31 files changed, 2841 insertions, 255 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index f9054f7c1d52..1381db853ef0 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -53,6 +53,7 @@ source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/mid-x86/Kconfig"
+source "sound/soc/mxs/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/samsung/Kconfig"
 source "sound/soc/s6000/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 4f913876f332..9ea8ac827adc 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -12,6 +12,7 @@ obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)   += imx/
 obj-$(CONFIG_SND_SOC)   += jz4740/
 obj-$(CONFIG_SND_SOC)   += mid-x86/
+obj-$(CONFIG_SND_SOC)   += mxs/
 obj-$(CONFIG_SND_SOC)   += nuc900/
 obj-$(CONFIG_SND_SOC)   += omap/
 obj-$(CONFIG_SND_SOC)   += kirkwood/
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 4b67140fdec3..6d592546e8fc 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -18,10 +18,38 @@ config SND_SOC_AU1XPSC_AC97
         select SND_AC97_CODEC
         select SND_SOC_AC97_BUS
 
+##
+## Au1000/1500/1100 DMA + AC97C/I2SC
+##
+config SND_SOC_AU1XAUDIO
+        tristate "SoC Audio for Au1000/Au1500/Au1100"
+        depends on MIPS_ALCHEMY
+        help
+          This is a driver set for the AC97 unit and the
+          old DMA controller as found on the Au1000/Au1500/Au1100 chips.
+
+config SND_SOC_AU1XAC97C
+        tristate
+        select AC97_BUS
+        select SND_AC97_CODEC
+        select SND_SOC_AC97_BUS
+
+config SND_SOC_AU1XI2SC
+        tristate
+
 
 ##
 ## Boards
 ##
+config SND_SOC_DB1000
+        tristate "DB1000 Audio support"
+        depends on SND_SOC_AU1XAUDIO
+        select SND_SOC_AU1XAC97C
+        select SND_SOC_AC97_CODEC
+        help
+          Select this option to enable AC97 audio on the early DB1x00 series
+          of boards (DB1000/DB1500/DB1100).
+
 config SND_SOC_DB1200
         tristate "DB1200 AC97+I2S audio support"
         depends on SND_SOC_AU1XPSC
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 16873076e8c4..920710514ea0 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -3,11 +3,21 @@ snd-soc-au1xpsc-dbdma-objs := dbdma2.o
 snd-soc-au1xpsc-i2s-objs := psc-i2s.o
 snd-soc-au1xpsc-ac97-objs := psc-ac97.o
 
+# Au1000/1500/1100 Audio units
+snd-soc-au1x-dma-objs := dma.o
+snd-soc-au1x-ac97c-objs := ac97c.o
+snd-soc-au1x-i2sc-objs := i2sc.o
+
 obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
 obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
 obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
+obj-$(CONFIG_SND_SOC_AU1XAUDIO) += snd-soc-au1x-dma.o
+obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o
+obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o
 
 # Boards
+snd-soc-db1000-objs := db1000.o
 snd-soc-db1200-objs := db1200.o
 
+obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o
 obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
new file mode 100644
index 000000000000..9c05f381d95e
--- /dev/null
+++ b/sound/soc/au1x/ac97c.c
@@ -0,0 +1,365 @@
+/*
+ * Au1000/Au1500/Au1100 AC97C controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * based on the old ALSA driver originally written by
+ *                      Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/platform_device.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+/* register offsets and bits */
+#define AC97_CONFIG     0x00
+#define AC97_STATUS     0x04
+#define AC97_DATA       0x08
+#define AC97_CMDRESP    0x0c
+#define AC97_ENABLE     0x10
+
+#define CFG_RC(x)       (((x) & 0x3ff) << 13)   /* valid rx slots mask */
+#define CFG_XS(x)       (((x) & 0x3ff) << 3)    /* valid tx slots mask */
+#define CFG_SG          (1 << 2)        /* sync gate */
+#define CFG_SN          (1 << 1)        /* sync control */
+#define CFG_RS          (1 << 0)        /* acrst# control */
+#define STAT_XU         (1 << 11)       /* tx underflow */
+#define STAT_XO         (1 << 10)       /* tx overflow */
+#define STAT_RU         (1 << 9)        /* rx underflow */
+#define STAT_RO         (1 << 8)        /* rx overflow */
+#define STAT_RD         (1 << 7)        /* codec ready */
+#define STAT_CP         (1 << 6)        /* command pending */
+#define STAT_TE         (1 << 4)        /* tx fifo empty */
+#define STAT_TF         (1 << 3)        /* tx fifo full */
+#define STAT_RE         (1 << 1)        /* rx fifo empty */
+#define STAT_RF         (1 << 0)        /* rx fifo full */
+#define CMD_SET_DATA(x) (((x) & 0xffff) << 16)
+#define CMD_GET_DATA(x) ((x) & 0xffff)
+#define CMD_READ        (1 << 7)
+#define CMD_WRITE       (0 << 7)
+#define CMD_IDX(x)      ((x) & 0x7f)
+#define EN_D            (1 << 1)        /* DISable bit */
+#define EN_CE           (1 << 0)        /* clock enable bit */
+
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES 5
+
+#define AC97_RATES      \
+        SNDRV_PCM_RATE_CONTINUOUS
+
+#define AC97_FMTS       \
+        (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE)
+
+/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only
+ * once AC97C on early Alchemy chips. The newer ones aren't so lucky.
+ */
+static struct au1xpsc_audio_data *ac97c_workdata;
+#define ac97_to_ctx(x)          ac97c_workdata
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+        return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+        __raw_writel(v, ctx->mmio + reg);
+        wmb();
+}
+
+static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97,
+                                          unsigned short r)
+{
+        struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+        unsigned int tmo, retry;
+        unsigned long data;
+
+        data = ~0;
+        retry = AC97_RW_RETRIES;
+        do {
+                mutex_lock(&ctx->lock);
+
+                tmo = 5;
+                while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+                        udelay(21);     /* wait an ac97 frame time */
+                if (!tmo) {
+                        pr_debug("ac97rd timeout #1\n");
+                        goto next;
+                }
+
+                WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ);
+
+                /* stupid errata: data is only valid for 21us, so
+                 * poll, Forrest, poll...
+                 */
+                tmo = 0x10000;
+                while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+                        asm volatile ("nop");
+                data = RD(ctx, AC97_CMDRESP);
+
+                if (!tmo)
+                        pr_debug("ac97rd timeout #2\n");
+
+next:
+                mutex_unlock(&ctx->lock);
+        } while (--retry && !tmo);
+
+        pr_debug("AC97RD %04x %04lx %d\n", r, data, retry);
+
+        return retry ? data & 0xffff : 0xffff;
+}
+
+static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r,
+                                 unsigned short v)
+{
+        struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+        unsigned int tmo, retry;
+
+        retry = AC97_RW_RETRIES;
+        do {
+                mutex_lock(&ctx->lock);
+
+                for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+                        udelay(21);
+                if (!tmo) {
+                        pr_debug("ac97wr timeout #1\n");
+                        goto next;
+                }
+
+                WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v));
+
+                for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+                        udelay(21);
+                if (!tmo)
+                        pr_debug("ac97wr timeout #2\n");
+next:
+                mutex_unlock(&ctx->lock);
+        } while (--retry && !tmo);
+
+        pr_debug("AC97WR %04x %04x %d\n", r, v, retry);
+}
+
+static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+        struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+
+        WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN);
+        msleep(20);
+        WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG);
+        WR(ctx, AC97_CONFIG, ctx->cfg);
+}
+
+static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+        struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+        int i;
+
+        WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS);
+        msleep(500);
+        WR(ctx, AC97_CONFIG, ctx->cfg);
+
+        /* wait for codec ready */
+        i = 50;
+        while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i)
+                msleep(20);
+        if (!i)
+                printk(KERN_ERR "ac97c: codec not ready after cold reset\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+        .read           = au1xac97c_ac97_read,
+        .write          = au1xac97c_ac97_write,
+        .reset          = au1xac97c_ac97_cold_reset,
+        .warm_reset     = au1xac97c_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);        /* globals be gone! */
+
+static int alchemy_ac97c_startup(struct snd_pcm_substream *substream,
+                                 struct snd_soc_dai *dai)
+{
+        struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+        snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+        return 0;
+}
+
+static struct snd_soc_dai_ops alchemy_ac97c_ops = {
+        .startup                = alchemy_ac97c_startup,
+};
+
+static int au1xac97c_dai_probe(struct snd_soc_dai *dai)
+{
+        return ac97c_workdata ? 0 : -ENODEV;
+}
+
+static struct snd_soc_dai_driver au1xac97c_dai_driver = {
+        .name                   = "alchemy-ac97c",
+        .ac97_control           = 1,
+        .probe                  = au1xac97c_dai_probe,
+        .playback = {
+                .rates          = AC97_RATES,
+                .formats        = AC97_FMTS,
+                .channels_min   = 2,
+                .channels_max   = 2,
+        },
+        .capture = {
+                .rates          = AC97_RATES,
+                .formats        = AC97_FMTS,
+                .channels_min   = 2,
+                .channels_max   = 2,
+        },
+        .ops                    = &alchemy_ac97c_ops,
+};
+
+static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
+{
+        int ret;
+        struct resource *r;
+        struct au1xpsc_audio_data *ctx;
+
+        ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+        if (!ctx)
+                return -ENOMEM;
+
+        mutex_init(&ctx->lock);
+
+        r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!r) {
+                ret = -ENODEV;
+                goto out0;
+        }
+
+        ret = -EBUSY;
+        if (!request_mem_region(r->start, resource_size(r), pdev->name))
+                goto out0;
+
+        ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+        if (!ctx->mmio)
+                goto out1;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!r)
+                goto out1;
+        ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (!r)
+                goto out1;
+        ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+        /* switch it on */
+        WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+        WR(ctx, AC97_ENABLE, EN_CE);
+
+        ctx->cfg = CFG_RC(3) | CFG_XS(3);
+        WR(ctx, AC97_CONFIG, ctx->cfg);
+
+        platform_set_drvdata(pdev, ctx);
+
+        ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
+        if (ret)
+                goto out1;
+
+        ac97c_workdata = ctx;
+        return 0;
+
+
+        snd_soc_unregister_dai(&pdev->dev);
+out1:
+        release_mem_region(r->start, resource_size(r));
+out0:
+        kfree(ctx);
+        return ret;
+}
+
+static int __devexit au1xac97c_drvremove(struct platform_device *pdev)
+{
+        struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+        struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+        snd_soc_unregister_dai(&pdev->dev);
+
+        WR(ctx, AC97_ENABLE, EN_D);     /* clock off, disable */
+
+        iounmap(ctx->mmio);
+        release_mem_region(r->start, resource_size(r));
+        kfree(ctx);
+
+        ac97c_workdata = NULL;  /* MDEV */
+
+        return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xac97c_drvsuspend(struct device *dev)
+{
+        struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+        WR(ctx, AC97_ENABLE, EN_D);     /* clock off, disable */
+
+        return 0;
+}
+
+static int au1xac97c_drvresume(struct device *dev)
+{
+        struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+        WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+        WR(ctx, AC97_ENABLE, EN_CE);
+        WR(ctx, AC97_CONFIG, ctx->cfg);
+
+        return 0;
+}
+
+static const struct dev_pm_ops au1xpscac97_pmops = {
+        .suspend        = au1xac97c_drvsuspend,
+        .resume         = au1xac97c_drvresume,
+};
+
+#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops)
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xac97c_driver = {
+        .driver = {
+                .name   = "alchemy-ac97c",
+                .owner  = THIS_MODULE,
+                .pm     = AU1XPSCAC97_PMOPS,
+        },
+        .probe          = au1xac97c_drvprobe,
+        .remove         = __devexit_p(au1xac97c_drvremove),
+};
+
+static int __init au1xac97c_load(void)
+{
+        ac97c_workdata = NULL;
+        return platform_driver_register(&au1xac97c_driver);
+}
+
+static void __exit au1xac97c_unload(void)
+{
+        platform_driver_unregister(&au1xac97c_driver);
+}
+
+module_init(au1xac97c_load);
+module_exit(au1xac97c_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
new file mode 100644
index 000000000000..127477a5e0c7
--- /dev/null
+++ b/sound/soc/au1x/db1000.c
@@ -0,0 +1,75 @@
+/*
+ * DB1000/DB1500/DB1100 ASoC audio fabric support code.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.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 <asm/mach-au1x00/au1000.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "psc.h"
+
+static struct snd_soc_dai_link db1000_ac97_dai = {
+        .name           = "AC97",
+        .stream_name    = "AC97 HiFi",
+        .codec_dai_name = "ac97-hifi",
+        .cpu_dai_name   = "alchemy-ac97c",
+        .platform_name  = "alchemy-pcm-dma.0",
+        .codec_name     = "ac97-codec",
+};
+
+static struct snd_soc_card db1000_ac97 = {
+        .name           = "DB1000_AC97",
+        .dai_link       = &db1000_ac97_dai,
+        .num_links      = 1,
+};
+
+static int __devinit db1000_audio_probe(struct platform_device *pdev)
+{
+        struct snd_soc_card *card = &db1000_ac97;
+        card->dev = &pdev->dev;
+        return snd_soc_register_card(card);
+}
+
+static int __devexit db1000_audio_remove(struct platform_device *pdev)
+{
+        struct snd_soc_card *card = platform_get_drvdata(pdev);
+        snd_soc_unregister_card(card);
+        return 0;
+}
+
+static struct platform_driver db1000_audio_driver = {
+        .driver = {
+                .name   = "db1000-audio",
+                .owner  = THIS_MODULE,
+                .pm     = &snd_soc_pm_ops,
+        },
+        .probe          = db1000_audio_probe,
+        .remove         = __devexit_p(db1000_audio_remove),
+};
+
+static int __init db1000_audio_load(void)
+{
+        return platform_driver_register(&db1000_audio_driver);
+}
+
+static void __exit db1000_audio_unload(void)
+{
+        platform_driver_unregister(&db1000_audio_driver);
+}
+
+module_init(db1000_audio_load);
+module_exit(db1000_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 1d3e258c9ea8..289312c14b99 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -1,7 +1,7 @@
 /*
  * DB1200 ASoC audio fabric support code.
  *
- * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com>
  *
  */
 
@@ -21,6 +21,17 @@
 #include "../codecs/wm8731.h"
 #include "psc.h"
 
+static struct platform_device_id db1200_pids[] = {
+        {
+                .name           = "db1200-ac97",
+                .driver_data    = 0,
+        }, {
+                .name           = "db1200-i2s",
+                .driver_data    = 1,
+        },
+        {},
+};
+
 /*-------------------------  AC97 PART  ---------------------------*/
 
 static struct snd_soc_dai_link db1200_ac97_dai = {
@@ -89,36 +100,47 @@ static struct snd_soc_card db1200_i2s_machine = {
 
 /*-------------------------  COMMON PART  ---------------------------*/
 
-static struct platform_device *db1200_asoc_dev;
+static struct snd_soc_card *db1200_cards[] __devinitdata = {
+        &db1200_ac97_machine,
+        &db1200_i2s_machine,
+};
 
-static int __init db1200_audio_load(void)
+static int __devinit db1200_audio_probe(struct platform_device *pdev)
 {
-        int ret;
+        const struct platform_device_id *pid = platform_get_device_id(pdev);
+        struct snd_soc_card *card;
 
-        ret = -ENOMEM;
-        db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */
-        if (!db1200_asoc_dev)
-                goto out;
+        card = db1200_cards[pid->driver_data];
+        card->dev = &pdev->dev;
+        return snd_soc_register_card(card);
+}
 
-        /* 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_machine);
-        else
-                platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine);
+static int __devexit db1200_audio_remove(struct platform_device *pdev)
+{
+        struct snd_soc_card *card = platform_get_drvdata(pdev);
+        snd_soc_unregister_card(card);
+        return 0;
+}
 
-        ret = platform_device_add(db1200_asoc_dev);
+static struct platform_driver db1200_audio_driver = {
+        .driver = {
+                .name   = "db1200-ac97",
+                .owner  = THIS_MODULE,
+                .pm     = &snd_soc_pm_ops,
+        },
+        .id_table       = db1200_pids,
+        .probe          = db1200_audio_probe,
+        .remove         = __devexit_p(db1200_audio_remove),
+};
 
-        if (ret) {
-                platform_device_put(db1200_asoc_dev);
-                db1200_asoc_dev = NULL;
-        }
-out:
-        return ret;
+static int __init db1200_audio_load(void)
+{
+        return platform_driver_register(&db1200_audio_driver);
 }
 
 static void __exit db1200_audio_unload(void)
 {
-        platform_device_unregister(db1200_asoc_dev);
+        platform_driver_unregister(&db1200_audio_driver);
 }
 
 module_init(db1200_audio_load);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 20bb53a837b1..d7d04e26eee5 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -169,7 +169,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
 
         au1x_pcm_dbdma_free(pcd);
 
-        if (stype == PCM_RX)
+        if (stype == SNDRV_PCM_STREAM_CAPTURE)
                 pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
                                         DSCR_CMD0_ALWAYS,
                                         au1x_pcm_dmarx_cb, (void *)pcd);
@@ -198,7 +198,7 @@ static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream
         struct snd_soc_pcm_runtime *rtd = ss->private_data;
         struct au1xpsc_audio_dmadata *pcd =
                                 snd_soc_platform_get_drvdata(rtd->platform);
-        return &pcd[SUBSTREAM_TYPE(ss)];
+        return &pcd[ss->stream];
 }
 
 static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -212,7 +212,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
         if (ret < 0)
                 goto out;
 
-        stype = SUBSTREAM_TYPE(substream);
+        stype = substream->stream;
         pcd = to_dmadata(substream);
 
         DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
@@ -255,7 +255,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
 
         au1xxx_dbdma_reset(pcd->ddma_chan);
 
-        if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+        if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
                 au1x_pcm_queue_rx(pcd);
                 au1x_pcm_queue_rx(pcd);
         } else {
@@ -293,6 +293,16 @@ au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
 
 static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
 {
+        struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        int stype = substream->stream, *dmaids;
+
+        dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+        if (!dmaids)
+                return -ENODEV; /* whoa, has ordering changed? */
+
+        pcd->ddma_id = dmaids[stype];
+
         snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
         return 0;
 }
@@ -340,36 +350,18 @@ struct snd_soc_platform_driver au1xpsc_soc_platform = {
 static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
 {
         struct au1xpsc_audio_dmadata *dmadata;
-        struct resource *r;
         int ret;
 
         dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
         if (!dmadata)
                 return -ENOMEM;
 
-        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-        if (!r) {
-                ret = -ENODEV;
-                goto out1;
-        }
-        dmadata[PCM_TX].ddma_id = r->start;
-
-        /* RX DMA */
-        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-        if (!r) {
-                ret = -ENODEV;
-                goto out1;
-        }
-        dmadata[PCM_RX].ddma_id = r->start;
-
         platform_set_drvdata(pdev, dmadata);
 
         ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform);
-        if (!ret)
-                return ret;
+        if (ret)
+                kfree(dmadata);
 
-out1:
-        kfree(dmadata);
         return ret;
 }
 
@@ -405,57 +397,6 @@ static void __exit au1xpsc_audio_dbdma_unload(void)
 module_init(au1xpsc_audio_dbdma_load);
 module_exit(au1xpsc_audio_dbdma_unload);
 
-
-struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev)
-{
-        struct resource *res, *r;
-        struct platform_device *pd;
-        int id[2];
-        int ret;
-
-        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-        if (!r)
-                return NULL;
-        id[0] = r->start;
-
-        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-        if (!r)
-                return NULL;
-        id[1] = r->start;
-
-        res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL);
-        if (!res)
-                return NULL;
-
-        res[0].start = res[0].end = id[0];
-        res[1].start = res[1].end = id[1];
-        res[0].flags = res[1].flags = IORESOURCE_DMA;
-
-        pd = platform_device_alloc("au1xpsc-pcm", pdev->id);
-        if (!pd)
-                goto out;
-
-        pd->resource = res;
-        pd->num_resources = 2;
-
-        ret = platform_device_add(pd);
-        if (!ret)
-                return pd;
-
-        platform_device_put(pd);
-out:
-        kfree(res);
-        return NULL;
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_add);
-
-void au1xpsc_pcm_destroy(struct platform_device *dmapd)
-{
-        if (dmapd)
-                platform_device_unregister(dmapd);
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy);
-
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
 MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
new file mode 100644
index 000000000000..7aa5b7606777
--- /dev/null
+++ b/sound/soc/au1x/dma.c
@@ -0,0 +1,377 @@
+/*
+ * Au1000/Au1500/Au1100 Audio DMA support.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * copied almost verbatim from the old ALSA driver, written by
+ *                      Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
+
+#include "psc.h"
+
+#define ALCHEMY_PCM_FMTS                                        \
+        (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_U8 |        \
+         SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |    \
+         SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |    \
+         SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |    \
+         SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE |    \
+         0)
+
+struct pcm_period {
+        u32 start;
+        u32 relative_end;       /* relative to start of buffer */
+        struct pcm_period *next;
+};
+
+struct audio_stream {
+        struct snd_pcm_substream *substream;
+        int dma;
+        struct pcm_period *buffer;
+        unsigned int period_size;
+        unsigned int periods;
+};
+
+struct alchemy_pcm_ctx {
+        struct audio_stream stream[2];  /* playback & capture */
+};
+
+static void au1000_release_dma_link(struct audio_stream *stream)
+{
+        struct pcm_period *pointer;
+        struct pcm_period *pointer_next;
+
+        stream->period_size = 0;
+        stream->periods = 0;
+        pointer = stream->buffer;
+        if (!pointer)
+                return;
+        do {
+                pointer_next = pointer->next;
+                kfree(pointer);
+                pointer = pointer_next;
+        } while (pointer != stream->buffer);
+        stream->buffer = NULL;
+}
+
+static int au1000_setup_dma_link(struct audio_stream *stream,
+                                 unsigned int period_bytes,
+                                 unsigned int periods)
+{
+        struct snd_pcm_substream *substream = stream->substream;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct pcm_period *pointer;
+        unsigned long dma_start;
+        int i;
+
+        dma_start = virt_to_phys(runtime->dma_area);
+
+        if (stream->period_size == period_bytes &&
+            stream->periods == periods)
+                return 0; /* not changed */
+
+        au1000_release_dma_link(stream);
+
+        stream->period_size = period_bytes;
+        stream->periods = periods;
+
+        stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL);
+        if (!stream->buffer)
+                return -ENOMEM;
+        pointer = stream->buffer;
+        for (i = 0; i < periods; i++) {
+                pointer->start = (u32)(dma_start + (i * period_bytes));
+                pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+                if (i < periods - 1) {
+                        pointer->next = kmalloc(sizeof(struct pcm_period),
+                                                GFP_KERNEL);
+                        if (!pointer->next) {
+                                au1000_release_dma_link(stream);
+                                return -ENOMEM;
+                        }
+                        pointer = pointer->next;
+                }
+        }
+        pointer->next = stream->buffer;
+        return 0;
+}
+
+static void au1000_dma_stop(struct audio_stream *stream)
+{
+        if (stream->buffer)
+                disable_dma(stream->dma);
+}
+
+static void au1000_dma_start(struct audio_stream *stream)
+{
+        if (!stream->buffer)
+                return;
+
+        init_dma(stream->dma);
+        if (get_dma_active_buffer(stream->dma) == 0) {
+                clear_dma_done0(stream->dma);
+                set_dma_addr0(stream->dma, stream->buffer->start);
+                set_dma_count0(stream->dma, stream->period_size >> 1);
+                set_dma_addr1(stream->dma, stream->buffer->next->start);
+                set_dma_count1(stream->dma, stream->period_size >> 1);
+        } else {
+                clear_dma_done1(stream->dma);
+                set_dma_addr1(stream->dma, stream->buffer->start);
+                set_dma_count1(stream->dma, stream->period_size >> 1);
+                set_dma_addr0(stream->dma, stream->buffer->next->start);
+                set_dma_count0(stream->dma, stream->period_size >> 1);
+        }
+        enable_dma_buffers(stream->dma);
+        start_dma(stream->dma);
+}
+
+static irqreturn_t au1000_dma_interrupt(int irq, void *ptr)
+{
+        struct audio_stream *stream = (struct audio_stream *)ptr;
+        struct snd_pcm_substream *substream = stream->substream;
+
+        switch (get_dma_buffer_done(stream->dma)) {
+        case DMA_D0:
+                stream->buffer = stream->buffer->next;
+                clear_dma_done0(stream->dma);
+                set_dma_addr0(stream->dma, stream->buffer->next->start);
+                set_dma_count0(stream->dma, stream->period_size >> 1);
+                enable_dma_buffer0(stream->dma);
+                break;
+        case DMA_D1:
+                stream->buffer = stream->buffer->next;
+                clear_dma_done1(stream->dma);
+                set_dma_addr1(stream->dma, stream->buffer->next->start);
+                set_dma_count1(stream->dma, stream->period_size >> 1);
+                enable_dma_buffer1(stream->dma);
+                break;
+        case (DMA_D0 | DMA_D1):
+                pr_debug("DMA %d missed interrupt.\n", stream->dma);
+                au1000_dma_stop(stream);
+                au1000_dma_start(stream);
+                break;
+        case (~DMA_D0 & ~DMA_D1):
+                pr_debug("DMA %d empty irq.\n", stream->dma);
+        }
+        snd_pcm_period_elapsed(substream);
+        return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_hardware alchemy_pcm_hardware = {
+        .info             = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+                            SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
+        .formats          = ALCHEMY_PCM_FMTS,
+        .rates            = SNDRV_PCM_RATE_8000_192000,
+        .rate_min         = SNDRV_PCM_RATE_8000,
+        .rate_max         = SNDRV_PCM_RATE_192000,
+        .channels_min     = 2,
+        .channels_max     = 2,
+        .period_bytes_min = 1024,
+        .period_bytes_max = 16 * 1024 - 1,
+        .periods_min      = 4,
+        .periods_max      = 255,
+        .buffer_bytes_max = 128 * 1024,
+        .fifo_size        = 16,
+};
+
+static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss)
+{
+        struct snd_soc_pcm_runtime *rtd = ss->private_data;
+        return snd_soc_platform_get_drvdata(rtd->platform);
+}
+
+static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss)
+{
+        struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss);
+        return &(ctx->stream[ss->stream]);
+}
+
+static int alchemy_pcm_open(struct snd_pcm_substream *substream)
+{
+        struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        int *dmaids, s = substream->stream;
+        char *name;
+
+        dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+        if (!dmaids)
+                return -ENODEV; /* whoa, has ordering changed? */
+
+        /* DMA setup */
+        name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
+        ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
+                                        au1000_dma_interrupt, IRQF_DISABLED,
+                                        &ctx->stream[s]);
+        set_dma_mode(ctx->stream[s].dma,
+                     get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
+
+        ctx->stream[s].substream = substream;
+        ctx->stream[s].buffer = NULL;
+        snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware);
+
+        return 0;
+}
+
+static int alchemy_pcm_close(struct snd_pcm_substream *substream)
+{
+        struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+        int stype = substream->stream;
+
+        ctx->stream[stype].substream = NULL;
+        free_au1000_dma(ctx->stream[stype].dma);
+
+        return 0;
+}
+
+static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *hw_params)
+{
+        struct audio_stream *stream = ss_to_as(substream);
+        int err;
+
+        err = snd_pcm_lib_malloc_pages(substream,
+                                       params_buffer_bytes(hw_params));
+        if (err < 0)
+                return err;
+        err = au1000_setup_dma_link(stream,
+                                    params_period_bytes(hw_params),
+                                    params_periods(hw_params));
+        if (err)
+                snd_pcm_lib_free_pages(substream);
+
+        return err;
+}
+
+static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        struct audio_stream *stream = ss_to_as(substream);
+        au1000_release_dma_link(stream);
+        return snd_pcm_lib_free_pages(substream);
+}
+
+static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct audio_stream *stream = ss_to_as(substream);
+        int err = 0;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+                au1000_dma_start(stream);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+                au1000_dma_stop(stream);
+                break;
+        default:
+                err = -EINVAL;
+                break;
+        }
+        return err;
+}
+
+static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss)
+{
+        struct audio_stream *stream = ss_to_as(ss);
+        long location;
+
+        location = get_dma_residue(stream->dma);
+        location = stream->buffer->relative_end - location;
+        if (location == -1)
+                location = 0;
+        return bytes_to_frames(ss->runtime, location);
+}
+
+static struct snd_pcm_ops alchemy_pcm_ops = {
+        .open                   = alchemy_pcm_open,
+        .close                  = alchemy_pcm_close,
+        .ioctl                  = snd_pcm_lib_ioctl,
+        .hw_params              = alchemy_pcm_hw_params,
+        .hw_free                = alchemy_pcm_hw_free,
+        .trigger                = alchemy_pcm_trigger,
+        .pointer                = alchemy_pcm_pointer,
+};
+
+static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+        snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_pcm *pcm = rtd->pcm;
+
+        snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+                snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1);
+
+        return 0;
+}
+
+struct snd_soc_platform_driver alchemy_pcm_soc_platform = {
+        .ops            = &alchemy_pcm_ops,
+        .pcm_new        = alchemy_pcm_new,
+        .pcm_free       = alchemy_pcm_free_dma_buffers,
+};
+
+static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev)
+{
+        struct alchemy_pcm_ctx *ctx;
+        int ret;
+
+        ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+        if (!ctx)
+                return -ENOMEM;
+
+        platform_set_drvdata(pdev, ctx);
+
+        ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform);
+        if (ret)
+                kfree(ctx);
+
+        return ret;
+}
+
+static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev)
+{
+        struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev);
+
+        snd_soc_unregister_platform(&pdev->dev);
+        kfree(ctx);
+
+        return 0;
+}
+
+static struct platform_driver alchemy_pcmdma_driver = {
+        .driver = {
+                .name   = "alchemy-pcm-dma",
+                .owner  = THIS_MODULE,
+        },
+        .probe          = alchemy_pcm_drvprobe,
+        .remove         = __devexit_p(alchemy_pcm_drvremove),
+};
+
+static int __init alchemy_pcmdma_load(void)
+{
+        return platform_driver_register(&alchemy_pcmdma_driver);
+}
+
+static void __exit alchemy_pcmdma_unload(void)
+{
+        platform_driver_unregister(&alchemy_pcmdma_driver);
+}
+
+module_init(alchemy_pcmdma_load);
+module_exit(alchemy_pcmdma_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
new file mode 100644
index 000000000000..b4172fdd2c48
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,347 @@
+/*
+ * Au1000/Au1500/Au1100 I2S controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * Note: clock supplied to the I2S controller must be 256x samplerate.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+#define I2S_RXTX        0x00
+#define I2S_CFG         0x04
+#define I2S_ENABLE      0x08
+
+#define CFG_XU          (1 << 25)       /* tx underflow */
+#define CFG_XO          (1 << 24)
+#define CFG_RU          (1 << 23)
+#define CFG_RO          (1 << 22)
+#define CFG_TR          (1 << 21)
+#define CFG_TE          (1 << 20)
+#define CFG_TF          (1 << 19)
+#define CFG_RR          (1 << 18)
+#define CFG_RF          (1 << 17)
+#define CFG_ICK         (1 << 12)       /* clock invert */
+#define CFG_PD          (1 << 11)       /* set to make I2SDIO INPUT */
+#define CFG_LB          (1 << 10)       /* loopback */
+#define CFG_IC          (1 << 9)        /* word select invert */
+#define CFG_FM_I2S      (0 << 7)        /* I2S format */
+#define CFG_FM_LJ       (1 << 7)        /* left-justified */
+#define CFG_FM_RJ       (2 << 7)        /* right-justified */
+#define CFG_FM_MASK     (3 << 7)
+#define CFG_TN          (1 << 6)        /* tx fifo en */
+#define CFG_RN          (1 << 5)        /* rx fifo en */
+#define CFG_SZ_8        (0x08)
+#define CFG_SZ_16       (0x10)
+#define CFG_SZ_18       (0x12)
+#define CFG_SZ_20       (0x14)
+#define CFG_SZ_24       (0x18)
+#define CFG_SZ_MASK     (0x1f)
+#define EN_D            (1 << 1)        /* DISable */
+#define EN_CE           (1 << 0)        /* clock enable */
+
+/* only limited by clock generator and board design */
+#define AU1XI2SC_RATES \
+        SNDRV_PCM_RATE_CONTINUOUS
+
+#define AU1XI2SC_FMTS \
+        (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |            \
+        SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |     \
+        SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |     \
+        SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE |   \
+        SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE |   \
+        SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE |   \
+        SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE |   \
+        SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |     \
+        SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE |     \
+        0)
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+        return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+        __raw_writel(v, ctx->mmio + reg);
+        wmb();
+}
+
+static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+        struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai);
+        unsigned long c;
+        int ret;
+
+        ret = -EINVAL;
+        c = ctx->cfg;
+
+        c &= ~CFG_FM_MASK;
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                c |= CFG_FM_I2S;
+                break;
+        case SND_SOC_DAIFMT_MSB:
+                c |= CFG_FM_RJ;
+                break;
+        case SND_SOC_DAIFMT_LSB:
+                c |= CFG_FM_LJ;
+                break;
+        default:
+                goto out;
+        }
+
+        c &= ~(CFG_IC | CFG_ICK);               /* IB-IF */
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                c |= CFG_IC | CFG_ICK;
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                c |= CFG_IC;
+                break;
+        case SND_SOC_DAIFMT_IB_NF:
+                c |= CFG_ICK;
+                break;
+        case SND_SOC_DAIFMT_IB_IF:
+                break;
+        default:
+                goto out;
+        }
+
+        /* I2S controller only supports master */
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBS_CFS:    /* CODEC slave */
+                break;
+        default:
+                goto out;
+        }
+
+        ret = 0;
+        ctx->cfg = c;
+out:
+        return ret;
+}
+
+static int au1xi2s_trigger(struct snd_pcm_substream *substream,
+                           int cmd, struct snd_soc_dai *dai)
+{
+        struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+        int stype = SUBSTREAM_TYPE(substream);
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+                /* power up */
+                WR(ctx, I2S_ENABLE, EN_D | EN_CE);
+                WR(ctx, I2S_ENABLE, EN_CE);
+                ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN;
+                WR(ctx, I2S_CFG, ctx->cfg);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN);
+                WR(ctx, I2S_CFG, ctx->cfg);
+                WR(ctx, I2S_ENABLE, EN_D);              /* power off */
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static unsigned long msbits_to_reg(int msbits)
+{
+        switch (msbits) {
+        case 8:
+                return CFG_SZ_8;
+        case 16:
+                return CFG_SZ_16;
+        case 18:
+                return CFG_SZ_18;
+        case 20:
+                return CFG_SZ_20;
+        case 24:
+                return CFG_SZ_24;
+        }
+        return 0;
+}
+
+static int au1xi2s_hw_params(struct snd_pcm_substream *substream,
+                             struct snd_pcm_hw_params *params,
+                             struct snd_soc_dai *dai)
+{
+        struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+        unsigned long v;
+
+        v = msbits_to_reg(params->msbits);
+        if (!v)
+                return -EINVAL;
+
+        ctx->cfg &= ~CFG_SZ_MASK;
+        ctx->cfg |= v;
+        return 0;
+}
+
+static int au1xi2s_startup(struct snd_pcm_substream *substream,
+                           struct snd_soc_dai *dai)
+{
+        struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+        snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+        return 0;
+}
+
+static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+        .startup        = au1xi2s_startup,
+        .trigger        = au1xi2s_trigger,
+        .hw_params      = au1xi2s_hw_params,
+        .set_fmt        = au1xi2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver au1xi2s_dai_driver = {
+        .symmetric_rates        = 1,
+        .playback = {
+                .rates          = AU1XI2SC_RATES,
+                .formats        = AU1XI2SC_FMTS,
+                .channels_min   = 2,
+                .channels_max   = 2,
+        },
+        .capture = {
+                .rates          = AU1XI2SC_RATES,
+                .formats        = AU1XI2SC_FMTS,
+                .channels_min   = 2,
+                .channels_max   = 2,
+        },
+        .ops = &au1xi2s_dai_ops,
+};
+
+static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
+{
+        int ret;
+        struct resource *r;
+        struct au1xpsc_audio_data *ctx;
+
+        ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+        if (!ctx)
+                return -ENOMEM;
+
+        r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!r) {
+                ret = -ENODEV;
+                goto out0;
+        }
+
+        ret = -EBUSY;
+        if (!request_mem_region(r->start, resource_size(r), pdev->name))
+                goto out0;
+
+        ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+        if (!ctx->mmio)
+                goto out1;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!r)
+                goto out1;
+        ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (!r)
+                goto out1;
+        ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+        platform_set_drvdata(pdev, ctx);
+
+        ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+        if (ret)
+                goto out1;
+
+        return 0;
+
+        snd_soc_unregister_dai(&pdev->dev);
+out1:
+        release_mem_region(r->start, resource_size(r));
+out0:
+        kfree(ctx);
+        return ret;
+}
+
+static int __devexit au1xi2s_drvremove(struct platform_device *pdev)
+{
+        struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+        struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+        snd_soc_unregister_dai(&pdev->dev);
+
+        WR(ctx, I2S_ENABLE, EN_D);      /* clock off, disable */
+
+        iounmap(ctx->mmio);
+        release_mem_region(r->start, resource_size(r));
+        kfree(ctx);
+
+        return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xi2s_drvsuspend(struct device *dev)
+{
+        struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+        WR(ctx, I2S_ENABLE, EN_D);      /* clock off, disable */
+
+        return 0;
+}
+
+static int au1xi2s_drvresume(struct device *dev)
+{
+        return 0;
+}
+
+static const struct dev_pm_ops au1xi2sc_pmops = {
+        .suspend        = au1xi2s_drvsuspend,
+        .resume         = au1xi2s_drvresume,
+};
+
+#define AU1XI2SC_PMOPS (&au1xi2sc_pmops)
+
+#else
+
+#define AU1XI2SC_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xi2s_driver = {
+        .driver = {
+                .name   = "alchemy-i2sc",
+                .owner  = THIS_MODULE,
+                .pm     = AU1XI2SC_PMOPS,
+        },
+        .probe          = au1xi2s_drvprobe,
+        .remove         = __devexit_p(au1xi2s_drvremove),
+};
+
+static int __init au1xi2s_load(void)
+{
+        return platform_driver_register(&au1xi2s_driver);
+}
+
+static void __exit au1xi2s_unload(void)
+{
+        platform_driver_unregister(&au1xi2s_driver);
+}
+
+module_init(au1xi2s_load);
+module_exit(au1xi2s_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index d0db66f24a00..172eefd38b2d 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -41,14 +41,14 @@
         (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
 
 #define AC97PCR_START(stype)    \
-        ((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
 #define AC97PCR_STOP(stype)     \
-        ((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
 #define AC97PCR_CLRFIFO(stype)  \
-        ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
 
 #define AC97STAT_BUSY(stype)    \
-        ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
 
 /* instance data. There can be only one, MacLeod!!!! */
 static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
@@ -215,7 +215,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 {
         struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
         unsigned long r, ro, stat;
-        int chans, t, stype = SUBSTREAM_TYPE(substream);
+        int chans, t, stype = substream->stream;
 
         chans = params_channels(params);
 
@@ -235,7 +235,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
                 r |= PSC_AC97CFG_SET_LEN(params->msbits);
 
                 /* channels: enable slots for front L/R channel */
-                if (stype == PCM_TX) {
+                if (stype == SNDRV_PCM_STREAM_PLAYBACK) {
                         r &= ~PSC_AC97CFG_TXSLOT_MASK;
                         r |= PSC_AC97CFG_TXSLOT_ENA(3);
                         r |= PSC_AC97CFG_TXSLOT_ENA(4);
@@ -294,7 +294,7 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
                                 int cmd, struct snd_soc_dai *dai)
 {
         struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
-        int ret, stype = SUBSTREAM_TYPE(substream);
+        int ret, stype = substream->stream;
 
         ret = 0;
 
@@ -324,12 +324,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
         return ret;
 }
 
+static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai)
+{
+        struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+        snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+        return 0;
+}
+
 static int au1xpsc_ac97_probe(struct snd_soc_dai *dai)
 {
         return au1xpsc_ac97_workdata ? 0 : -ENODEV;
 }
 
 static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+        .startup        = au1xpsc_ac97_startup,
         .trigger        = au1xpsc_ac97_trigger,
         .hw_params      = au1xpsc_ac97_hw_params,
 };
@@ -379,6 +388,16 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
         if (!wd->mmio)
                 goto out1;
 
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!r)
+                goto out2;
+        wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (!r)
+                goto out2;
+        wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
         /* configuration: max dma trigger threshold, enable ac97 */
         wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
                   PSC_AC97CFG_DE_ENABLE;
@@ -401,15 +420,13 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 
         ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
         if (ret)
-                goto out1;
+                goto out2;
 
-        wd->dmapd = au1xpsc_pcm_add(pdev);
-        if (wd->dmapd) {
-                au1xpsc_ac97_workdata = wd;
-                return 0;
-        }
+        au1xpsc_ac97_workdata = wd;
+        return 0;
 
-        snd_soc_unregister_dai(&pdev->dev);
+out2:
+        iounmap(wd->mmio);
 out1:
         release_mem_region(r->start, resource_size(r));
 out0:
@@ -422,9 +439,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
         struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
         struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
-        if (wd->dmapd)
-                au1xpsc_pcm_destroy(wd->dmapd);
-
         snd_soc_unregister_dai(&pdev->dev);
 
         /* disable PSC completely */
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index fca091276320..7c5ae920544f 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -42,13 +42,13 @@
         (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 #define I2SSTAT_BUSY(stype)     \
-        ((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
 #define I2SPCR_START(stype)     \
-        ((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
 #define I2SPCR_STOP(stype)      \
-        ((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
 #define I2SPCR_CLRFIFO(stype)   \
-        ((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+        ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
 
 
 static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
@@ -240,7 +240,7 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
                                struct snd_soc_dai *dai)
 {
         struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
-        int ret, stype = SUBSTREAM_TYPE(substream);
+        int ret, stype = substream->stream;
 
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
@@ -257,7 +257,16 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
         return ret;
 }
 
+static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
+                               struct snd_soc_dai *dai)
+{
+        struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+        snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+        return 0;
+}
+
 static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+        .startup        = au1xpsc_i2s_startup,
         .trigger        = au1xpsc_i2s_trigger,
         .hw_params      = au1xpsc_i2s_hw_params,
         .set_fmt        = au1xpsc_i2s_set_fmt,
@@ -304,6 +313,16 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
         if (!wd->mmio)
                 goto out1;
 
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!r)
+                goto out2;
+        wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+        r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (!r)
+                goto out2;
+        wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
         /* preserve PSC clock source set up by platform (dev.platform_data
          * is already occupied by soc layer)
          */
@@ -330,15 +349,11 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
         platform_set_drvdata(pdev, wd);
 
         ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
-        if (ret)
-                goto out1;
-
-        /* finally add the DMA device for this PSC */
-        wd->dmapd = au1xpsc_pcm_add(pdev);
-        if (wd->dmapd)
+        if (!ret)
                 return 0;
 
-        snd_soc_unregister_dai(&pdev->dev);
+out2:
+        iounmap(wd->mmio);
 out1:
         release_mem_region(r->start, resource_size(r));
 out0:
@@ -351,9 +366,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
         struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
         struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
-        if (wd->dmapd)
-                au1xpsc_pcm_destroy(wd->dmapd);
-
         snd_soc_unregister_dai(&pdev->dev);
 
         au_writel(0, I2S_CFG(wd));
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index b30eadd422a7..b16b2e02e0c9 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -1,7 +1,7 @@
 /*
- * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ * Alchemy ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * (c) 2007-2011 MSC Vertriebsges.m.b.H.,
  *      Manuel Lauss <manuel.lauss@gmail.com>
  *
  * This program is free software; you can redistribute it and/or modify
@@ -13,10 +13,6 @@
 #ifndef _AU1X_PCM_H
 #define _AU1X_PCM_H
 
-/* DBDMA helpers */
-extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev);
-extern void au1xpsc_pcm_destroy(struct platform_device *dmapd);
-
 struct au1xpsc_audio_data {
         void __iomem *mmio;
 
@@ -27,15 +23,9 @@ struct au1xpsc_audio_data {
 
         unsigned long pm[2];
         struct mutex lock;
-        struct platform_device *dmapd;
+        int dmaids[2];
 };
 
-#define PCM_TX  0
-#define PCM_RX  1
-
-#define SUBSTREAM_TYPE(substream) \
-        ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
-
 /* easy access macros */
 #define PSC_CTRL(x)     ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
 #define PSC_SEL(x)      ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 76258f2a2ffb..666fae6e148d 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -20,6 +20,7 @@
 #include <linux/regulator/driver.h>
 #include <linux/regulator/machine.h>
 #include <linux/regulator/consumer.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/tlv.h>
 #include <sound/pcm.h>
@@ -33,73 +34,31 @@
 #define SGTL5000_DAP_REG_OFFSET 0x0100
 #define SGTL5000_MAX_REG_OFFSET 0x013A
 
-/* default value of sgtl5000 registers except DAP */
-static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] =  {
-        0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */
-        0x0000, /* 0x0002, CHIP_DIG_POWER. */
-        0x0008, /* 0x0004, CHIP_CKL_CTRL */
-        0x0010, /* 0x0006, CHIP_I2S_CTRL */
-        0x0000, /* 0x0008, reserved */
-        0x0008, /* 0x000A, CHIP_SSS_CTRL */
-        0x0000, /* 0x000C, reserved */
-        0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */
-        0x3c3c, /* 0x0010, CHIP_DAC_VOL */
-        0x0000, /* 0x0012, reserved */
-        0x015f, /* 0x0014, CHIP_PAD_STRENGTH */
-        0x0000, /* 0x0016, reserved */
-        0x0000, /* 0x0018, reserved */
-        0x0000, /* 0x001A, reserved */
-        0x0000, /* 0x001E, reserved */
-        0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */
-        0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */
-        0x0111, /* 0x0024, CHIP_ANN_CTRL */
-        0x0000, /* 0x0026, CHIP_LINREG_CTRL */
-        0x0000, /* 0x0028, CHIP_REF_CTRL */
-        0x0000, /* 0x002A, CHIP_MIC_CTRL */
-        0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */
-        0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */
-        0x7060, /* 0x0030, CHIP_ANA_POWER */
-        0x5000, /* 0x0032, CHIP_PLL_CTRL */
-        0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */
-        0x0000, /* 0x0036, CHIP_ANA_STATUS */
-        0x0000, /* 0x0038, reserved */
-        0x0000, /* 0x003A, CHIP_ANA_TEST2 */
-        0x0000, /* 0x003C, CHIP_SHORT_CTRL */
-        0x0000, /* reserved */
-};
-
-/* default value of dap registers */
-static const u16 sgtl5000_dap_regs[] = {
-        0x0000, /* 0x0100, DAP_CONTROL */
-        0x0000, /* 0x0102, DAP_PEQ */
-        0x0040, /* 0x0104, DAP_BASS_ENHANCE */
-        0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */
-        0x0000, /* 0x0108, DAP_AUDIO_EQ */
-        0x0040, /* 0x010A, DAP_SGTL_SURROUND */
-        0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */
-        0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */
-        0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */
-        0x0000, /* 0x0112, reserved */
-        0x0000, /* 0x0114, reserved */
-        0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */
-        0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */
-        0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */
-        0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */
-        0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */
-        0x8000, /* 0x0120, DAP_MAIN_CHAN */
-        0x0000, /* 0x0122, DAP_MIX_CHAN */
-        0x0510, /* 0x0124, DAP_AVC_CTRL */
-        0x1473, /* 0x0126, DAP_AVC_THRESHOLD */
-        0x0028, /* 0x0128, DAP_AVC_ATTACK */
-        0x0050, /* 0x012A, DAP_AVC_DECAY */
-        0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */
-        0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */
-        0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */
-        0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */
-        0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */
-        0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */
-        0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */
-        0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */
+/* default value of sgtl5000 registers */
+static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] =  {
+        [SGTL5000_CHIP_CLK_CTRL] = 0x0008,
+        [SGTL5000_CHIP_I2S_CTRL] = 0x0010,
+        [SGTL5000_CHIP_SSS_CTRL] = 0x0008,
+        [SGTL5000_CHIP_DAC_VOL] = 0x3c3c,
+        [SGTL5000_CHIP_PAD_STRENGTH] = 0x015f,
+        [SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818,
+        [SGTL5000_CHIP_ANA_CTRL] = 0x0111,
+        [SGTL5000_CHIP_LINE_OUT_VOL] = 0x0404,
+        [SGTL5000_CHIP_ANA_POWER] = 0x7060,
+        [SGTL5000_CHIP_PLL_CTRL] = 0x5000,
+        [SGTL5000_DAP_BASS_ENHANCE] = 0x0040,
+        [SGTL5000_DAP_BASS_ENHANCE_CTRL] = 0x051f,
+        [SGTL5000_DAP_SURROUND] = 0x0040,
+        [SGTL5000_DAP_EQ_BASS_BAND0] = 0x002f,
+        [SGTL5000_DAP_EQ_BASS_BAND1] = 0x002f,
+        [SGTL5000_DAP_EQ_BASS_BAND2] = 0x002f,
+        [SGTL5000_DAP_EQ_BASS_BAND3] = 0x002f,
+        [SGTL5000_DAP_EQ_BASS_BAND4] = 0x002f,
+        [SGTL5000_DAP_MAIN_CHAN] = 0x8000,
+        [SGTL5000_DAP_AVC_CTRL] = 0x0510,
+        [SGTL5000_DAP_AVC_THRESHOLD] = 0x1473,
+        [SGTL5000_DAP_AVC_ATTACK] = 0x0028,
+        [SGTL5000_DAP_AVC_DECAY] = 0x0050,
 };
 
 /* regulator supplies for sgtl5000, VDDD is an optional external supply */
@@ -1023,12 +982,10 @@ static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state)
 static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
 {
         u16 *cache = codec->reg_cache;
-        int i;
-        int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1;
+        u16 reg;
 
         /* restore regular registers */
-        for (i = 0; i < regular_regs; i++) {
-                int reg = i << 1;
+        for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) {
 
                 /* this regs depends on the others */
                 if (reg == SGTL5000_CHIP_ANA_POWER ||
@@ -1038,35 +995,31 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
                         reg == SGTL5000_CHIP_CLK_CTRL)
                         continue;
 
-                snd_soc_write(codec, reg, cache[i]);
+                snd_soc_write(codec, reg, cache[reg]);
         }
 
         /* restore dap registers */
-        for (i = SGTL5000_DAP_REG_OFFSET >> 1;
-                        i < SGTL5000_MAX_REG_OFFSET >> 1; i++) {
-                int reg = i << 1;
-
-                snd_soc_write(codec, reg, cache[i]);
-        }
+        for (reg = SGTL5000_DAP_REG_OFFSET; reg < SGTL5000_MAX_REG_OFFSET; reg += 2)
+                snd_soc_write(codec, reg, cache[reg]);
 
         /*
          * restore power and other regs according
          * to set_power() and set_clock()
          */
         snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
-                        cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
+                        cache[SGTL5000_CHIP_LINREG_CTRL]);
 
         snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER,
-                        cache[SGTL5000_CHIP_ANA_POWER >> 1]);
+                        cache[SGTL5000_CHIP_ANA_POWER]);
 
         snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL,
-                        cache[SGTL5000_CHIP_CLK_CTRL >> 1]);
+                        cache[SGTL5000_CHIP_CLK_CTRL]);
 
         snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL,
-                        cache[SGTL5000_CHIP_REF_CTRL >> 1]);
+                        cache[SGTL5000_CHIP_REF_CTRL]);
 
         snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
-                        cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]);
+                        cache[SGTL5000_CHIP_LINE_OUT_CTRL]);
         return 0;
 }
 
@@ -1454,16 +1407,6 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client,
         if (!sgtl5000)
                 return -ENOMEM;
 
-        /*
-         * copy DAP default values to default value array.
-         * sgtl5000 register space has a big hole, merge it
-         * at init phase makes life easy.
-         * FIXME: should we drop 'const' of sgtl5000_regs?
-         */
-        memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)),
-                        sgtl5000_dap_regs,
-                        SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET);
-
         i2c_set_clientdata(client, sgtl5000);
 
         ret = snd_soc_register_codec(&client->dev,
@@ -1494,10 +1437,17 @@ static const struct i2c_device_id sgtl5000_id[] = {
 
 MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
 
+static const struct of_device_id sgtl5000_dt_ids[] = {
+        { .compatible = "fsl,sgtl5000", },
+        { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids);
+
 static struct i2c_driver sgtl5000_i2c_driver = {
         .driver = {
                    .name = "sgtl5000",
                    .owner = THIS_MODULE,
+                   .of_match_table = sgtl5000_dt_ids,
                    },
         .probe = sgtl5000_i2c_probe,
         .remove = __devexit_p(sgtl5000_i2c_remove),
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index bcc208967917..bbcf9ec34759 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -56,8 +56,26 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = {
 };
 
 static int __devinit wm1250_ev1_probe(struct i2c_client *i2c,
-                                      const struct i2c_device_id *id)
+                                      const struct i2c_device_id *i2c_id)
 {
+        int ret, id, board, rev;
+
+        board = i2c_smbus_read_byte_data(i2c, 0);
+        if (board < 0) {
+                dev_err(&i2c->dev, "Failed to read ID: %d\n", ret);
+                return ret;
+        }
+
+        id = (board & 0xfe) >> 2;
+        rev = board & 0x3;
+
+        if (id != 1) {
+                dev_err(&i2c->dev, "Unknown board ID %d\n", id);
+                return -ENODEV;
+        }
+
+        dev_info(&i2c->dev, "revision %d\n", rev);
+
         return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1,
                                       &wm1250_ev1_dai, 1);
 }
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 4fd4d8dca0fc..131200917c56 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -551,7 +551,7 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id);
 
 static struct i2c_driver wm8523_i2c_driver = {
         .driver = {
-                .name = "wm8523-codec",
+                .name = "wm8523",
                 .owner = THIS_MODULE,
         },
         .probe =    wm8523_i2c_probe,
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 4bbc0a79f01e..95ac6651094f 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -943,7 +943,7 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
 static struct i2c_driver wm8580_i2c_driver = {
         .driver = {
-                .name = "wm8580-codec",
+                .name = "wm8580",
                 .owner = THIS_MODULE,
         },
         .probe =    wm8580_i2c_probe,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 76b4361e9b80..f76b6fc6766a 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -22,6 +22,7 @@
 #include <linux/platform_device.h>
 #include <linux/regulator/consumer.h>
 #include <linux/spi/spi.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -607,6 +608,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = {
         .num_dapm_routes = ARRAY_SIZE(wm8731_intercon),
 };
 
+static const struct of_device_id wm8731_of_match[] = {
+        { .compatible = "wlf,wm8731", },
+        { }
+};
+
+MODULE_DEVICE_TABLE(of, wm8731_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8731_spi_probe(struct spi_device *spi)
 {
@@ -638,6 +646,7 @@ static struct spi_driver wm8731_spi_driver = {
         .driver = {
                 .name   = "wm8731",
                 .owner  = THIS_MODULE,
+                .of_match_table = wm8731_of_match,
         },
         .probe          = wm8731_spi_probe,
         .remove         = __devexit_p(wm8731_spi_remove),
@@ -682,6 +691,7 @@ static struct i2c_driver wm8731_i2c_driver = {
         .driver = {
                 .name = "wm8731",
                 .owner = THIS_MODULE,
+                .of_match_table = wm8731_of_match,
         },
         .probe =    wm8731_i2c_probe,
         .remove =   __devexit_p(wm8731_i2c_remove),
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 6e85b8869af7..f014e5676d20 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1433,7 +1433,8 @@ static int wm8993_probe(struct snd_soc_codec *codec)
         int ret, i, val;
 
         wm8993->hubs_data.hp_startup_mode = 1;
-        wm8993->hubs_data.dcs_codes = -2;
+        wm8993->hubs_data.dcs_codes_l = -2;
+        wm8993->hubs_data.dcs_codes_r = -2;
         wm8993->hubs_data.series_startup = 1;
 
         ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 09e680ae88b2..fb5c96163610 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -107,6 +107,7 @@ static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg)
         case WM8994_LDO_2:
         case WM8958_DSP2_EXECCONTROL:
         case WM8958_MIC_DETECT_3:
+        case WM8994_DC_SERVO_4E:
                 return 1;
         default:
                 return 0;
@@ -2972,13 +2973,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
                 switch (wm8994->revision) {
                 case 2:
                 case 3:
-                        wm8994->hubs.dcs_codes = -5;
+                        wm8994->hubs.dcs_codes_l = -5;
+                        wm8994->hubs.dcs_codes_r = -5;
                         wm8994->hubs.hp_startup_mode = 1;
                         wm8994->hubs.dcs_readback_mode = 1;
                         wm8994->hubs.series_startup = 1;
                         break;
                 default:
-                        wm8994->hubs.dcs_readback_mode = 1;
+                        wm8994->hubs.dcs_readback_mode = 2;
                         break;
                 }
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 4cc2d567f22f..017522e7cef9 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -18,6 +18,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/mfd/wm8994/registers.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -116,14 +117,23 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
         struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
         s8 offset;
-        u16 reg, reg_l, reg_r, dcs_cfg;
+        u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
+
+        switch (hubs->dcs_readback_mode) {
+        case 2:
+                dcs_reg = WM8994_DC_SERVO_4E;
+                break;
+        default:
+                dcs_reg = WM8993_DC_SERVO_3;
+                break;
+        }
 
         /* If we're using a digital only path and have a previously
          * callibrated DC servo offset stored then use that. */
         if (hubs->class_w && hubs->class_w_dcs) {
                 dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
                         hubs->class_w_dcs);
-                snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+                snd_soc_write(codec, dcs_reg, hubs->class_w_dcs);
                 wait_for_dc_servo(codec,
                                   WM8993_DCS_TRIG_DAC_WR_0 |
                                   WM8993_DCS_TRIG_DAC_WR_1);
@@ -154,8 +164,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
                 reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
                         & WM8993_DCS_INTEG_CHAN_1_MASK;
                 break;
+        case 2:
         case 1:
-                reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+                reg = snd_soc_read(codec, dcs_reg);
                 reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
                         >> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
                 reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
@@ -168,24 +179,25 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
         dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
 
         /* Apply correction to DC servo result */
-        if (hubs->dcs_codes) {
-                dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
-                        hubs->dcs_codes);
+        if (hubs->dcs_codes_l || hubs->dcs_codes_r) {
+                dev_dbg(codec->dev,
+                        "Applying %d/%d code DC servo correction\n",
+                        hubs->dcs_codes_l, hubs->dcs_codes_r);
 
                 /* HPOUT1R */
                 offset = reg_r;
-                offset += hubs->dcs_codes;
+                offset += hubs->dcs_codes_r;
                 dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
                 /* HPOUT1L */
                 offset = reg_l;
-                offset += hubs->dcs_codes;
+                offset += hubs->dcs_codes_l;
                 dcs_cfg |= (u8)offset;
 
                 dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
 
                 /* Do it */
-                snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
+                snd_soc_write(codec, dcs_reg, dcs_cfg);
                 wait_for_dc_servo(codec,
                                   WM8993_DCS_TRIG_DAC_WR_0 |
                                   WM8993_DCS_TRIG_DAC_WR_1);
@@ -217,7 +229,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 
         /* If we're applying an offset correction then updating the
          * callibration would be likely to introduce further offsets. */
-        if (hubs->dcs_codes || hubs->no_series_update)
+        if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update)
                 return ret;
 
         /* Only need to do this if the outputs are active */
@@ -440,9 +452,8 @@ static int hp_event(struct snd_soc_dapm_widget *w,
                 reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY;
                 snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
 
-                /* Smallest supported update interval */
                 snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
-                                    WM8993_DCS_TIMER_PERIOD_01_MASK, 1);
+                                    WM8993_DCS_TIMER_PERIOD_01_MASK, 0);
 
                 calibrate_dc_servo(codec);
 
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 676b1252ab91..c674c7a502a6 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,7 +23,8 @@ extern const unsigned int wm_hubs_spkmix_tlv[];
 
 /* This *must* be the first element of the codec->private_data struct */
 struct wm_hubs_data {
-        int dcs_codes;
+        int dcs_codes_l;
+        int dcs_codes_r;
         int dcs_readback_mode;
         int hp_startup_mode;
         int series_startup;
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
new file mode 100644
index 000000000000..e4ba8d5f25fa
--- /dev/null
+++ b/sound/soc/mxs/Kconfig
@@ -0,0 +1,20 @@
+menuconfig SND_MXS_SOC
+        tristate "SoC Audio for Freescale MXS CPUs"
+        depends on ARCH_MXS
+        select SND_PCM
+        help
+          Say Y or M if you want to add support for codecs attached to
+          the MXS SAIF interface.
+
+
+if SND_MXS_SOC
+
+config SND_SOC_MXS_SGTL5000
+        tristate "SoC Audio support for i.MX boards with sgtl5000"
+        depends on I2C
+        select SND_SOC_SGTL5000
+        help
+          Say Y if you want to add support for SoC audio on an MXS board with
+          a sgtl5000 codec.
+
+endif   # SND_MXS_SOC
diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile
new file mode 100644
index 000000000000..565b5b51e8b7
--- /dev/null
+++ b/sound/soc/mxs/Makefile
@@ -0,0 +1,10 @@
+# MXS Platform Support
+snd-soc-mxs-objs := mxs-saif.o
+snd-soc-mxs-pcm-objs := mxs-pcm.o
+
+obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o
+
+# i.MX Machine Support
+snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o
+
+obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
new file mode 100644
index 000000000000..dea5aa4aa647
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -0,0 +1,359 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * Based on sound/soc/imx/imx-pcm-dma-mx2.c
+ *
+ * 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/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/dma.h>
+#include "mxs-pcm.h"
+
+static struct snd_pcm_hardware snd_mxs_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,
+        .channels_min           = 2,
+        .channels_max           = 2,
+        .period_bytes_min       = 32,
+        .period_bytes_max       = 8192,
+        .periods_min            = 1,
+        .periods_max            = 52,
+        .buffer_bytes_max       = 64 * 1024,
+        .fifo_size              = 32,
+
+};
+
+static void audio_dma_irq(void *data)
+{
+        struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+        iprtd->offset += iprtd->period_bytes;
+        iprtd->offset %= iprtd->period_bytes * iprtd->periods;
+        snd_pcm_period_elapsed(substream);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+        struct mxs_pcm_runtime_data *iprtd = param;
+        struct mxs_pcm_dma_params *dma_params = iprtd->dma_params;
+
+        if (!mxs_dma_is_apbx(chan))
+                return false;
+
+        if (chan->chan_id != dma_params->chan_num)
+                return false;
+
+        chan->private = &iprtd->dma_data;
+
+        return true;
+}
+
+static int mxs_dma_alloc(struct snd_pcm_substream *substream,
+                                struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+        dma_cap_mask_t mask;
+
+        iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+        dma_cap_zero(mask);
+        dma_cap_set(DMA_SLAVE, mask);
+        iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq;
+        iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+        if (!iprtd->dma_chan)
+                return -EINVAL;
+
+        return 0;
+}
+
+static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
+                                struct snd_pcm_hw_params *params)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+        unsigned long dma_addr;
+        struct dma_chan *chan;
+        int ret;
+
+        ret = mxs_dma_alloc(substream, params);
+        if (ret)
+                return ret;
+        chan = iprtd->dma_chan;
+
+        iprtd->size = params_buffer_bytes(params);
+        iprtd->periods = params_periods(params);
+        iprtd->period_bytes = params_period_bytes(params);
+        iprtd->offset = 0;
+        iprtd->period_time = HZ / (params_rate(params) /
+                        params_period_size(params));
+
+        snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+        dma_addr = runtime->dma_addr;
+
+        iprtd->buf = substream->dma_buffer.area;
+
+        iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+                        iprtd->period_bytes * iprtd->periods,
+                        iprtd->period_bytes,
+                        substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+                        DMA_TO_DEVICE : DMA_FROM_DEVICE);
+        if (!iprtd->desc) {
+                dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+                return -EINVAL;
+        }
+
+        iprtd->desc->callback = audio_dma_irq;
+        iprtd->desc->callback_param = substream;
+
+        return 0;
+}
+
+static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+        if (iprtd->dma_chan) {
+                dma_release_channel(iprtd->dma_chan);
+                iprtd->dma_chan = NULL;
+        }
+
+        return 0;
+}
+
+static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                dmaengine_submit(iprtd->desc);
+
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                dmaengine_terminate_all(iprtd->dma_chan);
+
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static snd_pcm_uframes_t snd_mxs_pcm_pointer(
+                struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+        return bytes_to_frames(substream->runtime, iprtd->offset);
+}
+
+static int snd_mxs_open(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd;
+        int ret;
+
+        iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+        if (iprtd == NULL)
+                return -ENOMEM;
+        runtime->private_data = iprtd;
+
+        ret = snd_pcm_hw_constraint_integer(substream->runtime,
+                        SNDRV_PCM_HW_PARAM_PERIODS);
+        if (ret < 0) {
+                kfree(iprtd);
+                return ret;
+        }
+
+        snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware);
+
+        return 0;
+}
+
+static int snd_mxs_close(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+        kfree(iprtd);
+
+        return 0;
+}
+
+static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream,
+                struct vm_area_struct *vma)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+
+        return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+                                        runtime->dma_area,
+                                        runtime->dma_addr,
+                                        runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops mxs_pcm_ops = {
+        .open           = snd_mxs_open,
+        .close          = snd_mxs_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = snd_mxs_pcm_hw_params,
+        .hw_free        = snd_mxs_pcm_hw_free,
+        .trigger        = snd_mxs_pcm_trigger,
+        .pointer        = snd_mxs_pcm_pointer,
+        .mmap           = snd_mxs_pcm_mmap,
+};
+
+static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+        struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+        struct snd_dma_buffer *buf = &substream->dma_buffer;
+        size_t size = snd_mxs_hardware.buffer_bytes_max;
+
+        buf->dev.type = SNDRV_DMA_TYPE_DEV;
+        buf->dev.dev = pcm->card->dev;
+        buf->private_data = NULL;
+        buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+                                           &buf->addr, GFP_KERNEL);
+        if (!buf->area)
+                return -ENOMEM;
+        buf->bytes = size;
+
+        return 0;
+}
+
+static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32);
+static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_card *card = rtd->card->snd_card;
+        struct snd_pcm *pcm = rtd->pcm;
+        int ret = 0;
+
+        if (!card->dev->dma_mask)
+                card->dev->dma_mask = &mxs_pcm_dmamask;
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+        if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+                ret = mxs_pcm_preallocate_dma_buffer(pcm,
+                        SNDRV_PCM_STREAM_PLAYBACK);
+                if (ret)
+                        goto out;
+        }
+
+        if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+                ret = mxs_pcm_preallocate_dma_buffer(pcm,
+                        SNDRV_PCM_STREAM_CAPTURE);
+                if (ret)
+                        goto out;
+        }
+
+out:
+        return ret;
+}
+
+static void mxs_pcm_free(struct snd_pcm *pcm)
+{
+        struct snd_pcm_substream *substream;
+        struct snd_dma_buffer *buf;
+        int stream;
+
+        for (stream = 0; stream < 2; stream++) {
+                substream = pcm->streams[stream].substream;
+                if (!substream)
+                        continue;
+
+                buf = &substream->dma_buffer;
+                if (!buf->area)
+                        continue;
+
+                dma_free_writecombine(pcm->card->dev, buf->bytes,
+                                      buf->area, buf->addr);
+                buf->area = NULL;
+        }
+}
+
+static struct snd_soc_platform_driver mxs_soc_platform = {
+        .ops            = &mxs_pcm_ops,
+        .pcm_new        = mxs_pcm_new,
+        .pcm_free       = mxs_pcm_free,
+};
+
+static int __devinit mxs_soc_platform_probe(struct platform_device *pdev)
+{
+        return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform);
+}
+
+static int __devexit mxs_soc_platform_remove(struct platform_device *pdev)
+{
+        snd_soc_unregister_platform(&pdev->dev);
+
+        return 0;
+}
+
+static struct platform_driver mxs_pcm_driver = {
+        .driver = {
+                .name = "mxs-pcm-audio",
+                .owner = THIS_MODULE,
+        },
+        .probe = mxs_soc_platform_probe,
+        .remove = __devexit_p(mxs_soc_platform_remove),
+};
+
+static int __init snd_mxs_pcm_init(void)
+{
+        return platform_driver_register(&mxs_pcm_driver);
+}
+module_init(snd_mxs_pcm_init);
+
+static void __exit snd_mxs_pcm_exit(void)
+{
+        platform_driver_unregister(&mxs_pcm_driver);
+}
+module_exit(snd_mxs_pcm_exit);
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
new file mode 100644
index 000000000000..f55ac4f7a76a
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2011 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
+
+#include <mach/dma.h>
+
+struct mxs_pcm_dma_params {
+        int chan_irq;
+        int chan_num;
+};
+
+struct mxs_pcm_runtime_data {
+        int period_bytes;
+        int periods;
+        int dma;
+        unsigned long offset;
+        unsigned long size;
+        void *buf;
+        int period_time;
+        struct dma_async_tx_descriptor *desc;
+        struct dma_chan *dma_chan;
+        struct mxs_dma_data dma_data;
+        struct mxs_pcm_dma_params *dma_params;
+};
+
+#endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
new file mode 100644
index 000000000000..0b3adaec9f4c
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.c
@@ -0,0 +1,677 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * 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/mxs.h>
+
+#include "mxs-saif.h"
+
+static struct mxs_saif *mxs_saif[2];
+
+static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+                        int clk_id, unsigned int freq, int dir)
+{
+        struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+        switch (clk_id) {
+        case MXS_SAIF_MCLK:
+                saif->mclk = freq;
+                break;
+        default:
+                return -EINVAL;
+        }
+        return 0;
+}
+
+/*
+ * Set SAIF clock and MCLK
+ */
+static int mxs_saif_set_clk(struct mxs_saif *saif,
+                                  unsigned int mclk,
+                                  unsigned int rate)
+{
+        u32 scr;
+        int ret;
+
+        scr = __raw_readl(saif->base + SAIF_CTRL);
+        scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE;
+        scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+
+        /*
+         * Set SAIF clock
+         *
+         * The SAIF clock should be either 384*fs or 512*fs.
+         * If MCLK is used, the SAIF clk ratio need to match mclk ratio.
+         *  For 32x mclk, set saif clk as 512*fs.
+         *  For 48x mclk, set saif clk as 384*fs.
+         *
+         * If MCLK is not used, we just set saif clk to 512*fs.
+         */
+        if (saif->mclk_in_use) {
+                if (mclk % 32 == 0) {
+                        scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+                        ret = clk_set_rate(saif->clk, 512 * rate);
+                } else if (mclk % 48 == 0) {
+                        scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+                        ret = clk_set_rate(saif->clk, 384 * rate);
+                } else {
+                        /* SAIF MCLK should be either 32x or 48x */
+                        return -EINVAL;
+                }
+        } else {
+                ret = clk_set_rate(saif->clk, 512 * rate);
+                scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+        }
+
+        if (ret)
+                return ret;
+
+        if (!saif->mclk_in_use) {
+                __raw_writel(scr, saif->base + SAIF_CTRL);
+                return 0;
+        }
+
+        /*
+         * Program the over-sample rate for MCLK output
+         *
+         * The available MCLK range is 32x, 48x... 512x. The rate
+         * could be from 8kHz to 192kH.
+         */
+        switch (mclk / rate) {
+        case 32:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4);
+                break;
+        case 64:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+                break;
+        case 128:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+                break;
+        case 256:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+                break;
+        case 512:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+                break;
+        case 48:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+                break;
+        case 96:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+                break;
+        case 192:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+                break;
+        case 384:
+                scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        __raw_writel(scr, saif->base + SAIF_CTRL);
+
+        return 0;
+}
+
+/*
+ * Put and disable MCLK.
+ */
+int mxs_saif_put_mclk(unsigned int saif_id)
+{
+        struct mxs_saif *saif = mxs_saif[saif_id];
+        u32 stat;
+
+        if (!saif)
+                return -EINVAL;
+
+        stat = __raw_readl(saif->base + SAIF_STAT);
+        if (stat & BM_SAIF_STAT_BUSY) {
+                dev_err(saif->dev, "error: busy\n");
+                return -EBUSY;
+        }
+
+        clk_disable(saif->clk);
+
+        /* disable MCLK output */
+        __raw_writel(BM_SAIF_CTRL_CLKGATE,
+                saif->base + SAIF_CTRL + MXS_SET_ADDR);
+        __raw_writel(BM_SAIF_CTRL_RUN,
+                saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+        saif->mclk_in_use = 0;
+        return 0;
+}
+
+/*
+ * Get MCLK and set clock rate, then enable it
+ *
+ * This interface is used for codecs who are using MCLK provided
+ * by saif.
+ */
+int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+                                        unsigned int rate)
+{
+        struct mxs_saif *saif = mxs_saif[saif_id];
+        u32 stat;
+        int ret;
+
+        if (!saif)
+                return -EINVAL;
+
+        stat = __raw_readl(saif->base + SAIF_STAT);
+        if (stat & BM_SAIF_STAT_BUSY) {
+                dev_err(saif->dev, "error: busy\n");
+                return -EBUSY;
+        }
+
+        /* Clear Reset */
+        __raw_writel(BM_SAIF_CTRL_SFTRST,
+                saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+        saif->mclk_in_use = 1;
+        ret = mxs_saif_set_clk(saif, mclk, rate);
+        if (ret)
+                return ret;
+
+        ret = clk_enable(saif->clk);
+        if (ret)
+                return ret;
+
+        /* enable MCLK output */
+        __raw_writel(BM_SAIF_CTRL_CLKGATE,
+                saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+        __raw_writel(BM_SAIF_CTRL_RUN,
+                saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+        return 0;
+}
+
+/*
+ * SAIF DAI format configuration.
+ * Should only be called when port is inactive.
+ */
+static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+        u32 scr, stat;
+        u32 scr0;
+        struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+        stat = __raw_readl(saif->base + SAIF_STAT);
+        if (stat & BM_SAIF_STAT_BUSY) {
+                dev_err(cpu_dai->dev, "error: busy\n");
+                return -EBUSY;
+        }
+
+        scr0 = __raw_readl(saif->base + SAIF_CTRL);
+        scr0 = scr0 & ~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;
+        default:
+                return -EINVAL;
+        }
+
+        /* 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;
+        }
+
+        /*
+         * Note: We simply just support master mode since SAIF TX can only
+         * work as master.
+         */
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBS_CFS:
+                scr &= ~BM_SAIF_CTRL_SLAVE_MODE;
+                __raw_writel(scr | scr0, saif->base + SAIF_CTRL);
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static int mxs_saif_startup(struct snd_pcm_substream *substream,
+                           struct snd_soc_dai *cpu_dai)
+{
+        struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+        snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param);
+
+        /* clear error status to 0 for each re-open */
+        saif->fifo_underrun = 0;
+        saif->fifo_overrun = 0;
+
+        /* Clear Reset for normal operations */
+        __raw_writel(BM_SAIF_CTRL_SFTRST,
+                saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+        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)
+{
+        struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+        u32 scr, stat;
+        int ret;
+
+        /* mclk should already be set */
+        if (!saif->mclk && saif->mclk_in_use) {
+                dev_err(cpu_dai->dev, "set mclk first\n");
+                return -EINVAL;
+        }
+
+        stat = __raw_readl(saif->base + SAIF_STAT);
+        if (stat & BM_SAIF_STAT_BUSY) {
+                dev_err(cpu_dai->dev, "error: busy\n");
+                return -EBUSY;
+        }
+
+        /*
+         * Set saif clk based on sample rate.
+         * If mclk is used, we also set mclk, if not, saif->mclk is
+         * default 0, means not used.
+         */
+        ret = mxs_saif_set_clk(saif, saif->mclk, params_rate(params));
+        if (ret) {
+                dev_err(cpu_dai->dev, "unable to get proper clk\n");
+                return ret;
+        }
+
+        scr = __raw_readl(saif->base + SAIF_CTRL);
+
+        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;
+        default:
+                return -EINVAL;
+        }
+
+        /* 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;
+        }
+
+        __raw_writel(scr, saif->base + SAIF_CTRL);
+        return 0;
+}
+
+static int mxs_saif_prepare(struct snd_pcm_substream *substream,
+                           struct snd_soc_dai *cpu_dai)
+{
+        struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+        /* clear clock gate */
+        __raw_writel(BM_SAIF_CTRL_CLKGATE,
+                saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+        /* enable FIFO error irqs */
+        __raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN,
+                saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+        return 0;
+}
+
+static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
+                                struct snd_soc_dai *cpu_dai)
+{
+        struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                dev_dbg(cpu_dai->dev, "start\n");
+
+                clk_enable(saif->clk);
+                if (!saif->mclk_in_use)
+                        __raw_writel(BM_SAIF_CTRL_RUN,
+                                saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                        /*
+                         * write a data to saif data register to trigger
+                         * the transfer
+                         */
+                        __raw_writel(0, saif->base + SAIF_DATA);
+                } else {
+                        /*
+                         * read a data from saif data register to trigger
+                         * the receive
+                         */
+                        __raw_readl(saif->base + SAIF_DATA);
+                }
+
+                dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n",
+                        __raw_readl(saif->base + SAIF_CTRL),
+                        __raw_readl(saif->base + SAIF_STAT));
+
+                break;
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                dev_dbg(cpu_dai->dev, "stop\n");
+
+                clk_disable(saif->clk);
+                if (!saif->mclk_in_use)
+                        __raw_writel(BM_SAIF_CTRL_RUN,
+                                saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+#define MXS_SAIF_RATES          SNDRV_PCM_RATE_8000_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,
+        .trigger = mxs_saif_trigger,
+        .prepare = mxs_saif_prepare,
+        .hw_params = mxs_saif_hw_params,
+        .set_sysclk = mxs_saif_set_dai_sysclk,
+        .set_fmt = mxs_saif_set_dai_fmt,
+};
+
+static int mxs_saif_dai_probe(struct snd_soc_dai *dai)
+{
+        struct mxs_saif *saif = dev_get_drvdata(dai->dev);
+
+        snd_soc_dai_set_drvdata(dai, saif);
+
+        return 0;
+}
+
+static struct snd_soc_dai_driver mxs_saif_dai = {
+        .name = "mxs-saif",
+        .probe = mxs_saif_dai_probe,
+        .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,
+};
+
+static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
+{
+        struct mxs_saif *saif = dev_id;
+        unsigned int stat;
+
+        stat = __raw_readl(saif->base + SAIF_STAT);
+        if (!(stat & (BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ |
+                        BM_SAIF_STAT_FIFO_OVERFLOW_IRQ)))
+                return IRQ_NONE;
+
+        if (stat & BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ) {
+                dev_dbg(saif->dev, "underrun!!! %d\n", ++saif->fifo_underrun);
+                __raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ,
+                                saif->base + SAIF_STAT + MXS_CLR_ADDR);
+        }
+
+        if (stat & BM_SAIF_STAT_FIFO_OVERFLOW_IRQ) {
+                dev_dbg(saif->dev, "overrun!!! %d\n", ++saif->fifo_overrun);
+                __raw_writel(BM_SAIF_STAT_FIFO_OVERFLOW_IRQ,
+                                saif->base + SAIF_STAT + MXS_CLR_ADDR);
+        }
+
+        dev_dbg(saif->dev, "SAIF_CTRL %x SAIF_STAT %x\n",
+               __raw_readl(saif->base + SAIF_CTRL),
+               __raw_readl(saif->base + SAIF_STAT));
+
+        return IRQ_HANDLED;
+}
+
+static int mxs_saif_probe(struct platform_device *pdev)
+{
+        struct resource *res;
+        struct mxs_saif *saif;
+        int ret = 0;
+
+        saif = kzalloc(sizeof(*saif), GFP_KERNEL);
+        if (!saif)
+                return -ENOMEM;
+
+        if (pdev->id >= ARRAY_SIZE(mxs_saif))
+                return -EINVAL;
+        mxs_saif[pdev->id] = saif;
+
+        saif->clk = clk_get(&pdev->dev, NULL);
+        if (IS_ERR(saif->clk)) {
+                ret = PTR_ERR(saif->clk);
+                dev_err(&pdev->dev, "Cannot get the clock: %d\n",
+                        ret);
+                goto failed_clk;
+        }
+
+        res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!res) {
+                ret = -ENODEV;
+                dev_err(&pdev->dev, "failed to get io resource: %d\n",
+                        ret);
+                goto failed_get_resource;
+        }
+
+        if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) {
+                dev_err(&pdev->dev, "request_mem_region failed\n");
+                ret = -EBUSY;
+                goto failed_get_resource;
+        }
+
+        saif->base = ioremap(res->start, resource_size(res));
+        if (!saif->base) {
+                dev_err(&pdev->dev, "ioremap failed\n");
+                ret = -ENODEV;
+                goto failed_ioremap;
+        }
+
+        res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!res) {
+                ret = -ENODEV;
+                dev_err(&pdev->dev, "failed to get dma resource: %d\n",
+                        ret);
+                goto failed_ioremap;
+        }
+        saif->dma_param.chan_num = res->start;
+
+        saif->irq = platform_get_irq(pdev, 0);
+        if (saif->irq < 0) {
+                ret = saif->irq;
+                dev_err(&pdev->dev, "failed to get irq resource: %d\n",
+                        ret);
+                goto failed_get_irq1;
+        }
+
+        saif->dev = &pdev->dev;
+        ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif);
+        if (ret) {
+                dev_err(&pdev->dev, "failed to request irq\n");
+                goto failed_get_irq1;
+        }
+
+        saif->dma_param.chan_irq = platform_get_irq(pdev, 1);
+        if (saif->dma_param.chan_irq < 0) {
+                ret = saif->dma_param.chan_irq;
+                dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
+                        ret);
+                goto failed_get_irq2;
+        }
+
+        platform_set_drvdata(pdev, saif);
+
+        ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai);
+        if (ret) {
+                dev_err(&pdev->dev, "register DAI failed\n");
+                goto failed_register;
+        }
+
+        saif->soc_platform_pdev = platform_device_alloc(
+                                        "mxs-pcm-audio", pdev->id);
+        if (!saif->soc_platform_pdev) {
+                ret = -ENOMEM;
+                goto failed_pdev_alloc;
+        }
+
+        platform_set_drvdata(saif->soc_platform_pdev, saif);
+        ret = platform_device_add(saif->soc_platform_pdev);
+        if (ret) {
+                dev_err(&pdev->dev, "failed to add soc platform device\n");
+                goto failed_pdev_add;
+        }
+
+        return 0;
+
+failed_pdev_add:
+        platform_device_put(saif->soc_platform_pdev);
+failed_pdev_alloc:
+        snd_soc_unregister_dai(&pdev->dev);
+failed_register:
+failed_get_irq2:
+        free_irq(saif->irq, saif);
+failed_get_irq1:
+        iounmap(saif->base);
+failed_ioremap:
+        release_mem_region(res->start, resource_size(res));
+failed_get_resource:
+        clk_put(saif->clk);
+failed_clk:
+        kfree(saif);
+
+        return ret;
+}
+
+static int __devexit mxs_saif_remove(struct platform_device *pdev)
+{
+        struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        struct mxs_saif *saif = platform_get_drvdata(pdev);
+
+        platform_device_unregister(saif->soc_platform_pdev);
+
+        snd_soc_unregister_dai(&pdev->dev);
+
+        iounmap(saif->base);
+        release_mem_region(res->start, resource_size(res));
+        free_irq(saif->irq, saif);
+
+        clk_put(saif->clk);
+        kfree(saif);
+
+        return 0;
+}
+
+static struct platform_driver mxs_saif_driver = {
+        .probe = mxs_saif_probe,
+        .remove = __devexit_p(mxs_saif_remove),
+
+        .driver = {
+                .name = "mxs-saif",
+                .owner = THIS_MODULE,
+        },
+};
+
+static int __init mxs_saif_init(void)
+{
+        return platform_driver_register(&mxs_saif_driver);
+}
+
+static void __exit mxs_saif_exit(void)
+{
+        platform_driver_unregister(&mxs_saif_driver);
+}
+
+module_init(mxs_saif_init);
+module_exit(mxs_saif_exit);
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ASoC SAIF driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
new file mode 100644
index 000000000000..0e2ff8cdbfee
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2011 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
+
+#define SAIF_CTRL       0x0
+#define SAIF_STAT       0x10
+#define SAIF_DATA       0x20
+#define SAIF_VERSION    0X30
+
+/* SAIF_CTRL */
+#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 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
+
+/* SAIF_STAT */
+#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
+
+/* SAFI_DATA */
+#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)
+
+/* SAIF_VERSION */
+#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)
+
+#define MXS_SAIF_MCLK           0
+
+#include "mxs-pcm.h"
+
+struct mxs_saif {
+        struct device *dev;
+        struct clk *clk;
+        unsigned int mclk;
+        unsigned int mclk_in_use;
+        void __iomem *base;
+        int irq;
+        struct mxs_pcm_dma_params dma_param;
+
+        struct platform_device *soc_platform_pdev;
+        u32 fifo_underrun;
+        u32 fifo_overrun;
+};
+
+extern int mxs_saif_put_mclk(unsigned int saif_id);
+extern int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+                                        unsigned int rate);
+#endif
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
new file mode 100644
index 000000000000..a0d89c93df0f
--- /dev/null
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -0,0 +1,165 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * 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/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/sgtl5000.h"
+#include "mxs-saif.h"
+
+static int mxs_sgtl5000_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 *codec_dai = rtd->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+        unsigned int rate = params_rate(params);
+        u32 dai_format, mclk;
+        int ret;
+
+        /* sgtl5000 does not support 512*rate when in 96000 fs */
+        switch (rate) {
+        case 96000:
+                mclk = 256 * rate;
+                break;
+        default:
+                mclk = 512 * rate;
+                break;
+        }
+
+        /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */
+        if (mclk < 8000000 || mclk > 27000000)
+                return -EINVAL;
+
+        /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
+        ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
+        if (ret)
+                return ret;
+
+        /* The SAIF MCLK should be the same as SGTL5000_SYSCLK */
+        ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0);
+        if (ret)
+                return ret;
+
+        /* 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)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
+        if (ret)
+                return ret;
+
+        return 0;
+}
+
+static struct snd_soc_ops mxs_sgtl5000_hifi_ops = {
+        .hw_params = mxs_sgtl5000_hw_params,
+};
+
+static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
+        {
+                .name           = "HiFi",
+                .stream_name    = "HiFi Playback",
+                .codec_dai_name = "sgtl5000",
+                .codec_name     = "sgtl5000.0-000a",
+                .cpu_dai_name   = "mxs-saif.0",
+                .platform_name  = "mxs-pcm-audio.0",
+                .ops            = &mxs_sgtl5000_hifi_ops,
+        },
+};
+
+static struct snd_soc_card mxs_sgtl5000 = {
+        .name           = "mxs_sgtl5000",
+        .dai_link       = mxs_sgtl5000_dai,
+        .num_links      = ARRAY_SIZE(mxs_sgtl5000_dai),
+};
+
+static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev)
+{
+        struct snd_soc_card *card = &mxs_sgtl5000;
+        int ret;
+
+        /*
+         * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
+         * The Sgtl5000 sysclk is derived from saif0 mclk and it's range
+         * should be >= 8MHz and <= 27M.
+         */
+        ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
+        if (ret)
+                return ret;
+
+        card->dev = &pdev->dev;
+        platform_set_drvdata(pdev, card);
+
+        ret = snd_soc_register_card(card);
+        if (ret) {
+                dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+                        ret);
+                return ret;
+        }
+
+        return 0;
+}
+
+static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev)
+{
+        struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+        mxs_saif_put_mclk(0);
+
+        snd_soc_unregister_card(card);
+
+        return 0;
+}
+
+static struct platform_driver mxs_sgtl5000_audio_driver = {
+        .driver = {
+                .name = "mxs-sgtl5000",
+                .owner = THIS_MODULE,
+        },
+        .probe = mxs_sgtl5000_probe,
+        .remove = __devexit_p(mxs_sgtl5000_remove),
+};
+
+static int __init mxs_sgtl5000_init(void)
+{
+        return platform_driver_register(&mxs_sgtl5000_audio_driver);
+}
+module_init(mxs_sgtl5000_init);
+
+static void __exit mxs_sgtl5000_exit(void)
+{
+        platform_driver_unregister(&mxs_sgtl5000_audio_driver);
+}
+module_exit(mxs_sgtl5000_exit);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca27490..ae93aa81244c 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -105,7 +105,7 @@ static int format_register_str(struct snd_soc_codec *codec,
         if (wordsize + regsize + 2 + 1 != len)
                 return -EINVAL;
 
-        ret = snd_soc_read(codec , reg);
+        ret = snd_soc_read(codec, reg);
         if (ret < 0) {
                 memset(regbuf, 'X', regsize);
                 regbuf[regsize] = '\0';
@@ -3141,6 +3141,7 @@ int snd_soc_register_platform(struct device *dev,
         platform->driver = platform_drv;
         platform->dapm.dev = dev;
         platform->dapm.platform = platform;
+        platform->dapm.stream_event = platform_drv->stream_event;
 
         mutex_lock(&client_mutex);
         list_add(&platform->list, &platform_list);
@@ -3253,6 +3254,7 @@ int snd_soc_register_codec(struct device *dev,
         codec->dapm.dev = dev;
         codec->dapm.codec = codec;
         codec->dapm.seq_notifier = codec_drv->seq_notifier;
+        codec->dapm.stream_event = codec_drv->stream_event;
         codec->dev = dev;
         codec->driver = codec_drv;
         codec->num_dai = num_dai;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7e15914b3633..c26531132c66 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2584,7 +2584,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
         {
                 if (!w->sname || w->dapm != dapm)
                         continue;
-                dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
+                dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
                         w->name, w->sname, stream, event);
                 if (strstr(w->sname, stream)) {
                         switch(event) {
@@ -2604,6 +2604,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
         }
 
         dapm_power_widgets(dapm, event);
+
+        /* do we need to notify any clients that DAPM stream is complete */
+        if (dapm->stream_event)
+                dapm->stream_event(dapm, event);
 }
 
 /**