ASoC: Alchemy AC97C/I2SC audio support

This patch adds ASoC support for the AC97 and I2S controllers
on the old Au1000/Au1500/Au1100 chips,

AC97 Tested on a Db1500.  I2S untested since none of the boards
actually have an I2S codec wired up (just test pins).

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

6 files changed, 1126 insertions, 9 deletions

diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 4b67140fdec3..0460b428862c 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -18,6 +18,25 @@ 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
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 16873076e8c4..ff5531eee613 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -3,9 +3,17 @@ 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-db1200-objs := 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/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.h b/sound/soc/au1x/psc.h
index b30eadd422a7..c59b9e544e72 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
@@ -13,7 +13,13 @@
 #ifndef _AU1X_PCM_H
 #define _AU1X_PCM_H
 
-/* DBDMA helpers */
+#define PCM_TX  0
+#define PCM_RX  1
+
+#define SUBSTREAM_TYPE(substream) \
+        ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
+
+/* PSC/DBDMA helpers */
 extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev);
 extern void au1xpsc_pcm_destroy(struct platform_device *dmapd);
 
@@ -27,15 +33,10 @@ struct au1xpsc_audio_data {
 
         unsigned long pm[2];
         struct mutex lock;
+        int dmaids[2];
         struct platform_device *dmapd;
 };
 
-#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)