Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
  ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
  ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
  ALSA: usb-audio: add Yamaha MOX6/MOX8 support
  ALSA: virtuoso: add S/PDIF input support for all Xonars
  ALSA: ice1724 - Support for ooAoo SQ210a
  ALSA: ice1724 - Allow card info based on model only
  ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
  ALSA: hdspm - Provide unique driver id based on card serial
  ASoC: Dynamically allocate the rtd device for a non-empty release()
  ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
  ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
  ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
  ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
  ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
  ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
  ALSA: hda/cirrus - support for iMac12,2 model
  ASoC: cx20442: add bias control over a platform provided regulator
  ALSA: usb-audio - Avoid flood of frame-active debug messages
  ALSA: snd-usb-us122l: Delete calls to preempt_disable
  mfd: Put WM8994 into cache only mode when suspending
  ...

Fix up trivial conflicts in:
 - arch/arm/mach-s3c64xx/mach-crag6410.c:
	renamed speyside_wm8962 to tobermory, added littlemill right
	next to it
 - drivers/base/regmap/{regcache.c,regmap.c}:
	duplicate diff that had already come in with other changes in
	the regmap tree

12 files changed, 815 insertions, 11 deletions

diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index 802947f60915..6df30ed1581c 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
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
new file mode 100644
index 000000000000..48f2a1ff2bbc
--- /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 000000000000..05341a43fedf
--- /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
diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
new file mode 100644
index 000000000000..d97d69f81a7d
--- /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
diff --git a/include/sound/control.h b/include/sound/control.h
index 1a94a216ed99..b2796e83c7ac 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/include/sound/core.h b/include/sound/core.h
index 3be5ab782b99..5ab255f196cc 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 8f764204a856..5978f9a8c8b2 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/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index 9a155f9d0a12..9b1aacaa82fe 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/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 17a4c17f19f5..d26a9b784772 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 }
@@ -380,6 +383,7 @@ 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);
@@ -409,6 +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/include/sound/soc.h b/include/sound/soc.h
index 11cfb5953e06..0992dff55959 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;
@@ -266,8 +267,6 @@ enum snd_soc_control_type {
 
 enum snd_soc_compress_type {
 	SND_SOC_FLAT_COMPRESSION = 1,
-	SND_SOC_LZO_COMPRESSION,
-	SND_SOC_RBTREE_COMPRESSION
 };
 
 enum snd_soc_pcm_subclass {
@@ -318,6 +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);
@@ -593,8 +593,7 @@ 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 */
@@ -706,8 +705,11 @@ 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 */
@@ -718,6 +720,9 @@ 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);
 
@@ -813,6 +818,7 @@ 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;
 
@@ -840,8 +846,8 @@ struct snd_soc_card {
 };
 
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
-struct snd_soc_pcm_runtime  {
-	struct device dev;
+struct snd_soc_pcm_runtime {
+	struct device *dev;
 	struct snd_soc_card *card;
 	struct snd_soc_dai_link *dai_link;
 	struct mutex pcm_mutex;
@@ -927,12 +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)
@@ -960,6 +966,11 @@ 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);
+int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
+				   const char *propname);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS
diff --git a/include/sound/sta32x.h b/include/sound/sta32x.h
new file mode 100644
index 000000000000..8d93b0357a14
--- /dev/null
+++ b/include/sound/sta32x.h
@@ -0,0 +1,35 @@
+/*
+ * 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;
+	int needs_esd_watchdog;
+};
+
+#endif /* __LINUX_SND__STA32X_H */
diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index cf7ccb76a8de..b310c5a3a958 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