Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...

19 files changed, 282 insertions, 39 deletions

diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index fdeb8dc..d315a08 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -422,6 +422,7 @@
  */
 
 struct snd_ac97;
+struct snd_pcm_chmap;
 
 struct snd_ac97_build_ops {
 	int (*build_3d) (struct snd_ac97 *ac97);
@@ -528,6 +529,8 @@ struct snd_ac97 {
 	struct delayed_work power_work;
 #endif
 	struct device dev;
+
+	struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */
 };
 
 #define to_ac97_t(d) container_of(d, struct snd_ac97, dev)
diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h
index a7d8dc7..abdf609 100644
--- a/include/sound/ad1816a.h
+++ b/include/sound/ad1816a.h
@@ -147,6 +147,9 @@ struct snd_ad1816a {
 	unsigned int c_dma_size;
 
 	struct snd_timer *timer;
+#ifdef CONFIG_PM
+	unsigned short image[48];
+#endif
 };
 
 
@@ -165,11 +168,15 @@ struct snd_ad1816a {
 
 extern int snd_ad1816a_create(struct snd_card *card, unsigned long port,
 			      int irq, int dma1, int dma2,
-			      struct snd_ad1816a **chip);
+			      struct snd_ad1816a *chip);
 
 extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm);
 extern int snd_ad1816a_mixer(struct snd_ad1816a *chip);
 extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device,
 			     struct snd_timer **rtimer);
+#ifdef CONFIG_PM
+extern void snd_ad1816a_suspend(struct snd_ad1816a *chip);
+extern void snd_ad1816a_resume(struct snd_ad1816a *chip);
+#endif
 
 #endif	/* __SOUND_AD1816A_H */
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 0876a1e..dfe7d44 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -472,6 +472,45 @@ enum {
 	SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
 };
 
+/* channel positions */
+enum {
+	SNDRV_CHMAP_UNKNOWN = 0,
+	SNDRV_CHMAP_NA,		/* N/A, silent */
+	SNDRV_CHMAP_MONO,	/* mono stream */
+	/* this follows the alsa-lib mixer channel value + 3 */
+	SNDRV_CHMAP_FL,		/* front left */
+	SNDRV_CHMAP_FR,		/* front right */
+	SNDRV_CHMAP_RL,		/* rear left */
+	SNDRV_CHMAP_RR,		/* rear right */
+	SNDRV_CHMAP_FC,		/* front center */
+	SNDRV_CHMAP_LFE,	/* LFE */
+	SNDRV_CHMAP_SL,		/* side left */
+	SNDRV_CHMAP_SR,		/* side right */
+	SNDRV_CHMAP_RC,		/* rear center */
+	/* new definitions */
+	SNDRV_CHMAP_FLC,	/* front left center */
+	SNDRV_CHMAP_FRC,	/* front right center */
+	SNDRV_CHMAP_RLC,	/* rear left center */
+	SNDRV_CHMAP_RRC,	/* rear right center */
+	SNDRV_CHMAP_FLW,	/* front left wide */
+	SNDRV_CHMAP_FRW,	/* front right wide */
+	SNDRV_CHMAP_FLH,	/* front left high */
+	SNDRV_CHMAP_FCH,	/* front center high */
+	SNDRV_CHMAP_FRH,	/* front right high */
+	SNDRV_CHMAP_TC,		/* top center */
+	SNDRV_CHMAP_TFL,	/* top front left */
+	SNDRV_CHMAP_TFR,	/* top front right */
+	SNDRV_CHMAP_TFC,	/* top front center */
+	SNDRV_CHMAP_TRL,	/* top rear left */
+	SNDRV_CHMAP_TRR,	/* top rear right */
+	SNDRV_CHMAP_TRC,	/* top rear center */
+	SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC,
+};
+
+#define SNDRV_CHMAP_POSITION_MASK	0xffff
+#define SNDRV_CHMAP_PHASE_INVERSE	(0x01 << 16)
+#define SNDRV_CHMAP_DRIVER_SPEC		(0x02 << 16)
+
 #define SNDRV_PCM_IOCTL_PVERSION	_IOR('A', 0x00, int)
 #define SNDRV_PCM_IOCTL_INFO		_IOR('A', 0x01, struct snd_pcm_info)
 #define SNDRV_PCM_IOCTL_TSTAMP		_IOW('A', 0x02, int)
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index 48f2a1f..f2912ab 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -61,6 +61,7 @@ struct snd_compr_runtime {
 	u64 total_bytes_available;
 	u64 total_bytes_transferred;
 	wait_queue_head_t sleep;
+	void *private_data;
 };
 
 /**
diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
index da4a456..602dc6c 100644
--- a/include/sound/compress_params.h
+++ b/include/sound/compress_params.h
@@ -72,6 +72,7 @@
 #define SND_AUDIOCODEC_IEC61937              ((__u32) 0x0000000B)
 #define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
 #define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_G729
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
diff --git a/include/sound/da9055.h b/include/sound/da9055.h
new file mode 100644
index 0000000..cf1241b
--- /dev/null
+++ b/include/sound/da9055.h
@@ -0,0 +1,33 @@
+/*
+ * DA9055 ALSA Soc codec driver
+ *
+ * Copyright (c) 2012 Dialog Semiconductor
+ *
+ * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C
+ * Written by David Chen <david.chen@diasemi.com> and
+ * Ashish Chavan <ashish.chavan@kpitcummins.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; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __SOUND_DA9055_H__
+#define __SOUND_DA9055_H__
+
+enum da9055_micbias_voltage {
+	DA9055_MICBIAS_1_6V = 0,
+	DA9055_MICBIAS_1_8V = 1,
+	DA9055_MICBIAS_2_1V = 2,
+	DA9055_MICBIAS_2_2V = 3,
+};
+
+struct da9055_platform_data {
+	/* Selects which of the two MicBias pins acts as the bias source */
+	bool micbias_source;
+	/* Selects the micbias voltage */
+	enum da9055_micbias_voltage micbias;
+};
+
+#endif
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 4f865df..1a33f48 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1788,7 +1788,7 @@ struct snd_emu10k1 {
 	unsigned int efx_voices_mask[2];
 	unsigned int next_free_voice;
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 	unsigned int *saved_ptr;
 	unsigned int *saved_gpr;
 	unsigned int *tram_val_saved;
@@ -1856,7 +1856,7 @@ unsigned short snd_emu10k1_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
 void snd_emu10k1_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short data);
 unsigned int snd_emu10k1_rate_to_pitch(unsigned int rate);
 
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
 void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu);
 void snd_emu10k1_resume_init(struct snd_emu10k1 *emu);
 void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu);
diff --git a/include/sound/initval.h b/include/sound/initval.h
index f99a0d2..ac62c67 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -50,6 +50,20 @@
 #define SNDRV_DEFAULT_DMA_SIZE	{ [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA_SIZE }
 #define SNDRV_DEFAULT_PTR	SNDRV_DEFAULT_STR
 
+#ifdef SNDRV_LEGACY_FIND_FREE_IOPORT
+static long snd_legacy_find_free_ioport(long *port_table, long size)
+{
+	while (*port_table != -1) {
+		if (request_region(*port_table, size, "ALSA test")) {
+			release_region(*port_table, size);
+			return *port_table;
+		}
+		port_table++;
+	}
+	return -1;
+}
+#endif
+
 #ifdef SNDRV_LEGACY_FIND_FREE_IRQ
 #include <linux/interrupt.h>
 
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index c425062..844af65 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -98,8 +98,10 @@ static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
 /*
  * return the physical address at the corresponding offset
  */
-static inline dma_addr_t snd_sgbuf_get_addr(struct snd_sg_buf *sgbuf, size_t offset)
+static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
+					   size_t offset)
 {
+	struct snd_sg_buf *sgbuf = dmab->private_data;
 	dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
 	addr &= PAGE_MASK;
 	return addr + offset % PAGE_SIZE;
@@ -108,10 +110,31 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_sg_buf *sgbuf, size_t off
 /*
  * return the virtual address at the corresponding offset
  */
-static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset)
+static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
+				     size_t offset)
 {
+	struct snd_sg_buf *sgbuf = dmab->private_data;
 	return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
 }
+
+unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
+				      unsigned int ofs, unsigned int size);
+#else
+/* non-SG versions */
+static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
+					    size_t offset)
+{
+	return dmab->addr + offset;
+}
+
+static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
+				      size_t offset)
+{
+	return dmab->area + offset;
+}
+
+#define snd_sgbuf_get_chunk_size(dmab, ofs, size)	(size)
+
 #endif /* CONFIG_SND_DMA_SGBUF */
 
 /* allocate/release a buffer */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index d0711bc..6268a41 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -437,6 +437,7 @@ struct snd_pcm_str {
 	struct snd_info_entry *proc_xrun_debug_entry;
 #endif
 #endif
+	struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */
 };
 
 struct snd_pcm {
@@ -982,53 +983,42 @@ static int snd_pcm_lib_alloc_vmalloc_32_buffer
 	_snd_pcm_lib_alloc_vmalloc_buffer \
 			(subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO)
 
+#define snd_pcm_get_dma_buf(substream) ((substream)->runtime->dma_buffer_p)
+
 #ifdef CONFIG_SND_DMA_SGBUF
 /*
  * SG-buffer handling
  */
 #define snd_pcm_substream_sgbuf(substream) \
-	((substream)->runtime->dma_buffer_p->private_data)
-
-static inline dma_addr_t
-snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
-{
-	struct snd_sg_buf *sg = snd_pcm_substream_sgbuf(substream);
-	return snd_sgbuf_get_addr(sg, ofs);
-}
-
-static inline void *
-snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
-{
-	struct snd_sg_buf *sg = snd_pcm_substream_sgbuf(substream);
-	return snd_sgbuf_get_ptr(sg, ofs);
-}
+	snd_pcm_get_dma_buf(substream)->private_data
 
 struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream,
 				    unsigned long offset);
-unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
-					  unsigned int ofs, unsigned int size);
-
 #else /* !SND_DMA_SGBUF */
 /*
  * fake using a continuous buffer
  */
+#define snd_pcm_sgbuf_ops_page	NULL
+#endif /* SND_DMA_SGBUF */
+
 static inline dma_addr_t
 snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
 {
-	return substream->runtime->dma_addr + ofs;
+	return snd_sgbuf_get_addr(snd_pcm_get_dma_buf(substream), ofs);
 }
 
 static inline void *
 snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
 {
-	return substream->runtime->dma_area + ofs;
+	return snd_sgbuf_get_ptr(snd_pcm_get_dma_buf(substream), ofs);
 }
 
-#define snd_pcm_sgbuf_ops_page	NULL
-
-#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size)	(size)
-
-#endif /* SND_DMA_SGBUF */
+static inline unsigned int
+snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
+			     unsigned int ofs, unsigned int size)
+{
+	return snd_sgbuf_get_chunk_size(snd_pcm_get_dma_buf(substream), ofs, size);
+}
 
 /* handle mmap counter - PCM mmap callback should handle this counter properly */
 static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area)
@@ -1086,4 +1076,51 @@ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream
 		return "Capture";
 }
 
+/*
+ * PCM channel-mapping control API
+ */
+/* array element of channel maps */
+struct snd_pcm_chmap_elem {
+	unsigned char channels;
+	unsigned char map[15];
+};
+
+/* channel map information; retrieved via snd_kcontrol_chip() */
+struct snd_pcm_chmap {
+	struct snd_pcm *pcm;	/* assigned PCM instance */
+	int stream;		/* PLAYBACK or CAPTURE */
+	struct snd_kcontrol *kctl;
+	const struct snd_pcm_chmap_elem *chmap;
+	unsigned int max_channels;
+	unsigned int channel_mask;	/* optional: active channels bitmask */
+	void *private_data;	/* optional: private data pointer */
+};
+
+/* get the PCM substream assigned to the given chmap info */
+static inline struct snd_pcm_substream *
+snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx)
+{
+	struct snd_pcm_substream *s;
+	for (s = info->pcm->streams[info->stream].substream; s; s = s->next)
+		if (s->number == idx)
+			return s;
+	return NULL;
+}
+
+/* ALSA-standard channel maps (RL/RR prior to C/LFE) */
+extern const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[];
+/* Other world's standard channel maps (C/LFE prior to RL/RR) */
+extern const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[];
+
+/* bit masks to be passed to snd_pcm_chmap.channel_mask field */
+#define SND_PCM_CHMAP_MASK_24	((1U << 2) | (1U << 4))
+#define SND_PCM_CHMAP_MASK_246	(SND_PCM_CHMAP_MASK_24 | (1U << 6))
+#define SND_PCM_CHMAP_MASK_2468	(SND_PCM_CHMAP_MASK_246 | (1U << 8))
+
+int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
+			   const struct snd_pcm_chmap_elem *chmap,
+			   int max_channels,
+			   unsigned long private_value,
+			   struct snd_pcm_chmap **info_ret);
+
 #endif /* __SOUND_PCM_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1f69e0a..628db7b 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -18,6 +18,7 @@
 
 struct snd_pcm_substream;
 struct snd_soc_dapm_widget;
+struct snd_compr_stream;
 
 /*
  * DAI hardware audio formats.
@@ -205,6 +206,8 @@ struct snd_soc_dai_driver {
 	int (*remove)(struct snd_soc_dai *dai);
 	int (*suspend)(struct snd_soc_dai *dai);
 	int (*resume)(struct snd_soc_dai *dai);
+	/* compress dai */
+	bool compress_dai;
 
 	/* ops */
 	const struct snd_soc_dai_ops *ops;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index abe373d..e1ef63d 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -244,10 +244,11 @@ struct device;
 {	.id = snd_soc_dapm_supply, .name = wname, .reg = wreg,	\
 	.shift = wshift, .invert = winvert, .event = wevent, \
 	.event_flags = wflags}
-#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \
+#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags)	    \
 {	.id = snd_soc_dapm_regulator_supply, .name = wname, \
 	.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
-	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
+	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+	.invert = wflags}
 
 
 /* dapm kcontrol types */
@@ -319,6 +320,9 @@ struct device;
 #define SND_SOC_DAPM_EVENT_OFF(e)	\
 	(e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD))
 
+/* regulator widget flags */
+#define SND_SOC_DAPM_REGULATOR_BYPASS     0x1     /* bypass when disabled */
+
 struct snd_soc_dapm_widget;
 enum snd_soc_dapm_type;
 struct snd_soc_dapm_path;
@@ -412,6 +416,7 @@ 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);
+void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
 
 /* dapm path query */
 int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
@@ -510,7 +515,6 @@ struct snd_soc_dapm_widget {
 	/* dapm control */
 	int reg;				/* negative reg = no direct dapm */
 	unsigned char shift;			/* bits to shift */
-	unsigned int saved_value;		/* widget saved value */
 	unsigned int value;				/* widget current value */
 	unsigned int mask;			/* non-shifted mask */
 	unsigned int on_val;			/* on state value */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index e063380..91244a0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -20,8 +20,10 @@
 #include <linux/interrupt.h>
 #include <linux/kernel.h>
 #include <linux/regmap.h>
+#include <linux/log2.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
+#include <sound/compress_driver.h>
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
 
@@ -159,7 +161,8 @@
 		 .platform_max = xmax} }
 #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \
 {	.reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
-	.max = xmax, .texts = xtexts }
+	.max = xmax, .texts = xtexts, \
+	.mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0}
 #define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \
 	SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts)
 #define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \
@@ -399,6 +402,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform,
 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);
+int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
 
 struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
 		const char *dai_link, int stream);
@@ -632,6 +636,13 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
+struct snd_soc_compr_ops {
+	int (*startup)(struct snd_compr_stream *);
+	void (*shutdown)(struct snd_compr_stream *);
+	int (*set_params)(struct snd_compr_stream *);
+	int (*trigger)(struct snd_compr_stream *);
+};
+
 /* SoC cache ops */
 struct snd_soc_cache_ops {
 	const char *name;
@@ -787,9 +798,12 @@ struct snd_soc_platform_driver {
 	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
 		struct snd_soc_dai *);
 
-	/* platform stream ops */
+	/* platform stream pcm ops */
 	struct snd_pcm_ops *ops;
 
+	/* platform stream compress ops */
+	struct snd_compr_ops *compr_ops;
+
 	/* platform stream completion event */
 	int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
 
@@ -891,6 +905,7 @@ struct snd_soc_dai_link {
 
 	/* machine stream operations */
 	struct snd_soc_ops *ops;
+	struct snd_soc_compr_ops *compr_ops;
 };
 
 struct snd_soc_codec_conf {
@@ -1027,6 +1042,7 @@ struct snd_soc_pcm_runtime {
 
 	/* runtime devices */
 	struct snd_pcm *pcm;
+	struct snd_compr *compr;
 	struct snd_soc_codec *codec;
 	struct snd_soc_platform *platform;
 	struct snd_soc_dai *codec_dai;
diff --git a/include/sound/tegra_wm8903.h b/include/sound/tegra_wm8903.h
new file mode 100644
index 0000000..57b202e
--- /dev/null
+++ b/include/sound/tegra_wm8903.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright 2011 NVIDIA, Inc.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SOUND_TEGRA_WM38903_H
+#define __SOUND_TEGRA_WM38903_H
+
+struct tegra_wm8903_platform_data {
+	int gpio_spkr_en;
+	int gpio_hp_det;
+	int gpio_hp_mute;
+	int gpio_int_mic_en;
+	int gpio_ext_mic_en;
+};
+
+#endif
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index a64d8fe..28c65e1 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -86,4 +86,12 @@
 
 #define TLV_DB_GAIN_MUTE	-9999999
 
+/*
+ * channel-mapping TLV items
+ *  TLV length must match with num_channels
+ */
+#define SNDRV_CTL_TLVT_CHMAP_FIXED	0x101	/* fixed channel position */
+#define SNDRV_CTL_TLVT_CHMAP_VAR	0x102	/* channels freely swappable */
+#define SNDRV_CTL_TLVT_CHMAP_PAIRED	0x103	/* pair-wise swappable */
+
 #endif /* __SOUND_TLV_H */
diff --git a/include/sound/version.h b/include/sound/version.h
deleted file mode 100644
index cc75024..0000000
--- a/include/sound/version.h
+++ /dev/null
@@ -1,3 +0,0 @@
-/* include/version.h */
-#define CONFIG_SND_VERSION "1.0.25"
-#define CONFIG_SND_DATE ""
diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h
new file mode 100644
index 0000000..3261e90
--- /dev/null
+++ b/include/sound/wm0010.h
@@ -0,0 +1,27 @@
+/*
+ * wm0010.h -- Platform data for WM0010 DSP Driver
+ *
+ * Copyright 2012 Wolfson Microelectronics PLC.
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.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;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#ifndef WM0010_PDATA_H
+#define WM0010_PDATA_H
+
+struct wm0010_pdata {
+	int gpio_reset;
+
+	/* Set if there is an inverter between the GPIO controlling
+	 * the reset signal and the device.
+	 */
+	int reset_active_high;
+	int irq_flags;
+};
+
+#endif
diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h
index 74e9a95..e8ce8ee 100644
--- a/include/sound/wm8960.h
+++ b/include/sound/wm8960.h
@@ -18,7 +18,7 @@
 struct wm8960_data {
 	bool capless;  /* Headphone outputs configured in capless mode */
 
-	int dres;  /* Discharge resistance for headphone outputs */
+	bool shared_lrclk;  /* DAC and ADC LRCLKs are wired together */
 };
 
 #endif
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
index eee19f6..8016fd8 100644
--- a/include/sound/wm8993.h
+++ b/include/sound/wm8993.h
@@ -32,6 +32,10 @@ struct wm8993_platform_data {
 	unsigned int lineout1fb:1;
 	unsigned int lineout2fb:1;
 
+	/* Delay to add for microphones to stabalise after power up */
+	int micbias1_delay;
+	int micbias2_delay;
+
 	/* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
 	unsigned int micbias1_lvl:1;
 	unsigned int micbias2_lvl:1;