evk_sgtl5000_init,
+	.ops = &mxs_evk_ops,
+};
+
+static int mxs_evk_card_remove(struct platform_device *pdev)
+{
+	struct mxs_evk_priv *priv = &card_priv;
+	struct mxs_audio_platform_data *plat;
+	if (priv->pdev) {
+		plat = priv->pdev->dev.platform_data;
+		if (plat->finit)
+			plat->finit();
+	}
+
+	return 0;
+}
+
+static struct snd_soc_card snd_soc_card_mxs_evk = {
+	.name = "mxs-evk",
+	.platform = &mxs_soc_platform,
+	.dai_link = &mxs_evk_dai,
+	.num_links = 1,
+	.remove = mxs_evk_card_remove,
+};
+
+static struct snd_soc_device mxs_evk_snd_devdata = {
+	.card = &snd_soc_card_mxs_evk,
+	.codec_dev = &soc_codec_dev_sgtl5000,
+};
+
+static int __devinit mxs_evk_sgtl5000_probe(struct platform_device *pdev)
+{
+	struct mxs_audio_platform_data *plat = pdev->dev.platform_data;
+	struct mxs_saif *saif_select;
+	int ret = -EINVAL;
+	if (plat->init && plat->init())
+		goto err_plat_init;
+	mxs_evk_dai.cpu_dai = &mxs_saif_dai[0];
+	saif_select = (struct mxs_saif *)mxs_evk_dai.cpu_dai->private_data;
+	saif_select->stream_mapping = PLAYBACK_SAIF0_CAPTURE_SAIF1;
+	saif_select->saif_mclk = plat->saif_mclock;
+	saif_select->saif_clk = SAIF0;
+	return 0;
+err_plat_init:
+	if (plat->finit)
+		plat->finit();
+	return ret;
+}
+
+static int mxs_evk_sgtl5000_remove(struct platform_device *pdev)
+{
+	struct mxs_audio_platform_data *plat = pdev->dev.platform_data;
+
+	if (plat->finit)
+		plat->finit();
+	return 0;
+}
+
+static struct platform_driver mxs_evk_sgtl5000_audio_driver = {
+	.probe = mxs_evk_sgtl5000_probe,
+	.remove = mxs_evk_sgtl5000_remove,
+	.driver = {
+		   .name = "mxs-sgtl5000",
+		   },
+};
+
+static struct platform_device *mxs_evk_snd_device;
+
+static int __init mxs_evk_init(void)
+{
+	int ret;
+
+	ret = platform_driver_register(&mxs_evk_sgtl5000_audio_driver);
+	if (ret)
+		return -ENOMEM;
+
+	mxs_evk_snd_device = platform_device_alloc("soc-audio", 1);
+	if (!mxs_evk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mxs_evk_snd_device, &mxs_evk_snd_devdata);
+	mxs_evk_snd_devdata.dev = &mxs_evk_snd_device->dev;
+	ret = platform_device_add(mxs_evk_snd_device);
+
+	if (ret)
+		platform_device_put(mxs_evk_snd_device);
+
+	return ret;
+}
+
+static void __exit mxs_evk_exit(void)
+{
+	platform_driver_unregister(&mxs_evk_sgtl5000_audio_driver);
+	platform_device_unregister(mxs_evk_snd_device);
+}
+
+module_init(mxs_evk_init);
+module_exit(mxs_evk_exit);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("SGTL5000 Driver for MXS EVK");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-evk-adc.c b/sound/soc/mxs/mxs-evk-adc.c
new file mode 100644
index 000000000000..16c1cb4fab96
--- /dev/null
+++ b/sound/soc/mxs/mxs-evk-adc.c
@@ -0,0 +1,198 @@
+/*
+ * ASoC driver for Freescale MXS EVK board
+ *
+ * Author: Vladislav Buzov <vbuzov@embeddedalley.com>
+ *
+ * Copyright 2008-2010 Freescale Semiconductor, Inc.
+ * Copyright 2008 Embedded Alley Solutions, Inc All Rights Reserved.
+ */
+
+/*
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+#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-types.h>
+#include <asm/dma.h>
+#include <mach/hardware.h>
+
+#include "../codecs/mxs-adc-codec.h"
+#include "mxs-adc.h"
+#include "mxs-pcm.h"
+
+/* mxs evk machine connections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* HPR/HPL OUT --> Headphone Jack */
+	{"Headphone Jack", NULL, "HPR"},
+	{"Headphone Jack", NULL, "HPL"},
+
+	/* SPEAKER OUT --> Ext Speaker */
+	{"Ext Spk", NULL, "SPEAKER"},
+};
+
+static int mxs_evk_jack_func;
+static int mxs_evk_spk_func;
+
+static const char *jack_function[] = { "off", "on"};
+
+static const char *spk_function[] = { "off", "on" };
+
+
+static const struct soc_enum mxs_evk_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static int mxs_evk_get_jack(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[0] = mxs_evk_jack_func;
+	return 0;
+}
+
+static int mxs_evk_set_jack(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (mxs_evk_jack_func == ucontrol->value.enumerated.item[0])
+		return 0;
+
+	mxs_evk_jack_func = ucontrol->value.enumerated.item[0];
+	if (mxs_evk_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+	snd_soc_dapm_sync(codec);
+	return 1;
+}
+
+static int mxs_evk_get_spk(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[0] = mxs_evk_spk_func;
+	return 0;
+}
+
+static int mxs_evk_set_spk(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (mxs_evk_spk_func == ucontrol->value.enumerated.item[0])
+		return 0;
+
+	mxs_evk_spk_func = ucontrol->value.enumerated.item[0];
+	if (mxs_evk_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+	snd_soc_dapm_sync(codec);
+	return 1;
+}
+/* mxs evk card dapm widgets */
+static const struct snd_soc_dapm_widget mxs_evk_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+};
+
+static const struct snd_kcontrol_new mxs_evk_controls[] = {
+	SOC_ENUM_EXT("HP Playback Switch", mxs_evk_enum[0], mxs_evk_get_jack,
+		     mxs_evk_set_jack),
+	SOC_ENUM_EXT("Speaker Playback Switch", mxs_evk_enum[1],
+		     mxs_evk_get_spk, mxs_evk_set_spk),
+};
+
+static int mxs_evk_codec_init(struct snd_soc_codec *codec)
+{
+	int i, ret;
+	/* Add mxs evk specific controls */
+	snd_soc_add_controls(codec, mxs_evk_controls,
+		ARRAY_SIZE(mxs_evk_controls));
+
+	/* Add mxs evk specific widgets */
+	snd_soc_dapm_new_controls(codec, mxs_evk_dapm_widgets,
+				  ARRAY_SIZE(mxs_evk_dapm_widgets));
+
+	/* Set up mxs evk specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+	/* default on */
+	mxs_evk_jack_func = 1;
+	mxs_evk_spk_func = 1;
+
+	return ret;
+}
+/* mxs evk dac/adc audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link mxs_evk_codec_dai = {
+	.name = "MXS ADC/DAC",
+	.stream_name = "MXS ADC/DAC",
+	.cpu_dai = &mxs_adc_dai,
+	.codec_dai = &mxs_codec_dai,
+	.init = mxs_evk_codec_init,
+};
+
+/* mxs evk audio machine driver */
+static struct snd_soc_card snd_soc_card_mxs_evk = {
+	.name = "MXS EVK",
+	.platform = &mxs_soc_platform,
+	.dai_link = &mxs_evk_codec_dai,
+	.num_links = 1,
+};
+
+/* mxs evk audio subsystem */
+static struct snd_soc_device mxs_evk_snd_devdata = {
+	.card = &snd_soc_card_mxs_evk,
+	.codec_dev = &soc_codec_dev_mxs,
+};
+
+static struct platform_device *mxs_evk_snd_device;
+
+static int __init mxs_evk_adc_init(void)
+{
+	int ret = 0;
+
+	mxs_evk_snd_device = platform_device_alloc("soc-audio", 0);
+	if (!mxs_evk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mxs_evk_snd_device,
+			&mxs_evk_snd_devdata);
+	mxs_evk_snd_devdata.dev = &mxs_evk_snd_device->dev;
+	mxs_evk_snd_device->dev.platform_data =
+		&mxs_evk_snd_devdata;
+
+	ret = platform_device_add(mxs_evk_snd_device);
+	if (ret)
+		platform_device_put(mxs_evk_snd_device);
+
+	return ret;
+}
+
+static void __exit mxs_evk_adc_exit(void)
+{
+	platform_device_unregister(mxs_evk_snd_device);
+}
+
+module_init(mxs_evk_adc_init);
+module_exit(mxs_evk_adc_exit);
+
+MODULE_AUTHOR("Vladislav Buzov");
+MODULE_DESCRIPTION("MXS EVK board ADC/DAC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
new file mode 100644
index 000000000000..b4d7ca00a68f
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -0,0 +1,504 @@
+/*
+ * Copyright (C) 2010 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/dma.h>
+#include <mach/hardware.h>
+#include <mach/dmaengine.h>
+
+#include "mxs-pcm.h"
+static const struct snd_pcm_hardware mxs_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S20_3LE |
+				  SNDRV_PCM_FMTBIT_S24_LE |
+				  SNDRV_PCM_FMTBIT_S32_LE,
+	.channels_min		= 2,
+	.channels_max		= 2,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 1,
+	.periods_max		= 255,
+	.buffer_bytes_max	= 64 * 1024,
+	.fifo_size		= 32,
+};
+
+/*
+ * Required to request DMA channels
+ */
+struct device *mxs_pcm_dev;
+
+static irqreturn_t mxs_pcm_dma_irq(int irq, void *dev_id)
+{
+	struct snd_pcm_substream *substream = dev_id;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = substream->runtime->private_data;
+	struct mxs_dma_info dma_info;
+	void *pdma;
+	unsigned long prev_appl_offset, appl_count, cont, appl_ptr_bytes;
+
+	mxs_dma_get_info(prtd->dma_ch, &dma_info);
+
+	if (dma_info.status) {
+		printk(KERN_WARNING "%s: DMA audio channel %d (%s) error\n",
+			__func__, prtd->params->dma_ch, prtd->params->name);
+		mxs_dma_ack_irq(prtd->dma_ch);
+	} else {
+		if ((prtd->params->dma_ch == MXS_DMA_CHANNEL_AHB_APBX_SPDIF) &&
+		    (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) &&
+		    (runtime->access == SNDRV_PCM_ACCESS_MMAP_INTERLEAVED) &&
+		    ((prtd->format == SNDRV_PCM_FORMAT_S24_LE)
+		     || (prtd->format == SNDRV_PCM_FORMAT_S20_3LE))) {
+
+			appl_ptr_bytes =
+			    frames_to_bytes(runtime,
+					    runtime->control->appl_ptr);
+
+			appl_count = appl_ptr_bytes - prtd->appl_ptr_bytes;
+			prev_appl_offset =
+			    prtd->appl_ptr_bytes % prtd->dma_totsize;
+			cont = prtd->dma_totsize - prev_appl_offset;
+
+			if (appl_count > cont) {
+				pdma = runtime->dma_area + prev_appl_offset;
+				memmove(pdma + 1, pdma, cont - 1);
+				pdma = runtime->dma_area;
+				memmove(pdma + 1, pdma, appl_count - cont - 1);
+			} else {
+				pdma = runtime->dma_area + prev_appl_offset;
+				memmove(pdma + 1, pdma, appl_count - 1);
+			}
+			prtd->appl_ptr_bytes = appl_ptr_bytes;
+		}
+		mxs_dma_ack_irq(prtd->dma_ch);
+		snd_pcm_period_elapsed(substream);
+	}
+	return IRQ_HANDLED;
+}
+
+/*
+ * Make a circular DMA descriptor list
+ */
+static int mxs_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = runtime->private_data;
+	dma_addr_t dma_buffer_phys;
+	int periods_num, playback, i;
+
+	playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	periods_num = prtd->dma_totsize / prtd->dma_period;
+
+	dma_buffer_phys = runtime->dma_addr;
+
+	/* Reset DMA channel, enable interrupt */
+	mxs_dma_reset(prtd->dma_ch);
+
+	/* Set up a DMA chain to sent DMA buffer */
+	for (i = 0; i < periods_num; i++) {
+		int ret;
+		/* Link with previous command */
+		prtd->dma_desc_array[i]->cmd.cmd.bits.bytes = prtd->dma_period;
+		prtd->dma_desc_array[i]->cmd.cmd.bits.irq = 1;
+		prtd->dma_desc_array[i]->cmd.cmd.bits.dec_sem = 0;
+		prtd->dma_desc_array[i]->cmd.cmd.bits.chain = 1;
+		/* Set DMA direction */
+		if (playback)
+			prtd->dma_desc_array[i]->cmd.cmd.bits.command = \
+				DMA_READ;
+		else
+			prtd->dma_desc_array[i]->cmd.cmd.bits.command = \
+				DMA_WRITE;
+
+		prtd->dma_desc_array[i]->cmd.address = dma_buffer_phys;
+
+		ret = mxs_dma_desc_append(prtd->dma_ch, \
+			prtd->dma_desc_array[i]);
+		if (ret) {
+			printk(KERN_ERR "%s: Failed to append DMA descriptor\n",
+		       __func__);
+			return ret;
+		}
+		/* Next data chunk */
+		dma_buffer_phys += prtd->dma_period;
+	}
+
+	return 0;
+}
+
+/*
+ * Stop circular DMA descriptor list
+ * We should not stop DMA in a middle of current transaction once we receive
+ * stop request from ALSA core. This function finds the next DMA descriptor
+ * and set it up to decrement DMA channel semaphore. So the current transaction
+ * is the last data transfer.
+ */
+static void mxs_pcm_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = runtime->private_data;
+	struct mxs_dma_info dma_info;
+	int desc;
+
+	int periods_num = prtd->dma_totsize / prtd->dma_period;
+	/* Freez DMA channel for a moment */
+	mxs_dma_freeze(prtd->dma_ch);
+	mxs_dma_get_info(prtd->dma_ch, &dma_info);
+
+	desc = (dma_info.buf_addr - runtime->dma_addr) / prtd->dma_period;
+
+	if (desc >= periods_num)
+		desc = 0;
+	else if (desc < 0)
+		desc = 0;
+
+	/* Set up the next descriptor to decrement DMA channel sempahore */
+	prtd->dma_desc_array[(desc + 1)%periods_num]->cmd.cmd.bits.bytes = 0;
+	prtd->dma_desc_array[(desc + 1)%periods_num]->cmd.cmd.bits.pio_words = \
+		0;
+	prtd->dma_desc_array[(desc + 1)%periods_num]->cmd.cmd.bits.dec_sem = 1;
+	prtd->dma_desc_array[(desc + 1)%periods_num]->cmd.cmd.bits.irq = 0;
+	prtd->dma_desc_array[(desc + 1)%periods_num]->cmd.cmd.bits.command = \
+		NO_DMA_XFER;
+
+	mxs_dma_unfreeze(prtd->dma_ch);
+
+	mxs_dma_disable(prtd->dma_ch);
+}
+
+static int mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = substream->runtime->private_data;
+	int ret = 0;
+	switch (cmd) {
+
+	case SNDRV_PCM_TRIGGER_START:
+		if ((prtd->params->dma_ch == MXS_DMA_CHANNEL_AHB_APBX_SPDIF) &&
+		    (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) &&
+		    (runtime->access == SNDRV_PCM_ACCESS_MMAP_INTERLEAVED) &&
+		    ((prtd->format == SNDRV_PCM_FORMAT_S24_LE)
+		     || (prtd->format == SNDRV_PCM_FORMAT_S20_3LE))) {
+			prtd->appl_ptr_bytes =
+			    frames_to_bytes(runtime,
+					    runtime->control->appl_ptr);
+			memmove(runtime->dma_area + 1, runtime->dma_area,
+				prtd->appl_ptr_bytes - 1);
+		}
+		mxs_dma_enable(prtd->dma_ch);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+		mxs_pcm_stop(substream);
+		break;
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		mxs_dma_unfreeze(prtd->dma_ch);
+		break;
+
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		mxs_dma_freeze(prtd->dma_ch);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static snd_pcm_uframes_t
+mxs_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = runtime->private_data;
+	struct mxs_dma_info dma_info;
+	unsigned int offset;
+	dma_addr_t pos;
+
+	mxs_dma_get_info(prtd->params->dma_ch, &dma_info);
+	pos = dma_info.buf_addr;
+
+	offset = bytes_to_frames(runtime, pos - runtime->dma_addr);
+
+	if (offset >= runtime->buffer_size)
+		offset = 0;
+
+	return offset;
+}
+
+static int mxs_pcm_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *hw_params)
+{
+	struct mxs_runtime_data *prtd = substream->runtime->private_data;
+
+	prtd->dma_period = params_period_bytes(hw_params);
+	prtd->dma_totsize = params_buffer_bytes(hw_params);
+	prtd->format = params_format(hw_params);
+
+	return snd_pcm_lib_malloc_pages(substream,
+					params_buffer_bytes(hw_params));
+}
+
+static int mxs_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int mxs_pcm_dma_request(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct mxs_runtime_data *prtd = runtime->private_data;
+	struct mxs_pcm_dma_params *dma_data =
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+	int desc_num = mxs_pcm_hardware.periods_max;
+	int desc;
+	int ret;
+
+	if (!dma_data)
+		return -ENODEV;
+
+	prtd->params = dma_data;
+	prtd->dma_ch = dma_data->dma_ch;
+
+	ret = mxs_dma_request(prtd->dma_ch, mxs_pcm_dev,
+				   prtd->params->name);
+	if (ret) {
+		printk(KERN_ERR "%s: Failed to request DMA channel (%d:%d)\n",
+		       __func__, dma_data->dma_bus, dma_data->dma_ch);
+		return ret;
+	}
+
+	/* Allocate memory for data and pio DMA descriptors */
+	for (desc = 0; desc < desc_num; desc++) {
+		prtd->dma_desc_array[desc] = mxs_dma_alloc_desc();
+		if (prtd->dma_desc_array[desc] == NULL) {
+			printk(KERN_ERR"%s Unable to allocate DMA command %d\n",
+			       __func__, desc);
+			goto err;
+		}
+	}
+
+	ret = request_irq(prtd->params->irq, mxs_pcm_dma_irq, 0,
+			  "MXS PCM DMA", substream);
+	if (ret) {
+		printk(KERN_ERR "%s: Unable to request DMA irq %d\n", __func__,
+		       prtd->params->irq);
+		goto err;
+	}
+	/* Enable completion interrupt */
+	mxs_dma_ack_irq(prtd->dma_ch);
+	mxs_dma_enable_irq(prtd->dma_ch, 1);
+
+	return 0;
+
+err:
+	while (--desc >= 0)
+		mxs_dma_free_desc(prtd->dma_desc_array[desc]);
+	mxs_dma_release(prtd->dma_ch, mxs_pcm_dev);
+
+	return ret;
+}
+
+static int mxs_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd;
+	int ret;
+
+	/* Ensure that buffer size is a multiple of the period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		return ret;
+
+	snd_soc_set_runtime_hwparams(substream, &mxs_pcm_hardware);
+
+	prtd = kzalloc(sizeof(struct mxs_runtime_data), GFP_KERNEL);
+	if (prtd == NULL)
+		return -ENOMEM;
+
+	runtime->private_data = prtd;
+
+	ret = mxs_pcm_dma_request(substream);
+	if (ret) {
+		printk(KERN_ERR "mxs_pcm: Failed to request channels\n");
+		kfree(prtd);
+		return ret;
+	}
+	return 0;
+}
+
+static int mxs_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = runtime->private_data;
+	int desc_num = mxs_pcm_hardware.periods_max;
+	int desc;
+
+	static LIST_HEAD(list);
+	mxs_dma_disable(prtd->dma_ch);
+	/* Free DMA irq */
+	free_irq(prtd->params->irq, substream);
+	mxs_dma_get_cooked(prtd->dma_ch, &list);
+	/* Free DMA channel*/
+	mxs_dma_reset(prtd->dma_ch);
+	for (desc = 0; desc < desc_num; desc++)
+		mxs_dma_free_desc(prtd->dma_desc_array[desc]);
+	mxs_dma_release(prtd->dma_ch, mxs_pcm_dev);
+
+	/* Free private runtime data */
+	kfree(prtd);
+	return 0;
+}
+
+static int mcs_pcm_copy(struct snd_pcm_substream *substream, int channel,
+			snd_pcm_uframes_t hwoff, void __user *buf,
+			snd_pcm_uframes_t frames)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_runtime_data *prtd = runtime->private_data;
+	char *hwbuf = runtime->dma_area + frames_to_bytes(runtime, hwoff);
+	unsigned long count = frames_to_bytes(runtime, frames);
+
+	/* For S/PDIF 24-bit playback, fix the buffer.  Code taken from
+	   snd_pcm_lib_write_transfer() and snd_pcm_lib_read_transfer()
+	   in sound/core/pcm_lib.c */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if ((prtd->params->dma_ch == MXS_DMA_CHANNEL_AHB_APBX_SPDIF) &&
+		    ((prtd->format == SNDRV_PCM_FORMAT_S24_LE)
+		     || (prtd->format == SNDRV_PCM_FORMAT_S20_3LE))) {
+			if (copy_from_user(hwbuf + 1, buf, count - 1))
+				return -EFAULT;
+		} else {
+			if (copy_from_user(hwbuf, buf, count))
+				return -EFAULT;
+		}
+	} else {
+		if (copy_to_user(buf, hwbuf, count))
+			return -EFAULT;
+	}
+
+	return 0;
+}
+
+static int mxs_pcm_mmap(struct snd_pcm_substream *substream,
+			     struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_coherent(NULL, vma, runtime->dma_area,
+				 runtime->dma_addr, runtime->dma_bytes);
+}
+
+struct snd_pcm_ops mxs_pcm_ops = {
+	.open		= mxs_pcm_open,
+	.close		= mxs_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= mxs_pcm_hw_params,
+	.hw_free	= mxs_pcm_hw_free,
+	.prepare	= mxs_pcm_prepare,
+	.trigger	= mxs_pcm_trigger,
+	.pointer	= mxs_pcm_pointer,
+	.copy		= mcs_pcm_copy,
+	.mmap		= mxs_pcm_mmap,
+};
+
+static u64 mxs_pcm_dma_mask = DMA_BIT_MASK(32);
+
+static int mxs_pcm_new(struct snd_card *card,
+			    struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	size_t size = mxs_pcm_hardware.buffer_bytes_max;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &mxs_pcm_dma_mask;
+
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, NULL,
+					      size, size);
+
+	return 0;
+}
+
+static void mxs_pcm_free(struct snd_pcm *pcm)
+{
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+/*
+ * We need probe/remove callbacks to setup mxs_pcm_dev
+ */
+static int mxs_pcm_probe(struct platform_device *pdev)
+{
+	mxs_pcm_dev = &pdev->dev;
+	return 0;
+}
+
+static int mxs_pcm_remove(struct platform_device *pdev)
+{
+	mxs_pcm_dev = NULL;
+	return 0;
+}
+
+struct snd_soc_platform mxs_soc_platform = {
+	.name		= "MXS Audio",
+	.pcm_ops	= &mxs_pcm_ops,
+	.probe		= mxs_pcm_probe,
+	.remove		= mxs_pcm_remove,
+	.pcm_new	= mxs_pcm_new,
+	.pcm_free	= mxs_pcm_free,
+};
+EXPORT_SYMBOL_GPL(mxs_soc_platform);
+
+static int __init mxs_pcm_init(void)
+{
+	return snd_soc_register_platform(&mxs_soc_platform);
+}
+
+static void __exit mxs_pcm_exit(void)
+{
+	snd_soc_unregister_platform(&mxs_soc_platform);
+}
+module_init(mxs_pcm_init);
+module_exit(mxs_pcm_exit);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS DMA Module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
new file mode 100644
index 000000000000..738e3fd82d73
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2009-2010 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef _MXS_PCM_H
+#define _MXS_PCM_H
+
+struct mxs_pcm_dma_params {
+	char *name;
+	int dma_bus;	/* DMA bus */
+	int dma_ch;	/* DMA channel number */
+	int irq;	/* DMA interrupt number */
+};
+struct mxs_runtime_data {
+	u32 dma_ch;
+	u32 dma_period;
+	u32 dma_totsize;
+	unsigned long appl_ptr_bytes;
+	int format;
+	struct mxs_pcm_dma_params *params;
+	struct mxs_dma_desc *dma_desc_array[255];
+};
+
+extern struct snd_soc_platform mxs_soc_platform;
+
+#endif
diff --git a/sound/soc/mxs/mxs-spdif-dai.c b/sound/soc/mxs/mxs-spdif-dai.c
new file mode 100644
index 000000000000..1b45b595b90d
--- /dev/null
+++ b/sound/soc/mxs/mxs-spdif-dai.c
@@ -0,0 +1,203 @@
+/*
+ * ALSA SoC SPDIF Audio Layer for MXS
+ *
+ * Copyright (C) 2008-2010 Freescale Semiconductor, Inc.
+ *
+ * Based on stmp3xxx_spdif_dai.c
+ * Vladimir Barinov <vbarinov@embeddedalley.com>
+ * Copyright 2008 SigmaTel, Inc
+ * Copyright 2008 Embedded Alley Solutions, Inc
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program  is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <mach/dma.h>
+#include <mach/hardware.h>
+
+#include "../codecs/mxs_spdif.h"
+#include "mxs-pcm.h"
+
+#define REGS_SPDIF_BASE IO_ADDRESS(SPDIF_PHYS_ADDR)
+
+struct mxs_pcm_dma_params mxs_spdif = {
+	.name = "mxs spdif",
+	.dma_ch	= MXS_DMA_CHANNEL_AHB_APBX_SPDIF,
+	.irq = IRQ_SPDIF_DMA,
+};
+
+static irqreturn_t mxs_err_irq(int irq, void *dev_id)
+{
+	struct snd_pcm_substream *substream = dev_id;
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	u32 ctrl_reg = 0;
+	u32 overflow_mask;
+	u32 underflow_mask;
+
+	if (playback) {
+		ctrl_reg = __raw_readl(REGS_SPDIF_BASE + HW_SPDIF_CTRL);
+		underflow_mask = BM_SPDIF_CTRL_FIFO_UNDERFLOW_IRQ;
+		overflow_mask = BM_SPDIF_CTRL_FIFO_OVERFLOW_IRQ;
+	}
+
+	if (ctrl_reg & underflow_mask) {
+		printk(KERN_DEBUG "underflow detected SPDIF\n");
+
+		if (playback)
+			__raw_writel(BM_SPDIF_CTRL_FIFO_UNDERFLOW_IRQ,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_CLR);
+	} else if (ctrl_reg & overflow_mask) {
+		printk(KERN_DEBUG "overflow detected SPDIF\n");
+
+		if (playback)
+			__raw_writel(BM_SPDIF_CTRL_FIFO_OVERFLOW_IRQ,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_CLR);
+	} else
+		printk(KERN_WARNING "Unknown SPDIF error interrupt\n");
+
+	return IRQ_HANDLED;
+}
+
+static int mxs_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
+{
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		if (playback)
+			__raw_writel(BM_SPDIF_CTRL_RUN,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_SET);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		if (playback)
+			__raw_writel(BM_SPDIF_CTRL_RUN,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_CLR);
+		break;
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int mxs_spdif_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 playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+	int irq;
+	int ret;
+
+	if (playback) {
+		irq = IRQ_SPDIF_ERROR;
+		snd_soc_dai_set_dma_data(cpu_dai, substream, &mxs_spdif);
+	}
+
+	ret = request_irq(irq, mxs_err_irq, 0, "Mxs SPDIF Error",
+			  substream);
+	if (ret) {
+		printk(KERN_ERR "%s: Unable to request SPDIF error irq %d\n",
+		       __func__, IRQ_SPDIF_ERROR);
+		return ret;
+	}
+
+	/* Enable error interrupt */
+	if (playback) {
+		__raw_writel(BM_SPDIF_CTRL_FIFO_OVERFLOW_IRQ,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_CLR);
+		__raw_writel(BM_SPDIF_CTRL_FIFO_UNDERFLOW_IRQ,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_CLR);
+		__raw_writel(BM_SPDIF_CTRL_FIFO_ERROR_IRQ_EN,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_SET);
+	}
+
+	return 0;
+}
+
+static void mxs_spdif_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
+{
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0;
+
+	/* Disable error interrupt */
+	if (playback) {
+		__raw_writel(BM_SPDIF_CTRL_FIFO_ERROR_IRQ_EN,
+				REGS_SPDIF_BASE + HW_SPDIF_CTRL_CLR);
+		free_irq(IRQ_SPDIF_ERROR, substream);
+	}
+}
+
+#ifdef CONFIG_PM
+static int mxs_spdif_suspend(struct snd_soc_dai *cpu_dai)
+{
+	return 0;
+}
+
+static int mxs_spdif_resume(struct snd_soc_dai *cpu_dai)
+{
+	return 0;
+}
+#else
+#define mxs_spdif_suspend	NULL
+#define mxs_spdif_resume	NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_dai_ops mxs_spdif_dai_ops = {
+	.startup = mxs_spdif_startup,
+	.shutdown = mxs_spdif_shutdown,
+	.trigger = mxs_spdif_trigger,
+};
+
+struct snd_soc_dai mxs_spdif_dai = {
+	.name = "mxs-spdif",
+	.id = 0,
+	.suspend = mxs_spdif_suspend,
+	.resume = mxs_spdif_resume,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_SPDIF_RATES,
+		.formats = MXS_SPDIF_FORMATS,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_SPDIF_RATES,
+		.formats = MXS_SPDIF_FORMATS,
+	},
+	.ops = &mxs_spdif_dai_ops,
+};
+EXPORT_SYMBOL_GPL(mxs_spdif_dai);
+
+static int __init mxs_spdif_dai_init(void)
+{
+	return snd_soc_register_dai(&mxs_spdif_dai);
+}
+
+static void __exit mxs_spdif_dai_exit(void)
+{
+	snd_soc_unregister_dai(&mxs_spdif_dai);
+}
+module_init(mxs_spdif_dai_init);
+module_exit(mxs_spdif_dai_exit);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("MXS SPDIF DAI");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-spdif-dai.h b/sound/soc/mxs/mxs-spdif-dai.h
new file mode 100644
index 000000000000..035369af9676
--- /dev/null
+++ b/sound/soc/mxs/mxs-spdif-dai.h
@@ -0,0 +1,21 @@
+/*
+ * ASoC Audio Layer for Freescale STMP3XXX SPDIF transmitter
+ *
+ * Author: Vladimir Barinov <vbarinov@embeddedalley.com>
+ *
+ * Copyright 2008-2010 Freescale Semiconductor, Inc.
+ * Copyright 2008 Embedded Alley Solutions, Inc All Rights Reserved.
+ */
+
+/*
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+#ifndef _STMP3XXX_SPDIF_H
+#define _STMP3XXX_SPDIF_H
+extern struct snd_soc_dai mxs_spdif_dai;
+#endif