Merge branch 'for-2.6.38' into for-2.6.39

16 files changed, 326 insertions, 340 deletions

diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt
index 37ba3a72cb76..bce23a4a7875 100644
--- a/Documentation/sound/alsa/soc/codec.txt
+++ b/Documentation/sound/alsa/soc/codec.txt
@@ -27,42 +27,38 @@ ASoC Codec driver breakdown
 
 1 - Codec DAI and PCM configuration
 -----------------------------------
-Each codec driver must have a struct snd_soc_codec_dai to define its DAI and
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
 PCM capabilities and operations. This struct is exported so that it can be
 registered with the core by your machine driver.
 
 e.g.
 
-struct snd_soc_codec_dai wm8731_dai = {
-	.name = "WM8731",
-	/* playback capabilities */
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+	.name = "wm8731-hifi",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8731_RATES,
 		.formats = WM8731_FORMATS,},
-	/* capture capabilities */
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8731_RATES,
 		.formats = WM8731_FORMATS,},
-	/* pcm operations - see section 4 below */
-	.ops = {
-		.prepare = wm8731_pcm_prepare,
-		.hw_params = wm8731_hw_params,
-		.shutdown = wm8731_shutdown,
-	},
-	/* DAI operations - see DAI.txt */
-	.dai_ops = {
-		.digital_mute = wm8731_mute,
-		.set_sysclk = wm8731_set_dai_sysclk,
-		.set_fmt = wm8731_set_dai_fmt,
-	}
+	.ops = &wm8731_dai_ops,
+	.symmetric_rates = 1,
 };
-EXPORT_SYMBOL_GPL(wm8731_dai);
 
 
 2 - Codec control IO
@@ -186,13 +182,14 @@ when the mute is applied or freed.
 
 i.e.
 
-static int wm8974_mute(struct snd_soc_codec *codec,
-	struct snd_soc_codec_dai *dai, int mute)
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
 {
-	u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf;
-	if(mute)
-		wm8974_write(codec, WM8974_DAC, mute_reg | 0x40);
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+	if (mute)
+		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
 	else
-		wm8974_write(codec, WM8974_DAC, mute_reg);
+		snd_soc_write(codec, WM8974_DAC, mute_reg);
 	return 0;
 }
diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt
index 2524c75557df..3e2ec9cbf397 100644
--- a/Documentation/sound/alsa/soc/machine.txt
+++ b/Documentation/sound/alsa/soc/machine.txt
@@ -12,6 +12,8 @@ the following struct:-
 struct snd_soc_card {
 	char *name;
 
+	...
+
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
 
@@ -22,12 +24,13 @@ struct snd_soc_card {
 	int (*resume_pre)(struct platform_device *pdev);
 	int (*resume_post)(struct platform_device *pdev);
 
-	/* machine stream operations */
-	struct snd_soc_ops *ops;
+	...
 
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;
 	int num_links;
+
+	...
 };
 
 probe()/remove()
@@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs
 and DMA is suspended and resumed. Optional.
 
 
-Machine operations
-------------------
-The machine specific audio operations can be set here. Again this is optional.
-
-
 Machine DAI Configuration
 -------------------------
 The machine DAI configuration glues all the codec and CPU DAIs together. It can
@@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
 static struct snd_soc_dai_link corgi_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &wm8731_dai,
+	.cpu_dai_name = "pxa-is2-dai",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8713-codec.0-001a",
 	.init = corgi_wm8731_init,
 	.ops = &corgi_ops,
 };
@@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = {
 };
 
 
-Machine Audio Subsystem
------------------------
-
-The machine soc device glues the platform, machine and codec driver together.
-Private data can also be set here. e.g.
-
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
-	.i2c_address = 0x1b,
-};
-
-/* corgi audio subsystem */
-static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_corgi,
-	.platform = &pxa2xx_soc_platform,
-	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &corgi_wm8731_setup,
-};
-
-
 Machine Power Map
 -----------------
 
diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt
index 06d835987c6a..d57efad37e0a 100644
--- a/Documentation/sound/alsa/soc/platform.txt
+++ b/Documentation/sound/alsa/soc/platform.txt
@@ -20,9 +20,10 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-The platform driver exports its DMA functionality via struct snd_soc_platform:-
+The platform driver exports its DMA functionality via struct
+snd_soc_platform_driver:-
 
-struct snd_soc_platform {
+struct snd_soc_platform_driver {
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -34,6 +35,13 @@ struct snd_soc_platform {
 	int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *);
 	void (*pcm_free)(struct snd_pcm *);
 
+	/*
+	 * For platform caused delay reporting.
+	 * Optional.
+	 */
+	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+
 	/* platform stream ops */
 	struct snd_pcm_ops *pcm_ops;
 };
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index fa42be529a73..61e36efbf279 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -168,6 +168,9 @@ config SND_SOC_L3
 config SND_SOC_DA7210
         tristate
 
+config SND_SOC_DMIC
+	tristate
+
 config SND_SOC_MAX98088
        tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 76304d478912..333910a9f8fb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
+snd-soc-dmic-objs := dmic.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-pcm3008-objs := pcm3008.o
@@ -93,6 +94,7 @@ obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 3a582caa6ef9..8b51245f2318 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -106,6 +106,21 @@
 #define CS4270_MUTE_DAC_A	0x01
 #define CS4270_MUTE_DAC_B	0x02
 
+/* Power-on default values for the registers
+ *
+ * This array contains the power-on default values of the registers, with the
+ * exception of the "CHIPID" register (01h).  The lower four bits of that
+ * register contain the hardware revision, so it is treated as volatile.
+ *
+ * Also note that on the CS4270, the first readable register is 1, but ASoC
+ * assumes the first register is 0.  Therfore, the array must have an entry for
+ * register 0, but we use cs4270_reg_is_readable() to tell ASoC that it can't
+ * be read.
+ */
+static const u8 cs4270_default_reg_cache[CS4270_LASTREG + 1] = {
+	0x00, 0x00, 0x00, 0x30, 0x00, 0x60, 0x20, 0x00, 0x00
+};
+
 static const char *supply_names[] = {
 	"va", "vd", "vlc"
 };
@@ -178,6 +193,20 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
 /* The number of MCLK/LRCK ratios supported by the CS4270 */
 #define NUM_MCLK_RATIOS		ARRAY_SIZE(cs4270_mode_ratios)
 
+static int cs4270_reg_is_readable(unsigned int reg)
+{
+	return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG);
+}
+
+static int cs4270_reg_is_volatile(unsigned int reg)
+{
+	/* Unreadable registers are considered volatile */
+	if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
+		return 1;
+
+	return reg == CS4270_CHIPID;
+}
+
 /**
  * cs4270_set_dai_sysclk - determine the CS4270 samples rates.
  * @codec_dai: the codec DAI
@@ -263,97 +292,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 /**
- * cs4270_fill_cache - pre-fill the CS4270 register cache.
- * @codec: the codec for this CS4270
- *
- * This function fills in the CS4270 register cache by reading the register
- * values from the hardware.
- *
- * This CS4270 registers are cached to avoid excessive I2C I/O operations.
- * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherency problem.
- *
- * We use the auto-increment feature of the CS4270 to read all registers in
- * one shot.
- */
-static int cs4270_fill_cache(struct snd_soc_codec *codec)
-{
-	u8 *cache = codec->reg_cache;
-	struct i2c_client *i2c_client = codec->control_data;
-	s32 length;
-
-	length = i2c_smbus_read_i2c_block_data(i2c_client,
-		CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
-
-	if (length != CS4270_NUMREGS) {
-		dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
-		       i2c_client->addr);
-		return -EIO;
-	}
-
-	return 0;
-}
-
-/**
- * cs4270_read_reg_cache - read from the CS4270 register cache.
- * @codec: the codec for this CS4270
- * @reg: the register to read
- *
- * This function returns the value for a given register.  It reads only from
- * the register cache, not the hardware itself.
- *
- * This CS4270 registers are cached to avoid excessive I2C I/O operations.
- * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherency problem.
- */
-static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
-	unsigned int reg)
-{
-	u8 *cache = codec->reg_cache;
-
-	if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
-		return -EIO;
-
-	return cache[reg - CS4270_FIRSTREG];
-}
-
-/**
- * cs4270_i2c_write - write to a CS4270 register via the I2C bus.
- * @codec: the codec for this CS4270
- * @reg: the register to write
- * @value: the value to write to the register
- *
- * This function writes the given value to the given CS4270 register, and
- * also updates the register cache.
- *
- * Note that we don't use the hw_write function pointer of snd_soc_codec.
- * That's because it's too clunky: the hw_write_t prototype does not match
- * i2c_smbus_write_byte_data(), and it's just another layer of overhead.
- */
-static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
-			    unsigned int value)
-{
-	u8 *cache = codec->reg_cache;
-
-	if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG))
-		return -EIO;
-
-	/* Only perform an I2C operation if the new value is different */
-	if (cache[reg - CS4270_FIRSTREG] != value) {
-		struct i2c_client *client = codec->control_data;
-		if (i2c_smbus_write_byte_data(client, reg, value)) {
-			dev_err(codec->dev, "i2c write failed\n");
-			return -EIO;
-		}
-
-		/* We've written to the hardware, so update the cache */
-		cache[reg - CS4270_FIRSTREG] = value;
-	}
-
-	return 0;
-}
-
-/**
  * cs4270_hw_params - program the CS4270 with the given hardware parameters.
  * @substream: the audio stream
  * @params: the hardware parameters to set
@@ -550,15 +488,16 @@ static struct snd_soc_dai_driver cs4270_dai = {
 static int cs4270_probe(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	int i, ret, reg;
+	int i, ret;
 
 	codec->control_data = cs4270->control_data;
 
-	/* The I2C interface is set up, so pre-fill our register cache */
-
-	ret = cs4270_fill_cache(codec);
+	/* Tell ASoC what kind of I/O to use to read the registers.  ASoC will
+	 * then do the I2C transactions itself.
+	 */
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs4270->control_type);
 	if (ret < 0) {
-		dev_err(codec->dev, "failed to fill register cache\n");
+		dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
 		return ret;
 	}
 
@@ -567,10 +506,7 @@ static int cs4270_probe(struct snd_soc_codec *codec)
 	 * this feature disabled by default.  An application (e.g. alsactl) can
 	 * re-enabled it by using the controls.
 	 */
-
-	reg = cs4270_read_reg_cache(codec, CS4270_MUTE);
-	reg &= ~CS4270_MUTE_AUTO;
-	ret = cs4270_i2c_write(codec, CS4270_MUTE, reg);
+	ret = snd_soc_update_bits(codec, CS4270_MUTE, CS4270_MUTE_AUTO, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
@@ -581,10 +517,8 @@ static int cs4270_probe(struct snd_soc_codec *codec)
 	 * playback has started.  An application (e.g. alsactl) can
 	 * re-enabled it by using the controls.
 	 */
-
-	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
-	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
-	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+	ret = snd_soc_update_bits(codec, CS4270_TRANS,
+		CS4270_TRANS_SOFT | CS4270_TRANS_ZERO, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
@@ -707,15 +641,16 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
  * Assign this variable to the codec_dev field of the machine driver's
  * snd_soc_device structure.
  */
-static struct snd_soc_codec_driver soc_codec_device_cs4270 = {
-	.probe =	cs4270_probe,
-	.remove =	cs4270_remove,
-	.suspend =	cs4270_soc_suspend,
-	.resume =	cs4270_soc_resume,
-	.read = cs4270_read_reg_cache,
-	.write = cs4270_i2c_write,
-	.reg_cache_size = CS4270_NUMREGS,
-	.reg_word_size = sizeof(u8),
+static const struct snd_soc_codec_driver soc_codec_device_cs4270 = {
+	.probe =		cs4270_probe,
+	.remove =		cs4270_remove,
+	.suspend =		cs4270_soc_suspend,
+	.resume =		cs4270_soc_resume,
+	.volatile_register =	cs4270_reg_is_volatile,
+	.readable_register =	cs4270_reg_is_readable,
+	.reg_cache_size =	CS4270_LASTREG + 1,
+	.reg_word_size =	sizeof(u8),
+	.reg_cache_default =	cs4270_default_reg_cache,
 };
 
 /**
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
new file mode 100644
index 000000000000..57e9dac88d38
--- /dev/null
+++ b/sound/soc/codecs/dmic.c
@@ -0,0 +1,81 @@
+/*
+ * dmic.c  --  SoC audio for Generic Digital MICs
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+static struct snd_soc_dai_driver dmic_dai = {
+	.name = "dmic-hifi",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 8,
+		.rates = SNDRV_PCM_RATE_CONTINUOUS,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE
+			| SNDRV_PCM_FMTBIT_S24_LE
+			| SNDRV_PCM_FMTBIT_S16_LE,
+	},
+};
+
+static struct snd_soc_codec_driver soc_dmic = {};
+
+static int __devinit dmic_dev_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev,
+			&soc_dmic, &dmic_dai, 1);
+}
+
+static int __devexit dmic_dev_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+MODULE_ALIAS("platform:dmic-codec");
+
+static struct platform_driver dmic_driver = {
+	.driver = {
+		.name = "dmic-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = dmic_dev_probe,
+	.remove = __devexit_p(dmic_dev_remove),
+};
+
+static int __init dmic_init(void)
+{
+	return platform_driver_register(&dmic_driver);
+}
+module_init(dmic_init);
+
+static void __exit dmic_exit(void)
+{
+	platform_driver_unregister(&dmic_driver);
+}
+module_exit(dmic_exit);
+
+MODULE_DESCRIPTION("Generic DMIC driver");
+MODULE_AUTHOR("Liam Girdwood <lrg@slimlogic.co.uk>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 776ac80cc1a8..71d7be8ac488 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -42,14 +42,15 @@
 #include <sound/tlv320dac33-plat.h>
 #include "tlv320dac33.h"
 
-#define DAC33_BUFFER_SIZE_BYTES		24576	/* bytes, 12288 16 bit words,
-						 * 6144 stereo */
-#define DAC33_BUFFER_SIZE_SAMPLES	6144
-
-#define NSAMPLE_MAX		5700
-
-#define MODE7_LTHR		10
-#define MODE7_UTHR		(DAC33_BUFFER_SIZE_SAMPLES - 10)
+/*
+ * The internal FIFO is 24576 bytes long
+ * It can be configured to hold 16bit or 24bit samples
+ * In 16bit configuration the FIFO can hold 6144 stereo samples
+ * In 24bit configuration the FIFO can hold 4096 stereo samples
+ */
+#define DAC33_FIFO_SIZE_16BIT	6144
+#define DAC33_FIFO_SIZE_24BIT	4096
+#define DAC33_MODE7_MARGIN	10	/* Safety margin for FIFO in Mode7 */
 
 #define BURST_BASEFREQ_HZ	49152000
 
@@ -99,16 +100,11 @@ struct tlv320dac33_priv {
 	unsigned int refclk;
 
 	unsigned int alarm_threshold;	/* set to be half of LATENCY_TIME_MS */
-	unsigned int nsample_min;	/* nsample should not be lower than
-					 * this */
-	unsigned int nsample_max;	/* nsample should not be higher than
-					 * this */
 	enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
+	unsigned int fifo_size;		/* Size of the FIFO in samples */
 	unsigned int nsample;		/* burst read amount from host */
 	int mode1_latency;		/* latency caused by the i2c writes in
 					 * us */
-	int auto_fifo_config; 		/* Configure the FIFO based on the
-					 * period size */
 	u8 burst_bclkdiv;		/* BCLK divider value in burst mode */
 	unsigned int burst_rate;	/* Interface speed in Burst modes */
 
@@ -302,7 +298,6 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 	if (unlikely(!dac33->chip_power))
 		return;
 
-	/* 44-46: DAC Control Registers */
 	/* A : DAC sample rate Fsref/1.5 */
 	dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0));
 	/* B : DAC src=normal, not muted */
@@ -325,6 +320,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 		    dac33_read_reg_cache(codec, DAC33_LINEL_TO_LLO_VOL));
 	dac33_write(codec, DAC33_LINER_TO_RLO_VOL,
 		    dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL));
+
+	dac33_write(codec, DAC33_OUT_AMP_CTRL,
+		    dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL));
+
 }
 
 static inline int dac33_read_id(struct snd_soc_codec *codec)
@@ -436,73 +435,6 @@ static int dac33_playback_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
-			 struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-
-	ucontrol->value.integer.value[0] = dac33->nsample;
-
-	return 0;
-}
-
-static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
-			 struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
-
-	if (dac33->nsample == ucontrol->value.integer.value[0])
-		return 0;
-
-	if (ucontrol->value.integer.value[0] < dac33->nsample_min ||
-	    ucontrol->value.integer.value[0] > dac33->nsample_max) {
-		ret = -EINVAL;
-	} else {
-		dac33->nsample = ucontrol->value.integer.value[0];
-		/* Re calculate the burst time */
-		dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
-						      dac33->nsample);
-	}
-
-	return ret;
-}
-
-static int dac33_get_uthr(struct snd_kcontrol *kcontrol,
-			 struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-
-	ucontrol->value.integer.value[0] = dac33->uthr;
-
-	return 0;
-}
-
-static int dac33_set_uthr(struct snd_kcontrol *kcontrol,
-			 struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
-
-	if (dac33->substream)
-		return -EBUSY;
-
-	if (dac33->uthr == ucontrol->value.integer.value[0])
-		return 0;
-
-	if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) ||
-	    ucontrol->value.integer.value[0] > MODE7_UTHR)
-		ret = -EINVAL;
-	else
-		dac33->uthr = ucontrol->value.integer.value[0];
-
-	return ret;
-}
-
 static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
@@ -587,13 +519,6 @@ static const struct snd_kcontrol_new dac33_mode_snd_controls[] = {
 		 dac33_get_fifo_mode, dac33_set_fifo_mode),
 };
 
-static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = {
-	SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
-		dac33_get_nsample, dac33_set_nsample),
-	SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0,
-		 dac33_get_uthr, dac33_set_uthr),
-};
-
 /* Analog bypass */
 static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
 	SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1);
@@ -601,6 +526,25 @@ static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
 static const struct snd_kcontrol_new dac33_dapm_abypassr_control =
 	SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1);
 
+/* LOP L/R invert selection */
+static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"};
+
+static const struct soc_enum dac33_left_lom_enum =
+	SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3,
+			ARRAY_SIZE(dac33_lr_lom_texts),
+			dac33_lr_lom_texts);
+
+static const struct snd_kcontrol_new dac33_dapm_left_lom_control =
+SOC_DAPM_ENUM("Route", dac33_left_lom_enum);
+
+static const struct soc_enum dac33_right_lom_enum =
+	SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2,
+			ARRAY_SIZE(dac33_lr_lom_texts),
+			dac33_lr_lom_texts);
+
+static const struct snd_kcontrol_new dac33_dapm_right_lom_control =
+SOC_DAPM_ENUM("Route", dac33_right_lom_enum);
+
 static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("LEFT_LO"),
 	SND_SOC_DAPM_OUTPUT("RIGHT_LO"),
@@ -617,6 +561,18 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
 	SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0,
 				&dac33_dapm_abypassr_control),
 
+	SND_SOC_DAPM_MUX("Left LOM Inverted From", SND_SOC_NOPM, 0, 0,
+		&dac33_dapm_left_lom_control),
+	SND_SOC_DAPM_MUX("Right LOM Inverted From", SND_SOC_NOPM, 0, 0,
+		&dac33_dapm_right_lom_control),
+	/*
+	 * For DAPM path, when only the anlog bypass path is enabled, and the
+	 * LOP inverted from the corresponding DAC side.
+	 * This is needed, so we can attach the DAC power supply in this case.
+	 */
+	SND_SOC_DAPM_PGA("Left Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+
 	SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amplifier",
 			 DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0),
 	SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amplifier",
@@ -639,11 +595,22 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Output Left Amplifier", NULL, "DACL"},
 	{"Output Right Amplifier", NULL, "DACR"},
 
-	{"Output Left Amplifier", NULL, "Analog Left Bypass"},
-	{"Output Right Amplifier", NULL, "Analog Right Bypass"},
+	{"Left Bypass PGA", NULL, "Analog Left Bypass"},
+	{"Right Bypass PGA", NULL, "Analog Right Bypass"},
 
-	{"Output Left Amplifier", NULL, "Left DAC Power"},
-	{"Output Right Amplifier", NULL, "Right DAC Power"},
+	{"Left LOM Inverted From", "DAC", "Left Bypass PGA"},
+	{"Right LOM Inverted From", "DAC", "Right Bypass PGA"},
+	{"Left LOM Inverted From", "LOP", "Analog Left Bypass"},
+	{"Right LOM Inverted From", "LOP", "Analog Right Bypass"},
+
+	{"Output Left Amplifier", NULL, "Left LOM Inverted From"},
+	{"Output Right Amplifier", NULL, "Right LOM Inverted From"},
+
+	{"DACL", NULL, "Left DAC Power"},
+	{"DACR", NULL, "Right DAC Power"},
+
+	{"Left Bypass PGA", NULL, "Left DAC Power"},
+	{"Right Bypass PGA", NULL, "Right DAC Power"},
 
 	/* output */
 	{"LEFT_LO", NULL, "Output Left Amplifier"},
@@ -732,7 +699,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 		spin_unlock_irq(&dac33->lock);
 
 		dac33_write16(codec, DAC33_PREFILL_MSB,
-				DAC33_THRREG(MODE7_LTHR));
+				DAC33_THRREG(DAC33_MODE7_MARGIN));
 
 		/* Enable Upper Threshold IRQ */
 		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MUT);
@@ -842,6 +809,8 @@ static int dac33_startup(struct snd_pcm_substream *substream,
 	/* Stream started, save the substream pointer */
 	dac33->substream = substream;
 
+	snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
+
 	return 0;
 }
 
@@ -853,18 +822,17 @@ static void dac33_shutdown(struct snd_pcm_substream *substream,
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	dac33->substream = NULL;
-
-	/* Reset the nSample restrictions */
-	dac33->nsample_min = 0;
-	dac33->nsample_max = NSAMPLE_MAX;
 }
 
+#define CALC_BURST_RATE(bclkdiv, bclk_per_sample) \
+	(BURST_BASEFREQ_HZ / bclkdiv / bclk_per_sample)
 static int dac33_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
+	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
 	/* Check parameters for validity */
 	switch (params_rate(params)) {
@@ -879,6 +847,12 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
+		dac33->fifo_size = DAC33_FIFO_SIZE_16BIT;
+		dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 32);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		dac33->fifo_size = DAC33_FIFO_SIZE_24BIT;
+		dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 64);
 		break;
 	default:
 		dev_err(codec->dev, "unsupported format %d\n",
@@ -933,6 +907,9 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16);
 		fifoctrl_a |= DAC33_WIDTH;
 		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		aictrl_a |= (DAC33_NCYCL_32 | DAC33_WLEN_24);
+		break;
 	default:
 		dev_err(codec->dev, "unsupported format %d\n",
 			substream->runtime->format);
@@ -1067,7 +1044,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C,
 							dac33->burst_bclkdiv);
 	else
-		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+		if (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE)
+			dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+		else
+			dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 16);
 
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
@@ -1080,7 +1060,8 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		 * at the bottom, and also at the top of the FIFO
 		 */
 		dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr));
-		dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
+		dac33_write16(codec, DAC33_LTHR_MSB,
+			      DAC33_THRREG(DAC33_MODE7_MARGIN));
 		break;
 	default:
 		break;
@@ -1109,42 +1090,21 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 		/* Number of samples under i2c latency */
 		dac33->alarm_threshold = US_TO_SAMPLES(rate,
 						dac33->mode1_latency);
-		nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
-				dac33->alarm_threshold;
-
-		if (dac33->auto_fifo_config) {
-			if (period_size <= dac33->alarm_threshold)
-				/*
-				 * Configure nSamaple to number of periods,
-				 * which covers the latency requironment.
-				 */
-				dac33->nsample = period_size *
-				       ((dac33->alarm_threshold / period_size) +
-				       (dac33->alarm_threshold % period_size ?
-				       1 : 0));
-			else if (period_size > nsample_limit)
-				dac33->nsample = nsample_limit;
-			else
-				dac33->nsample = period_size;
-		} else {
-			/* nSample time shall not be shorter than i2c latency */
-			dac33->nsample_min = dac33->alarm_threshold;
+		nsample_limit = dac33->fifo_size - dac33->alarm_threshold;
+
+		if (period_size <= dac33->alarm_threshold)
 			/*
-			 * nSample should not be bigger than alsa buffer minus
-			 * size of one period to avoid overruns
+			 * Configure nSamaple to number of periods,
+			 * which covers the latency requironment.
 			 */
-			dac33->nsample_max = substream->runtime->buffer_size -
-						period_size;
-
-			if (dac33->nsample_max > nsample_limit)
-				dac33->nsample_max = nsample_limit;
-
-			/* Correct the nSample if it is outside of the ranges */
-			if (dac33->nsample < dac33->nsample_min)
-				dac33->nsample = dac33->nsample_min;
-			if (dac33->nsample > dac33->nsample_max)
-				dac33->nsample = dac33->nsample_max;
-		}
+			dac33->nsample = period_size *
+				((dac33->alarm_threshold / period_size) +
+				(dac33->alarm_threshold % period_size ?
+				1 : 0));
+		else if (period_size > nsample_limit)
+			dac33->nsample = nsample_limit;
+		else
+			dac33->nsample = period_size;
 
 		dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
 						      dac33->nsample);
@@ -1152,19 +1112,16 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
 		dac33->t_stamp2 = 0;
 		break;
 	case DAC33_FIFO_MODE7:
-		if (dac33->auto_fifo_config) {
-			dac33->uthr = UTHR_FROM_PERIOD_SIZE(
-					period_size,
-					rate,
-					dac33->burst_rate) + 9;
-			if (dac33->uthr > MODE7_UTHR)
-				dac33->uthr = MODE7_UTHR;
-			if (dac33->uthr < (MODE7_LTHR + 10))
-				dac33->uthr = (MODE7_LTHR + 10);
-		}
+		dac33->uthr = UTHR_FROM_PERIOD_SIZE(period_size, rate,
+						    dac33->burst_rate) + 9;
+		if (dac33->uthr > (dac33->fifo_size - DAC33_MODE7_MARGIN))
+			dac33->uthr = dac33->fifo_size - DAC33_MODE7_MARGIN;
+		if (dac33->uthr < (DAC33_MODE7_MARGIN + 10))
+			dac33->uthr = (DAC33_MODE7_MARGIN + 10);
+
 		dac33->mode7_us_to_lthr =
 				SAMPLES_TO_US(substream->runtime->rate,
-					dac33->uthr - MODE7_LTHR + 1);
+					dac33->uthr - DAC33_MODE7_MARGIN + 1);
 		dac33->t_stamp1 = 0;
 		break;
 	default:
@@ -1282,8 +1239,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
 			samples += (samples_in - samples_out);
 
 			if (likely(samples > 0))
-				delay = samples > DAC33_BUFFER_SIZE_SAMPLES ?
-					DAC33_BUFFER_SIZE_SAMPLES : samples;
+				delay = samples > dac33->fifo_size ?
+					dac33->fifo_size : samples;
 			else
 				delay = 0;
 		}
@@ -1335,7 +1292,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
 			samples_in = US_TO_SAMPLES(
 					dac33->burst_rate,
 					time_delta);
-			delay = MODE7_LTHR + samples_in - samples_out;
+			delay = DAC33_MODE7_MARGIN + samples_in - samples_out;
 
 			if (unlikely(delay > uthr))
 				delay = uthr;
@@ -1486,14 +1443,10 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
 	snd_soc_add_controls(codec, dac33_snd_controls,
 			     ARRAY_SIZE(dac33_snd_controls));
 	/* Only add the FIFO controls, if we have valid IRQ number */
-	if (dac33->irq >= 0) {
+	if (dac33->irq >= 0)
 		snd_soc_add_controls(codec, dac33_mode_snd_controls,
 				     ARRAY_SIZE(dac33_mode_snd_controls));
-		/* FIFO usage controls only, if autoio config is not selected */
-		if (!dac33->auto_fifo_config)
-			snd_soc_add_controls(codec, dac33_fifo_snd_controls,
-					ARRAY_SIZE(dac33_fifo_snd_controls));
-	}
+
 	dac33_add_widgets(codec);
 
 err_power:
@@ -1542,7 +1495,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
 
 #define DAC33_RATES	(SNDRV_PCM_RATE_44100 | \
 			 SNDRV_PCM_RATE_48000)
-#define DAC33_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+#define DAC33_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static struct snd_soc_dai_ops dac33_dai_ops = {
 	.startup	= dac33_startup,
@@ -1590,17 +1543,11 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
 
 	dac33->power_gpio = pdata->power_gpio;
 	dac33->burst_bclkdiv = pdata->burst_bclkdiv;
-	/* Pre calculate the burst rate */
-	dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32;
 	dac33->keep_bclk = pdata->keep_bclk;
-	dac33->auto_fifo_config = pdata->auto_fifo_config;
 	dac33->mode1_latency = pdata->mode1_latency;
 	if (!dac33->mode1_latency)
 		dac33->mode1_latency = 10000; /* 10ms */
 	dac33->irq = client->irq;
-	dac33->nsample = NSAMPLE_MAX;
-	dac33->nsample_max = NSAMPLE_MAX;
-	dac33->uthr = MODE7_UTHR;
 	/* Disable FIFO use by default */
 	dac33->fifo_mode = DAC33_FIFO_BYPASS;
 
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 0a99f313e218..1f1ac8110bef 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -339,7 +339,6 @@ EXPORT_SYMBOL_GPL(tpa6130a2_stereo_enable);
 int tpa6130a2_add_controls(struct snd_soc_codec *codec)
 {
 	struct	tpa6130a2_data *data;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	if (tpa6130a2_client == NULL)
 		return -ENODEV;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 2f68f5949a63..4bbf1b15a493 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1138,19 +1138,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
 			SND_SOC_NOPM, 0, 0, &hsr_mux_controls),
 
 	/* Analog playback drivers */
-	SND_SOC_DAPM_PGA_E("Handsfree Left Driver",
+	SND_SOC_DAPM_OUT_DRV_E("Handsfree Left Driver",
 			TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
 			pga_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_PGA_E("Handsfree Right Driver",
+	SND_SOC_DAPM_OUT_DRV_E("Handsfree Right Driver",
 			TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
 			pga_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_PGA_E("Headset Left Driver",
+	SND_SOC_DAPM_OUT_DRV_E("Headset Left Driver",
 			TWL6040_REG_HSLCTL, 2, 0, NULL, 0,
 			pga_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-	SND_SOC_DAPM_PGA_E("Headset Right Driver",
+	SND_SOC_DAPM_OUT_DRV_E("Headset Right Driver",
 			TWL6040_REG_HSRCTL, 2, 0, NULL, 0,
 			pga_event,
 			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 96e0dc09f203..ac210ccebd4b 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -30,7 +30,7 @@
 
 #include "wm8995.h"
 
-static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] __devinitconst = {
+static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = {
 	[0]     = 0x8995, [5]     = 0x0100, [16]    = 0x000b, [17]    = 0x000b,
 	[24]    = 0x02c0, [25]    = 0x02c0, [26]    = 0x02c0, [27]    = 0x02c0,
 	[28]    = 0x000f, [32]    = 0x0005, [33]    = 0x0005, [40]    = 0x0003,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 7e84f24b9a88..d203f4da18a0 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -102,6 +102,17 @@ static const int omap24xx_dma_reqs[][2] = {
 static const int omap24xx_dma_reqs[][2] = {};
 #endif
 
+#if defined(CONFIG_ARCH_OMAP4)
+static const int omap44xx_dma_reqs[][2] = {
+	{ OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX },
+	{ OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX },
+	{ OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX },
+	{ OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX },
+};
+#else
+static const int omap44xx_dma_reqs[][2] = {};
+#endif
+
 #if defined(CONFIG_ARCH_OMAP2420)
 static const unsigned long omap2420_mcbsp_port[][2] = {
 	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
@@ -147,6 +158,21 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
 static const unsigned long omap34xx_mcbsp_port[][2] = {};
 #endif
 
+#if defined(CONFIG_ARCH_OMAP4)
+static const unsigned long omap44xx_mcbsp_port[][2] = {
+	{ OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap44xx_mcbsp_port[][2] = {};
+#endif
+
 static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -224,7 +250,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
 	 * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
 	 * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
 	 */
-	if (cpu_is_omap343x()) {
+	if (cpu_is_omap343x() || cpu_is_omap44xx()) {
 		/*
 		* Rule for the buffer size. We should not allow
 		* smaller buffer than the FIFO size to avoid underruns
@@ -332,6 +358,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	} else if (cpu_is_omap343x()) {
 		dma = omap24xx_dma_reqs[bus_id][substream->stream];
 		port = omap34xx_mcbsp_port[bus_id][substream->stream];
+	 } else if (cpu_is_omap44xx()) {
+		dma = omap44xx_dma_reqs[bus_id][substream->stream];
+		port = omap44xx_mcbsp_port[bus_id][substream->stream];
 	} else {
 		return -ENODEV;
 	}
@@ -498,11 +527,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	regs->spcr2	|= XINTM(3) | FREE;
 	regs->spcr1	|= RINTM(3);
 	/* RFIG and XFIG are not defined in 34xx */
-	if (!cpu_is_omap34xx()) {
+	if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) {
 		regs->rcr2	|= RFIG;
 		regs->xcr2	|= XFIG;
 	}
-	if (cpu_is_omap2430() || cpu_is_omap34xx()) {
+	if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) {
 		regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
 		regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
 	}
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index ffdcc5abb7b9..110c106611d3 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -50,6 +50,10 @@ enum omap_mcbsp_div {
 #undef  NUM_LINKS
 #define NUM_LINKS	3
 #endif
+#if defined(CONFIG_ARCH_OMAP4)
+#undef  NUM_LINKS
+#define NUM_LINKS	4
+#endif
 #if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3)
 #undef  NUM_LINKS
 #define NUM_LINKS	5
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 5f2479ce9dde..1e574a5d440d 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -235,6 +235,7 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 
 	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
 	snd_soc_dapm_enable_pin(dapm, "Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
 	snd_soc_dapm_sync(dapm);
 
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index df13338cb3e2..6088a6a3238a 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -8,11 +8,11 @@
  * published by the Free Software Foundation.
  */
 
+#include <linux/clkdev.h>
 #include <linux/device.h>
 #include <linux/firmware.h>
 #include <linux/module.h>
 
-#include <asm/clkdev.h>
 #include <asm/clock.h>
 
 #include <cpu/sh7722.h>
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index b2e333f5a388..1a36b36c5baa 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -1339,7 +1339,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec)
 			goto err;
 		}
 		lzo_blocks[i]->sync_bmp = sync_bmp;
-		lzo_blocks[i]->sync_bmp_nbits = reg_size;
+		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)