Adding upstream version 4.19.249.upstream/4.19.249

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

11 files changed, 2677 insertions, 0 deletions

diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 000000000..a56104040
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,64 @@
+##
+## Au1200/Au1550/Au1300 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+	tristate "SoC Audio for Au12xx/Au13xx/Au1550"
+	depends on MIPS_ALCHEMY
+	help
+	  This option enables support for the Programmable Serial
+	  Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
+	  Controller (DBDMA) as found on the Au12xx/Au13xx/Au1550 SoC.
+
+config SND_SOC_AU1XPSC_I2S
+	tristate
+
+config SND_SOC_AU1XPSC_AC97
+	tristate
+	select AC97_BUS
+	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/DB1300/DB1550 Audio support"
+	depends on SND_SOC_AU1XPSC
+	select SND_SOC_AU1XPSC_AC97
+	select SND_SOC_AC97_CODEC
+	select SND_SOC_WM9712
+	select SND_SOC_AU1XPSC_I2S
+	select SND_SOC_WM8731
+	help
+	  Select this option to enable audio (AC97 and I2S) on the
+	  Alchemy/AMD/RMI/NetLogic Db1200, Db1550 and Db1300 evaluation boards.
+	  If you need Db1300 touchscreen support, you definitely want to say Y.
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 000000000..33183d7fe
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -0,0 +1,24 @@
+# SPDX-License-Identifier: GPL-2.0
+# Au1200/Au1550 PSC audio
+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 000000000..66d6c52e7
--- /dev/null
+++ b/sound/soc/au1x/ac97c.c
@@ -0,0 +1,347 @@
+/*
+ * 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 = 6;
+		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 */
+static struct snd_ac97_bus_ops ac97c_bus_ops = {
+	.read		= au1xac97c_ac97_read,
+	.write		= au1xac97c_ac97_write,
+	.reset		= au1xac97c_ac97_cold_reset,
+	.warm_reset	= au1xac97c_ac97_warm_reset,
+};
+
+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 const 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",
+	.bus_control		= true,
+	.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 const struct snd_soc_component_driver au1xac97c_component = {
+	.name		= "au1xac97c",
+};
+
+static int au1xac97c_drvprobe(struct platform_device *pdev)
+{
+	int ret;
+	struct resource *iores, *dmares;
+	struct au1xpsc_audio_data *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	mutex_init(&ctx->lock);
+
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!iores)
+		return -ENODEV;
+
+	if (!devm_request_mem_region(&pdev->dev, iores->start,
+				     resource_size(iores),
+				     pdev->name))
+		return -EBUSY;
+
+	ctx->mmio = devm_ioremap_nocache(&pdev->dev, iores->start,
+					 resource_size(iores));
+	if (!ctx->mmio)
+		return -EBUSY;
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmares)
+		return -EBUSY;
+	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!dmares)
+		return -EBUSY;
+	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->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_set_ac97_ops(&ac97c_bus_ops);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component,
+					 &au1xac97c_dai_driver, 1);
+	if (ret)
+		return ret;
+
+	ac97c_workdata = ctx;
+	return 0;
+}
+
+static int au1xac97c_drvremove(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_component(&pdev->dev);
+
+	WR(ctx, AC97_ENABLE, EN_D);	/* clock off, disable */
+
+	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",
+		.pm	= AU1XPSCAC97_PMOPS,
+	},
+	.probe		= au1xac97c_drvprobe,
+	.remove		= au1xac97c_drvremove,
+};
+
+module_platform_driver(au1xac97c_driver);
+
+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 000000000..e97c32798
--- /dev/null
+++ b/sound/soc/au1x/db1000.c
@@ -0,0 +1,56 @@
+/*
+ * 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",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1000_ac97_dai,
+	.num_links	= 1,
+};
+
+static int db1000_audio_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &db1000_ac97;
+	card->dev = &pdev->dev;
+	return devm_snd_soc_register_card(&pdev->dev, card);
+}
+
+static struct platform_driver db1000_audio_driver = {
+	.driver	= {
+		.name	= "db1000-audio",
+		.pm	= &snd_soc_pm_ops,
+	},
+	.probe		= db1000_audio_probe,
+};
+
+module_platform_driver(db1000_audio_driver);
+
+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
new file mode 100644
index 000000000..301e1fc9a
--- /dev/null
+++ b/sound/soc/au1x/db1200.c
@@ -0,0 +1,197 @@
+/*
+ * DB1200/DB1300/DB1550 ASoC audio fabric support code.
+ *
+ * (c) 2008-2011 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-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "../codecs/wm8731.h"
+#include "psc.h"
+
+static const struct platform_device_id db1200_pids[] = {
+	{
+		.name		= "db1200-ac97",
+		.driver_data	= 0,
+	}, {
+		.name		= "db1200-i2s",
+		.driver_data	= 1,
+	}, {
+		.name		= "db1300-ac97",
+		.driver_data	= 2,
+	}, {
+		.name		= "db1300-i2s",
+		.driver_data	= 3,
+	}, {
+		.name		= "db1550-ac97",
+		.driver_data	= 4,
+	}, {
+		.name		= "db1550-i2s",
+		.driver_data	= 5,
+	},
+	{},
+};
+
+/*-------------------------  AC97 PART  ---------------------------*/
+
+static struct snd_soc_dai_link db1200_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.codec_dai_name	= "ac97-hifi",
+	.cpu_dai_name	= "au1xpsc_ac97.1",
+	.platform_name	= "au1xpsc-pcm.1",
+	.codec_name	= "ac97-codec.1",
+};
+
+static struct snd_soc_card db1200_ac97_machine = {
+	.name		= "DB1200_AC97",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1200_ac97_dai,
+	.num_links	= 1,
+};
+
+static struct snd_soc_dai_link db1300_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.codec_dai_name	= "wm9712-hifi",
+	.cpu_dai_name	= "au1xpsc_ac97.1",
+	.platform_name	= "au1xpsc-pcm.1",
+	.codec_name	= "wm9712-codec.1",
+};
+
+static struct snd_soc_card db1300_ac97_machine = {
+	.name		= "DB1300_AC97",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1300_ac97_dai,
+	.num_links	= 1,
+};
+
+static struct snd_soc_card db1550_ac97_machine = {
+	.name		= "DB1550_AC97",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1200_ac97_dai,
+	.num_links	= 1,
+};
+
+/*-------------------------  I2S PART  ---------------------------*/
+
+static int db1200_i2s_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	/* WM8731 has its own 12MHz crystal */
+	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+				12000000, SND_SOC_CLOCK_IN);
+
+	return 0;
+}
+
+static const struct snd_soc_ops db1200_i2s_wm8731_ops = {
+	.startup	= db1200_i2s_startup,
+};
+
+static struct snd_soc_dai_link db1200_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.codec_dai_name	= "wm8731-hifi",
+	.cpu_dai_name	= "au1xpsc_i2s.1",
+	.platform_name	= "au1xpsc-pcm.1",
+	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1200_i2s_machine = {
+	.name		= "DB1200_I2S",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1200_i2s_dai,
+	.num_links	= 1,
+};
+
+static struct snd_soc_dai_link db1300_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.codec_dai_name	= "wm8731-hifi",
+	.cpu_dai_name	= "au1xpsc_i2s.2",
+	.platform_name	= "au1xpsc-pcm.2",
+	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1300_i2s_machine = {
+	.name		= "DB1300_I2S",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1300_i2s_dai,
+	.num_links	= 1,
+};
+
+static struct snd_soc_dai_link db1550_i2s_dai = {
+	.name		= "WM8731",
+	.stream_name	= "WM8731 PCM",
+	.codec_dai_name	= "wm8731-hifi",
+	.cpu_dai_name	= "au1xpsc_i2s.3",
+	.platform_name	= "au1xpsc-pcm.3",
+	.codec_name	= "wm8731.0-001b",
+	.dai_fmt	= SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+			  SND_SOC_DAIFMT_CBM_CFM,
+	.ops		= &db1200_i2s_wm8731_ops,
+};
+
+static struct snd_soc_card db1550_i2s_machine = {
+	.name		= "DB1550_I2S",
+	.owner		= THIS_MODULE,
+	.dai_link	= &db1550_i2s_dai,
+	.num_links	= 1,
+};
+
+/*-------------------------  COMMON PART  ---------------------------*/
+
+static struct snd_soc_card *db1200_cards[] = {
+	&db1200_ac97_machine,
+	&db1200_i2s_machine,
+	&db1300_ac97_machine,
+	&db1300_i2s_machine,
+	&db1550_ac97_machine,
+	&db1550_i2s_machine,
+};
+
+static int db1200_audio_probe(struct platform_device *pdev)
+{
+	const struct platform_device_id *pid = platform_get_device_id(pdev);
+	struct snd_soc_card *card;
+
+	card = db1200_cards[pid->driver_data];
+	card->dev = &pdev->dev;
+	return devm_snd_soc_register_card(&pdev->dev, card);
+}
+
+static struct platform_driver db1200_audio_driver = {
+	.driver	= {
+		.name	= "db1200-ac97",
+		.pm	= &snd_soc_pm_ops,
+	},
+	.id_table	= db1200_pids,
+	.probe		= db1200_audio_probe,
+};
+
+module_platform_driver(db1200_audio_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1200/DB1300/DB1550 ASoC audio support");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 000000000..a906560d0
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -0,0 +1,365 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * DMA glue for Au1x-PSC audio.
+ *
+ */
+
+
+#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/au1xxx_dbdma.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/*#define PCM_DEBUG*/
+
+#define DRV_NAME "dbdma2"
+
+#define MSG(x...)	printk(KERN_INFO "au1xpsc_pcm: " x)
+#ifdef PCM_DEBUG
+#define DBG		MSG
+#else
+#define DBG(x...)	do {} while (0)
+#endif
+
+struct au1xpsc_audio_dmadata {
+	/* DDMA control data */
+	unsigned int ddma_id;		/* DDMA direction ID for this PSC */
+	u32 ddma_chan;			/* DDMA context */
+
+	/* PCM context (for irq handlers) */
+	struct snd_pcm_substream *substream;
+	unsigned long curr_period;	/* current segment DDMA is working on */
+	unsigned long q_period;		/* queue period(s) */
+	dma_addr_t dma_area;		/* address of queued DMA area */
+	dma_addr_t dma_area_s;		/* start address of DMA area */
+	unsigned long pos;		/* current byte position being played */
+	unsigned long periods;		/* number of SG segments in total */
+	unsigned long period_bytes;	/* size in bytes of one SG segment */
+
+	/* runtime data */
+	int msbits;
+};
+
+/*
+ * These settings are somewhat okay, at least on my machine audio plays
+ * almost skip-free. Especially the 64kB buffer seems to help a LOT.
+ */
+#define AU1XPSC_PERIOD_MIN_BYTES	1024
+#define AU1XPSC_BUFFER_MIN_BYTES	65536
+
+/* PCM hardware DMA capabilities - platform specific */
+static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
+	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
+	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
+	.period_bytes_max = 4096 * 1024 - 1,
+	.periods_min	  = 2,
+	.periods_max	  = 4096,	/* 2 to as-much-as-you-like */
+	.buffer_bytes_max = 4096 * 1024 - 1,
+	.fifo_size	  = 16,		/* fifo entries of AC97/I2S PSC */
+};
+
+static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
+{
+	au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area,
+				cd->period_bytes, DDMA_FLAGS_IE);
+
+	/* update next-to-queue period */
+	++cd->q_period;
+	cd->dma_area += cd->period_bytes;
+	if (cd->q_period >= cd->periods) {
+		cd->q_period = 0;
+		cd->dma_area = cd->dma_area_s;
+	}
+}
+
+static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
+{
+	au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area,
+			      cd->period_bytes, DDMA_FLAGS_IE);
+
+	/* update next-to-queue period */
+	++cd->q_period;
+	cd->dma_area += cd->period_bytes;
+	if (cd->q_period >= cd->periods) {
+		cd->q_period = 0;
+		cd->dma_area = cd->dma_area_s;
+	}
+}
+
+static void au1x_pcm_dmatx_cb(int irq, void *dev_id)
+{
+	struct au1xpsc_audio_dmadata *cd = dev_id;
+
+	cd->pos += cd->period_bytes;
+	if (++cd->curr_period >= cd->periods) {
+		cd->pos = 0;
+		cd->curr_period = 0;
+	}
+	snd_pcm_period_elapsed(cd->substream);
+	au1x_pcm_queue_tx(cd);
+}
+
+static void au1x_pcm_dmarx_cb(int irq, void *dev_id)
+{
+	struct au1xpsc_audio_dmadata *cd = dev_id;
+
+	cd->pos += cd->period_bytes;
+	if (++cd->curr_period >= cd->periods) {
+		cd->pos = 0;
+		cd->curr_period = 0;
+	}
+	snd_pcm_period_elapsed(cd->substream);
+	au1x_pcm_queue_rx(cd);
+}
+
+static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd)
+{
+	if (pcd->ddma_chan) {
+		au1xxx_dbdma_stop(pcd->ddma_chan);
+		au1xxx_dbdma_reset(pcd->ddma_chan);
+		au1xxx_dbdma_chan_free(pcd->ddma_chan);
+		pcd->ddma_chan = 0;
+		pcd->msbits = 0;
+	}
+}
+
+/* in case of missing DMA ring or changed TX-source / RX-dest bit widths,
+ * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according
+ * to ALSA-supplied sample depth.  This is due to limitations in the dbdma api
+ * (cannot adjust source/dest widths of already allocated descriptor ring).
+ */
+static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
+				 int stype, int msbits)
+{
+	/* DMA only in 8/16/32 bit widths */
+	if (msbits == 24)
+		msbits = 32;
+
+	/* check current config: correct bits and descriptors allocated? */
+	if ((pcd->ddma_chan) && (msbits == pcd->msbits))
+		goto out;	/* all ok! */
+
+	au1x_pcm_dbdma_free(pcd);
+
+	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);
+	else
+		pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS,
+					pcd->ddma_id,
+					au1x_pcm_dmatx_cb, (void *)pcd);
+
+	if (!pcd->ddma_chan)
+		return -ENOMEM;
+
+	au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits);
+	au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2);
+
+	pcd->msbits = msbits;
+
+	au1xxx_dbdma_stop(pcd->ddma_chan);
+	au1xxx_dbdma_reset(pcd->ddma_chan);
+
+out:
+	return 0;
+}
+
+static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_pcm_runtime *rtd = ss->private_data;
+	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	struct au1xpsc_audio_dmadata *pcd = snd_soc_component_get_drvdata(component);
+	return &pcd[ss->stream];
+}
+
+static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct au1xpsc_audio_dmadata *pcd;
+	int stype, ret;
+
+	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	if (ret < 0)
+		goto out;
+
+	stype = substream->stream;
+	pcd = to_dmadata(substream);
+
+	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu "
+	    "runtime->min_align %lu\n",
+		(unsigned long)runtime->dma_area,
+		(unsigned long)runtime->dma_addr, runtime->dma_bytes,
+		runtime->min_align);
+
+	DBG("bits %d  frags %d  frag_bytes %d  is_rx %d\n", params->msbits,
+		params_periods(params), params_period_bytes(params), stype);
+
+	ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits);
+	if (ret) {
+		MSG("DDMA channel (re)alloc failed!\n");
+		goto out;
+	}
+
+	pcd->substream = substream;
+	pcd->period_bytes = params_period_bytes(params);
+	pcd->periods = params_periods(params);
+	pcd->dma_area_s = pcd->dma_area = runtime->dma_addr;
+	pcd->q_period = 0;
+	pcd->curr_period = 0;
+	pcd->pos = 0;
+
+	ret = 0;
+out:
+	return ret;
+}
+
+static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_lib_free_pages(substream);
+	return 0;
+}
+
+static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
+
+	au1xxx_dbdma_reset(pcd->ddma_chan);
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		au1x_pcm_queue_rx(pcd);
+		au1x_pcm_queue_rx(pcd);
+	} else {
+		au1x_pcm_queue_tx(pcd);
+		au1x_pcm_queue_tx(pcd);
+	}
+
+	return 0;
+}
+
+static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	u32 c = to_dmadata(substream)->ddma_chan;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		au1xxx_dbdma_start(c);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		au1xxx_dbdma_stop(c);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t
+au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	return bytes_to_frames(substream->runtime, to_dmadata(substream)->pos);
+}
+
+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;
+}
+
+static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
+{
+	au1x_pcm_dbdma_free(to_dmadata(substream));
+	return 0;
+}
+
+static const struct snd_pcm_ops au1xpsc_pcm_ops = {
+	.open		= au1xpsc_pcm_open,
+	.close		= au1xpsc_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= au1xpsc_pcm_hw_params,
+	.hw_free	= au1xpsc_pcm_hw_free,
+	.prepare	= au1xpsc_pcm_prepare,
+	.trigger	= au1xpsc_pcm_trigger,
+	.pointer	= au1xpsc_pcm_pointer,
+};
+
+static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+		card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
+
+	return 0;
+}
+
+/* au1xpsc audio platform */
+static struct snd_soc_component_driver au1xpsc_soc_component = {
+	.name		= DRV_NAME,
+	.ops		= &au1xpsc_pcm_ops,
+	.pcm_new	= au1xpsc_pcm_new,
+};
+
+static int au1xpsc_pcm_drvprobe(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_dmadata *dmadata;
+
+	dmadata = devm_kcalloc(&pdev->dev,
+			       2, sizeof(struct au1xpsc_audio_dmadata),
+			       GFP_KERNEL);
+	if (!dmadata)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, dmadata);
+
+	return devm_snd_soc_register_component(&pdev->dev,
+					&au1xpsc_soc_component, NULL, 0);
+}
+
+static struct platform_driver au1xpsc_pcm_driver = {
+	.driver	= {
+		.name	= "au1xpsc-pcm",
+	},
+	.probe		= au1xpsc_pcm_drvprobe,
+};
+
+module_platform_driver(au1xpsc_pcm_driver);
+
+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 000000000..efff1e293
--- /dev/null
+++ b/sound/soc/au1x/dma.c
@@ -0,0 +1,334 @@
+/*
+ * 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 DRV_NAME "au1x_dma"
+
+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,
+	.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;
+	struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+	return snd_soc_component_get_drvdata(component);
+}
+
+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, 0,
+					&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 const 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 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;
+}
+
+static struct snd_soc_component_driver alchemy_pcm_soc_component = {
+	.name		= DRV_NAME,
+	.ops		= &alchemy_pcm_ops,
+	.pcm_new	= alchemy_pcm_new,
+};
+
+static int alchemy_pcm_drvprobe(struct platform_device *pdev)
+{
+	struct alchemy_pcm_ctx *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, ctx);
+
+	return devm_snd_soc_register_component(&pdev->dev,
+					&alchemy_pcm_soc_component, NULL, 0);
+}
+
+static struct platform_driver alchemy_pcmdma_driver = {
+	.driver	= {
+		.name	= "alchemy-pcm-dma",
+	},
+	.probe		= alchemy_pcm_drvprobe,
+};
+
+module_platform_driver(alchemy_pcmdma_driver);
+
+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 000000000..450c842c7
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,323 @@
+/*
+ * 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 const struct snd_soc_component_driver au1xi2s_component = {
+	.name		= "au1xi2s",
+};
+
+static int au1xi2s_drvprobe(struct platform_device *pdev)
+{
+	struct resource *iores, *dmares;
+	struct au1xpsc_audio_data *ctx;
+
+	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!iores)
+		return -ENODEV;
+
+	if (!devm_request_mem_region(&pdev->dev, iores->start,
+				     resource_size(iores),
+				     pdev->name))
+		return -EBUSY;
+
+	ctx->mmio = devm_ioremap_nocache(&pdev->dev, iores->start,
+					 resource_size(iores));
+	if (!ctx->mmio)
+		return -EBUSY;
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmares)
+		return -EBUSY;
+	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!dmares)
+		return -EBUSY;
+	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
+
+	platform_set_drvdata(pdev, ctx);
+
+	return snd_soc_register_component(&pdev->dev, &au1xi2s_component,
+					  &au1xi2s_dai_driver, 1);
+}
+
+static int au1xi2s_drvremove(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_component(&pdev->dev);
+
+	WR(ctx, I2S_ENABLE, EN_D);	/* clock off, disable */
+
+	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",
+		.pm	= AU1XI2SC_PMOPS,
+	},
+	.probe		= au1xi2s_drvprobe,
+	.remove		= au1xi2s_drvremove,
+};
+
+module_platform_driver(au1xi2s_driver);
+
+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
new file mode 100644
index 000000000..a2050ae5a
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,504 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2009 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Au1xxx-PSC AC97 glue.
+ *
+ */
+
+#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/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 <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES	5
+
+#define AC97_DIR	\
+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES	\
+	SNDRV_PCM_RATE_8000_48000
+
+#define AC97_FMTS	\
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
+
+#define AC97PCR_START(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+#define AC97PCR_STOP(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+#define AC97PCR_CLRFIFO(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+
+#define AC97STAT_BUSY(stype)	\
+	((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;
+
+#if 0
+
+/* this could theoretically work, but ac97->bus->card->private_data can be NULL
+ * when snd_ac97_mixer() is called; I don't know if the rest further down the
+ * chain are always valid either.
+ */
+static inline struct au1xpsc_audio_data *ac97_to_pscdata(struct snd_ac97 *x)
+{
+	struct snd_soc_card *c = x->bus->card->private_data;
+	return snd_soc_dai_get_drvdata(c->rtd->cpu_dai);
+}
+
+#else
+
+#define ac97_to_pscdata(x)	au1xpsc_ac97_workdata
+
+#endif
+
+/* AC97 controller reads codec register */
+static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
+					unsigned short reg)
+{
+	struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97);
+	unsigned short retry, tmo;
+	unsigned long data;
+
+	__raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+	wmb(); /* drain writebuffer */
+
+	retry = AC97_RW_RETRIES;
+	do {
+		mutex_lock(&pscdata->lock);
+
+		__raw_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg),
+			  AC97_CDC(pscdata));
+		wmb(); /* drain writebuffer */
+
+		tmo = 20;
+		do {
+			udelay(21);
+			if (__raw_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
+				break;
+		} while (--tmo);
+
+		data = __raw_readl(AC97_CDC(pscdata));
+
+		__raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+		wmb(); /* drain writebuffer */
+
+		mutex_unlock(&pscdata->lock);
+
+		if (reg != ((data >> 16) & 0x7f))
+			tmo = 1;	/* wrong register, try again */
+
+	} while (--retry && !tmo);
+
+	return retry ? data & 0xffff : 0xffff;
+}
+
+/* AC97 controller writes to codec register */
+static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+				unsigned short val)
+{
+	struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97);
+	unsigned int tmo, retry;
+
+	__raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+	wmb(); /* drain writebuffer */
+
+	retry = AC97_RW_RETRIES;
+	do {
+		mutex_lock(&pscdata->lock);
+
+		__raw_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff),
+			  AC97_CDC(pscdata));
+		wmb(); /* drain writebuffer */
+
+		tmo = 20;
+		do {
+			udelay(21);
+			if (__raw_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
+				break;
+		} while (--tmo);
+
+		__raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+		wmb(); /* drain writebuffer */
+
+		mutex_unlock(&pscdata->lock);
+	} while (--retry && !tmo);
+}
+
+/* AC97 controller asserts a warm reset */
+static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97);
+
+	__raw_writel(PSC_AC97RST_SNC, AC97_RST(pscdata));
+	wmb(); /* drain writebuffer */
+	msleep(10);
+	__raw_writel(0, AC97_RST(pscdata));
+	wmb(); /* drain writebuffer */
+}
+
+static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97);
+	int i;
+
+	/* disable PSC during cold reset */
+	__raw_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* issue cold reset */
+	__raw_writel(PSC_AC97RST_RST, AC97_RST(pscdata));
+	wmb(); /* drain writebuffer */
+	msleep(500);
+	__raw_writel(0, AC97_RST(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* enable PSC */
+	__raw_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* wait for PSC to indicate it's ready */
+	i = 1000;
+	while (!((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
+		msleep(1);
+
+	if (i == 0) {
+		printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
+		return;
+	}
+
+	/* enable the ac97 function */
+	__raw_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* wait for AC97 core to become ready */
+	i = 1000;
+	while (!((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
+		msleep(1);
+	if (i == 0)
+		printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
+}
+
+/* AC97 controller operations */
+static struct snd_ac97_bus_ops psc_ac97_ops = {
+	.read		= au1xpsc_ac97_read,
+	.write		= au1xpsc_ac97_write,
+	.reset		= au1xpsc_ac97_cold_reset,
+	.warm_reset	= au1xpsc_ac97_warm_reset,
+};
+
+static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+	unsigned long r, ro, stat;
+	int chans, t, stype = substream->stream;
+
+	chans = params_channels(params);
+
+	r = ro = __raw_readl(AC97_CFG(pscdata));
+	stat = __raw_readl(AC97_STAT(pscdata));
+
+	/* already active? */
+	if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) {
+		/* reject parameters not currently set up */
+		if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) ||
+		    (pscdata->rate != params_rate(params)))
+			return -EINVAL;
+	} else {
+
+		/* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
+		r &= ~PSC_AC97CFG_LEN_MASK;
+		r |= PSC_AC97CFG_SET_LEN(params->msbits);
+
+		/* channels: enable slots for front L/R channel */
+		if (stype == SNDRV_PCM_STREAM_PLAYBACK) {
+			r &= ~PSC_AC97CFG_TXSLOT_MASK;
+			r |= PSC_AC97CFG_TXSLOT_ENA(3);
+			r |= PSC_AC97CFG_TXSLOT_ENA(4);
+		} else {
+			r &= ~PSC_AC97CFG_RXSLOT_MASK;
+			r |= PSC_AC97CFG_RXSLOT_ENA(3);
+			r |= PSC_AC97CFG_RXSLOT_ENA(4);
+		}
+
+		/* do we need to poke the hardware? */
+		if (!(r ^ ro))
+			goto out;
+
+		/* ac97 engine is about to be disabled */
+		mutex_lock(&pscdata->lock);
+
+		/* disable AC97 device controller first... */
+		__raw_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+		wmb(); /* drain writebuffer */
+
+		/* ...wait for it... */
+		t = 100;
+		while ((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t)
+			msleep(1);
+
+		if (!t)
+			printk(KERN_ERR "PSC-AC97: can't disable!\n");
+
+		/* ...write config... */
+		__raw_writel(r, AC97_CFG(pscdata));
+		wmb(); /* drain writebuffer */
+
+		/* ...enable the AC97 controller again... */
+		__raw_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+		wmb(); /* drain writebuffer */
+
+		/* ...and wait for ready bit */
+		t = 100;
+		while ((!(__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t)
+			msleep(1);
+
+		if (!t)
+			printk(KERN_ERR "PSC-AC97: can't enable!\n");
+
+		mutex_unlock(&pscdata->lock);
+
+		pscdata->cfg = r;
+		pscdata->rate = params_rate(params);
+	}
+
+out:
+	return 0;
+}
+
+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->stream;
+
+	ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		__raw_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+		wmb(); /* drain writebuffer */
+		__raw_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
+		wmb(); /* drain writebuffer */
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		__raw_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
+		wmb(); /* drain writebuffer */
+
+		while (__raw_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype))
+			asm volatile ("nop");
+
+		__raw_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+		wmb(); /* drain writebuffer */
+
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	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 const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.startup	= au1xpsc_ac97_startup,
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
+static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = {
+	.bus_control		= true,
+	.probe			= au1xpsc_ac97_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 = &au1xpsc_ac97_dai_ops,
+};
+
+static const struct snd_soc_component_driver au1xpsc_ac97_component = {
+	.name		= "au1xpsc-ac97",
+};
+
+static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
+{
+	int ret;
+	struct resource *iores, *dmares;
+	unsigned long sel;
+	struct au1xpsc_audio_data *wd;
+
+	wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),
+			  GFP_KERNEL);
+	if (!wd)
+		return -ENOMEM;
+
+	mutex_init(&wd->lock);
+
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
+	if (IS_ERR(wd->mmio))
+		return PTR_ERR(wd->mmio);
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmares)
+		return -EBUSY;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!dmares)
+		return -EBUSY;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
+
+	/* configuration: max dma trigger threshold, enable ac97 */
+	wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
+		  PSC_AC97CFG_DE_ENABLE;
+
+	/* preserve PSC clock source set up by platform	 */
+	sel = __raw_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(0, PSC_SEL(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd));
+	wmb(); /* drain writebuffer */
+
+	/* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */
+	memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template,
+	       sizeof(struct snd_soc_dai_driver));
+	wd->dai_drv.name = dev_name(&pdev->dev);
+
+	platform_set_drvdata(pdev, wd);
+
+	ret = snd_soc_set_ac97_ops(&psc_ac97_ops);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component,
+					 &wd->dai_drv, 1);
+	if (ret)
+		return ret;
+
+	au1xpsc_ac97_workdata = wd;
+	return 0;
+}
+
+static int au1xpsc_ac97_drvremove(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_component(&pdev->dev);
+
+	/* disable PSC completely */
+	__raw_writel(0, AC97_CFG(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+
+	au1xpsc_ac97_workdata = NULL;	/* MDEV */
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xpsc_ac97_drvsuspend(struct device *dev)
+{
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
+	/* save interesting registers and disable PSC */
+	wd->pm[0] = __raw_readl(PSC_SEL(wd));
+
+	__raw_writel(0, AC97_CFG(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+
+	return 0;
+}
+
+static int au1xpsc_ac97_drvresume(struct device *dev)
+{
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
+	/* restore PSC clock config */
+	__raw_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd));
+	wmb(); /* drain writebuffer */
+
+	/* after this point the ac97 core will cold-reset the codec.
+	 * During cold-reset the PSC is reinitialized and the last
+	 * configuration set up in hw_params() is restored.
+	 */
+	return 0;
+}
+
+static const struct dev_pm_ops au1xpscac97_pmops = {
+	.suspend	= au1xpsc_ac97_drvsuspend,
+	.resume		= au1xpsc_ac97_drvresume,
+};
+
+#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xpsc_ac97_driver = {
+	.driver	= {
+		.name	= "au1xpsc_ac97",
+		.pm	= AU1XPSCAC97_PMOPS,
+	},
+	.probe		= au1xpsc_ac97_drvprobe,
+	.remove		= au1xpsc_ac97_drvremove,
+};
+
+module_platform_driver(au1xpsc_ac97_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
+MODULE_AUTHOR("Manuel Lauss");
+
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 000000000..e6eec081e
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,421 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <manuel.lauss@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Au1xxx-PSC I2S glue.
+ *
+ * NOTE: so far only PSC slave mode (bit- and frameclock) is supported.
+ */
+
+#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 <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/* supported I2S DAI hardware formats */
+#define AU1XPSC_I2S_DAIFMT \
+	(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J |	\
+	 SND_SOC_DAIFMT_NB_NF)
+
+/* supported I2S direction */
+#define AU1XPSC_I2S_DIR \
+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AU1XPSC_I2S_RATES \
+	SNDRV_PCM_RATE_8000_192000
+
+#define AU1XPSC_I2S_FMTS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+#define I2SSTAT_BUSY(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+#define I2SPCR_START(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+#define I2SPCR_STOP(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+#define I2SPCR_CLRFIFO(stype)	\
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+
+
+static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+			       unsigned int fmt)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned long ct;
+	int ret;
+
+	ret = -EINVAL;
+
+	ct = pscdata->cfg;
+
+	ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ);	/* left-justified */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ct |= PSC_I2SCFG_XM;	/* enable I2S mode */
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		ct |= PSC_I2SCFG_MLJ;	/* LSB (right-) justified */
+		break;
+	default:
+		goto out;
+	}
+
+	ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI);		/* IB-IF */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		ct |= PSC_I2SCFG_BI;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		ct |= PSC_I2SCFG_WI;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		break;
+	default:
+		goto out;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:	/* CODEC master */
+		ct |= PSC_I2SCFG_MS;	/* PSC I2S slave mode */
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:	/* CODEC slave */
+		ct &= ~PSC_I2SCFG_MS;	/* PSC I2S Master mode */
+		break;
+	default:
+		goto out;
+	}
+
+	pscdata->cfg = ct;
+	ret = 0;
+out:
+	return ret;
+}
+
+static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+
+	int cfgbits;
+	unsigned long stat;
+
+	/* check if the PSC is already streaming data */
+	stat = __raw_readl(I2S_STAT(pscdata));
+	if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) {
+		/* reject parameters not currently set up in hardware */
+		cfgbits = __raw_readl(I2S_CFG(pscdata));
+		if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) ||
+		    (params_rate(params) != pscdata->rate))
+			return -EINVAL;
+	} else {
+		/* set sample bitdepth */
+		pscdata->cfg &= ~(0x1f << 4);
+		pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits);
+		/* remember current rate for other stream */
+		pscdata->rate = params_rate(params);
+	}
+	return 0;
+}
+
+/* Configure PSC late:  on my devel systems the codec  is I2S master and
+ * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit.  ASoC
+ * uses aggressive PM and  switches the codec off  when it is not in use
+ * which also means the PSC unit doesn't get any clocks and is therefore
+ * dead. That's why this chunk here gets called from the trigger callback
+ * because I can be reasonably certain the codec is driving the clocks.
+ */
+static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata)
+{
+	unsigned long tmo;
+
+	/* bring PSC out of sleep, and configure I2S unit */
+	__raw_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+	wmb(); /* drain writebuffer */
+
+	tmo = 1000000;
+	while (!(__raw_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo)
+		tmo--;
+
+	if (!tmo)
+		goto psc_err;
+
+	__raw_writel(0, I2S_CFG(pscdata));
+	wmb(); /* drain writebuffer */
+	__raw_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* wait for I2S controller to become ready */
+	tmo = 1000000;
+	while (!(__raw_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo)
+		tmo--;
+
+	if (tmo)
+		return 0;
+
+psc_err:
+	__raw_writel(0, I2S_CFG(pscdata));
+	__raw_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+	wmb(); /* drain writebuffer */
+	return -ETIMEDOUT;
+}
+
+static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype)
+{
+	unsigned long tmo, stat;
+	int ret;
+
+	ret = 0;
+
+	/* if both TX and RX are idle, configure the PSC  */
+	stat = __raw_readl(I2S_STAT(pscdata));
+	if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
+		ret = au1xpsc_i2s_configure(pscdata);
+		if (ret)
+			goto out;
+	}
+
+	__raw_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata));
+	wmb(); /* drain writebuffer */
+	__raw_writel(I2SPCR_START(stype), I2S_PCR(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* wait for start confirmation */
+	tmo = 1000000;
+	while (!(__raw_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+		tmo--;
+
+	if (!tmo) {
+		__raw_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+		wmb(); /* drain writebuffer */
+		ret = -ETIMEDOUT;
+	}
+out:
+	return ret;
+}
+
+static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
+{
+	unsigned long tmo, stat;
+
+	__raw_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+	wmb(); /* drain writebuffer */
+
+	/* wait for stop confirmation */
+	tmo = 1000000;
+	while ((__raw_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+		tmo--;
+
+	/* if both TX and RX are idle, disable PSC */
+	stat = __raw_readl(I2S_STAT(pscdata));
+	if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
+		__raw_writel(0, I2S_CFG(pscdata));
+		wmb(); /* drain writebuffer */
+		__raw_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+		wmb(); /* drain writebuffer */
+	}
+	return 0;
+}
+
+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->stream;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		ret = au1xpsc_i2s_start(pscdata, stype);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ret = au1xpsc_i2s_stop(pscdata, stype);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	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 const 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,
+};
+
+static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = {
+	.playback = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.capture = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.ops = &au1xpsc_i2s_dai_ops,
+};
+
+static const struct snd_soc_component_driver au1xpsc_i2s_component = {
+	.name		= "au1xpsc-i2s",
+};
+
+static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
+{
+	struct resource *iores, *dmares;
+	unsigned long sel;
+	struct au1xpsc_audio_data *wd;
+
+	wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),
+			  GFP_KERNEL);
+	if (!wd)
+		return -ENOMEM;
+
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	wd->mmio = devm_ioremap_resource(&pdev->dev, iores);
+	if (IS_ERR(wd->mmio))
+		return PTR_ERR(wd->mmio);
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmares)
+		return -EBUSY;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
+
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!dmares)
+		return -EBUSY;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
+
+	/* preserve PSC clock source set up by platform (dev.platform_data
+	 * is already occupied by soc layer)
+	 */
+	sel = __raw_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd));
+	__raw_writel(0, I2S_CFG(wd));
+	wmb(); /* drain writebuffer */
+
+	/* preconfigure: set max rx/tx fifo depths */
+	wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
+
+	/* don't wait for I2S core to become ready now; clocks may not
+	 * be running yet; depending on clock input for PSC a wait might
+	 * time out.
+	 */
+
+	/* name the DAI like this device instance ("au1xpsc-i2s.PSCINDEX") */
+	memcpy(&wd->dai_drv, &au1xpsc_i2s_dai_template,
+	       sizeof(struct snd_soc_dai_driver));
+	wd->dai_drv.name = dev_name(&pdev->dev);
+
+	platform_set_drvdata(pdev, wd);
+
+	return snd_soc_register_component(&pdev->dev, &au1xpsc_i2s_component,
+					  &wd->dai_drv, 1);
+}
+
+static int au1xpsc_i2s_drvremove(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_component(&pdev->dev);
+
+	__raw_writel(0, I2S_CFG(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xpsc_i2s_drvsuspend(struct device *dev)
+{
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
+	/* save interesting register and disable PSC */
+	wd->pm[0] = __raw_readl(PSC_SEL(wd));
+
+	__raw_writel(0, I2S_CFG(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+
+	return 0;
+}
+
+static int au1xpsc_i2s_drvresume(struct device *dev)
+{
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
+	/* select I2S mode and PSC clock */
+	__raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(0, PSC_SEL(wd));
+	wmb(); /* drain writebuffer */
+	__raw_writel(wd->pm[0], PSC_SEL(wd));
+	wmb(); /* drain writebuffer */
+
+	return 0;
+}
+
+static const struct dev_pm_ops au1xpsci2s_pmops = {
+	.suspend	= au1xpsc_i2s_drvsuspend,
+	.resume		= au1xpsc_i2s_drvresume,
+};
+
+#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops
+
+#else
+
+#define AU1XPSCI2S_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xpsc_i2s_driver = {
+	.driver		= {
+		.name	= "au1xpsc_i2s",
+		.pm	= AU1XPSCI2S_PMOPS,
+	},
+	.probe		= au1xpsc_i2s_drvprobe,
+	.remove		= au1xpsc_i2s_drvremove,
+};
+
+module_platform_driver(au1xpsc_i2s_driver);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 000000000..74dffeb64
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -0,0 +1,42 @@
+/*
+ * Alchemy ALSA ASoC audio support.
+ *
+ * (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
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _AU1X_PCM_H
+#define _AU1X_PCM_H
+
+struct au1xpsc_audio_data {
+	void __iomem *mmio;
+
+	unsigned long cfg;
+	unsigned long rate;
+
+	struct snd_soc_dai_driver dai_drv;
+
+	unsigned long pm[2];
+	struct mutex lock;
+	int dmaids[2];
+};
+
+/* easy access macros */
+#define PSC_CTRL(x)	((x)->mmio + PSC_CTRL_OFFSET)
+#define PSC_SEL(x)	((x)->mmio + PSC_SEL_OFFSET)
+#define I2S_STAT(x)	((x)->mmio + PSC_I2SSTAT_OFFSET)
+#define I2S_CFG(x)	((x)->mmio + PSC_I2SCFG_OFFSET)
+#define I2S_PCR(x)	((x)->mmio + PSC_I2SPCR_OFFSET)
+#define AC97_CFG(x)	((x)->mmio + PSC_AC97CFG_OFFSET)
+#define AC97_CDC(x)	((x)->mmio + PSC_AC97CDC_OFFSET)
+#define AC97_EVNT(x)	((x)->mmio + PSC_AC97EVNT_OFFSET)
+#define AC97_PCR(x)	((x)->mmio + PSC_AC97PCR_OFFSET)
+#define AC97_RST(x)	((x)->mmio + PSC_AC97RST_OFFSET)
+#define AC97_STAT(x)	((x)->mmio + PSC_AC97STAT_OFFSET)
+
+#endif