From e50fad4f029c36ed85a71fe7413684cfd3c7d78c Mon Sep 17 00:00:00 2001
From: "ramesh.babu@linux.intel.com" <ramesh.babu@linux.intel.com>
Date: Thu, 27 Oct 2011 12:12:33 +0530
Subject: ASoC: Allow machines to ignore pmdown_time per-link

With this flag, each dai_link in machine driver can choose
to ignore pmdown_time during DAPM shut down sequence.

If the ignore_pmdown_time is set, the DAPM for corresponding DAI
will be executed immediately.

Signed-off-by: Ramesh Babu K V <ramesh.babu@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 11cfb5953e0..877fcc1e016 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -718,6 +718,9 @@ struct snd_soc_dai_link {
 	/* Symmetry requirements */
 	unsigned int symmetric_rates:1;
 
+	/* pmdown_time is ignored at stop */
+	unsigned int ignore_pmdown_time:1;
+
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_pcm_runtime *rtd);
 
-- 
cgit v1.2.3-70-g09d2


From d66a327ddad647fd1678fd24d9070846737c6834 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 10 Nov 2011 11:46:13 +0000
Subject: ASoC: Remove extra space in runtime struct definition

My usual technique for finding definitions is to search for "name {"
which breaks with the extra space.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 877fcc1e016..02a5c5519f3 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -843,7 +843,7 @@ struct snd_soc_card {
 };
 
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
-struct snd_soc_pcm_runtime  {
+struct snd_soc_pcm_runtime {
 	struct device dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
-- 
cgit v1.2.3-70-g09d2


From c9016a7937122b72d87ff2037664b7bd717d3e4b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 10 Nov 2011 13:06:52 +0000
Subject: ASoC: Remove LZO cache type

There are no current users and new drivers ought to be using the regmap
API and its cache implementation directly so just delete the ASoC copy.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h   |   1 -
 sound/soc/Kconfig     |  15 --
 sound/soc/soc-cache.c | 384 --------------------------------------------------
 3 files changed, 400 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 02a5c5519f3..b21b3047e91 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -266,7 +266,6 @@ enum snd_soc_control_type {
 
 enum snd_soc_compress_type {
 	SND_SOC_FLAT_COMPRESSION = 1,
-	SND_SOC_LZO_COMPRESSION,
 	SND_SOC_RBTREE_COMPRESSION
 };
 
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 1381db853ef..35e662d270e 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -22,21 +22,6 @@ menuconfig SND_SOC
 
 if SND_SOC
 
-config SND_SOC_CACHE_LZO
-	bool "Support LZO compression for register caches"
-	select LZO_COMPRESS
-	select LZO_DECOMPRESS
-	---help---
-	   Select this to enable LZO compression for register caches.
-	   This will allow machine or CODEC drivers to compress register
-	   caches in memory, reducing the memory consumption at the
-	   expense of performance.  If this is not present and is used
-	   the system will fall back to uncompressed caches.
-
-	   Usually it is safe to disable this option, where cache
-	   compression in used the rbtree option will typically perform
-	   better.
-
 config SND_SOC_AC97_BUS
 	bool
 
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 9077aa4b3b4..18bb6b3335e 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -14,7 +14,6 @@
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
 #include <sound/soc.h>
-#include <linux/lzo.h>
 #include <linux/bitmap.h>
 #include <linux/rbtree.h>
 #include <linux/export.h>
@@ -439,378 +438,6 @@ err:
 	return ret;
 }
 
-#ifdef CONFIG_SND_SOC_CACHE_LZO
-struct snd_soc_lzo_ctx {
-	void *wmem;
-	void *dst;
-	const void *src;
-	size_t src_len;
-	size_t dst_len;
-	size_t decompressed_size;
-	unsigned long *sync_bmp;
-	int sync_bmp_nbits;
-};
-
-#define LZO_BLOCK_NUM 8
-static int snd_soc_lzo_block_count(void)
-{
-	return LZO_BLOCK_NUM;
-}
-
-static int snd_soc_lzo_prepare(struct snd_soc_lzo_ctx *lzo_ctx)
-{
-	lzo_ctx->wmem = kmalloc(LZO1X_MEM_COMPRESS, GFP_KERNEL);
-	if (!lzo_ctx->wmem)
-		return -ENOMEM;
-	return 0;
-}
-
-static int snd_soc_lzo_compress(struct snd_soc_lzo_ctx *lzo_ctx)
-{
-	size_t compress_size;
-	int ret;
-
-	ret = lzo1x_1_compress(lzo_ctx->src, lzo_ctx->src_len,
-			       lzo_ctx->dst, &compress_size, lzo_ctx->wmem);
-	if (ret != LZO_E_OK || compress_size > lzo_ctx->dst_len)
-		return -EINVAL;
-	lzo_ctx->dst_len = compress_size;
-	return 0;
-}
-
-static int snd_soc_lzo_decompress(struct snd_soc_lzo_ctx *lzo_ctx)
-{
-	size_t dst_len;
-	int ret;
-
-	dst_len = lzo_ctx->dst_len;
-	ret = lzo1x_decompress_safe(lzo_ctx->src, lzo_ctx->src_len,
-				    lzo_ctx->dst, &dst_len);
-	if (ret != LZO_E_OK || dst_len != lzo_ctx->dst_len)
-		return -EINVAL;
-	return 0;
-}
-
-static int snd_soc_lzo_compress_cache_block(struct snd_soc_codec *codec,
-		struct snd_soc_lzo_ctx *lzo_ctx)
-{
-	int ret;
-
-	lzo_ctx->dst_len = lzo1x_worst_compress(PAGE_SIZE);
-	lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL);
-	if (!lzo_ctx->dst) {
-		lzo_ctx->dst_len = 0;
-		return -ENOMEM;
-	}
-
-	ret = snd_soc_lzo_compress(lzo_ctx);
-	if (ret < 0)
-		return ret;
-	return 0;
-}
-
-static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec,
-		struct snd_soc_lzo_ctx *lzo_ctx)
-{
-	int ret;
-
-	lzo_ctx->dst_len = lzo_ctx->decompressed_size;
-	lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL);
-	if (!lzo_ctx->dst) {
-		lzo_ctx->dst_len = 0;
-		return -ENOMEM;
-	}
-
-	ret = snd_soc_lzo_decompress(lzo_ctx);
-	if (ret < 0)
-		return ret;
-	return 0;
-}
-
-static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec,
-		unsigned int reg)
-{
-	const struct snd_soc_codec_driver *codec_drv;
-
-	codec_drv = codec->driver;
-	return (reg * codec_drv->reg_word_size) /
-	       DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count());
-}
-
-static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec,
-		unsigned int reg)
-{
-	const struct snd_soc_codec_driver *codec_drv;
-
-	codec_drv = codec->driver;
-	return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) /
-		      codec_drv->reg_word_size);
-}
-
-static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec)
-{
-	return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count());
-}
-
-static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec)
-{
-	struct snd_soc_lzo_ctx **lzo_blocks;
-	unsigned int val;
-	int i;
-	int ret;
-
-	lzo_blocks = codec->reg_cache;
-	for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) {
-		WARN_ON(!snd_soc_codec_writable_register(codec, i));
-		ret = snd_soc_cache_read(codec, i, &val);
-		if (ret)
-			return ret;
-		codec->cache_bypass = 1;
-		ret = snd_soc_write(codec, i, val);
-		codec->cache_bypass = 0;
-		if (ret)
-			return ret;
-		dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
-			i, val);
-	}
-
-	return 0;
-}
-
-static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec,
-				   unsigned int reg, unsigned int value)
-{
-	struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks;
-	int ret, blkindex, blkpos;
-	size_t blksize, tmp_dst_len;
-	void *tmp_dst;
-
-	/* index of the compressed lzo block */
-	blkindex = snd_soc_lzo_get_blkindex(codec, reg);
-	/* register index within the decompressed block */
-	blkpos = snd_soc_lzo_get_blkpos(codec, reg);
-	/* size of the compressed block */
-	blksize = snd_soc_lzo_get_blksize(codec);
-	lzo_blocks = codec->reg_cache;
-	lzo_block = lzo_blocks[blkindex];
-
-	/* save the pointer and length of the compressed block */
-	tmp_dst = lzo_block->dst;
-	tmp_dst_len = lzo_block->dst_len;
-
-	/* prepare the source to be the compressed block */
-	lzo_block->src = lzo_block->dst;
-	lzo_block->src_len = lzo_block->dst_len;
-
-	/* decompress the block */
-	ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block);
-	if (ret < 0) {
-		kfree(lzo_block->dst);
-		goto out;
-	}
-
-	/* write the new value to the cache */
-	if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value,
-				  codec->driver->reg_word_size)) {
-		kfree(lzo_block->dst);
-		goto out;
-	}
-
-	/* prepare the source to be the decompressed block */
-	lzo_block->src = lzo_block->dst;
-	lzo_block->src_len = lzo_block->dst_len;
-
-	/* compress the block */
-	ret = snd_soc_lzo_compress_cache_block(codec, lzo_block);
-	if (ret < 0) {
-		kfree(lzo_block->dst);
-		kfree(lzo_block->src);
-		goto out;
-	}
-
-	/* set the bit so we know we have to sync this register */
-	set_bit(reg, lzo_block->sync_bmp);
-	kfree(tmp_dst);
-	kfree(lzo_block->src);
-	return 0;
-out:
-	lzo_block->dst = tmp_dst;
-	lzo_block->dst_len = tmp_dst_len;
-	return ret;
-}
-
-static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec,
-				  unsigned int reg, unsigned int *value)
-{
-	struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks;
-	int ret, blkindex, blkpos;
-	size_t blksize, tmp_dst_len;
-	void *tmp_dst;
-
-	*value = 0;
-	/* index of the compressed lzo block */
-	blkindex = snd_soc_lzo_get_blkindex(codec, reg);
-	/* register index within the decompressed block */
-	blkpos = snd_soc_lzo_get_blkpos(codec, reg);
-	/* size of the compressed block */
-	blksize = snd_soc_lzo_get_blksize(codec);
-	lzo_blocks = codec->reg_cache;
-	lzo_block = lzo_blocks[blkindex];
-
-	/* save the pointer and length of the compressed block */
-	tmp_dst = lzo_block->dst;
-	tmp_dst_len = lzo_block->dst_len;
-
-	/* prepare the source to be the compressed block */
-	lzo_block->src = lzo_block->dst;
-	lzo_block->src_len = lzo_block->dst_len;
-
-	/* decompress the block */
-	ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block);
-	if (ret >= 0)
-		/* fetch the value from the cache */
-		*value = snd_soc_get_cache_val(lzo_block->dst, blkpos,
-					       codec->driver->reg_word_size);
-
-	kfree(lzo_block->dst);
-	/* restore the pointer and length of the compressed block */
-	lzo_block->dst = tmp_dst;
-	lzo_block->dst_len = tmp_dst_len;
-	return 0;
-}
-
-static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec)
-{
-	struct snd_soc_lzo_ctx **lzo_blocks;
-	int i, blkcount;
-
-	lzo_blocks = codec->reg_cache;
-	if (!lzo_blocks)
-		return 0;
-
-	blkcount = snd_soc_lzo_block_count();
-	/*
-	 * the pointer to the bitmap used for syncing the cache
-	 * is shared amongst all lzo_blocks.  Ensure it is freed
-	 * only once.
-	 */
-	if (lzo_blocks[0])
-		kfree(lzo_blocks[0]->sync_bmp);
-	for (i = 0; i < blkcount; ++i) {
-		if (lzo_blocks[i]) {
-			kfree(lzo_blocks[i]->wmem);
-			kfree(lzo_blocks[i]->dst);
-		}
-		/* each lzo_block is a pointer returned by kmalloc or NULL */
-		kfree(lzo_blocks[i]);
-	}
-	kfree(lzo_blocks);
-	codec->reg_cache = NULL;
-	return 0;
-}
-
-static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
-{
-	struct snd_soc_lzo_ctx **lzo_blocks;
-	size_t bmp_size;
-	const struct snd_soc_codec_driver *codec_drv;
-	int ret, tofree, i, blksize, blkcount;
-	const char *p, *end;
-	unsigned long *sync_bmp;
-
-	ret = 0;
-	codec_drv = codec->driver;
-
-	/*
-	 * If we have not been given a default register cache
-	 * then allocate a dummy zero-ed out region, compress it
-	 * and remember to free it afterwards.
-	 */
-	tofree = 0;
-	if (!codec->reg_def_copy)
-		tofree = 1;
-
-	if (!codec->reg_def_copy) {
-		codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL);
-		if (!codec->reg_def_copy)
-			return -ENOMEM;
-	}
-
-	blkcount = snd_soc_lzo_block_count();
-	codec->reg_cache = kzalloc(blkcount * sizeof *lzo_blocks,
-				   GFP_KERNEL);
-	if (!codec->reg_cache) {
-		ret = -ENOMEM;
-		goto err_tofree;
-	}
-	lzo_blocks = codec->reg_cache;
-
-	/*
-	 * allocate a bitmap to be used when syncing the cache with
-	 * the hardware.  Each time a register is modified, the corresponding
-	 * bit is set in the bitmap, so we know that we have to sync
-	 * that register.
-	 */
-	bmp_size = codec_drv->reg_cache_size;
-	sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof(long),
-			   GFP_KERNEL);
-	if (!sync_bmp) {
-		ret = -ENOMEM;
-		goto err;
-	}
-	bitmap_zero(sync_bmp, bmp_size);
-
-	/* allocate the lzo blocks and initialize them */
-	for (i = 0; i < blkcount; ++i) {
-		lzo_blocks[i] = kzalloc(sizeof **lzo_blocks,
-					GFP_KERNEL);
-		if (!lzo_blocks[i]) {
-			kfree(sync_bmp);
-			ret = -ENOMEM;
-			goto err;
-		}
-		lzo_blocks[i]->sync_bmp = sync_bmp;
-		lzo_blocks[i]->sync_bmp_nbits = bmp_size;
-		/* alloc the working space for the compressed block */
-		ret = snd_soc_lzo_prepare(lzo_blocks[i]);
-		if (ret < 0)
-			goto err;
-	}
-
-	blksize = snd_soc_lzo_get_blksize(codec);
-	p = codec->reg_def_copy;
-	end = codec->reg_def_copy + codec->reg_size;
-	/* compress the register map and fill the lzo blocks */
-	for (i = 0; i < blkcount; ++i, p += blksize) {
-		lzo_blocks[i]->src = p;
-		if (p + blksize > end)
-			lzo_blocks[i]->src_len = end - p;
-		else
-			lzo_blocks[i]->src_len = blksize;
-		ret = snd_soc_lzo_compress_cache_block(codec,
-						       lzo_blocks[i]);
-		if (ret < 0)
-			goto err;
-		lzo_blocks[i]->decompressed_size =
-			lzo_blocks[i]->src_len;
-	}
-
-	if (tofree) {
-		kfree(codec->reg_def_copy);
-		codec->reg_def_copy = NULL;
-	}
-	return 0;
-err:
-	snd_soc_cache_exit(codec);
-err_tofree:
-	if (tofree) {
-		kfree(codec->reg_def_copy);
-		codec->reg_def_copy = NULL;
-	}
-	return ret;
-}
-#endif
-
 static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 {
 	int i;
@@ -889,17 +516,6 @@ static const struct snd_soc_cache_ops cache_types[] = {
 		.write = snd_soc_flat_cache_write,
 		.sync = snd_soc_flat_cache_sync
 	},
-#ifdef CONFIG_SND_SOC_CACHE_LZO
-	{
-		.id = SND_SOC_LZO_COMPRESSION,
-		.name = "LZO",
-		.init = snd_soc_lzo_cache_init,
-		.exit = snd_soc_lzo_cache_exit,
-		.read = snd_soc_lzo_cache_read,
-		.write = snd_soc_lzo_cache_write,
-		.sync = snd_soc_lzo_cache_sync
-	},
-#endif
 	{
 		.id = SND_SOC_RBTREE_COMPRESSION,
 		.name = "rbtree",
-- 
cgit v1.2.3-70-g09d2


From e012ba249171a205c5735a76b947bdae9cf34c6e Mon Sep 17 00:00:00 2001
From: Johannes Stezenbach <js@sig21.net>
Date: Mon, 14 Nov 2011 17:23:17 +0100
Subject: ASoC: sta32x: add platform data definition

Add a structure for platform specific configuration and use it,
thereby removing a few FIXMEs which marked hard-coded values.

Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/sta32x.h    | 34 ++++++++++++++++++++++++++++++++++
 sound/soc/codecs/sta32x.c | 30 +++++++++++++++++++++---------
 2 files changed, 55 insertions(+), 9 deletions(-)
 create mode 100644 include/sound/sta32x.h

(limited to 'include/sound')

diff --git a/include/sound/sta32x.h b/include/sound/sta32x.h
new file mode 100644
index 00000000000..45d7477c049
--- /dev/null
+++ b/include/sound/sta32x.h
@@ -0,0 +1,34 @@
+/*
+ * Platform data for ST STA32x ASoC codec driver.
+ *
+ * Copyright: 2011 Raumfeld GmbH
+ * Author: Johannes Stezenbach <js@sig21.net>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+#ifndef __LINUX_SND__STA32X_H
+#define __LINUX_SND__STA32X_H
+
+#define STA32X_OCFG_2CH		0
+#define STA32X_OCFG_2_1CH	1
+#define STA32X_OCFG_1CH		3
+
+#define STA32X_OM_CH1		0
+#define STA32X_OM_CH2		1
+#define STA32X_OM_CH3		2
+
+#define STA32X_THERMAL_ADJUSTMENT_ENABLE	1
+#define STA32X_THERMAL_RECOVERY_ENABLE		2
+
+struct sta32x_platform_data {
+	int output_conf;
+	int ch1_output_mapping;
+	int ch2_output_mapping;
+	int ch3_output_mapping;
+	int thermal_conf;
+};
+
+#endif /* __LINUX_SND__STA32X_H */
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index d2f37152f94..97091e3b9a0 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -35,6 +35,7 @@
 #include <sound/initval.h>
 #include <sound/tlv.h>
 
+#include <sound/sta32x.h>
 #include "sta32x.h"
 
 #define STA32X_RATES (SNDRV_PCM_RATE_32000 | \
@@ -73,6 +74,7 @@ static const char *sta32x_supply_names[] = {
 struct sta32x_priv {
 	struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)];
 	struct snd_soc_codec *codec;
+	struct sta32x_platform_data *pdata;
 
 	unsigned int mclk;
 	unsigned int format;
@@ -775,9 +777,10 @@ static int sta32x_resume(struct snd_soc_codec *codec)
 static int sta32x_probe(struct snd_soc_codec *codec)
 {
 	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
-	int i, ret = 0;
+	int i, ret = 0, thermal = 0;
 
 	sta32x->codec = codec;
+	sta32x->pdata = dev_get_platdata(codec->dev);
 
 	/* regulators */
 	for (i = 0; i < ARRAY_SIZE(sta32x->supplies); i++)
@@ -820,25 +823,34 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 	snd_soc_cache_write(codec, STA32X_AUTO3, 0x00);
 	snd_soc_cache_write(codec, STA32X_C3CFG, 0x40);
 
-	/* FIXME enable thermal warning adjustment and recovery  */
+	/* set thermal warning adjustment and recovery */
+	if (!(sta32x->pdata->thermal_conf & STA32X_THERMAL_ADJUSTMENT_ENABLE))
+		thermal |= STA32X_CONFA_TWAB;
+	if (!(sta32x->pdata->thermal_conf & STA32X_THERMAL_RECOVERY_ENABLE))
+		thermal |= STA32X_CONFA_TWRB;
 	snd_soc_update_bits(codec, STA32X_CONFA,
-			    STA32X_CONFA_TWAB | STA32X_CONFA_TWRB, 0);
+			    STA32X_CONFA_TWAB | STA32X_CONFA_TWRB,
+			    thermal);
 
-	/* FIXME select 2.1 mode  */
+	/* select output configuration  */
 	snd_soc_update_bits(codec, STA32X_CONFF,
 			    STA32X_CONFF_OCFG_MASK,
-			    1 << STA32X_CONFF_OCFG_SHIFT);
+			    sta32x->pdata->output_conf
+			    << STA32X_CONFF_OCFG_SHIFT);
 
-	/* FIXME channel to output mapping */
+	/* channel to output mapping */
 	snd_soc_update_bits(codec, STA32X_C1CFG,
 			    STA32X_CxCFG_OM_MASK,
-			    0 << STA32X_CxCFG_OM_SHIFT);
+			    sta32x->pdata->ch1_output_mapping
+			    << STA32X_CxCFG_OM_SHIFT);
 	snd_soc_update_bits(codec, STA32X_C2CFG,
 			    STA32X_CxCFG_OM_MASK,
-			    1 << STA32X_CxCFG_OM_SHIFT);
+			    sta32x->pdata->ch2_output_mapping
+			    << STA32X_CxCFG_OM_SHIFT);
 	snd_soc_update_bits(codec, STA32X_C3CFG,
 			    STA32X_CxCFG_OM_MASK,
-			    2 << STA32X_CxCFG_OM_SHIFT);
+			    sta32x->pdata->ch3_output_mapping
+			    << STA32X_CxCFG_OM_SHIFT);
 
 	/* initialize coefficient shadow RAM with reset values */
 	for (i = 4; i <= 49; i += 5)
-- 
cgit v1.2.3-70-g09d2


From 3fb5eac50d66cab4a41177269432ffffcc3e67ac Mon Sep 17 00:00:00 2001
From: Johannes Stezenbach <js@sig21.net>
Date: Mon, 14 Nov 2011 17:23:18 +0100
Subject: ASoC: sta32x: add workaround for ESD reset issue

sta32x resets and loses all configuration during ESD test.
Work around by polling the CONFA register once a second
and restore all coeffcients and registers when CONFA
changes unexpectedly.

Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/sta32x.h    |  1 +
 sound/soc/codecs/sta32x.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++-
 2 files changed, 50 insertions(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/sta32x.h b/include/sound/sta32x.h
index 45d7477c049..8d93b0357a1 100644
--- a/include/sound/sta32x.h
+++ b/include/sound/sta32x.h
@@ -29,6 +29,7 @@ struct sta32x_platform_data {
 	int ch2_output_mapping;
 	int ch3_output_mapping;
 	int thermal_conf;
+	int needs_esd_watchdog;
 };
 
 #endif /* __LINUX_SND__STA32X_H */
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 97091e3b9a0..3b0deafd766 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -27,6 +27,7 @@
 #include <linux/platform_device.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
+#include <linux/workqueue.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -80,6 +81,8 @@ struct sta32x_priv {
 	unsigned int format;
 
 	u32 coef_shadow[STA32X_COEF_COUNT];
+	struct delayed_work watchdog_work;
+	int shutdown;
 };
 
 static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
@@ -304,6 +307,46 @@ int sta32x_cache_sync(struct snd_soc_codec *codec)
 	return rc;
 }
 
+/* work around ESD issue where sta32x resets and loses all configuration */
+static void sta32x_watchdog(struct work_struct *work)
+{
+	struct sta32x_priv *sta32x = container_of(work, struct sta32x_priv,
+						  watchdog_work.work);
+	struct snd_soc_codec *codec = sta32x->codec;
+	unsigned int confa, confa_cached;
+
+	/* check if sta32x has reset itself */
+	confa_cached = snd_soc_read(codec, STA32X_CONFA);
+	codec->cache_bypass = 1;
+	confa = snd_soc_read(codec, STA32X_CONFA);
+	codec->cache_bypass = 0;
+	if (confa != confa_cached) {
+		codec->cache_sync = 1;
+		sta32x_cache_sync(codec);
+	}
+
+	if (!sta32x->shutdown)
+		schedule_delayed_work(&sta32x->watchdog_work,
+				      round_jiffies_relative(HZ));
+}
+
+static void sta32x_watchdog_start(struct sta32x_priv *sta32x)
+{
+	if (sta32x->pdata->needs_esd_watchdog) {
+		sta32x->shutdown = 0;
+		schedule_delayed_work(&sta32x->watchdog_work,
+				      round_jiffies_relative(HZ));
+	}
+}
+
+static void sta32x_watchdog_stop(struct sta32x_priv *sta32x)
+{
+	if (sta32x->pdata->needs_esd_watchdog) {
+		sta32x->shutdown = 1;
+		cancel_delayed_work_sync(&sta32x->watchdog_work);
+	}
+}
+
 #define SINGLE_COEF(xname, index) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = sta32x_coefficient_info, \
@@ -714,6 +757,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 			}
 
 			sta32x_cache_sync(codec);
+			sta32x_watchdog_start(sta32x);
 		}
 
 		/* Power up to mute */
@@ -730,7 +774,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
 				    STA32X_CONFF_PWDN | STA32X_CONFF_EAPD,
 				    STA32X_CONFF_PWDN);
 		msleep(300);
-
+		sta32x_watchdog_stop(sta32x);
 		regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies),
 				       sta32x->supplies);
 		break;
@@ -863,6 +907,9 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 	sta32x->coef_shadow[60] = 0x400000;
 	sta32x->coef_shadow[61] = 0x400000;
 
+	if (sta32x->pdata->needs_esd_watchdog)
+		INIT_DELAYED_WORK(&sta32x->watchdog_work, sta32x_watchdog);
+
 	sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
@@ -879,6 +926,7 @@ static int sta32x_remove(struct snd_soc_codec *codec)
 {
 	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
 
+	sta32x_watchdog_stop(sta32x);
 	sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
 	regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
-- 
cgit v1.2.3-70-g09d2


From 35be544af367170a9c6bf63adcf9d0cb2d569dbb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Nov 2011 08:36:06 +0100
Subject: ALSA: Introduce common helper functions for jack-detection control

Now move the helper function for creating and reporting the jack-detection
to the common place.  The driver that needs this functionality should
select CONFIG_SND_KCTL_JACK kconfig.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/control.h  |  8 +++++++
 sound/core/Kconfig       |  3 +++
 sound/core/Makefile      |  1 +
 sound/core/ctljack.c     | 55 +++++++++++++++++++++++++++++++++++++++++++++++
 sound/pci/hda/Kconfig    |  1 +
 sound/pci/hda/hda_jack.c | 56 +++++++-----------------------------------------
 sound/pci/hda/hda_jack.h |  1 -
 7 files changed, 76 insertions(+), 49 deletions(-)
 create mode 100644 sound/core/ctljack.c

(limited to 'include/sound')

diff --git a/include/sound/control.h b/include/sound/control.h
index 1a94a216ed9..b2796e83c7a 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -227,4 +227,12 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
 	return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
 }
 
+/*
+ * Helper functions for jack-detection controls
+ */
+struct snd_kcontrol *
+snd_kctl_jack_new(const char *name, int idx, void *private_data);
+void snd_kctl_jack_report(struct snd_card *card,
+			  struct snd_kcontrol *kctl, bool status);
+
 #endif	/* __SOUND_CONTROL_H */
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 475455c7661..66f287f2f75 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -207,6 +207,9 @@ config SND_PCM_XRUN_DEBUG
 config SND_VMASTER
 	bool
 
+config SND_KCTL_JACK
+	bool
+
 config SND_DMA_SGBUF
 	def_bool y
 	depends on X86
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 350a08d277f..b4637c3e776 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -7,6 +7,7 @@ snd-y     := sound.o init.o memory.o info.o control.o misc.o device.o
 snd-$(CONFIG_ISA_DMA_API) += isadma.o
 snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o info_oss.o
 snd-$(CONFIG_SND_VMASTER) += vmaster.o
+snd-$(CONFIG_SND_KCTL_JACK) += ctljack.o
 snd-$(CONFIG_SND_JACK)	  += jack.o
 
 snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
diff --git a/sound/core/ctljack.c b/sound/core/ctljack.c
new file mode 100644
index 00000000000..af0e78a3809
--- /dev/null
+++ b/sound/core/ctljack.c
@@ -0,0 +1,55 @@
+/*
+ * Helper functions for jack-detection kcontrols
+ *
+ * Copyright (c) 2011 Takashi Iwai <tiwai@suse.de>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the Free
+ * Software Foundation; either version 2 of the License, or (at your option)
+ * any later version.
+ */
+
+#include <linux/kernel.h>
+#include <sound/core.h>
+#include <sound/control.h>
+
+#define jack_detect_kctl_info	snd_ctl_boolean_mono_info
+
+static int jack_detect_kctl_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = kcontrol->private_value;
+	return 0;
+}
+
+static struct snd_kcontrol_new jack_detect_kctl = {
+	/* name is filled later */
+	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
+	.access = SNDRV_CTL_ELEM_ACCESS_READ,
+	.info = jack_detect_kctl_info,
+	.get = jack_detect_kctl_get,
+};
+
+struct snd_kcontrol *
+snd_kctl_jack_new(const char *name, int idx, void *private_data)
+{
+	struct snd_kcontrol *kctl;
+	kctl = snd_ctl_new1(&jack_detect_kctl, private_data);
+	if (!kctl)
+		return NULL;
+	snprintf(kctl->id.name, sizeof(kctl->id.name), "%s Jack", name);
+	kctl->id.index = idx;
+	kctl->private_value = 0;
+	return kctl;
+}
+EXPORT_SYMBOL_GPL(snd_kctl_jack_new);
+
+void snd_kctl_jack_report(struct snd_card *card,
+			  struct snd_kcontrol *kctl, bool status)
+{
+	if (kctl->private_value == status)
+		return;
+	kctl->private_value = status;
+	snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &kctl->id);
+}
+EXPORT_SYMBOL_GPL(snd_kctl_jack_report);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index bb7e102d672..163b6b5de3e 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -2,6 +2,7 @@ menuconfig SND_HDA_INTEL
 	tristate "Intel HD Audio"
 	select SND_PCM
 	select SND_VMASTER
+	select SND_KCTL_JACK
 	help
 	  Say Y here to include support for Intel "High Definition
 	  Audio" (Azalia) and its compatible devices.
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 8829d5c83fe..a2ab52b2726 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -97,12 +97,8 @@ static void jack_detect_update(struct hda_codec *codec,
 			       struct hda_jack_tbl *jack)
 {
 	if (jack->jack_dirty || !jack->jack_detect) {
-		unsigned int val = read_pin_sense(codec, jack->nid);
+		jack->pin_sense = read_pin_sense(codec, jack->nid);
 		jack->jack_dirty = 0;
-		if (val != jack->pin_sense) {
-			jack->need_notify = 1;
-			jack->pin_sense = val;
-		}
 	}
 }
 
@@ -142,6 +138,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 }
 EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
 
+#define get_jack_plug_state(sense) !!(sense & AC_PINSENSE_PRESENCE)
+
 /**
  * snd_hda_jack_detect - query pin Presence Detect status
  * @codec: the CODEC to sense
@@ -152,7 +150,7 @@ EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
 int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
 {
 	u32 sense = snd_hda_pin_sense(codec, nid);
-	return !!(sense & AC_PINSENSE_PRESENCE);
+	return get_jack_plug_state(sense);
 }
 EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
 
@@ -176,58 +174,23 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
 }
 EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
 
-/* queue the notification when needed */
-static void jack_detect_report(struct hda_codec *codec,
-			       struct hda_jack_tbl *jack)
-{
-	jack_detect_update(codec, jack);
-	if (jack->need_notify) {
-		snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE,
-			       &jack->kctl->id);
-		jack->need_notify = 0;
-	}
-}
-
 /**
  * snd_hda_jack_report_sync - sync the states of all jacks and report if changed
  */
 void snd_hda_jack_report_sync(struct hda_codec *codec)
 {
 	struct hda_jack_tbl *jack = codec->jacktbl.list;
-	int i;
+	int i, state;
 
 	for (i = 0; i < codec->jacktbl.used; i++, jack++)
 		if (jack->nid) {
 			jack_detect_update(codec, jack);
-			jack_detect_report(codec, jack);
+			state = get_jack_plug_state(jack->pin_sense);
+			snd_kctl_jack_notify(codec->bus->card, jack->kctl, state);
 		}
 }
 EXPORT_SYMBOL_HDA(snd_hda_jack_report_sync);
 
-/*
- * jack-detection kcontrols
- */
-
-#define jack_detect_kctl_info	snd_ctl_boolean_mono_info
-
-static int jack_detect_kctl_get(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	hda_nid_t nid = kcontrol->private_value;
-
-	ucontrol->value.integer.value[0] = snd_hda_jack_detect(codec, nid);
-	return 0;
-}
-
-static struct snd_kcontrol_new jack_detect_kctl = {
-	/* name is filled later */
-	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
-	.access = SNDRV_CTL_ELEM_ACCESS_READ,
-	.info = jack_detect_kctl_info,
-	.get = jack_detect_kctl_get,
-};
-
 /**
  * snd_hda_jack_add_kctl - Add a kctl for the given pin
  *
@@ -245,12 +208,9 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
 		return 0;
 	if (jack->kctl)
 		return 0; /* already created */
-	kctl = snd_ctl_new1(&jack_detect_kctl, codec);
+	kctl = snd_kctl_jack_new(name, idx, codec);
 	if (!kctl)
 		return -ENOMEM;
-	snprintf(kctl->id.name, sizeof(kctl->id.name), "%s Jack", name);
-	kctl->id.index = idx;
-	kctl->private_value = nid;
 	if (snd_hda_ctl_add(codec, nid, kctl) < 0)
 		return -ENOMEM;
 	jack->kctl = kctl;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 69a67f8e4f4..4bb75ee16b5 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -21,7 +21,6 @@ struct hda_jack_tbl {
 	unsigned int pin_sense;		/* cached pin-sense value */
 	unsigned int jack_detect:1;	/* capable of jack-detection? */
 	unsigned int jack_dirty:1;	/* needs to update? */
-	unsigned int need_notify:1;	/* to be notified? */
 	struct snd_kcontrol *kctl;	/* assigned kctl for jack-detection */
 };
 
-- 
cgit v1.2.3-70-g09d2


From 1633281b79fd276f1c7c2fb37c3b97da74e42ae5 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 23 Nov 2011 12:42:04 -0700
Subject: ASoC: Implement fully_routed card property

A card is fully routed if the DAPM route table describes all connections on
the board.

When a card is fully routed, some operations can be automated by the ASoC
core. The first, and currently only, such operation is described below, and
implemented by this patch.

Codecs often have a large number of external pins, and not all of these pins
will be connected on all board designs. Some machine drivers therefore call
snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core
never to activate them.

However, when a card is fully routed, the information needed to derive the
set of unused pins is present in card->dapm_routes. In this case, have
the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused
codec pin.

This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dapm.h |  1 +
 include/sound/soc.h      |  1 +
 sound/soc/soc-core.c     |  4 +++
 sound/soc/soc-dapm.c     | 73 ++++++++++++++++++++++++++++++++++++++++++++++++
 4 files changed, 79 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 17a4c17f19f..0c159a7d211 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -380,6 +380,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
 				  const char *pin);
 int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
 				const char *pin);
+void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
 
 /* Mostly internal - should not normally be used */
 void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index b21b3047e91..737a4f4b18f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -815,6 +815,7 @@ struct snd_soc_card {
 	int num_dapm_widgets;
 	const struct snd_soc_dapm_route *dapm_routes;
 	int num_dapm_routes;
+	bool fully_routed;
 
 	struct work_struct deferred_resume_work;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b194842c4cc..2abaf6dcdb0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1488,6 +1488,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 
 	snd_soc_dapm_new_widgets(&card->dapm);
 
+	if (card->fully_routed)
+		list_for_each_entry(codec, &codec_list, list)
+			snd_soc_dapm_auto_nc_codec_pins(codec);
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f42e8b9fb17..1ecd1b4927f 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2947,6 +2947,79 @@ int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend);
 
+static bool snd_soc_dapm_widget_in_card_paths(struct snd_soc_card *card,
+					      struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *p;
+
+	list_for_each_entry(p, &card->paths, list) {
+		if ((p->source == w) || (p->sink == w)) {
+			dev_dbg(card->dev,
+			    "... Path %s(id:%d dapm:%p) - %s(id:%d dapm:%p)\n",
+			    p->source->name, p->source->id, p->source->dapm,
+			    p->sink->name, p->sink->id, p->sink->dapm);
+
+			/* Connected to something other than the codec */
+			if (p->source->dapm != p->sink->dapm)
+				return true;
+			/*
+			 * Loopback connection from codec external pin to
+			 * codec external pin
+			 */
+			if (p->sink->id == snd_soc_dapm_input) {
+				switch (p->source->id) {
+				case snd_soc_dapm_output:
+				case snd_soc_dapm_micbias:
+					return true;
+				default:
+					break;
+				}
+			}
+		}
+	}
+
+	return false;
+}
+
+/**
+ * snd_soc_dapm_auto_nc_codec_pins - call snd_soc_dapm_nc_pin for unused pins
+ * @codec: The codec whose pins should be processed
+ *
+ * Automatically call snd_soc_dapm_nc_pin() for any external pins in the codec
+ * which are unused. Pins are used if they are connected externally to the
+ * codec, whether that be to some other device, or a loop-back connection to
+ * the codec itself.
+ */
+void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec)
+{
+	struct snd_soc_card *card = codec->card;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_widget *w;
+
+	dev_dbg(card->dev, "Auto NC: DAPMs: card:%p codec:%p\n",
+		&card->dapm, &codec->dapm);
+
+	list_for_each_entry(w, &card->widgets, list) {
+		if (w->dapm != dapm)
+			continue;
+		switch (w->id) {
+		case snd_soc_dapm_input:
+		case snd_soc_dapm_output:
+		case snd_soc_dapm_micbias:
+			dev_dbg(card->dev, "Auto NC: Checking widget %s\n",
+				w->name);
+			if (!snd_soc_dapm_widget_in_card_paths(card, w)) {
+				dev_dbg(card->dev,
+					"... Not in map; disabling\n");
+				snd_soc_dapm_nc_pin(dapm, w->name);
+			}
+			break;
+		default:
+			break;
+		}
+	}
+}
+
 /**
  * snd_soc_dapm_free - free dapm resources
  * @dapm: DAPM context
-- 
cgit v1.2.3-70-g09d2


From 45f3121615b2b354f7d95d30f795bc5fe0043e92 Mon Sep 17 00:00:00 2001
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Date: Wed, 23 Nov 2011 16:55:34 -0800
Subject: ASoC: fsi-ak4642: modify specification method of FSI / ak464x

Current fsi-ak4642 was using id_entry name in order to specify
FSI port and ak464x codec.
But it was no sense, no flexibility.
Platform can specify FSI/ak464x pair by this patch.

Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/arm/mach-shmobile/board-ap4evb.c   |  15 ++++-
 arch/arm/mach-shmobile/board-mackerel.c |  14 +++-
 arch/sh/boards/mach-se/7724/setup.c     |  14 +++-
 include/sound/sh_fsi.h                  |  12 ++++
 sound/soc/sh/fsi-ak4642.c               | 114 +++-----------------------------
 5 files changed, 63 insertions(+), 106 deletions(-)

(limited to 'include/sound')

diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c
index a3aa0f6df96..ade98c2d9e8 100644
--- a/arch/arm/mach-shmobile/board-ap4evb.c
+++ b/arch/arm/mach-shmobile/board-ap4evb.c
@@ -762,9 +762,22 @@ static struct platform_device fsi_device = {
 	},
 };
 
+static struct fsi_ak4642_info fsi2_ak4643_info = {
+	.name		= "AK4643",
+	.card		= "FSI2A-AK4643",
+	.cpu_dai	= "fsia-dai",
+	.codec		= "ak4642-codec.0-0013",
+	.platform	= "sh_fsi2",
+	.id		= FSI_PORT_A,
+};
+
 static struct platform_device fsi_ak4643_device = {
-	.name		= "sh_fsi2_a_ak4643",
+	.name	= "fsi-ak4642-audio",
+	.dev	= {
+		.platform_data	= &fsi_info,
+	},
 };
+
 static struct sh_mobile_meram_cfg hdmi_meram_cfg = {
 	.icb[0] = {
 		.marker_icb     = 30,
diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c
index 9c5e598e0e3..a2908d49a6d 100644
--- a/arch/arm/mach-shmobile/board-mackerel.c
+++ b/arch/arm/mach-shmobile/board-mackerel.c
@@ -990,8 +990,20 @@ static struct platform_device fsi_device = {
 	},
 };
 
+static struct fsi_ak4642_info fsi2_ak4643_info = {
+	.name		= "AK4643",
+	.card		= "FSI2A-AK4643",
+	.cpu_dai	= "fsia-dai",
+	.codec		= "ak4642-codec.0-0013",
+	.platform	= "sh_fsi2",
+	.id		= FSI_PORT_A,
+};
+
 static struct platform_device fsi_ak4643_device = {
-	.name		= "sh_fsi2_a_ak4643",
+	.name	= "fsi-ak4642-audio",
+	.dev	= {
+		.platform_data	= &fsi2_ak4643_info,
+	},
 };
 
 /*
diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c
index b747c0ab926..b49723b2191 100644
--- a/arch/sh/boards/mach-se/7724/setup.c
+++ b/arch/sh/boards/mach-se/7724/setup.c
@@ -315,8 +315,20 @@ static struct platform_device fsi_device = {
 	},
 };
 
+static struct fsi_ak4642_info fsi_ak4642_info = {
+	.name		= "AK4642",
+	.card		= "FSIA-AK4642",
+	.cpu_dai	= "fsia-dai",
+	.codec		= "ak4642-codec.0-0012",
+	.platform	= "sh_fsi.0",
+	.id		= FSI_PORT_A,
+};
+
 static struct platform_device fsi_ak4642_device = {
-	.name		= "sh_fsi_a_ak4642",
+	.name	= "fsi-ak4642-audio",
+	.dev	= {
+		.platform_data	= &fsi_ak4642_info,
+	},
 };
 
 /* KEYSC in SoC (Needs SW33-2 set to ON) */
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index 9a155f9d0a1..9b1aacaa82f 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -78,4 +78,16 @@ struct sh_fsi_platform_info {
 	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 };
 
+/*
+ * for fsi-ak4642
+ */
+struct fsi_ak4642_info {
+	const char *name;
+	const char *card;
+	const char *cpu_dai;
+	const char *codec;
+	const char *platform;
+	int id;
+};
+
 #endif /* __SOUND_FSI_H */
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index dff64b95f5d..11d2d7ff29d 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -58,27 +58,23 @@ static struct platform_device *fsi_snd_device;
 static int fsi_ak4642_probe(struct platform_device *pdev)
 {
 	int ret = -ENOMEM;
-	const struct platform_device_id	*id_entry;
-	struct fsi_ak4642_data *pdata;
+	struct fsi_ak4642_info *pinfo = pdev->dev.platform_data;
 
-	id_entry = pdev->id_entry;
-	if (!id_entry) {
-		dev_err(&pdev->dev, "unknown fsi ak4642\n");
-		return -ENODEV;
+	if (!pinfo) {
+		dev_err(&pdev->dev, "no info for fsi ak4642\n");
+		goto out;
 	}
 
-	pdata = (struct fsi_ak4642_data *)id_entry->driver_data;
-
-	fsi_snd_device = platform_device_alloc("soc-audio", pdata->id);
+	fsi_snd_device = platform_device_alloc("soc-audio", pinfo->id);
 	if (!fsi_snd_device)
 		goto out;
 
-	fsi_dai_link.name		= pdata->name;
-	fsi_dai_link.stream_name	= pdata->name;
-	fsi_dai_link.cpu_dai_name	= pdata->cpu_dai;
-	fsi_dai_link.platform_name	= pdata->platform;
-	fsi_dai_link.codec_name		= pdata->codec;
-	fsi_soc_card.name		= pdata->card;
+	fsi_dai_link.name		= pinfo->name;
+	fsi_dai_link.stream_name	= pinfo->name;
+	fsi_dai_link.cpu_dai_name	= pinfo->cpu_dai;
+	fsi_dai_link.platform_name	= pinfo->platform;
+	fsi_dai_link.codec_name		= pinfo->codec;
+	fsi_soc_card.name		= pinfo->card;
 
 	platform_set_drvdata(fsi_snd_device, &fsi_soc_card);
 	ret = platform_device_add(fsi_snd_device);
@@ -96,100 +92,12 @@ static int fsi_ak4642_remove(struct platform_device *pdev)
 	return 0;
 }
 
-static struct fsi_ak4642_data fsi_a_ak4642 = {
-	.name		= "AK4642",
-	.card		= "FSIA-AK4642",
-	.cpu_dai	= "fsia-dai",
-	.codec		= "ak4642-codec.0-0012",
-	.platform	= "sh_fsi.0",
-	.id		= FSI_PORT_A,
-};
-
-static struct fsi_ak4642_data fsi_b_ak4642 = {
-	.name		= "AK4642",
-	.card		= "FSIB-AK4642",
-	.cpu_dai	= "fsib-dai",
-	.codec		= "ak4642-codec.0-0012",
-	.platform	= "sh_fsi.0",
-	.id		= FSI_PORT_B,
-};
-
-static struct fsi_ak4642_data fsi_a_ak4643 = {
-	.name		= "AK4643",
-	.card		= "FSIA-AK4643",
-	.cpu_dai	= "fsia-dai",
-	.codec		= "ak4642-codec.0-0013",
-	.platform	= "sh_fsi.0",
-	.id		= FSI_PORT_A,
-};
-
-static struct fsi_ak4642_data fsi_b_ak4643 = {
-	.name		= "AK4643",
-	.card		= "FSIB-AK4643",
-	.cpu_dai	= "fsib-dai",
-	.codec		= "ak4642-codec.0-0013",
-	.platform	= "sh_fsi.0",
-	.id		= FSI_PORT_B,
-};
-
-static struct fsi_ak4642_data fsi2_a_ak4642 = {
-	.name		= "AK4642",
-	.card		= "FSI2A-AK4642",
-	.cpu_dai	= "fsia-dai",
-	.codec		= "ak4642-codec.0-0012",
-	.platform	= "sh_fsi2",
-	.id		= FSI_PORT_A,
-};
-
-static struct fsi_ak4642_data fsi2_b_ak4642 = {
-	.name		= "AK4642",
-	.card		= "FSI2B-AK4642",
-	.cpu_dai	= "fsib-dai",
-	.codec		= "ak4642-codec.0-0012",
-	.platform	= "sh_fsi2",
-	.id		= FSI_PORT_B,
-};
-
-static struct fsi_ak4642_data fsi2_a_ak4643 = {
-	.name		= "AK4643",
-	.card		= "FSI2A-AK4643",
-	.cpu_dai	= "fsia-dai",
-	.codec		= "ak4642-codec.0-0013",
-	.platform	= "sh_fsi2",
-	.id		= FSI_PORT_A,
-};
-
-static struct fsi_ak4642_data fsi2_b_ak4643 = {
-	.name		= "AK4643",
-	.card		= "FSI2B-AK4643",
-	.cpu_dai	= "fsib-dai",
-	.codec		= "ak4642-codec.0-0013",
-	.platform	= "sh_fsi2",
-	.id		= FSI_PORT_B,
-};
-
-static struct platform_device_id fsi_id_table[] = {
-	/* FSI */
-	{ "sh_fsi_a_ak4642",	(kernel_ulong_t)&fsi_a_ak4642 },
-	{ "sh_fsi_b_ak4642",	(kernel_ulong_t)&fsi_b_ak4642 },
-	{ "sh_fsi_a_ak4643",	(kernel_ulong_t)&fsi_a_ak4643 },
-	{ "sh_fsi_b_ak4643",	(kernel_ulong_t)&fsi_b_ak4643 },
-
-	/* FSI 2 */
-	{ "sh_fsi2_a_ak4642",	(kernel_ulong_t)&fsi2_a_ak4642 },
-	{ "sh_fsi2_b_ak4642",	(kernel_ulong_t)&fsi2_b_ak4642 },
-	{ "sh_fsi2_a_ak4643",	(kernel_ulong_t)&fsi2_a_ak4643 },
-	{ "sh_fsi2_b_ak4643",	(kernel_ulong_t)&fsi2_b_ak4643 },
-	{},
-};
-
 static struct platform_driver fsi_ak4642 = {
 	.driver = {
 		.name	= "fsi-ak4642-audio",
 	},
 	.probe		= fsi_ak4642_probe,
 	.remove		= fsi_ak4642_remove,
-	.id_table	= fsi_id_table,
 };
 
 static int __init fsi_ak4642_init(void)
-- 
cgit v1.2.3-70-g09d2


From 1ab97c8cad98de016cb36a870e118feaf0a0caaf Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 27 Nov 2011 16:21:51 +0000
Subject: ASoC: Add signal generator widget type

A signal generator behaves as an input would but is not considered for
any of the special behaviour associated with external input pins. This
is especially useful when automatically working out not connected widgets.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
 include/sound/soc-dapm.h | 4 ++++
 sound/soc/soc-dapm.c     | 7 +++++++
 2 files changed, 11 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 0c159a7d211..d26a9b78477 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -43,6 +43,9 @@
 	.num_kcontrols = 0}
 
 /* platform domain */
+#define SND_SOC_DAPM_SIGGEN(wname) \
+{	.id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
+	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
 #define SND_SOC_DAPM_INPUT(wname) \
 {	.id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
@@ -410,6 +413,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_supply,		/* power/clock supply */
 	snd_soc_dapm_aif_in,		/* audio interface input */
 	snd_soc_dapm_aif_out,		/* audio interface output */
+	snd_soc_dapm_siggen,		/* signal generator */
 };
 
 /*
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index da5c1ae7cc3..6bb327e431a 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -339,6 +339,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_output:
 	case snd_soc_dapm_adc:
 	case snd_soc_dapm_input:
+	case snd_soc_dapm_siggen:
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_micbias:
 	case snd_soc_dapm_vmid:
@@ -772,6 +773,11 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 			return widget->inputs;
 		}
 
+		/* signal generator */
+		if (widget->id == snd_soc_dapm_siggen) {
+			widget->inputs = snd_soc_dapm_suspend_check(widget);
+			return widget->inputs;
+		}
 	}
 
 	list_for_each_entry(path, &widget->sources, list_sink) {
@@ -1982,6 +1988,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_out_drv:
 	case snd_soc_dapm_input:
 	case snd_soc_dapm_output:
+	case snd_soc_dapm_siggen:
 	case snd_soc_dapm_micbias:
 	case snd_soc_dapm_vmid:
 	case snd_soc_dapm_pre:
-- 
cgit v1.2.3-70-g09d2


From 84b315ee893676e9a9ce8ac42ab5ef44e2af3ee1 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Fri, 2 Dec 2011 10:18:28 +0100
Subject: ASoC: Drop unused state parameter from CODEC suspend callback

The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h              | 3 +--
 sound/soc/codecs/ac97.c          | 2 +-
 sound/soc/codecs/ad1836.c        | 2 +-
 sound/soc/codecs/adau1373.c      | 2 +-
 sound/soc/codecs/adav80x.c       | 2 +-
 sound/soc/codecs/ak4535.c        | 2 +-
 sound/soc/codecs/ak4641.c        | 2 +-
 sound/soc/codecs/alc5623.c       | 2 +-
 sound/soc/codecs/alc5632.c       | 2 +-
 sound/soc/codecs/cs4270.c        | 2 +-
 sound/soc/codecs/cs4271.c        | 2 +-
 sound/soc/codecs/cs42l73.c       | 2 +-
 sound/soc/codecs/jz4740.c        | 2 +-
 sound/soc/codecs/max98088.c      | 2 +-
 sound/soc/codecs/max98095.c      | 2 +-
 sound/soc/codecs/max9850.c       | 2 +-
 sound/soc/codecs/pcm3008.c       | 2 +-
 sound/soc/codecs/rt5631.c        | 2 +-
 sound/soc/codecs/sgtl5000.c      | 2 +-
 sound/soc/codecs/ssm2602.c       | 2 +-
 sound/soc/codecs/sta32x.c        | 2 +-
 sound/soc/codecs/stac9766.c      | 3 +--
 sound/soc/codecs/tlv320aic23.c   | 3 +--
 sound/soc/codecs/tlv320aic32x4.c | 2 +-
 sound/soc/codecs/tlv320aic3x.c   | 2 +-
 sound/soc/codecs/tlv320dac33.c   | 2 +-
 sound/soc/codecs/twl4030.c       | 2 +-
 sound/soc/codecs/twl6040.c       | 2 +-
 sound/soc/codecs/uda134x.c       | 3 +--
 sound/soc/codecs/uda1380.c       | 2 +-
 sound/soc/codecs/wm8350.c        | 2 +-
 sound/soc/codecs/wm8400.c        | 2 +-
 sound/soc/codecs/wm8510.c        | 2 +-
 sound/soc/codecs/wm8523.c        | 2 +-
 sound/soc/codecs/wm8711.c        | 2 +-
 sound/soc/codecs/wm8728.c        | 2 +-
 sound/soc/codecs/wm8731.c        | 2 +-
 sound/soc/codecs/wm8737.c        | 2 +-
 sound/soc/codecs/wm8750.c        | 2 +-
 sound/soc/codecs/wm8753.c        | 2 +-
 sound/soc/codecs/wm8770.c        | 2 +-
 sound/soc/codecs/wm8776.c        | 2 +-
 sound/soc/codecs/wm8804.c        | 2 +-
 sound/soc/codecs/wm8900.c        | 2 +-
 sound/soc/codecs/wm8903.c        | 2 +-
 sound/soc/codecs/wm8904.c        | 2 +-
 sound/soc/codecs/wm8940.c        | 2 +-
 sound/soc/codecs/wm8955.c        | 2 +-
 sound/soc/codecs/wm8960.c        | 2 +-
 sound/soc/codecs/wm8961.c        | 2 +-
 sound/soc/codecs/wm8971.c        | 2 +-
 sound/soc/codecs/wm8974.c        | 2 +-
 sound/soc/codecs/wm8978.c        | 2 +-
 sound/soc/codecs/wm8983.c        | 2 +-
 sound/soc/codecs/wm8985.c        | 2 +-
 sound/soc/codecs/wm8988.c        | 2 +-
 sound/soc/codecs/wm8990.c        | 2 +-
 sound/soc/codecs/wm8991.c        | 2 +-
 sound/soc/codecs/wm8993.c        | 2 +-
 sound/soc/codecs/wm8994.c        | 2 +-
 sound/soc/codecs/wm8995.c        | 2 +-
 sound/soc/codecs/wm9081.c        | 2 +-
 sound/soc/codecs/wm9090.c        | 2 +-
 sound/soc/codecs/wm9705.c        | 2 +-
 sound/soc/codecs/wm9712.c        | 3 +--
 sound/soc/codecs/wm9713.c        | 3 +--
 sound/soc/soc-core.c             | 2 +-
 67 files changed, 67 insertions(+), 73 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 737a4f4b18f..d9aa66be199 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -592,8 +592,7 @@ struct snd_soc_codec_driver {
 	/* driver ops */
 	int (*probe)(struct snd_soc_codec *);
 	int (*remove)(struct snd_soc_codec *);
-	int (*suspend)(struct snd_soc_codec *,
-			pm_message_t state);
+	int (*suspend)(struct snd_soc_codec *);
 	int (*resume)(struct snd_soc_codec *);
 
 	/* Default control and setup, added after probe() is run */
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 221ec29f68e..1bbad4c16d2 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -99,7 +99,7 @@ static int ac97_soc_remove(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int ac97_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg)
+static int ac97_soc_suspend(struct snd_soc_codec *codec)
 {
 	snd_ac97_suspend(codec->ac97);
 
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index fab0948f7a5..919322daf6d 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -223,7 +223,7 @@ static struct snd_soc_dai_driver ad183x_dais[] = {
 };
 
 #ifdef CONFIG_PM
-static int ad1836_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int ad1836_suspend(struct snd_soc_codec *codec)
 {
 	/* reset clock control mode */
 	return snd_soc_update_bits(codec, AD1836_ADC_CTRL2,
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 45c63028b40..637b114bea7 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1321,7 +1321,7 @@ static int adau1373_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int adau1373_suspend(struct snd_soc_codec *codec)
 {
 	return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
 }
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index f9f08948e5e..ebd7b37b902 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -798,7 +798,7 @@ static int adav80x_probe(struct snd_soc_codec *codec)
 	return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 }
 
-static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int adav80x_suspend(struct snd_soc_codec *codec)
 {
 	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 }
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index e1f53108545..96296fd172f 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -354,7 +354,7 @@ static struct snd_soc_dai_driver ak4535_dai = {
 	.ops = &ak4535_dai_ops,
 };
 
-static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int ak4535_suspend(struct snd_soc_codec *codec)
 {
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index f53f3148056..90184701480 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -498,7 +498,7 @@ static struct snd_soc_dai_driver ak4641_dai[] = {
 },
 };
 
-static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int ak4641_suspend(struct snd_soc_codec *codec)
 {
 	ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 6a5c001e8ba..da97f024ec7 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -868,7 +868,7 @@ static struct snd_soc_dai_driver alc5623_dai = {
 	.ops = &alc5623_dai_ops,
 };
 
-static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+static int alc5623_suspend(struct snd_soc_codec *codec)
 {
 	alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 3f750def896..08613c7e109 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -956,7 +956,7 @@ static struct snd_soc_dai_driver alc5632_dai = {
 };
 
 #ifdef CONFIG_PM
-static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+static int alc5632_suspend(struct snd_soc_codec *codec)
 {
 	alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index dc77ff7ba33..fef0f48330e 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -578,7 +578,7 @@ static int cs4270_remove(struct snd_soc_codec *codec)
  * and all registers are written back to the hardware when resuming.
  */
 
-static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+static int cs4270_soc_suspend(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int reg, ret;
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index a6f77a855f4..f6fe846b6a6 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -430,7 +430,7 @@ static struct snd_soc_dai_driver cs4271_dai = {
 };
 
 #ifdef CONFIG_PM
-static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
+static int cs4271_soc_suspend(struct snd_soc_codec *codec)
 {
 	int ret;
 	/* Set power-down bit */
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index da3125aa55f..9d38db8f191 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1262,7 +1262,7 @@ static struct snd_soc_dai_driver cs42l73_dai[] = {
 	 }
 };
 
-static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int cs42l73_suspend(struct snd_soc_codec *codec)
 {
 	cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 4fca8bccd53..517e2a516da 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -311,7 +311,7 @@ static int jz4740_codec_dev_remove(struct snd_soc_codec *codec)
 
 #ifdef CONFIG_PM_SLEEP
 
-static int jz4740_codec_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int jz4740_codec_suspend(struct snd_soc_codec *codec)
 {
 	return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
 }
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 9b6036e5738..ba4f6f167a1 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1946,7 +1946,7 @@ static void max98088_handle_pdata(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int max98088_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int max98088_suspend(struct snd_soc_codec *codec)
 {
        max98088_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 01f4ad72514..c69dd022bea 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -2174,7 +2174,7 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int max98095_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int max98095_suspend(struct snd_soc_codec *codec)
 {
 	max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 94c2b586ed5..7dfd6e84796 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -273,7 +273,7 @@ static struct snd_soc_dai_driver max9850_dai = {
 };
 
 #ifdef CONFIG_PM
-static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int max9850_suspend(struct snd_soc_codec *codec)
 {
 	max9850_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index b12d01f6799..edcaa7ea548 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -118,7 +118,7 @@ static int pcm3008_soc_remove(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int pcm3008_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg)
+static int pcm3008_soc_suspend(struct snd_soc_codec *codec)
 {
 	struct pcm3008_setup_data *setup = codec->dev->platform_data;
 
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 9fd50bd77c4..f6e4f5ed928 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1641,7 +1641,7 @@ static int rt5631_remove(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int rt5631_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int rt5631_suspend(struct snd_soc_codec *codec)
 {
 	rt5631_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index ff0a1079efe..250175755eb 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -967,7 +967,7 @@ static int sgtl5000_volatile_register(struct snd_soc_codec *codec,
 }
 
 #ifdef CONFIG_SUSPEND
-static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int sgtl5000_suspend(struct snd_soc_codec *codec)
 {
 	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 0d43e4b4a58..7dfc7b08114 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -523,7 +523,7 @@ static struct snd_soc_dai_driver ssm2602_dai = {
 	.ops = &ssm2602_dai_ops,
 };
 
-static int ssm2602_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int ssm2602_suspend(struct snd_soc_codec *codec)
 {
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index b3d1c78e361..6648af6656c 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -801,7 +801,7 @@ static struct snd_soc_dai_driver sta32x_dai = {
 };
 
 #ifdef CONFIG_PM
-static int sta32x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int sta32x_suspend(struct snd_soc_codec *codec)
 {
 	sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 55819537b67..e34969cdc0e 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -256,8 +256,7 @@ static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
 	return 0;
 }
 
-static int stac9766_codec_suspend(struct snd_soc_codec *codec,
-				  pm_message_t state)
+static int stac9766_codec_suspend(struct snd_soc_codec *codec)
 {
 	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index cba798e1a07..60d08aeac22 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -528,8 +528,7 @@ static struct snd_soc_dai_driver tlv320aic23_dai = {
 	.ops = &tlv320aic23_dai_ops,
 };
 
-static int tlv320aic23_suspend(struct snd_soc_codec *codec,
-			       pm_message_t state)
+static int tlv320aic23_suspend(struct snd_soc_codec *codec)
 {
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index f5533756737..81a26e1090b 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -621,7 +621,7 @@ static struct snd_soc_dai_driver aic32x4_dai = {
 	.symmetric_rates = 1,
 };
 
-static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int aic32x4_suspend(struct snd_soc_codec *codec)
 {
 	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 21625dddde2..6f963c50e76 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1268,7 +1268,7 @@ static struct snd_soc_dai_driver aic3x_dai = {
 	.symmetric_rates = 1,
 };
 
-static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int aic3x_suspend(struct snd_soc_codec *codec)
 {
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 6b0f0e220f8..c7a61fbdae4 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1460,7 +1460,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int dac33_soc_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int dac33_soc_suspend(struct snd_soc_codec *codec)
 {
 	dac33_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 61d8a9065ff..18e71014cc2 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -2202,7 +2202,7 @@ static struct snd_soc_dai_driver twl4030_dai[] = {
 },
 };
 
-static int twl4030_soc_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int twl4030_soc_suspend(struct snd_soc_codec *codec)
 {
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index a4a65dc9e33..3376e6fad2a 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1470,7 +1470,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = {
 };
 
 #ifdef CONFIG_PM
-static int twl6040_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int twl6040_suspend(struct snd_soc_codec *codec)
 {
 	twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index d0f9d904ce8..8f4f469d641 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -571,8 +571,7 @@ static int uda134x_soc_remove(struct snd_soc_codec *codec)
 }
 
 #if defined(CONFIG_PM)
-static int uda134x_soc_suspend(struct snd_soc_codec *codec,
-						pm_message_t state)
+static int uda134x_soc_suspend(struct snd_soc_codec *codec)
 {
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 6b933efc7ed..d08b91dde53 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -705,7 +705,7 @@ static struct snd_soc_dai_driver uda1380_dai[] = {
 },
 };
 
-static int uda1380_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int uda1380_suspend(struct snd_soc_codec *codec)
 {
 	uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3f1ed5f5ccf..f39497fc13e 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1315,7 +1315,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm8350_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8350_suspend(struct snd_soc_codec *codec)
 {
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index a1173eb7936..56a7b7256ef 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1352,7 +1352,7 @@ static struct snd_soc_dai_driver wm8400_dai = {
 	.ops = &wm8400_dai_ops,
 };
 
-static int wm8400_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8400_suspend(struct snd_soc_codec *codec)
 {
 	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 3a655719ba2..5e847506138 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -534,7 +534,7 @@ static struct snd_soc_dai_driver wm8510_dai = {
 	.symmetric_rates = 1,
 };
 
-static int wm8510_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8510_suspend(struct snd_soc_codec *codec)
 {
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 0c89f8e2daa..7fea2c3bf7e 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -384,7 +384,7 @@ static struct snd_soc_dai_driver wm8523_dai = {
 };
 
 #ifdef CONFIG_PM
-static int wm8523_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8523_suspend(struct snd_soc_codec *codec)
 {
 	wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 760080e4301..b9b1a2f8360 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -338,7 +338,7 @@ static struct snd_soc_dai_driver wm8711_dai = {
 	.ops = &wm8711_ops,
 };
 
-static int wm8711_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8711_suspend(struct snd_soc_codec *codec)
 {
 	snd_soc_write(codec, WM8711_ACTIVE, 0x0);
 	wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 085c2f81d8c..b1f01d9273b 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -214,7 +214,7 @@ static struct snd_soc_dai_driver wm8728_dai = {
 	.ops = &wm8728_dai_ops,
 };
 
-static int wm8728_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8728_suspend(struct snd_soc_codec *codec)
 {
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c18dee06f29..8821af70e66 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -490,7 +490,7 @@ static struct snd_soc_dai_driver wm8731_dai = {
 };
 
 #ifdef CONFIG_PM
-static int wm8731_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8731_suspend(struct snd_soc_codec *codec)
 {
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index c13e4f7809c..ff95e62c56b 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -539,7 +539,7 @@ static struct snd_soc_dai_driver wm8737_dai = {
 };
 
 #ifdef CONFIG_PM
-static int wm8737_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8737_suspend(struct snd_soc_codec *codec)
 {
 	wm8737_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index b312fccbf67..48cb78fd010 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -666,7 +666,7 @@ static struct snd_soc_dai_driver wm8750_dai = {
 	.ops = &wm8750_dai_ops,
 };
 
-static int wm8750_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8750_suspend(struct snd_soc_codec *codec)
 {
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index dc3153852d8..b114c19f530 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1380,7 +1380,7 @@ static void wm8753_work(struct work_struct *work)
 	wm8753_set_bias_level(codec, dapm->bias_level);
 }
 
-static int wm8753_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8753_suspend(struct snd_soc_codec *codec)
 {
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 391c385ec43..8976eb5796d 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -555,7 +555,7 @@ static struct snd_soc_dai_driver wm8770_dai = {
 };
 
 #ifdef CONFIG_PM
-static int wm8770_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8770_suspend(struct snd_soc_codec *codec)
 {
 	wm8770_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index af542a2f594..fbf80c5220d 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -371,7 +371,7 @@ static struct snd_soc_dai_driver wm8776_dai[] = {
 };
 
 #ifdef CONFIG_PM
-static int wm8776_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8776_suspend(struct snd_soc_codec *codec)
 {
 	wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index d99c6a0a0a2..ae4b8fb3c3e 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -542,7 +542,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec,
 }
 
 #ifdef CONFIG_PM
-static int wm8804_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8804_suspend(struct snd_soc_codec *codec)
 {
 	wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 6ac80cf80b3..85632ffcb87 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1106,7 +1106,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm8900_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8900_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec);
 	int fll_out = wm8900->fll_out;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 70a2268c549..d663c97785d 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1758,7 +1758,7 @@ static struct snd_soc_dai_driver wm8903_dai = {
 	.symmetric_rates = 1,
 };
 
-static int wm8903_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8903_suspend(struct snd_soc_codec *codec)
 {
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index babca49c876..f0ae01bbaa9 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2234,7 +2234,7 @@ static struct snd_soc_dai_driver wm8904_dai = {
 };
 
 #ifdef CONFIG_PM
-static int wm8904_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8904_suspend(struct snd_soc_codec *codec)
 {
 	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 9f1cce8d105..0fe4545eef8 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -672,7 +672,7 @@ static struct snd_soc_dai_driver wm8940_dai = {
 	.symmetric_rates = 1,
 };
 
-static int wm8940_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8940_suspend(struct snd_soc_codec *codec)
 {
 	return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
 }
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index ca38722bc3f..cdd51398e1f 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -878,7 +878,7 @@ static struct snd_soc_dai_driver wm8955_dai = {
 };
 
 #ifdef CONFIG_PM
-static int wm8955_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8955_suspend(struct snd_soc_codec *codec)
 {
 	wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index ed2773f623c..55b9a25cd1b 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -894,7 +894,7 @@ static struct snd_soc_dai_driver wm8960_dai = {
 	.symmetric_rates = 1,
 };
 
-static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8960_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
 
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index c0587013fdf..9bcf846e93b 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1038,7 +1038,7 @@ static int wm8961_remove(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm8961_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8961_suspend(struct snd_soc_codec *codec)
 {
 	wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index b01df56b824..aadd14a1466 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -599,7 +599,7 @@ static void wm8971_work(struct work_struct *work)
 	wm8971_set_bias_level(codec, codec->dapm.bias_level);
 }
 
-static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8971_suspend(struct snd_soc_codec *codec)
 {
 	wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index e41f9993c65..a5fd017c433 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -582,7 +582,7 @@ static struct snd_soc_dai_driver wm8974_dai = {
 	.symmetric_rates = 1,
 };
 
-static int wm8974_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8974_suspend(struct snd_soc_codec *codec)
 {
 	wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 649a2e3c02a..85d514d63a4 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -892,7 +892,7 @@ static struct snd_soc_dai_driver wm8978_dai = {
 	.ops = &wm8978_dai_ops,
 };
 
-static int wm8978_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8978_suspend(struct snd_soc_codec *codec)
 {
 	wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	/* Also switch PLL off */
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 362298cce92..cebde568d19 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -974,7 +974,7 @@ static int wm8983_set_bias_level(struct snd_soc_codec *codec,
 }
 
 #ifdef CONFIG_PM
-static int wm8983_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8983_suspend(struct snd_soc_codec *codec)
 {
 	wm8983_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 9e4481bb122..c0c86b3c6ad 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -945,7 +945,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec,
 }
 
 #ifdef CONFIG_PM
-static int wm8985_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8985_suspend(struct snd_soc_codec *codec)
 {
 	wm8985_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 608c6721e4f..093884705b0 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -728,7 +728,7 @@ static struct snd_soc_dai_driver wm8988_dai = {
 	.symmetric_rates = 1,
 };
 
-static int wm8988_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8988_suspend(struct snd_soc_codec *codec)
 {
 	wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 58d7f0bff99..b417d2e0cdf 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1313,7 +1313,7 @@ static struct snd_soc_dai_driver wm8990_dai = {
 	.ops = &wm8990_dai_ops,
 };
 
-static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8990_suspend(struct snd_soc_codec *codec)
 {
 	wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 35c5389e5ef..7ee40da8dbb 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1240,7 +1240,7 @@ static int wm8991_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8991_suspend(struct snd_soc_codec *codec)
 {
 	wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 780c24cdab6..0f8278b4f0a 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1544,7 +1544,7 @@ static int wm8993_remove(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm8993_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8993_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int fll_fout = wm8993->fll_fout;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0699ed2fb79..d9faa39d826 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -2801,7 +2801,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
 };
 
 #ifdef CONFIG_PM
-static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8994_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994 *control = wm8994->wm8994;
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 8f6a36d7c75..5863406b459 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -2009,7 +2009,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec,
 }
 
 #ifdef CONFIG_PM
-static int wm8995_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm8995_suspend(struct snd_soc_codec *codec)
 {
 	wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 8a4b9706044..1f2672b1e03 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1302,7 +1302,7 @@ static int wm9081_remove(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm9081_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm9081_suspend(struct snd_soc_codec *codec)
 {
 	wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index f94c06057c6..5cb8759868d 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -604,7 +604,7 @@ static int wm9090_probe(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm9090_suspend(struct snd_soc_codec *codec, pm_message_t state)
+static int wm9090_suspend(struct snd_soc_codec *codec)
 {
 	wm9090_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index b720a43c422..40c92ead85a 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -306,7 +306,7 @@ static int wm9705_reset(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm9705_soc_suspend(struct snd_soc_codec *codec, pm_message_t msg)
+static int wm9705_soc_suspend(struct snd_soc_codec *codec)
 {
 	soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
 
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 4ce73f59df2..b7b31f84c10 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -583,8 +583,7 @@ err:
 	return -EIO;
 }
 
-static int wm9712_soc_suspend(struct snd_soc_codec *codec,
-	pm_message_t state)
+static int wm9712_soc_suspend(struct snd_soc_codec *codec)
 {
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index edb598182c6..2b8479bfcd9 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1140,8 +1140,7 @@ static int wm9713_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm9713_soc_suspend(struct snd_soc_codec *codec,
-	pm_message_t state)
+static int wm9713_soc_suspend(struct snd_soc_codec *codec)
 {
 	u16 reg;
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ec783f0a27e..5195f0653b3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -572,7 +572,7 @@ int snd_soc_suspend(struct device *dev)
 			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 			case SND_SOC_BIAS_OFF:
-				codec->driver->suspend(codec, PMSG_SUSPEND);
+				codec->driver->suspend(codec);
 				codec->suspended = 1;
 				codec->cache_sync = 1;
 				break;
-- 
cgit v1.2.3-70-g09d2


From a0f203d384fadacba514748cd0095efeadeed96c Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Fri, 2 Dec 2011 15:08:37 -0700
Subject: ASoC: WM8903: Fix platform data gpio_cfg confusion

wm8903_platform_data.gpio_cfg[] was intended to be interpreted as follows:
0:       Don't touch this GPIO's configuration register
1..7fff: Write that value to the GPIO's configuration register
8000:    Write zero to the GPIO's configuration register
other:   Undefined (invalid)

The rationale is that platform data is usually global data, and a value of
zero means that the field wasn't explicitly set to anything (e.g. because
the field was new to the pdata type, and existing users weren't update to
initialize it) and hence the value zero should be ignored. 0x8000 is an
explicit way to get 0 in the register.

The code worked this way until commit 7cfe561 "ASoC: wm8903: Expose GPIOs
through gpiolib", where the behaviour was changed due to my lack of
awareness of the above rationale.

This patch reverts to the intended behaviour, and updates all in-tree users
to use the correct scheme. This also makes WM8903 consistent with other
devices that use a similar scheme.

WM8903_GPIO_NO_CONFIG is also renamed to WM8903_GPIO_CONFIG_ZERO so that
its name accurately reflects its purpose.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Cc: Olof Johansson <olof@lixom.net>
Cc: Colin Cross <ccross@android.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 arch/arm/mach-tegra/board-harmony.c  | 8 ++++----
 arch/arm/mach-tegra/board-seaboard.c | 8 ++++----
 include/sound/wm8903.h               | 7 +++++--
 sound/soc/codecs/wm8903.c            | 3 ++-
 4 files changed, 15 insertions(+), 11 deletions(-)

(limited to 'include/sound')

diff --git a/arch/arm/mach-tegra/board-harmony.c b/arch/arm/mach-tegra/board-harmony.c
index f0bdc5e3fe5..3c57cdcdff4 100644
--- a/arch/arm/mach-tegra/board-harmony.c
+++ b/arch/arm/mach-tegra/board-harmony.c
@@ -89,11 +89,11 @@ static struct wm8903_platform_data harmony_wm8903_pdata = {
 	.micdet_delay = 100,
 	.gpio_base = HARMONY_GPIO_WM8903(0),
 	.gpio_cfg = {
-		WM8903_GPIO_NO_CONFIG,
-		WM8903_GPIO_NO_CONFIG,
 		0,
-		WM8903_GPIO_NO_CONFIG,
-		WM8903_GPIO_NO_CONFIG,
+		0,
+		WM8903_GPIO_CONFIG_ZERO,
+		0,
+		0,
 	},
 };
 
diff --git a/arch/arm/mach-tegra/board-seaboard.c b/arch/arm/mach-tegra/board-seaboard.c
index bf13ea355ef..5c2f7751a33 100644
--- a/arch/arm/mach-tegra/board-seaboard.c
+++ b/arch/arm/mach-tegra/board-seaboard.c
@@ -171,11 +171,11 @@ static struct wm8903_platform_data wm8903_pdata = {
 	.micdet_delay = 100,
 	.gpio_base = SEABOARD_GPIO_WM8903(0),
 	.gpio_cfg = {
-		WM8903_GPIO_NO_CONFIG,
-		WM8903_GPIO_NO_CONFIG,
 		0,
-		WM8903_GPIO_NO_CONFIG,
-		WM8903_GPIO_NO_CONFIG,
+		0,
+		WM8903_GPIO_CONFIG_ZERO,
+		0,
+		0,
 	},
 };
 
diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index cf7ccb76a8d..b310c5a3a95 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -11,8 +11,11 @@
 #ifndef __LINUX_SND_WM8903_H
 #define __LINUX_SND_WM8903_H
 
-/* Used to enable configuration of a GPIO to all zeros */
-#define WM8903_GPIO_NO_CONFIG 0x8000
+/*
+ * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of
+ * zero in platform data means "don't touch this pin".
+ */
+#define WM8903_GPIO_CONFIG_ZERO 0x8000
 
 /*
  * R6 (0x06) - Mic Bias Control 0
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index e6ecede576d..184b67730c3 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1893,7 +1893,8 @@ static int wm8903_probe(struct snd_soc_codec *codec)
 		bool mic_gpio = false;
 
 		for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
-			if (pdata->gpio_cfg[i] > 0x7fff)
+			if ((!pdata->gpio_cfg[i]) ||
+			    (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
 				continue;
 
 			snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
-- 
cgit v1.2.3-70-g09d2


From 1dfb6efd87d63d2efef6e985770d5dd642f83146 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 10 Nov 2011 17:39:40 +0000
Subject: ASoC: Remove rbtree register cache

All users now use regmap directly so delete the ASoC version of the code.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h   |   1 -
 sound/soc/soc-cache.c | 381 --------------------------------------------------
 2 files changed, 382 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index d9aa66be199..a35cf14a8eb 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -266,7 +266,6 @@ enum snd_soc_control_type {
 
 enum snd_soc_compress_type {
 	SND_SOC_FLAT_COMPRESSION = 1,
-	SND_SOC_RBTREE_COMPRESSION
 };
 
 enum snd_soc_pcm_subclass {
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 18bb6b3335e..9d56f0218f4 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -66,378 +66,6 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx,
 	return -1;
 }
 
-struct snd_soc_rbtree_node {
-	struct rb_node node; /* the actual rbtree node holding this block */
-	unsigned int base_reg; /* base register handled by this block */
-	unsigned int word_size; /* number of bytes needed to represent the register index */
-	void *block; /* block of adjacent registers */
-	unsigned int blklen; /* number of registers available in the block */
-} __attribute__ ((packed));
-
-struct snd_soc_rbtree_ctx {
-	struct rb_root root;
-	struct snd_soc_rbtree_node *cached_rbnode;
-};
-
-static inline void snd_soc_rbtree_get_base_top_reg(
-	struct snd_soc_rbtree_node *rbnode,
-	unsigned int *base, unsigned int *top)
-{
-	*base = rbnode->base_reg;
-	*top = rbnode->base_reg + rbnode->blklen - 1;
-}
-
-static unsigned int snd_soc_rbtree_get_register(
-	struct snd_soc_rbtree_node *rbnode, unsigned int idx)
-{
-	unsigned int val;
-
-	switch (rbnode->word_size) {
-	case 1: {
-		u8 *p = rbnode->block;
-		val = p[idx];
-		return val;
-	}
-	case 2: {
-		u16 *p = rbnode->block;
-		val = p[idx];
-		return val;
-	}
-	default:
-		BUG();
-		break;
-	}
-	return -1;
-}
-
-static void snd_soc_rbtree_set_register(struct snd_soc_rbtree_node *rbnode,
-					unsigned int idx, unsigned int val)
-{
-	switch (rbnode->word_size) {
-	case 1: {
-		u8 *p = rbnode->block;
-		p[idx] = val;
-		break;
-	}
-	case 2: {
-		u16 *p = rbnode->block;
-		p[idx] = val;
-		break;
-	}
-	default:
-		BUG();
-		break;
-	}
-}
-
-static struct snd_soc_rbtree_node *snd_soc_rbtree_lookup(
-	struct rb_root *root, unsigned int reg)
-{
-	struct rb_node *node;
-	struct snd_soc_rbtree_node *rbnode;
-	unsigned int base_reg, top_reg;
-
-	node = root->rb_node;
-	while (node) {
-		rbnode = container_of(node, struct snd_soc_rbtree_node, node);
-		snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
-		if (reg >= base_reg && reg <= top_reg)
-			return rbnode;
-		else if (reg > top_reg)
-			node = node->rb_right;
-		else if (reg < base_reg)
-			node = node->rb_left;
-	}
-
-	return NULL;
-}
-
-static int snd_soc_rbtree_insert(struct rb_root *root,
-				 struct snd_soc_rbtree_node *rbnode)
-{
-	struct rb_node **new, *parent;
-	struct snd_soc_rbtree_node *rbnode_tmp;
-	unsigned int base_reg_tmp, top_reg_tmp;
-	unsigned int base_reg;
-
-	parent = NULL;
-	new = &root->rb_node;
-	while (*new) {
-		rbnode_tmp = container_of(*new, struct snd_soc_rbtree_node,
-					  node);
-		/* base and top registers of the current rbnode */
-		snd_soc_rbtree_get_base_top_reg(rbnode_tmp, &base_reg_tmp,
-						&top_reg_tmp);
-		/* base register of the rbnode to be added */
-		base_reg = rbnode->base_reg;
-		parent = *new;
-		/* if this register has already been inserted, just return */
-		if (base_reg >= base_reg_tmp &&
-		    base_reg <= top_reg_tmp)
-			return 0;
-		else if (base_reg > top_reg_tmp)
-			new = &((*new)->rb_right);
-		else if (base_reg < base_reg_tmp)
-			new = &((*new)->rb_left);
-	}
-
-	/* insert the node into the rbtree */
-	rb_link_node(&rbnode->node, parent, new);
-	rb_insert_color(&rbnode->node, root);
-
-	return 1;
-}
-
-static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec)
-{
-	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	struct rb_node *node;
-	struct snd_soc_rbtree_node *rbnode;
-	unsigned int regtmp;
-	unsigned int val, def;
-	int ret;
-	int i;
-
-	rbtree_ctx = codec->reg_cache;
-	for (node = rb_first(&rbtree_ctx->root); node; node = rb_next(node)) {
-		rbnode = rb_entry(node, struct snd_soc_rbtree_node, node);
-		for (i = 0; i < rbnode->blklen; ++i) {
-			regtmp = rbnode->base_reg + i;
-			val = snd_soc_rbtree_get_register(rbnode, i);
-			def = snd_soc_get_cache_val(codec->reg_def_copy, i,
-						    rbnode->word_size);
-			if (val == def)
-				continue;
-
-			WARN_ON(!snd_soc_codec_writable_register(codec, regtmp));
-
-			codec->cache_bypass = 1;
-			ret = snd_soc_write(codec, regtmp, val);
-			codec->cache_bypass = 0;
-			if (ret)
-				return ret;
-			dev_dbg(codec->dev, "Synced register %#x, value = %#x\n",
-				regtmp, val);
-		}
-	}
-
-	return 0;
-}
-
-static int snd_soc_rbtree_insert_to_block(struct snd_soc_rbtree_node *rbnode,
-					  unsigned int pos, unsigned int reg,
-					  unsigned int value)
-{
-	u8 *blk;
-
-	blk = krealloc(rbnode->block,
-		       (rbnode->blklen + 1) * rbnode->word_size, GFP_KERNEL);
-	if (!blk)
-		return -ENOMEM;
-
-	/* insert the register value in the correct place in the rbnode block */
-	memmove(blk + (pos + 1) * rbnode->word_size,
-		blk + pos * rbnode->word_size,
-		(rbnode->blklen - pos) * rbnode->word_size);
-
-	/* update the rbnode block, its size and the base register */
-	rbnode->block = blk;
-	rbnode->blklen++;
-	if (!pos)
-		rbnode->base_reg = reg;
-
-	snd_soc_rbtree_set_register(rbnode, pos, value);
-	return 0;
-}
-
-static int snd_soc_rbtree_cache_write(struct snd_soc_codec *codec,
-				      unsigned int reg, unsigned int value)
-{
-	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	struct snd_soc_rbtree_node *rbnode, *rbnode_tmp;
-	struct rb_node *node;
-	unsigned int val;
-	unsigned int reg_tmp;
-	unsigned int base_reg, top_reg;
-	unsigned int pos;
-	int i;
-	int ret;
-
-	rbtree_ctx = codec->reg_cache;
-	/* look up the required register in the cached rbnode */
-	rbnode = rbtree_ctx->cached_rbnode;
-	if (rbnode) {
-		snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
-		if (reg >= base_reg && reg <= top_reg) {
-			reg_tmp = reg - base_reg;
-			val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
-			if (val == value)
-				return 0;
-			snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
-			return 0;
-		}
-	}
-	/* if we can't locate it in the cached rbnode we'll have
-	 * to traverse the rbtree looking for it.
-	 */
-	rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
-	if (rbnode) {
-		reg_tmp = reg - rbnode->base_reg;
-		val = snd_soc_rbtree_get_register(rbnode, reg_tmp);
-		if (val == value)
-			return 0;
-		snd_soc_rbtree_set_register(rbnode, reg_tmp, value);
-		rbtree_ctx->cached_rbnode = rbnode;
-	} else {
-		/* bail out early, no need to create the rbnode yet */
-		if (!value)
-			return 0;
-		/* look for an adjacent register to the one we are about to add */
-		for (node = rb_first(&rbtree_ctx->root); node;
-		     node = rb_next(node)) {
-			rbnode_tmp = rb_entry(node, struct snd_soc_rbtree_node, node);
-			for (i = 0; i < rbnode_tmp->blklen; ++i) {
-				reg_tmp = rbnode_tmp->base_reg + i;
-				if (abs(reg_tmp - reg) != 1)
-					continue;
-				/* decide where in the block to place our register */
-				if (reg_tmp + 1 == reg)
-					pos = i + 1;
-				else
-					pos = i;
-				ret = snd_soc_rbtree_insert_to_block(rbnode_tmp, pos,
-								     reg, value);
-				if (ret)
-					return ret;
-				rbtree_ctx->cached_rbnode = rbnode_tmp;
-				return 0;
-			}
-		}
-		/* we did not manage to find a place to insert it in an existing
-		 * block so create a new rbnode with a single register in its block.
-		 * This block will get populated further if any other adjacent
-		 * registers get modified in the future.
-		 */
-		rbnode = kzalloc(sizeof *rbnode, GFP_KERNEL);
-		if (!rbnode)
-			return -ENOMEM;
-		rbnode->blklen = 1;
-		rbnode->base_reg = reg;
-		rbnode->word_size = codec->driver->reg_word_size;
-		rbnode->block = kmalloc(rbnode->blklen * rbnode->word_size,
-					GFP_KERNEL);
-		if (!rbnode->block) {
-			kfree(rbnode);
-			return -ENOMEM;
-		}
-		snd_soc_rbtree_set_register(rbnode, 0, value);
-		snd_soc_rbtree_insert(&rbtree_ctx->root, rbnode);
-		rbtree_ctx->cached_rbnode = rbnode;
-	}
-
-	return 0;
-}
-
-static int snd_soc_rbtree_cache_read(struct snd_soc_codec *codec,
-				     unsigned int reg, unsigned int *value)
-{
-	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	struct snd_soc_rbtree_node *rbnode;
-	unsigned int base_reg, top_reg;
-	unsigned int reg_tmp;
-
-	rbtree_ctx = codec->reg_cache;
-	/* look up the required register in the cached rbnode */
-	rbnode = rbtree_ctx->cached_rbnode;
-	if (rbnode) {
-		snd_soc_rbtree_get_base_top_reg(rbnode, &base_reg, &top_reg);
-		if (reg >= base_reg && reg <= top_reg) {
-			reg_tmp = reg - base_reg;
-			*value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
-			return 0;
-		}
-	}
-	/* if we can't locate it in the cached rbnode we'll have
-	 * to traverse the rbtree looking for it.
-	 */
-	rbnode = snd_soc_rbtree_lookup(&rbtree_ctx->root, reg);
-	if (rbnode) {
-		reg_tmp = reg - rbnode->base_reg;
-		*value = snd_soc_rbtree_get_register(rbnode, reg_tmp);
-		rbtree_ctx->cached_rbnode = rbnode;
-	} else {
-		/* uninitialized registers default to 0 */
-		*value = 0;
-	}
-
-	return 0;
-}
-
-static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec)
-{
-	struct rb_node *next;
-	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	struct snd_soc_rbtree_node *rbtree_node;
-
-	/* if we've already been called then just return */
-	rbtree_ctx = codec->reg_cache;
-	if (!rbtree_ctx)
-		return 0;
-
-	/* free up the rbtree */
-	next = rb_first(&rbtree_ctx->root);
-	while (next) {
-		rbtree_node = rb_entry(next, struct snd_soc_rbtree_node, node);
-		next = rb_next(&rbtree_node->node);
-		rb_erase(&rbtree_node->node, &rbtree_ctx->root);
-		kfree(rbtree_node->block);
-		kfree(rbtree_node);
-	}
-
-	/* release the resources */
-	kfree(codec->reg_cache);
-	codec->reg_cache = NULL;
-
-	return 0;
-}
-
-static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec)
-{
-	struct snd_soc_rbtree_ctx *rbtree_ctx;
-	unsigned int word_size;
-	unsigned int val;
-	int i;
-	int ret;
-
-	codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL);
-	if (!codec->reg_cache)
-		return -ENOMEM;
-
-	rbtree_ctx = codec->reg_cache;
-	rbtree_ctx->root = RB_ROOT;
-	rbtree_ctx->cached_rbnode = NULL;
-
-	if (!codec->reg_def_copy)
-		return 0;
-
-	word_size = codec->driver->reg_word_size;
-	for (i = 0; i < codec->driver->reg_cache_size; ++i) {
-		val = snd_soc_get_cache_val(codec->reg_def_copy, i,
-					    word_size);
-		if (!val)
-			continue;
-		ret = snd_soc_rbtree_cache_write(codec, i, val);
-		if (ret)
-			goto err;
-	}
-
-	return 0;
-
-err:
-	snd_soc_cache_exit(codec);
-	return ret;
-}
-
 static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec)
 {
 	int i;
@@ -516,15 +144,6 @@ static const struct snd_soc_cache_ops cache_types[] = {
 		.write = snd_soc_flat_cache_write,
 		.sync = snd_soc_flat_cache_sync
 	},
-	{
-		.id = SND_SOC_RBTREE_COMPRESSION,
-		.name = "rbtree",
-		.init = snd_soc_rbtree_cache_init,
-		.exit = snd_soc_rbtree_cache_exit,
-		.read = snd_soc_rbtree_cache_read,
-		.write = snd_soc_rbtree_cache_write,
-		.sync = snd_soc_rbtree_cache_sync
-	}
 };
 
 int snd_soc_cache_init(struct snd_soc_codec *codec)
-- 
cgit v1.2.3-70-g09d2


From bec4fa05e25f7e78ec67df389539acc6bb352a2a Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Mon, 12 Dec 2011 15:55:34 -0700
Subject: ASoC: Add utility to set a card's name from device tree

Implement snd_soc_of_parse_card_name(), a utility function that sets a
card's name from device tree. The machine driver specifies the DT
property to use, since this is binding-specific.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  3 +++
 sound/soc/soc-core.c | 25 +++++++++++++++++++++++++
 2 files changed, 28 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index a35cf14a8eb..278f3b892ca 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -961,6 +961,9 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
 int snd_soc_util_init(void);
 void snd_soc_util_exit(void);
 
+int snd_soc_of_parse_card_name(struct snd_soc_card *card,
+			       const char *propname);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1252ab1ebf6..51eef9b7b53 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -32,6 +32,7 @@
 #include <linux/platform_device.h>
 #include <linux/ctype.h>
 #include <linux/slab.h>
+#include <linux/of.h>
 #include <sound/ac97_codec.h>
 #include <sound/core.h>
 #include <sound/jack.h>
@@ -3317,6 +3318,30 @@ found:
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
 
+/* Retrieve a card's name from device tree */
+int snd_soc_of_parse_card_name(struct snd_soc_card *card,
+			       const char *propname)
+{
+	struct device_node *np = card->dev->of_node;
+	int ret;
+
+	ret = of_property_read_string_index(np, propname, 0, &card->name);
+	/*
+	 * EINVAL means the property does not exist. This is fine providing
+	 * card->name was previously set, which is checked later in
+	 * snd_soc_register_card.
+	 */
+	if (ret < 0 && ret != -EINVAL) {
+		dev_err(card->dev,
+			"Property '%s' could not be read: %d\n",
+			propname, ret);
+		return ret;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name);
+
 static int __init snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
-- 
cgit v1.2.3-70-g09d2


From a4a54dd5bb1bb01010f46147d6d8b452255957bf Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Mon, 12 Dec 2011 15:55:35 -0700
Subject: ASoC: Add utility to parse DAPM routes from device tree

Implement snd_soc_of_parse_audio_routing(), a utility function that can
parses a simple DAPM route table from device tree.The machine driver
specifies the DT property to use, since this is binding-specific.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  2 ++
 sound/soc/soc-core.c | 57 ++++++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 59 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 278f3b892ca..db8acd29904 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -963,6 +963,8 @@ void snd_soc_util_exit(void);
 
 int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname);
+int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
+				   const char *propname);
 
 #include <sound/soc-dai.h>
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 51eef9b7b53..42ad2db8f08 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3342,6 +3342,63 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 }
 EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name);
 
+int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
+				   const char *propname)
+{
+	struct device_node *np = card->dev->of_node;
+	int num_routes;
+	struct snd_soc_dapm_route *routes;
+	int i, ret;
+
+	num_routes = of_property_count_strings(np, propname);
+	if (num_routes & 1) {
+		dev_err(card->dev,
+			"Property '%s's length is not even\n",
+			propname);
+		return -EINVAL;
+	}
+	num_routes /= 2;
+	if (!num_routes) {
+		dev_err(card->dev,
+			"Property '%s's length is zero\n",
+			propname);
+		return -EINVAL;
+	}
+
+	routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes),
+			      GFP_KERNEL);
+	if (!routes) {
+		dev_err(card->dev,
+			"Could not allocate DAPM route table\n");
+		return -EINVAL;
+	}
+
+	for (i = 0; i < num_routes; i++) {
+		ret = of_property_read_string_index(np, propname,
+			2 * i, &routes[i].sink);
+		if (ret) {
+			dev_err(card->dev,
+				"Property '%s' index %d could not be read: %d\n",
+				propname, 2 * i, ret);
+			return -EINVAL;
+		}
+		ret = of_property_read_string_index(np, propname,
+			(2 * i) + 1, &routes[i].source);
+		if (ret) {
+			dev_err(card->dev,
+				"Property '%s' index %d could not be read: %d\n",
+				propname, (2 * i) + 1, ret);
+			return -EINVAL;
+		}
+	}
+
+	card->num_dapm_routes = num_routes;
+	card->dapm_routes = routes;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing);
+
 static int __init snd_soc_init(void)
 {
 #ifdef CONFIG_DEBUG_FS
-- 
cgit v1.2.3-70-g09d2


From 5a5049637cf08c4c17805be679c19544bb27fb92 Mon Sep 17 00:00:00 2001
From: Stephen Warren <swarren@nvidia.com>
Date: Wed, 21 Dec 2011 10:40:59 -0700
Subject: ASoC: Allow DAI links to be specified using device tree nodes

DAI link endpoints and platform (DMA) devices are currently specified
by name. When instantiating sound cards from device tree, it may be more
convenient to refer to these devices by phandle in the device tree, and
for code to describe DAI links using the "struct device_node *"
("of_node") those phandles map to.

This change adds new fields to snd_soc_dai_link which can "name" devices
using of_node, enhances soc_bind_dai_link() to allow binding based on
of_node, and enhances snd_soc_register_card() to ensure that illegal
combinations of name and of_node are not used.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  4 ++++
 sound/soc/soc-core.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++------
 2 files changed, 61 insertions(+), 7 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index db8acd29904..8391b0ec217 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -231,6 +231,7 @@ enum snd_soc_bias_level {
 	SND_SOC_BIAS_ON = 3,
 };
 
+struct device_node;
 struct snd_jack;
 struct snd_soc_card;
 struct snd_soc_pcm_stream;
@@ -703,8 +704,11 @@ struct snd_soc_dai_link {
 	const char *name;			/* Codec name */
 	const char *stream_name;		/* Stream name */
 	const char *codec_name;		/* for multi-codec */
+	const struct device_node *codec_of_node;
 	const char *platform_name;	/* for multi-platform */
+	const struct device_node *platform_of_node;
 	const char *cpu_dai_name;
+	const struct device_node *cpu_dai_of_node;
 	const char *codec_dai_name;
 
 	unsigned int dai_fmt;           /* format to set on init */
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 42ad2db8f08..a4592cbee49 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -764,8 +764,13 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	}
 	/* no, then find CPU DAI from registered DAIs*/
 	list_for_each_entry(cpu_dai, &dai_list, list) {
-		if (strcmp(cpu_dai->name, dai_link->cpu_dai_name))
-			continue;
+		if (dai_link->cpu_dai_of_node) {
+			if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node)
+				continue;
+		} else {
+			if (strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+				continue;
+		}
 
 		rtd->cpu_dai = cpu_dai;
 		goto find_codec;
@@ -781,8 +786,13 @@ find_codec:
 
 	/* no, then find CODEC from registered CODECs*/
 	list_for_each_entry(codec, &codec_list, list) {
-		if (strcmp(codec->name, dai_link->codec_name))
-			continue;
+		if (dai_link->codec_of_node) {
+			if (codec->dev->of_node != dai_link->codec_of_node)
+				continue;
+		} else {
+			if (strcmp(codec->name, dai_link->codec_name))
+				continue;
+		}
 
 		rtd->codec = codec;
 
@@ -814,13 +824,19 @@ find_platform:
 
 	/* if there's no platform we match on the empty platform */
 	platform_name = dai_link->platform_name;
-	if (!platform_name)
+	if (!platform_name && !dai_link->platform_of_node)
 		platform_name = "snd-soc-dummy";
 
 	/* no, then find one from the set of registered platforms */
 	list_for_each_entry(platform, &platform_list, list) {
-		if (strcmp(platform->name, platform_name))
-			continue;
+		if (dai_link->platform_of_node) {
+			if (platform->dev->of_node !=
+			    dai_link->platform_of_node)
+				continue;
+		} else {
+			if (strcmp(platform->name, platform_name))
+				continue;
+		}
 
 		rtd->platform = platform;
 		goto out;
@@ -2831,6 +2847,40 @@ int snd_soc_register_card(struct snd_soc_card *card)
 	if (!card->name || !card->dev)
 		return -EINVAL;
 
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai_link *link = &card->dai_link[i];
+
+		/*
+		 * Codec must be specified by 1 of name or OF node,
+		 * not both or neither.
+		 */
+		if (!!link->codec_name == !!link->codec_of_node) {
+			dev_err(card->dev,
+				"Neither/both codec name/of_node are set\n");
+			return -EINVAL;
+		}
+
+		/*
+		 * Platform may be specified by either name or OF node, but
+		 * can be left unspecified, and a dummy platform will be used.
+		 */
+		if (link->platform_name && link->platform_of_node) {
+			dev_err(card->dev,
+				"Both platform name/of_node are set\n");
+			return -EINVAL;
+		}
+
+		/*
+		 * CPU DAI must be specified by 1 of name or OF node,
+		 * not both or neither.
+		 */
+		if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) {
+			dev_err(card->dev,
+				"Neither/both cpu_dai name/of_node are set\n");
+			return -EINVAL;
+		}
+	}
+
 	dev_set_drvdata(card->dev, card);
 
 	snd_soc_initialize_card_lists(card);
-- 
cgit v1.2.3-70-g09d2


From 354a21423d09c2a6afe0fcea9dbbda9cdada6e45 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 22 Dec 2011 12:16:39 +0000
Subject: ASoC: Declare soc_new_pcm() properly

Ensure that everything is seeing the same declaration by moving it to
a header file rather than putting the declaration in soc-core.c

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
---
 include/sound/soc.h  | 1 +
 sound/soc/soc-core.c | 2 --
 2 files changed, 1 insertion(+), 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8391b0ec217..f75d1ccc5c5 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -317,6 +317,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform,
 					unsigned int reg);
 int snd_soc_platform_write(struct snd_soc_platform *platform,
 					unsigned int reg, unsigned int val);
+int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
 
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a4592cbee49..acbb96005a6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -59,8 +59,6 @@ static LIST_HEAD(dai_list);
 static LIST_HEAD(platform_list);
 static LIST_HEAD(codec_list);
 
-int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
-
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
  * It can be used to eliminate pops between different playback streams, e.g.
-- 
cgit v1.2.3-70-g09d2


From 3eafc959b32f71d3fe6b27c9eae7495a23acfc3a Mon Sep 17 00:00:00 2001
From: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Date: Fri, 23 Dec 2011 10:36:36 +0530
Subject: ALSA: core: add support for compressed devices

Use the minor numbers 2 and 3 for audio compressed offload devices.
Also add support for these devices in core

Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/core.h   | 1 +
 include/sound/minors.h | 4 +++-
 sound/core/sound.c     | 1 +
 3 files changed, 5 insertions(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/core.h b/include/sound/core.h
index 3be5ab782b9..5ab255f196c 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -62,6 +62,7 @@ typedef int __bitwise snd_device_type_t;
 #define	SNDRV_DEV_BUS		((__force snd_device_type_t) 0x1007)
 #define	SNDRV_DEV_CODEC		((__force snd_device_type_t) 0x1008)
 #define	SNDRV_DEV_JACK          ((__force snd_device_type_t) 0x1009)
+#define	SNDRV_DEV_COMPRESS	((__force snd_device_type_t) 0x100A)
 #define	SNDRV_DEV_LOWLEVEL	((__force snd_device_type_t) 0x2000)
 
 typedef int __bitwise snd_device_state_t;
diff --git a/include/sound/minors.h b/include/sound/minors.h
index 8f764204a85..5978f9a8c8b 100644
--- a/include/sound/minors.h
+++ b/include/sound/minors.h
@@ -35,7 +35,7 @@
 #define SNDRV_MINOR_TIMER		33	/* SNDRV_MINOR_GLOBAL + 1 * 32 */
 
 #ifndef CONFIG_SND_DYNAMIC_MINORS
-						/* 2 - 3 (reserved) */
+#define SNDRV_MINOR_COMPRESS		2	/* 2 - 3 */
 #define SNDRV_MINOR_HWDEP		4	/* 4 - 7 */
 #define SNDRV_MINOR_RAWMIDI		8	/* 8 - 15 */
 #define SNDRV_MINOR_PCM_PLAYBACK	16	/* 16 - 23 */
@@ -49,6 +49,7 @@
 #define SNDRV_DEVICE_TYPE_PCM_CAPTURE	SNDRV_MINOR_PCM_CAPTURE
 #define SNDRV_DEVICE_TYPE_SEQUENCER	SNDRV_MINOR_SEQUENCER
 #define SNDRV_DEVICE_TYPE_TIMER		SNDRV_MINOR_TIMER
+#define SNDRV_DEVICE_TYPE_COMPRESS	SNDRV_MINOR_COMPRESS
 
 #else /* CONFIG_SND_DYNAMIC_MINORS */
 
@@ -60,6 +61,7 @@ enum {
 	SNDRV_DEVICE_TYPE_RAWMIDI,
 	SNDRV_DEVICE_TYPE_PCM_PLAYBACK,
 	SNDRV_DEVICE_TYPE_PCM_CAPTURE,
+	SNDRV_DEVICE_TYPE_COMPRESS,
 };
 
 #endif /* CONFIG_SND_DYNAMIC_MINORS */
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 828af353ea9..28f35593a75 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -229,6 +229,7 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev)
 	case SNDRV_DEVICE_TYPE_RAWMIDI:
 	case SNDRV_DEVICE_TYPE_PCM_PLAYBACK:
 	case SNDRV_DEVICE_TYPE_PCM_CAPTURE:
+	case SNDRV_DEVICE_TYPE_COMPRESS:
 		if (snd_BUG_ON(!card))
 			return -EINVAL;
 		minor = SNDRV_MINOR(card->number, type + dev);
-- 
cgit v1.2.3-70-g09d2


From 50c34cfe7bbb5cef9d32de63286ff97d8d6877a9 Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@linux.intel.com>
Date: Fri, 23 Dec 2011 10:36:37 +0530
Subject: ALSA: core: add compress parameter definations

The patch adds the various definations used to define the encoder
and decoder parameters

Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/compress_params.h | 397 ++++++++++++++++++++++++++++++++++++++++
 1 file changed, 397 insertions(+)
 create mode 100644 include/sound/compress_params.h

(limited to 'include/sound')

diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
new file mode 100644
index 00000000000..d97d69f81a7
--- /dev/null
+++ b/include/sound/compress_params.h
@@ -0,0 +1,397 @@
+/*
+ *  compress_params.h - codec types and parameters for compressed data
+ *  streaming interface
+ *
+ *  Copyright (C) 2011 Intel Corporation
+ *  Authors:	Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *              Vinod Koul <vinod.koul@linux.intel.com>
+ *
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * The definitions in this file are derived from the OpenMAX AL version 1.1
+ * and OpenMAX IL v 1.1.2 header files which contain the copyright notice below.
+ *
+ * Copyright (c) 2007-2010 The Khronos Group Inc.
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and/or associated documentation files (the
+ * "Materials "), to deal in the Materials without restriction, including
+ * without limitation the rights to use, copy, modify, merge, publish,
+ * distribute, sublicense, and/or sell copies of the Materials, and to
+ * permit persons to whom the Materials are furnished to do so, subject to
+ * the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included
+ * in all copies or substantial portions of the Materials.
+ *
+ * THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
+ * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
+ * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
+ * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
+ * MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS.
+ *
+ */
+#ifndef __SND_COMPRESS_PARAMS_H
+#define __SND_COMPRESS_PARAMS_H
+
+/* AUDIO CODECS SUPPORTED */
+#define MAX_NUM_CODECS 32
+#define MAX_NUM_CODEC_DESCRIPTORS 32
+#define MAX_NUM_BITRATES 32
+
+/* Codecs are listed linearly to allow for extensibility */
+#define SND_AUDIOCODEC_PCM                   ((__u32) 0x00000001)
+#define SND_AUDIOCODEC_MP3                   ((__u32) 0x00000002)
+#define SND_AUDIOCODEC_AMR                   ((__u32) 0x00000003)
+#define SND_AUDIOCODEC_AMRWB                 ((__u32) 0x00000004)
+#define SND_AUDIOCODEC_AMRWBPLUS             ((__u32) 0x00000005)
+#define SND_AUDIOCODEC_AAC                   ((__u32) 0x00000006)
+#define SND_AUDIOCODEC_WMA                   ((__u32) 0x00000007)
+#define SND_AUDIOCODEC_REAL                  ((__u32) 0x00000008)
+#define SND_AUDIOCODEC_VORBIS                ((__u32) 0x00000009)
+#define SND_AUDIOCODEC_FLAC                  ((__u32) 0x0000000A)
+#define SND_AUDIOCODEC_IEC61937              ((__u32) 0x0000000B)
+#define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
+#define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
+
+/*
+ * Profile and modes are listed with bit masks. This allows for a
+ * more compact representation of fields that will not evolve
+ * (in contrast to the list of codecs)
+ */
+
+#define SND_AUDIOPROFILE_PCM                 ((__u32) 0x00000001)
+
+/* MP3 modes are only useful for encoders */
+#define SND_AUDIOCHANMODE_MP3_MONO           ((__u32) 0x00000001)
+#define SND_AUDIOCHANMODE_MP3_STEREO         ((__u32) 0x00000002)
+#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO    ((__u32) 0x00000004)
+#define SND_AUDIOCHANMODE_MP3_DUAL           ((__u32) 0x00000008)
+
+#define SND_AUDIOPROFILE_AMR                 ((__u32) 0x00000001)
+
+/* AMR modes are only useful for encoders */
+#define SND_AUDIOMODE_AMR_DTX_OFF            ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AMR_VAD1               ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AMR_VAD2               ((__u32) 0x00000004)
+
+#define SND_AUDIOSTREAMFORMAT_UNDEFINED	     ((__u32) 0x00000000)
+#define SND_AUDIOSTREAMFORMAT_CONFORMANCE    ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_IF1            ((__u32) 0x00000002)
+#define SND_AUDIOSTREAMFORMAT_IF2            ((__u32) 0x00000004)
+#define SND_AUDIOSTREAMFORMAT_FSF            ((__u32) 0x00000008)
+#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD     ((__u32) 0x00000010)
+#define SND_AUDIOSTREAMFORMAT_ITU            ((__u32) 0x00000020)
+
+#define SND_AUDIOPROFILE_AMRWB               ((__u32) 0x00000001)
+
+/* AMRWB modes are only useful for encoders */
+#define SND_AUDIOMODE_AMRWB_DTX_OFF          ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AMRWB_VAD1             ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AMRWB_VAD2             ((__u32) 0x00000004)
+
+#define SND_AUDIOPROFILE_AMRWBPLUS           ((__u32) 0x00000001)
+
+#define SND_AUDIOPROFILE_AAC                 ((__u32) 0x00000001)
+
+/* AAC modes are required for encoders and decoders */
+#define SND_AUDIOMODE_AAC_MAIN               ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AAC_LC                 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AAC_SSR                ((__u32) 0x00000004)
+#define SND_AUDIOMODE_AAC_LTP                ((__u32) 0x00000008)
+#define SND_AUDIOMODE_AAC_HE                 ((__u32) 0x00000010)
+#define SND_AUDIOMODE_AAC_SCALABLE           ((__u32) 0x00000020)
+#define SND_AUDIOMODE_AAC_ERLC               ((__u32) 0x00000040)
+#define SND_AUDIOMODE_AAC_LD                 ((__u32) 0x00000080)
+#define SND_AUDIOMODE_AAC_HE_PS              ((__u32) 0x00000100)
+#define SND_AUDIOMODE_AAC_HE_MPS             ((__u32) 0x00000200)
+
+/* AAC formats are required for encoders and decoders */
+#define SND_AUDIOSTREAMFORMAT_MP2ADTS        ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_MP4ADTS        ((__u32) 0x00000002)
+#define SND_AUDIOSTREAMFORMAT_MP4LOAS        ((__u32) 0x00000004)
+#define SND_AUDIOSTREAMFORMAT_MP4LATM        ((__u32) 0x00000008)
+#define SND_AUDIOSTREAMFORMAT_ADIF           ((__u32) 0x00000010)
+#define SND_AUDIOSTREAMFORMAT_MP4FF          ((__u32) 0x00000020)
+#define SND_AUDIOSTREAMFORMAT_RAW            ((__u32) 0x00000040)
+
+#define SND_AUDIOPROFILE_WMA7                ((__u32) 0x00000001)
+#define SND_AUDIOPROFILE_WMA8                ((__u32) 0x00000002)
+#define SND_AUDIOPROFILE_WMA9                ((__u32) 0x00000004)
+#define SND_AUDIOPROFILE_WMA10               ((__u32) 0x00000008)
+
+#define SND_AUDIOMODE_WMA_LEVEL1             ((__u32) 0x00000001)
+#define SND_AUDIOMODE_WMA_LEVEL2             ((__u32) 0x00000002)
+#define SND_AUDIOMODE_WMA_LEVEL3             ((__u32) 0x00000004)
+#define SND_AUDIOMODE_WMA_LEVEL4             ((__u32) 0x00000008)
+#define SND_AUDIOMODE_WMAPRO_LEVELM0         ((__u32) 0x00000010)
+#define SND_AUDIOMODE_WMAPRO_LEVELM1         ((__u32) 0x00000020)
+#define SND_AUDIOMODE_WMAPRO_LEVELM2         ((__u32) 0x00000040)
+#define SND_AUDIOMODE_WMAPRO_LEVELM3         ((__u32) 0x00000080)
+
+#define SND_AUDIOSTREAMFORMAT_WMA_ASF        ((__u32) 0x00000001)
+/*
+ * Some implementations strip the ASF header and only send ASF packets
+ * to the DSP
+ */
+#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR  ((__u32) 0x00000002)
+
+#define SND_AUDIOPROFILE_REALAUDIO           ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_REALAUDIO_G2           ((__u32) 0x00000001)
+#define SND_AUDIOMODE_REALAUDIO_8            ((__u32) 0x00000002)
+#define SND_AUDIOMODE_REALAUDIO_10           ((__u32) 0x00000004)
+#define SND_AUDIOMODE_REALAUDIO_SURROUND     ((__u32) 0x00000008)
+
+#define SND_AUDIOPROFILE_VORBIS              ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_VORBIS                 ((__u32) 0x00000001)
+
+#define SND_AUDIOPROFILE_FLAC                ((__u32) 0x00000001)
+
+/*
+ * Define quality levels for FLAC encoders, from LEVEL0 (fast)
+ * to LEVEL8 (best)
+ */
+#define SND_AUDIOMODE_FLAC_LEVEL0            ((__u32) 0x00000001)
+#define SND_AUDIOMODE_FLAC_LEVEL1            ((__u32) 0x00000002)
+#define SND_AUDIOMODE_FLAC_LEVEL2            ((__u32) 0x00000004)
+#define SND_AUDIOMODE_FLAC_LEVEL3            ((__u32) 0x00000008)
+#define SND_AUDIOMODE_FLAC_LEVEL4            ((__u32) 0x00000010)
+#define SND_AUDIOMODE_FLAC_LEVEL5            ((__u32) 0x00000020)
+#define SND_AUDIOMODE_FLAC_LEVEL6            ((__u32) 0x00000040)
+#define SND_AUDIOMODE_FLAC_LEVEL7            ((__u32) 0x00000080)
+#define SND_AUDIOMODE_FLAC_LEVEL8            ((__u32) 0x00000100)
+
+#define SND_AUDIOSTREAMFORMAT_FLAC           ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_FLAC_OGG       ((__u32) 0x00000002)
+
+/* IEC61937 payloads without CUVP and preambles */
+#define SND_AUDIOPROFILE_IEC61937            ((__u32) 0x00000001)
+/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */
+#define SND_AUDIOPROFILE_IEC61937_SPDIF      ((__u32) 0x00000002)
+
+/*
+ * IEC modes are mandatory for decoders. Format autodetection
+ * will only happen on the DSP side with mode 0. The PCM mode should
+ * not be used, the PCM codec should be used instead.
+ */
+#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER  ((__u32) 0x00000000)
+#define SND_AUDIOMODE_IEC_LPCM		     ((__u32) 0x00000001)
+#define SND_AUDIOMODE_IEC_AC3		     ((__u32) 0x00000002)
+#define SND_AUDIOMODE_IEC_MPEG1		     ((__u32) 0x00000004)
+#define SND_AUDIOMODE_IEC_MP3		     ((__u32) 0x00000008)
+#define SND_AUDIOMODE_IEC_MPEG2		     ((__u32) 0x00000010)
+#define SND_AUDIOMODE_IEC_AACLC		     ((__u32) 0x00000020)
+#define SND_AUDIOMODE_IEC_DTS		     ((__u32) 0x00000040)
+#define SND_AUDIOMODE_IEC_ATRAC		     ((__u32) 0x00000080)
+#define SND_AUDIOMODE_IEC_SACD		     ((__u32) 0x00000100)
+#define SND_AUDIOMODE_IEC_EAC3		     ((__u32) 0x00000200)
+#define SND_AUDIOMODE_IEC_DTS_HD	     ((__u32) 0x00000400)
+#define SND_AUDIOMODE_IEC_MLP		     ((__u32) 0x00000800)
+#define SND_AUDIOMODE_IEC_DST		     ((__u32) 0x00001000)
+#define SND_AUDIOMODE_IEC_WMAPRO	     ((__u32) 0x00002000)
+#define SND_AUDIOMODE_IEC_REF_CXT            ((__u32) 0x00004000)
+#define SND_AUDIOMODE_IEC_HE_AAC	     ((__u32) 0x00008000)
+#define SND_AUDIOMODE_IEC_HE_AAC2	     ((__u32) 0x00010000)
+#define SND_AUDIOMODE_IEC_MPEG_SURROUND	     ((__u32) 0x00020000)
+
+#define SND_AUDIOPROFILE_G723_1              ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_G723_1_ANNEX_A         ((__u32) 0x00000001)
+#define SND_AUDIOMODE_G723_1_ANNEX_B         ((__u32) 0x00000002)
+#define SND_AUDIOMODE_G723_1_ANNEX_C         ((__u32) 0x00000004)
+
+#define SND_AUDIOPROFILE_G729                ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_G729_ANNEX_A           ((__u32) 0x00000001)
+#define SND_AUDIOMODE_G729_ANNEX_B           ((__u32) 0x00000002)
+
+/* <FIXME: multichannel encoders aren't supported for now. Would need
+   an additional definition of channel arrangement> */
+
+/* VBR/CBR definitions */
+#define SND_RATECONTROLMODE_CONSTANTBITRATE  ((__u32) 0x00000001)
+#define SND_RATECONTROLMODE_VARIABLEBITRATE  ((__u32) 0x00000002)
+
+/* Encoder options */
+
+struct snd_enc_wma {
+	__u32 super_block_align; /* WMA Type-specific data */
+};
+
+
+/**
+ * struct snd_enc_vorbis
+ * @quality: Sets encoding quality to n, between -1 (low) and 10 (high).
+ * In the default mode of operation, the quality level is 3.
+ * Normal quality range is 0 - 10.
+ * @managed: Boolean. Set  bitrate  management  mode. This turns off the
+ * normal VBR encoding, but allows hard or soft bitrate constraints to be
+ * enforced by the encoder. This mode can be slower, and may also be
+ * lower quality. It is primarily useful for streaming.
+ * @max_bit_rate: Enabled only if managed is TRUE
+ * @min_bit_rate: Enabled only if managed is TRUE
+ * @downmix: Boolean. Downmix input from stereo to mono (has no effect on
+ * non-stereo streams). Useful for lower-bitrate encoding.
+ *
+ * These options were extracted from the OpenMAX IL spec and Gstreamer vorbisenc
+ * properties
+ *
+ * For best quality users should specify VBR mode and set quality levels.
+ */
+
+struct snd_enc_vorbis {
+	__s32 quality;
+	__u32 managed;
+	__u32 max_bit_rate;
+	__u32 min_bit_rate;
+	__u32 downmix;
+};
+
+
+/**
+ * struct snd_enc_real
+ * @quant_bits: number of coupling quantization bits in the stream
+ * @start_region: coupling start region in the stream
+ * @num_regions: number of regions value
+ *
+ * These options were extracted from the OpenMAX IL spec
+ */
+
+struct snd_enc_real {
+	__u32 quant_bits;
+	__u32 start_region;
+	__u32 num_regions;
+};
+
+/**
+ * struct snd_enc_flac
+ * @num: serial number, valid only for OGG formats
+ *	needs to be set by application
+ * @gain: Add replay gain tags
+ *
+ * These options were extracted from the FLAC online documentation
+ * at http://flac.sourceforge.net/documentation_tools_flac.html
+ *
+ * To make the API simpler, it is assumed that the user will select quality
+ * profiles. Additional options that affect encoding quality and speed can
+ * be added at a later stage if needed.
+ *
+ * By default the Subset format is used by encoders.
+ *
+ * TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are
+ * not supported in this API.
+ */
+
+struct snd_enc_flac {
+	__u32 num;
+	__u32 gain;
+};
+
+struct snd_enc_generic {
+	__u32 bw;	/* encoder bandwidth */
+	__s32 reserved[15];
+};
+
+union snd_codec_options {
+	struct snd_enc_wma wma;
+	struct snd_enc_vorbis vorbis;
+	struct snd_enc_real real;
+	struct snd_enc_flac flac;
+	struct snd_enc_generic generic;
+};
+
+/** struct snd_codec_desc - description of codec capabilities
+ * @max_ch: Maximum number of audio channels
+ * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this
+ * @bit_rate: Indexed array containing supported bit rates
+ * @num_bitrates: Number of valid values in bit_rate array
+ * @rate_control: value is specified by SND_RATECONTROLMODE defines.
+ * @profiles: Supported profiles. See SND_AUDIOPROFILE defines.
+ * @modes: Supported modes. See SND_AUDIOMODE defines
+ * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines
+ * @min_buffer: Minimum buffer size handled by codec implementation
+ * @reserved: reserved for future use
+ *
+ * This structure provides a scalar value for profiles, modes and stream
+ * format fields.
+ * If an implementation supports multiple combinations, they will be listed as
+ * codecs with different descriptors, for example there would be 2 descriptors
+ * for AAC-RAW and AAC-ADTS.
+ * This entails some redundancy but makes it easier to avoid invalid
+ * configurations.
+ *
+ */
+
+struct snd_codec_desc {
+	__u32 max_ch;
+	__u32 sample_rates;
+	__u32 bit_rate[MAX_NUM_BITRATES];
+	__u32 num_bitrates;
+	__u32 rate_control;
+	__u32 profiles;
+	__u32 modes;
+	__u32 formats;
+	__u32 min_buffer;
+	__u32 reserved[15];
+};
+
+/** struct snd_codec
+ * @id: Identifies the supported audio encoder/decoder.
+ *		See SND_AUDIOCODEC macros.
+ * @ch_in: Number of input audio channels
+ * @ch_out: Number of output channels. In case of contradiction between
+ *		this field and the channelMode field, the channelMode field
+ *		overrides.
+ * @sample_rate: Audio sample rate of input data
+ * @bit_rate: Bitrate of encoded data. May be ignored by decoders
+ * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines.
+ *               Encoders may rely on profiles for quality levels.
+ *		 May be ignored by decoders.
+ * @profile: Mandatory for encoders, can be mandatory for specific
+ *		decoders as well. See SND_AUDIOPROFILE defines.
+ * @level: Supported level (Only used by WMA at the moment)
+ * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines
+ * @format: Format of encoded bistream. Mandatory when defined.
+ *		See SND_AUDIOSTREAMFORMAT defines.
+ * @align: Block alignment in bytes of an audio sample.
+ *		Only required for PCM or IEC formats.
+ * @options: encoder-specific settings
+ * @reserved: reserved for future use
+ */
+
+struct snd_codec {
+	__u32 id;
+	__u32 ch_in;
+	__u32 ch_out;
+	__u32 sample_rate;
+	__u32 bit_rate;
+	__u32 rate_control;
+	__u32 profile;
+	__u32 level;
+	__u32 ch_mode;
+	__u32 format;
+	__u32 align;
+	union snd_codec_options options;
+	__u32 reserved[3];
+};
+
+#endif
-- 
cgit v1.2.3-70-g09d2


From e60061a379e1f679ff862acfe1be5819fb6d234c Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@linux.intel.com>
Date: Fri, 23 Dec 2011 10:36:38 +0530
Subject: ALSA: core: add API header and driver header files

This patch adds the header files for ioctl definitions and header file for
driver APIs for lower level device drivers to use

Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/compress_driver.h  | 167 +++++++++++++++++++++++++++++++++++++++
 include/sound/compress_offload.h | 161 +++++++++++++++++++++++++++++++++++++
 2 files changed, 328 insertions(+)
 create mode 100644 include/sound/compress_driver.h
 create mode 100644 include/sound/compress_offload.h

(limited to 'include/sound')

diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
new file mode 100644
index 00000000000..48f2a1ff2bb
--- /dev/null
+++ b/include/sound/compress_driver.h
@@ -0,0 +1,167 @@
+/*
+ *  compress_driver.h - compress offload driver definations
+ *
+ *  Copyright (C) 2011 Intel Corporation
+ *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
+ *		Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+#ifndef __COMPRESS_DRIVER_H
+#define __COMPRESS_DRIVER_H
+
+#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/compress_offload.h>
+#include <sound/asound.h>
+#include <sound/pcm.h>
+
+struct snd_compr_ops;
+
+/**
+ * struct snd_compr_runtime: runtime stream description
+ * @state: stream state
+ * @ops: pointer to DSP callbacks
+ * @buffer: pointer to kernel buffer, valid only when not in mmap mode or
+ *	DSP doesn't implement copy
+ * @buffer_size: size of the above buffer
+ * @fragment_size: size of buffer fragment in bytes
+ * @fragments: number of such fragments
+ * @hw_pointer: offset of last location in buffer where DSP copied data
+ * @app_pointer: offset of last location in buffer where app wrote data
+ * @total_bytes_available: cumulative number of bytes made available in
+ *	the ring buffer
+ * @total_bytes_transferred: cumulative bytes transferred by offload DSP
+ * @sleep: poll sleep
+ */
+struct snd_compr_runtime {
+	snd_pcm_state_t state;
+	struct snd_compr_ops *ops;
+	void *buffer;
+	u64 buffer_size;
+	u32 fragment_size;
+	u32 fragments;
+	u64 hw_pointer;
+	u64 app_pointer;
+	u64 total_bytes_available;
+	u64 total_bytes_transferred;
+	wait_queue_head_t sleep;
+};
+
+/**
+ * struct snd_compr_stream: compressed stream
+ * @name: device name
+ * @ops: pointer to DSP callbacks
+ * @runtime: pointer to runtime structure
+ * @device: device pointer
+ * @direction: stream direction, playback/recording
+ * @private_data: pointer to DSP private data
+ */
+struct snd_compr_stream {
+	const char *name;
+	struct snd_compr_ops *ops;
+	struct snd_compr_runtime *runtime;
+	struct snd_compr *device;
+	enum snd_compr_direction direction;
+	void *private_data;
+};
+
+/**
+ * struct snd_compr_ops: compressed path DSP operations
+ * @open: Open the compressed stream
+ * This callback is mandatory and shall keep dsp ready to receive the stream
+ * parameter
+ * @free: Close the compressed stream, mandatory
+ * @set_params: Sets the compressed stream parameters, mandatory
+ * This can be called in during stream creation only to set codec params
+ * and the stream properties
+ * @get_params: retrieve the codec parameters, mandatory
+ * @trigger: Trigger operations like start, pause, resume, drain, stop.
+ * This callback is mandatory
+ * @pointer: Retrieve current h/w pointer information. Mandatory
+ * @copy: Copy the compressed data to/from userspace, Optional
+ * Can't be implemented if DSP supports mmap
+ * @mmap: DSP mmap method to mmap DSP memory
+ * @ack: Ack for DSP when data is written to audio buffer, Optional
+ * Not valid if copy is implemented
+ * @get_caps: Retrieve DSP capabilities, mandatory
+ * @get_codec_caps: Retrieve capabilities for a specific codec, mandatory
+ */
+struct snd_compr_ops {
+	int (*open)(struct snd_compr_stream *stream);
+	int (*free)(struct snd_compr_stream *stream);
+	int (*set_params)(struct snd_compr_stream *stream,
+			struct snd_compr_params *params);
+	int (*get_params)(struct snd_compr_stream *stream,
+			struct snd_codec *params);
+	int (*trigger)(struct snd_compr_stream *stream, int cmd);
+	int (*pointer)(struct snd_compr_stream *stream,
+			struct snd_compr_tstamp *tstamp);
+	int (*copy)(struct snd_compr_stream *stream, const char __user *buf,
+		       size_t count);
+	int (*mmap)(struct snd_compr_stream *stream,
+			struct vm_area_struct *vma);
+	int (*ack)(struct snd_compr_stream *stream, size_t bytes);
+	int (*get_caps) (struct snd_compr_stream *stream,
+			struct snd_compr_caps *caps);
+	int (*get_codec_caps) (struct snd_compr_stream *stream,
+			struct snd_compr_codec_caps *codec);
+};
+
+/**
+ * struct snd_compr: Compressed device
+ * @name: DSP device name
+ * @dev: Device pointer
+ * @ops: pointer to DSP callbacks
+ * @private_data: pointer to DSP pvt data
+ * @card: sound card pointer
+ * @direction: Playback or capture direction
+ * @lock: device lock
+ * @device: device id
+ */
+struct snd_compr {
+	const char *name;
+	struct device *dev;
+	struct snd_compr_ops *ops;
+	void *private_data;
+	struct snd_card *card;
+	unsigned int direction;
+	struct mutex lock;
+	int device;
+};
+
+/* compress device register APIs */
+int snd_compress_register(struct snd_compr *device);
+int snd_compress_deregister(struct snd_compr *device);
+int snd_compress_new(struct snd_card *card, int device,
+			int type, struct snd_compr *compr);
+
+/* dsp driver callback apis
+ * For playback: driver should call snd_compress_fragment_elapsed() to let the
+ * framework know that a fragment has been consumed from the ring buffer
+ *
+ * For recording: we want to know when a frame is available or when
+ * at least one frame is available so snd_compress_frame_elapsed()
+ * callback should be called when a encodeded frame is available
+ */
+static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream)
+{
+	wake_up(&stream->runtime->sleep);
+}
+
+#endif
diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h
new file mode 100644
index 00000000000..05341a43fed
--- /dev/null
+++ b/include/sound/compress_offload.h
@@ -0,0 +1,161 @@
+/*
+ *  compress_offload.h - compress offload header definations
+ *
+ *  Copyright (C) 2011 Intel Corporation
+ *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
+ *		Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+#ifndef __COMPRESS_OFFLOAD_H
+#define __COMPRESS_OFFLOAD_H
+
+#include <linux/types.h>
+#include <sound/asound.h>
+#include <sound/compress_params.h>
+
+
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
+/**
+ * struct snd_compressed_buffer: compressed buffer
+ * @fragment_size: size of buffer fragment in bytes
+ * @fragments: number of such fragments
+ */
+struct snd_compressed_buffer {
+	__u32 fragment_size;
+	__u32 fragments;
+};
+
+/**
+ * struct snd_compr_params: compressed stream params
+ * @buffer: buffer description
+ * @codec: codec parameters
+ * @no_wake_mode: dont wake on fragment elapsed
+ */
+struct snd_compr_params {
+	struct snd_compressed_buffer buffer;
+	struct snd_codec codec;
+	__u8 no_wake_mode;
+};
+
+/**
+ * struct snd_compr_tstamp: timestamp descriptor
+ * @byte_offset: Byte offset in ring buffer to DSP
+ * @copied_total: Total number of bytes copied from/to ring buffer to/by DSP
+ * @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by
+ *	large steps and should only be used to monitor encoding/decoding
+ *	progress. It shall not be used for timing estimates.
+ * @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio
+ * output/input. This field should be used for A/V sync or time estimates.
+ * @sampling_rate: sampling rate of audio
+ */
+struct snd_compr_tstamp {
+	__u32 byte_offset;
+	__u32 copied_total;
+	snd_pcm_uframes_t pcm_frames;
+	snd_pcm_uframes_t pcm_io_frames;
+	__u32 sampling_rate;
+};
+
+/**
+ * struct snd_compr_avail: avail descriptor
+ * @avail: Number of bytes available in ring buffer for writing/reading
+ * @tstamp: timestamp infomation
+ */
+struct snd_compr_avail {
+	__u64 avail;
+	struct snd_compr_tstamp tstamp;
+};
+
+enum snd_compr_direction {
+	SND_COMPRESS_PLAYBACK = 0,
+	SND_COMPRESS_CAPTURE
+};
+
+/**
+ * struct snd_compr_caps: caps descriptor
+ * @codecs: pointer to array of codecs
+ * @direction: direction supported. Of type snd_compr_direction
+ * @min_fragment_size: minimum fragment supported by DSP
+ * @max_fragment_size: maximum fragment supported by DSP
+ * @min_fragments: min fragments supported by DSP
+ * @max_fragments: max fragments supported by DSP
+ * @num_codecs: number of codecs supported
+ * @reserved: reserved field
+ */
+struct snd_compr_caps {
+	__u32 num_codecs;
+	__u32 direction;
+	__u32 min_fragment_size;
+	__u32 max_fragment_size;
+	__u32 min_fragments;
+	__u32 max_fragments;
+	__u32 codecs[MAX_NUM_CODECS];
+	__u32 reserved[11];
+};
+
+/**
+ * struct snd_compr_codec_caps: query capability of codec
+ * @codec: codec for which capability is queried
+ * @num_descriptors: number of codec descriptors
+ * @descriptor: array of codec capability descriptor
+ */
+struct snd_compr_codec_caps {
+	__u32 codec;
+	__u32 num_descriptors;
+	struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS];
+};
+
+/**
+ * compress path ioctl definitions
+ * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
+ * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec
+ * SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters
+ * Note: only codec params can be changed runtime and stream params cant be
+ * SNDRV_COMPRESS_GET_PARAMS: Query codec params
+ * SNDRV_COMPRESS_TSTAMP: get the current timestamp value
+ * SNDRV_COMPRESS_AVAIL: get the current buffer avail value.
+ * This also queries the tstamp properties
+ * SNDRV_COMPRESS_PAUSE: Pause the running stream
+ * SNDRV_COMPRESS_RESUME: resume a paused stream
+ * SNDRV_COMPRESS_START: Start a stream
+ * SNDRV_COMPRESS_STOP: stop a running stream, discarding ring buffer content
+ * and the buffers currently with DSP
+ * SNDRV_COMPRESS_DRAIN: Play till end of buffers and stop after that
+ * SNDRV_COMPRESS_IOCTL_VERSION: Query the API version
+ */
+#define SNDRV_COMPRESS_IOCTL_VERSION	_IOR('C', 0x00, int)
+#define SNDRV_COMPRESS_GET_CAPS		_IOWR('C', 0x10, struct snd_compr_caps)
+#define SNDRV_COMPRESS_GET_CODEC_CAPS	_IOWR('C', 0x11,\
+						struct snd_compr_codec_caps)
+#define SNDRV_COMPRESS_SET_PARAMS	_IOW('C', 0x12, struct snd_compr_params)
+#define SNDRV_COMPRESS_GET_PARAMS	_IOR('C', 0x13, struct snd_codec)
+#define SNDRV_COMPRESS_TSTAMP		_IOR('C', 0x20, struct snd_compr_tstamp)
+#define SNDRV_COMPRESS_AVAIL		_IOR('C', 0x21, struct snd_compr_avail)
+#define SNDRV_COMPRESS_PAUSE		_IO('C', 0x30)
+#define SNDRV_COMPRESS_RESUME		_IO('C', 0x31)
+#define SNDRV_COMPRESS_START		_IO('C', 0x32)
+#define SNDRV_COMPRESS_STOP		_IO('C', 0x33)
+#define SNDRV_COMPRESS_DRAIN		_IO('C', 0x34)
+/*
+ * TODO
+ * 1. add mmap support
+ *
+ */
+#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */
+#endif
-- 
cgit v1.2.3-70-g09d2


From de21eee9608f67a8a648bbd1a5358f819644501e Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@linux.intel.com>
Date: Thu, 29 Dec 2011 18:42:31 +0530
Subject: ALSA: export compress headers

Export compress_offload.h and compress_params.h for userland to use

Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/Kbuild | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index 802947f6091..6df30ed1581 100644
--- a/include/sound/Kbuild
+++ b/include/sound/Kbuild
@@ -6,3 +6,5 @@ header-y += hdsp.h
 header-y += hdspm.h
 header-y += sb16_csp.h
 header-y += sfnt_info.h
+header-y += compress_params.h
+header-y += compress_offload.h
-- 
cgit v1.2.3-70-g09d2


From 36ae1a96c4dcb0f6581d595cc5d43cf3a7e648c7 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 6 Jan 2012 17:12:45 -0800
Subject: ASoC: Dynamically allocate the rtd device for a non-empty release()

The device model needs a release() function so it can free devices when
they become dereferenced.  Do that for rtds.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h  |  6 +++---
 sound/soc/soc-core.c | 44 +++++++++++++++++++++++++-------------------
 sound/soc/soc-dapm.c |  3 +--
 3 files changed, 29 insertions(+), 24 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index f75d1ccc5c5..0992dff5595 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -847,7 +847,7 @@ struct snd_soc_card {
 
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
 struct snd_soc_pcm_runtime {
-	struct device dev;
+	struct device *dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
 	struct mutex pcm_mutex;
@@ -933,12 +933,12 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo
 static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd,
 		void *data)
 {
-	dev_set_drvdata(&rtd->dev, data);
+	dev_set_drvdata(rtd->dev, data);
 }
 
 static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
 {
-	return dev_get_drvdata(&rtd->dev);
+	return dev_get_drvdata(rtd->dev);
 }
 
 static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index acbb96005a6..3986520b467 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -169,8 +169,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
 static ssize_t codec_reg_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 
 	return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0);
 }
@@ -180,8 +179,7 @@ static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
 static ssize_t pmdown_time_show(struct device *dev,
 				struct device_attribute *attr, char *buf)
 {
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 
 	return sprintf(buf, "%ld\n", rtd->pmdown_time);
 }
@@ -190,8 +188,7 @@ static ssize_t pmdown_time_set(struct device *dev,
 			       struct device_attribute *attr,
 			       const char *buf, size_t count)
 {
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	int ret;
 
 	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
@@ -884,9 +881,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
 
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
-		device_remove_file(&rtd->dev, &dev_attr_pmdown_time);
-		device_remove_file(&rtd->dev, &dev_attr_codec_reg);
-		device_unregister(&rtd->dev);
+		device_remove_file(rtd->dev, &dev_attr_pmdown_time);
+		device_remove_file(rtd->dev, &dev_attr_codec_reg);
+		device_unregister(rtd->dev);
 		rtd->dev_registered = 0;
 	}
 
@@ -1061,7 +1058,10 @@ err_probe:
 	return ret;
 }
 
-static void rtd_release(struct device *dev) {}
+static void rtd_release(struct device *dev)
+{
+	kfree(dev);
+}
 
 static int soc_post_component_init(struct snd_soc_card *card,
 				   struct snd_soc_codec *codec,
@@ -1104,11 +1104,17 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
 	/* register the rtd device */
 	rtd->codec = codec;
-	rtd->dev.parent = card->dev;
-	rtd->dev.release = rtd_release;
-	rtd->dev.init_name = name;
+
+	rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL);
+	if (!rtd->dev)
+		return -ENOMEM;
+	device_initialize(rtd->dev);
+	rtd->dev->parent = card->dev;
+	rtd->dev->release = rtd_release;
+	rtd->dev->init_name = name;
+	dev_set_drvdata(rtd->dev, rtd);
 	mutex_init(&rtd->pcm_mutex);
-	ret = device_register(&rtd->dev);
+	ret = device_add(rtd->dev);
 	if (ret < 0) {
 		dev_err(card->dev,
 			"asoc: failed to register runtime device: %d\n", ret);
@@ -1117,14 +1123,14 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	rtd->dev_registered = 1;
 
 	/* add DAPM sysfs entries for this codec */
-	ret = snd_soc_dapm_sys_add(&rtd->dev);
+	ret = snd_soc_dapm_sys_add(rtd->dev);
 	if (ret < 0)
 		dev_err(codec->dev,
 			"asoc: failed to add codec dapm sysfs entries: %d\n",
 			ret);
 
 	/* add codec sysfs entries */
-	ret = device_create_file(&rtd->dev, &dev_attr_codec_reg);
+	ret = device_create_file(rtd->dev, &dev_attr_codec_reg);
 	if (ret < 0)
 		dev_err(codec->dev,
 			"asoc: failed to add codec sysfs files: %d\n", ret);
@@ -1213,7 +1219,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 	if (ret)
 		return ret;
 
-	ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time);
+	ret = device_create_file(rtd->dev, &dev_attr_pmdown_time);
 	if (ret < 0)
 		printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n");
 
@@ -1311,8 +1317,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
 
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
-		device_remove_file(&rtd->dev, &dev_attr_codec_reg);
-		device_unregister(&rtd->dev);
+		device_remove_file(rtd->dev, &dev_attr_codec_reg);
+		device_del(rtd->dev);
 		rtd->dev_registered = 0;
 	}
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index e174d0811da..3ad1f59b802 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1738,8 +1738,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 static ssize_t dapm_widget_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(dev, struct snd_soc_pcm_runtime, dev);
+	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	struct snd_soc_codec *codec =rtd->codec;
 	struct snd_soc_dapm_widget *w;
 	int count = 0;
-- 
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