From b3c70c9ea62a3ae6c63536e43fa28f965a56de91 Mon Sep 17 00:00:00 2001
From: Manuel Lauss <manuel.lauss@googlemail.com>
Date: Mon, 25 Jul 2011 13:44:45 +0200
Subject: 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>
---
 sound/soc/au1x/Kconfig  |  19 +++
 sound/soc/au1x/Makefile |   8 +
 sound/soc/au1x/ac97c.c  | 365 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/au1x/dma.c    | 377 ++++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/au1x/i2sc.c   | 347 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/au1x/psc.h    |  19 +--
 6 files changed, 1126 insertions(+), 9 deletions(-)
 create mode 100644 sound/soc/au1x/ac97c.c
 create mode 100644 sound/soc/au1x/dma.c
 create mode 100644 sound/soc/au1x/i2sc.c

(limited to 'sound/soc/au1x')

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)
-- 
cgit v1.2.3


From b2ce305dda483e59a78d5aa6e4211034c0cea38d Mon Sep 17 00:00:00 2001
From: Manuel Lauss <manuel.lauss@googlemail.com>
Date: Mon, 25 Jul 2011 13:44:46 +0200
Subject: ASoC: Add a DB1x00 AC97 machine driver

Add a machine driver suitable for the AC97 part on the DB1000/DB1500/DB1100
boards.

Run-tested on DB1500.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/mips/alchemy/devboards/db1x00/platform.c | 48 +++++++++++++++++
 sound/soc/au1x/Kconfig                        |  9 ++++
 sound/soc/au1x/Makefile                       |  2 +
 sound/soc/au1x/db1000.c                       | 75 +++++++++++++++++++++++++++
 4 files changed, 134 insertions(+)
 create mode 100644 sound/soc/au1x/db1000.c

(limited to 'sound/soc/au1x')

diff --git a/arch/mips/alchemy/devboards/db1x00/platform.c b/arch/mips/alchemy/devboards/db1x00/platform.c
index 978d5ab3d678..7057d28f7301 100644
--- a/arch/mips/alchemy/devboards/db1x00/platform.c
+++ b/arch/mips/alchemy/devboards/db1x00/platform.c
@@ -19,8 +19,11 @@
  */
 
 #include <linux/init.h>
+#include <linux/interrupt.h>
 #include <linux/platform_device.h>
 
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
 #include <asm/mach-au1x00/au1xxx.h>
 #include <asm/mach-db1x00/bcsr.h>
 #include "../platform.h"
@@ -85,6 +88,45 @@
 #endif
 #endif
 
+static struct resource alchemy_ac97c_res[] = {
+	[0] = {
+		.start	= AU1000_AC97_PHYS_ADDR,
+		.end	= AU1000_AC97_PHYS_ADDR + 0xfff,
+		.flags	= IORESOURCE_MEM,
+	},
+	[1] = {
+		.start	= DMA_ID_AC97C_TX,
+		.end	= DMA_ID_AC97C_TX,
+		.flags	= IORESOURCE_DMA,
+	},
+	[2] = {
+		.start	= DMA_ID_AC97C_RX,
+		.end	= DMA_ID_AC97C_RX,
+		.flags	= IORESOURCE_DMA,
+	},
+};
+
+static struct platform_device alchemy_ac97c_dev = {
+	.name		= "alchemy-ac97c",
+	.id		= -1,
+	.resource	= alchemy_ac97c_res,
+	.num_resources	= ARRAY_SIZE(alchemy_ac97c_res),
+};
+
+static struct platform_device alchemy_ac97c_dma_dev = {
+	.name		= "alchemy-pcm-dma",
+	.id		= 0,
+};
+
+static struct platform_device db1x00_codec_dev = {
+	.name		= "ac97-codec",
+	.id		= -1,
+};
+
+static struct platform_device db1x00_audio_dev = {
+	.name		= "db1000-audio",
+};
+
 static int __init db1xxx_dev_init(void)
 {
 #ifdef DB1XXX_HAS_PCMCIA
@@ -113,6 +155,12 @@ static int __init db1xxx_dev_init(void)
 				    1);
 #endif
 	db1x_register_norflash(BOARD_FLASH_SIZE, BOARD_FLASH_WIDTH, F_SWAPPED);
+
+	platform_device_register(&db1x00_codec_dev);
+	platform_device_register(&alchemy_ac97c_dma_dev);
+	platform_device_register(&alchemy_ac97c_dev);
+	platform_device_register(&db1x00_audio_dev);
+
 	return 0;
 }
 device_initcall(db1xxx_dev_init);
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 0460b428862c..6d592546e8fc 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -41,6 +41,15 @@ config SND_SOC_AU1XI2SC
 ##
 ## Boards
 ##
+config SND_SOC_DB1000
+	tristate "DB1000 Audio support"
+	depends on SND_SOC_AU1XAUDIO
+	select SND_SOC_AU1XAC97C
+	select SND_SOC_AC97_CODEC
+	help
+	  Select this option to enable AC97 audio on the early DB1x00 series
+	  of boards (DB1000/DB1500/DB1100).
+
 config SND_SOC_DB1200
 	tristate "DB1200 AC97+I2S audio support"
 	depends on SND_SOC_AU1XPSC
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index ff5531eee613..920710514ea0 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -16,6 +16,8 @@ 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/db1000.c b/sound/soc/au1x/db1000.c
new file mode 100644
index 000000000000..127477a5e0c7
--- /dev/null
+++ b/sound/soc/au1x/db1000.c
@@ -0,0 +1,75 @@
+/*
+ * DB1000/DB1500/DB1100 ASoC audio fabric support code.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "psc.h"
+
+static struct snd_soc_dai_link db1000_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.codec_dai_name	= "ac97-hifi",
+	.cpu_dai_name	= "alchemy-ac97c",
+	.platform_name	= "alchemy-pcm-dma.0",
+	.codec_name	= "ac97-codec",
+};
+
+static struct snd_soc_card db1000_ac97 = {
+	.name		= "DB1000_AC97",
+	.dai_link	= &db1000_ac97_dai,
+	.num_links	= 1,
+};
+
+static int __devinit db1000_audio_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &db1000_ac97;
+	card->dev = &pdev->dev;
+	return snd_soc_register_card(card);
+}
+
+static int __devexit db1000_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver db1000_audio_driver = {
+	.driver	= {
+		.name	= "db1000-audio",
+		.owner	= THIS_MODULE,
+		.pm	= &snd_soc_pm_ops,
+	},
+	.probe		= db1000_audio_probe,
+	.remove		= __devexit_p(db1000_audio_remove),
+};
+
+static int __init db1000_audio_load(void)
+{
+	return platform_driver_register(&db1000_audio_driver);
+}
+
+static void __exit db1000_audio_unload(void)
+{
+	platform_driver_unregister(&db1000_audio_driver);
+}
+
+module_init(db1000_audio_load);
+module_exit(db1000_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio");
+MODULE_AUTHOR("Manuel Lauss");
-- 
cgit v1.2.3


From 5b0912be7a8ff1dbfe56358c5f933d65445bb8af Mon Sep 17 00:00:00 2001
From: Manuel Lauss <manuel.lauss@googlemail.com>
Date: Mon, 25 Jul 2011 13:45:02 +0200
Subject: ASoC: au1x: remove automatic DMA device registration from PSC drivers

The PSC audio drivers (psc-ac97/psc-i2s) register the DMA platform_device
on their own.  This is frowned upon, from now on board code must
register a simple pcm dma platform device for each PSC with sound duties.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/mips/alchemy/devboards/db1200/platform.c |  6 ++
 sound/soc/au1x/dbdma2.c                       | 83 ++++-----------------------
 sound/soc/au1x/psc-ac97.c                     | 34 +++++++----
 sound/soc/au1x/psc-i2s.c                      | 32 +++++++----
 sound/soc/au1x/psc.h                          |  5 --
 5 files changed, 64 insertions(+), 96 deletions(-)

(limited to 'sound/soc/au1x')

diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c
index fbb55935b99e..cfb71aed4cb9 100644
--- a/arch/mips/alchemy/devboards/db1200/platform.c
+++ b/arch/mips/alchemy/devboards/db1200/platform.c
@@ -434,12 +434,18 @@ static struct platform_device db1200_stac_dev = {
 	.id		= 1,	/* on PSC1 */
 };
 
+static struct platform_device db1200_audiodma_dev = {
+	.name		= "au1xpsc-pcm",
+	.id		= 1,	/* PSC ID */
+};
+
 static struct platform_device *db1200_devs[] __initdata = {
 	NULL,		/* PSC0, selected by S6.8 */
 	&db1200_ide_dev,
 	&db1200_eth_dev,
 	&db1200_rtc_dev,
 	&db1200_nand_dev,
+	&db1200_audiodma_dev,
 	&db1200_audio_dev,
 	&db1200_stac_dev,
 };
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 20bb53a837b1..fd5378f7dece 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -293,6 +293,16 @@ au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
 
 static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
 {
+	struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int stype = SUBSTREAM_TYPE(substream), *dmaids;
+
+	dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dmaids)
+		return -ENODEV;	/* whoa, has ordering changed? */
+
+	pcd->ddma_id = dmaids[stype];
+
 	snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
 	return 0;
 }
@@ -340,36 +350,18 @@ struct snd_soc_platform_driver au1xpsc_soc_platform = {
 static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
 {
 	struct au1xpsc_audio_dmadata *dmadata;
-	struct resource *r;
 	int ret;
 
 	dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
 	if (!dmadata)
 		return -ENOMEM;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r) {
-		ret = -ENODEV;
-		goto out1;
-	}
-	dmadata[PCM_TX].ddma_id = r->start;
-
-	/* RX DMA */
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r) {
-		ret = -ENODEV;
-		goto out1;
-	}
-	dmadata[PCM_RX].ddma_id = r->start;
-
 	platform_set_drvdata(pdev, dmadata);
 
 	ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform);
-	if (!ret)
-		return ret;
+	if (ret)
+		kfree(dmadata);
 
-out1:
-	kfree(dmadata);
 	return ret;
 }
 
@@ -405,57 +397,6 @@ static void __exit au1xpsc_audio_dbdma_unload(void)
 module_init(au1xpsc_audio_dbdma_load);
 module_exit(au1xpsc_audio_dbdma_unload);
 
-
-struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev)
-{
-	struct resource *res, *r;
-	struct platform_device *pd;
-	int id[2];
-	int ret;
-
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r)
-		return NULL;
-	id[0] = r->start;
-
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r)
-		return NULL;
-	id[1] = r->start;
-
-	res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL);
-	if (!res)
-		return NULL;
-
-	res[0].start = res[0].end = id[0];
-	res[1].start = res[1].end = id[1];
-	res[0].flags = res[1].flags = IORESOURCE_DMA;
-
-	pd = platform_device_alloc("au1xpsc-pcm", pdev->id);
-	if (!pd)
-		goto out;
-
-	pd->resource = res;
-	pd->num_resources = 2;
-
-	ret = platform_device_add(pd);
-	if (!ret)
-		return pd;
-
-	platform_device_put(pd);
-out:
-	kfree(res);
-	return NULL;
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_add);
-
-void au1xpsc_pcm_destroy(struct platform_device *dmapd)
-{
-	if (dmapd)
-		platform_device_unregister(dmapd);
-}
-EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy);
-
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
 MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index d0db66f24a00..44296abfc38f 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -324,12 +324,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+	return 0;
+}
+
 static int au1xpsc_ac97_probe(struct snd_soc_dai *dai)
 {
 	return au1xpsc_ac97_workdata ? 0 : -ENODEV;
 }
 
 static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.startup	= au1xpsc_ac97_startup,
 	.trigger	= au1xpsc_ac97_trigger,
 	.hw_params	= au1xpsc_ac97_hw_params,
 };
@@ -379,6 +388,16 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 	if (!wd->mmio)
 		goto out1;
 
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		goto out2;
+	wd->dmaids[PCM_TX] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		goto out2;
+	wd->dmaids[PCM_RX] = r->start;
+
 	/* configuration: max dma trigger threshold, enable ac97 */
 	wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
 		  PSC_AC97CFG_DE_ENABLE;
@@ -401,15 +420,13 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 
 	ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
 	if (ret)
-		goto out1;
+		goto out2;
 
-	wd->dmapd = au1xpsc_pcm_add(pdev);
-	if (wd->dmapd) {
-		au1xpsc_ac97_workdata = wd;
-		return 0;
-	}
+	au1xpsc_ac97_workdata = wd;
+	return 0;
 
-	snd_soc_unregister_dai(&pdev->dev);
+out2:
+	iounmap(wd->mmio);
 out1:
 	release_mem_region(r->start, resource_size(r));
 out0:
@@ -422,9 +439,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
 	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
 	struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
-	if (wd->dmapd)
-		au1xpsc_pcm_destroy(wd->dmapd);
-
 	snd_soc_unregister_dai(&pdev->dev);
 
 	/* disable PSC completely */
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index fca091276320..1b7ab5d422e0 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -257,7 +257,16 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+	return 0;
+}
+
 static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.startup	= au1xpsc_i2s_startup,
 	.trigger	= au1xpsc_i2s_trigger,
 	.hw_params	= au1xpsc_i2s_hw_params,
 	.set_fmt	= au1xpsc_i2s_set_fmt,
@@ -304,6 +313,16 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	if (!wd->mmio)
 		goto out1;
 
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		goto out2;
+	wd->dmaids[PCM_TX] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		goto out2;
+	wd->dmaids[PCM_RX] = r->start;
+
 	/* preserve PSC clock source set up by platform (dev.platform_data
 	 * is already occupied by soc layer)
 	 */
@@ -330,15 +349,11 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, wd);
 
 	ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
-	if (ret)
-		goto out1;
-
-	/* finally add the DMA device for this PSC */
-	wd->dmapd = au1xpsc_pcm_add(pdev);
-	if (wd->dmapd)
+	if (!ret)
 		return 0;
 
-	snd_soc_unregister_dai(&pdev->dev);
+out2:
+	iounmap(wd->mmio);
 out1:
 	release_mem_region(r->start, resource_size(r));
 out0:
@@ -351,9 +366,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
 	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
 	struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
-	if (wd->dmapd)
-		au1xpsc_pcm_destroy(wd->dmapd);
-
 	snd_soc_unregister_dai(&pdev->dev);
 
 	au_writel(0, I2S_CFG(wd));
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index c59b9e544e72..1b21c4ffae12 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -19,10 +19,6 @@
 #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);
-
 struct au1xpsc_audio_data {
 	void __iomem *mmio;
 
@@ -34,7 +30,6 @@ struct au1xpsc_audio_data {
 	unsigned long pm[2];
 	struct mutex lock;
 	int dmaids[2];
-	struct platform_device *dmapd;
 };
 
 /* easy access macros */
-- 
cgit v1.2.3


From adbc7a5a61ee9225e0b80d3f5719e05a88db2b4c Mon Sep 17 00:00:00 2001
From: Manuel Lauss <manuel.lauss@googlemail.com>
Date: Mon, 25 Jul 2011 13:45:03 +0200
Subject: ASoC: au1x: update db1200 machine to the new way of things

The use of the "soc-audio" platform device is no longer en vogue,
update the code to the newer, simpler way of doing things.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/mips/alchemy/devboards/db1200/platform.c | 10 +++++
 sound/soc/au1x/db1200.c                       | 64 ++++++++++++++++++---------
 2 files changed, 53 insertions(+), 21 deletions(-)

(limited to 'sound/soc/au1x')

diff --git a/arch/mips/alchemy/devboards/db1200/platform.c b/arch/mips/alchemy/devboards/db1200/platform.c
index cfb71aed4cb9..dda090bf74e6 100644
--- a/arch/mips/alchemy/devboards/db1200/platform.c
+++ b/arch/mips/alchemy/devboards/db1200/platform.c
@@ -422,6 +422,7 @@ static struct resource au1200_psc1_res[] = {
 	},
 };
 
+/* AC97 or I2S device */
 static struct platform_device db1200_audio_dev = {
 	/* name assigned later based on switch setting */
 	.id		= 1,	/* PSC ID */
@@ -429,6 +430,12 @@ static struct platform_device db1200_audio_dev = {
 	.resource	= au1200_psc1_res,
 };
 
+/* DB1200 ASoC card device */
+static struct platform_device db1200_sound_dev = {
+	/* name assigned later based on switch setting */
+	.id		= 1,	/* PSC ID */
+};
+
 static struct platform_device db1200_stac_dev = {
 	.name		= "ac97-codec",
 	.id		= 1,	/* on PSC1 */
@@ -448,6 +455,7 @@ static struct platform_device *db1200_devs[] __initdata = {
 	&db1200_audiodma_dev,
 	&db1200_audio_dev,
 	&db1200_stac_dev,
+	&db1200_sound_dev,
 };
 
 static int __init db1200_dev_init(void)
@@ -507,10 +515,12 @@ static int __init db1200_dev_init(void)
 	if (sw == BCSR_SWITCHES_DIP_8) {
 		bcsr_mod(BCSR_RESETS, 0, BCSR_RESETS_PSC1MUX);
 		db1200_audio_dev.name = "au1xpsc_i2s";
+		db1200_sound_dev.name = "db1200-i2s";
 		printk(KERN_INFO " S6.7 ON : PSC1 mode I2S\n");
 	} else {
 		bcsr_mod(BCSR_RESETS, BCSR_RESETS_PSC1MUX, 0);
 		db1200_audio_dev.name = "au1xpsc_ac97";
+		db1200_sound_dev.name = "db1200-ac97";
 		printk(KERN_INFO " S6.7 OFF: PSC1 mode AC97\n");
 	}
 
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 1d3e258c9ea8..289312c14b99 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -1,7 +1,7 @@
 /*
  * DB1200 ASoC audio fabric support code.
  *
- * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com>
  *
  */
 
@@ -21,6 +21,17 @@
 #include "../codecs/wm8731.h"
 #include "psc.h"
 
+static struct platform_device_id db1200_pids[] = {
+	{
+		.name		= "db1200-ac97",
+		.driver_data	= 0,
+	}, {
+		.name		= "db1200-i2s",
+		.driver_data	= 1,
+	},
+	{},
+};
+
 /*-------------------------  AC97 PART  ---------------------------*/
 
 static struct snd_soc_dai_link db1200_ac97_dai = {
@@ -89,36 +100,47 @@ static struct snd_soc_card db1200_i2s_machine = {
 
 /*-------------------------  COMMON PART  ---------------------------*/
 
-static struct platform_device *db1200_asoc_dev;
+static struct snd_soc_card *db1200_cards[] __devinitdata = {
+	&db1200_ac97_machine,
+	&db1200_i2s_machine,
+};
 
-static int __init db1200_audio_load(void)
+static int __devinit db1200_audio_probe(struct platform_device *pdev)
 {
-	int ret;
+	const struct platform_device_id *pid = platform_get_device_id(pdev);
+	struct snd_soc_card *card;
 
-	ret = -ENOMEM;
-	db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */
-	if (!db1200_asoc_dev)
-		goto out;
+	card = db1200_cards[pid->driver_data];
+	card->dev = &pdev->dev;
+	return snd_soc_register_card(card);
+}
 
-	/* DB1200 board setup set PSC1MUX to preferred audio device */
-	if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
-		platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine);
-	else
-		platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine);
+static int __devexit db1200_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	snd_soc_unregister_card(card);
+	return 0;
+}
 
-	ret = platform_device_add(db1200_asoc_dev);
+static struct platform_driver db1200_audio_driver = {
+	.driver	= {
+		.name	= "db1200-ac97",
+		.owner	= THIS_MODULE,
+		.pm	= &snd_soc_pm_ops,
+	},
+	.id_table	= db1200_pids,
+	.probe		= db1200_audio_probe,
+	.remove		= __devexit_p(db1200_audio_remove),
+};
 
-	if (ret) {
-		platform_device_put(db1200_asoc_dev);
-		db1200_asoc_dev = NULL;
-	}
-out:
-	return ret;
+static int __init db1200_audio_load(void)
+{
+	return platform_driver_register(&db1200_audio_driver);
 }
 
 static void __exit db1200_audio_unload(void)
 {
-	platform_device_unregister(db1200_asoc_dev);
+	platform_driver_unregister(&db1200_audio_driver);
 }
 
 module_init(db1200_audio_load);
-- 
cgit v1.2.3


From 25942fdc824a709c48190356ed058ef7be19fb6a Mon Sep 17 00:00:00 2001
From: Manuel Lauss <manuel.lauss@googlemail.com>
Date: Mon, 25 Jul 2011 13:45:04 +0200
Subject: ASoC: au1x: use substream stream info directly

PCM_TX/RX are the same as SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.  Use
them directly.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/au1x/dbdma2.c   | 10 +++++-----
 sound/soc/au1x/psc-ac97.c | 18 +++++++++---------
 sound/soc/au1x/psc-i2s.c  | 14 +++++++-------
 sound/soc/au1x/psc.h      |  6 ------
 4 files changed, 21 insertions(+), 27 deletions(-)

(limited to 'sound/soc/au1x')

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index fd5378f7dece..d7d04e26eee5 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -169,7 +169,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
 
 	au1x_pcm_dbdma_free(pcd);
 
-	if (stype == PCM_RX)
+	if (stype == SNDRV_PCM_STREAM_CAPTURE)
 		pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
 					DSCR_CMD0_ALWAYS,
 					au1x_pcm_dmarx_cb, (void *)pcd);
@@ -198,7 +198,7 @@ static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream
 	struct snd_soc_pcm_runtime *rtd = ss->private_data;
 	struct au1xpsc_audio_dmadata *pcd =
 				snd_soc_platform_get_drvdata(rtd->platform);
-	return &pcd[SUBSTREAM_TYPE(ss)];
+	return &pcd[ss->stream];
 }
 
 static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -212,7 +212,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		goto out;
 
-	stype = SUBSTREAM_TYPE(substream);
+	stype = substream->stream;
 	pcd = to_dmadata(substream);
 
 	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
@@ -255,7 +255,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
 
 	au1xxx_dbdma_reset(pcd->ddma_chan);
 
-	if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		au1x_pcm_queue_rx(pcd);
 		au1x_pcm_queue_rx(pcd);
 	} else {
@@ -295,7 +295,7 @@ 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_TYPE(substream), *dmaids;
+	int stype = substream->stream, *dmaids;
 
 	dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 	if (!dmaids)
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 44296abfc38f..172eefd38b2d 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -41,14 +41,14 @@
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
 
 #define AC97PCR_START(stype)	\
-	((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
 #define AC97PCR_STOP(stype)	\
-	((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
 #define AC97PCR_CLRFIFO(stype)	\
-	((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
 
 #define AC97STAT_BUSY(stype)	\
-	((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
 
 /* instance data. There can be only one, MacLeod!!!! */
 static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
@@ -215,7 +215,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
 	unsigned long r, ro, stat;
-	int chans, t, stype = SUBSTREAM_TYPE(substream);
+	int chans, t, stype = substream->stream;
 
 	chans = params_channels(params);
 
@@ -235,7 +235,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 		r |= PSC_AC97CFG_SET_LEN(params->msbits);
 
 		/* channels: enable slots for front L/R channel */
-		if (stype == PCM_TX) {
+		if (stype == SNDRV_PCM_STREAM_PLAYBACK) {
 			r &= ~PSC_AC97CFG_TXSLOT_MASK;
 			r |= PSC_AC97CFG_TXSLOT_ENA(3);
 			r |= PSC_AC97CFG_TXSLOT_ENA(4);
@@ -294,7 +294,7 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 				int cmd, struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
-	int ret, stype = SUBSTREAM_TYPE(substream);
+	int ret, stype = substream->stream;
 
 	ret = 0;
 
@@ -391,12 +391,12 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!r)
 		goto out2;
-	wd->dmaids[PCM_TX] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
 
 	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!r)
 		goto out2;
-	wd->dmaids[PCM_RX] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
 
 	/* configuration: max dma trigger threshold, enable ac97 */
 	wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 1b7ab5d422e0..7c5ae920544f 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -42,13 +42,13 @@
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 #define I2SSTAT_BUSY(stype)	\
-	((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
 #define I2SPCR_START(stype)	\
-	((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
 #define I2SPCR_STOP(stype)	\
-	((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
 #define I2SPCR_CLRFIFO(stype)	\
-	((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
 
 
 static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
@@ -240,7 +240,7 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			       struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
-	int ret, stype = SUBSTREAM_TYPE(substream);
+	int ret, stype = substream->stream;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -316,12 +316,12 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!r)
 		goto out2;
-	wd->dmaids[PCM_TX] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
 
 	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!r)
 		goto out2;
-	wd->dmaids[PCM_RX] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
 
 	/* preserve PSC clock source set up by platform (dev.platform_data
 	 * is already occupied by soc layer)
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index 1b21c4ffae12..b16b2e02e0c9 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -13,12 +13,6 @@
 #ifndef _AU1X_PCM_H
 #define _AU1X_PCM_H
 
-#define PCM_TX	0
-#define PCM_RX	1
-
-#define SUBSTREAM_TYPE(substream) \
-	((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
-
 struct au1xpsc_audio_data {
 	void __iomem *mmio;
 
-- 
cgit v1.2.3


From 4f7e7954a7f66735b0ee4b304c075c24ffae091a Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@gmail.com>
Date: Wed, 17 Aug 2011 15:14:33 +0800
Subject: ASoC: Remove unreachable code in au1xac97c_drvprobe and
 au1xi2s_drvprobe

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/au1x/ac97c.c | 2 --
 sound/soc/au1x/i2sc.c  | 1 -
 2 files changed, 3 deletions(-)

(limited to 'sound/soc/au1x')

diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
index 9c05f381d95e..13802ff7cf05 100644
--- a/sound/soc/au1x/ac97c.c
+++ b/sound/soc/au1x/ac97c.c
@@ -275,8 +275,6 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
 	ac97c_workdata = ctx;
 	return 0;
 
-
-	snd_soc_unregister_dai(&pdev->dev);
 out1:
 	release_mem_region(r->start, resource_size(r));
 out0:
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
index b4172fdd2c48..19e0d2a9c828 100644
--- a/sound/soc/au1x/i2sc.c
+++ b/sound/soc/au1x/i2sc.c
@@ -267,7 +267,6 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
 
 	return 0;
 
-	snd_soc_unregister_dai(&pdev->dev);
 out1:
 	release_mem_region(r->start, resource_size(r));
 out0:
-- 
cgit v1.2.3