Merge remote-tracking branch 'airlied/drm-fixes' into drm-intel-next-queued

Back-merge from drm-fixes into drm-intel-next to sort out two things:

- interlaced support: -fixes contains a bugfix to correctly clear
  interlaced configuration bits in case the bios sets up an interlaced
  mode and we want to set up the progressive mode (current kernels
  don't support interlaced). The actual feature work to support
  interlaced depends upon (and conflicts with) this bugfix.

- forcewake voodoo to workaround missed IRQ issues: -fixes only enabled
  this for ivybridge, but some recent bug reports indicate that we
  need this on Sandybridge, too. But in a slightly different flavour
  and with other fixes and reworks on top. Additionally there are some
  forcewake cleanup patches heading to -next that would conflict with
  currrent -fixes.

Signed-Off-by: Daniel Vetter <daniel.vetter@ffwll.ch>

14 files changed, 818 insertions, 14 deletions

diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index 802947f6091..6df30ed1581 100644
--- a/include/sound/Kbuild
+++ b/include/sound/Kbuild
@@ -6,3 +6,5 @@ header-y += hdsp.h
 header-y += hdspm.h
 header-y += sb16_csp.h
 header-y += sfnt_info.h
+header-y += compress_params.h
+header-y += compress_offload.h
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
new file mode 100644
index 00000000000..48f2a1ff2bb
--- /dev/null
+++ b/include/sound/compress_driver.h
@@ -0,0 +1,167 @@
+/*
+ *  compress_driver.h - compress offload driver definations
+ *
+ *  Copyright (C) 2011 Intel Corporation
+ *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
+ *		Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+#ifndef __COMPRESS_DRIVER_H
+#define __COMPRESS_DRIVER_H
+
+#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/compress_offload.h>
+#include <sound/asound.h>
+#include <sound/pcm.h>
+
+struct snd_compr_ops;
+
+/**
+ * struct snd_compr_runtime: runtime stream description
+ * @state: stream state
+ * @ops: pointer to DSP callbacks
+ * @buffer: pointer to kernel buffer, valid only when not in mmap mode or
+ *	DSP doesn't implement copy
+ * @buffer_size: size of the above buffer
+ * @fragment_size: size of buffer fragment in bytes
+ * @fragments: number of such fragments
+ * @hw_pointer: offset of last location in buffer where DSP copied data
+ * @app_pointer: offset of last location in buffer where app wrote data
+ * @total_bytes_available: cumulative number of bytes made available in
+ *	the ring buffer
+ * @total_bytes_transferred: cumulative bytes transferred by offload DSP
+ * @sleep: poll sleep
+ */
+struct snd_compr_runtime {
+	snd_pcm_state_t state;
+	struct snd_compr_ops *ops;
+	void *buffer;
+	u64 buffer_size;
+	u32 fragment_size;
+	u32 fragments;
+	u64 hw_pointer;
+	u64 app_pointer;
+	u64 total_bytes_available;
+	u64 total_bytes_transferred;
+	wait_queue_head_t sleep;
+};
+
+/**
+ * struct snd_compr_stream: compressed stream
+ * @name: device name
+ * @ops: pointer to DSP callbacks
+ * @runtime: pointer to runtime structure
+ * @device: device pointer
+ * @direction: stream direction, playback/recording
+ * @private_data: pointer to DSP private data
+ */
+struct snd_compr_stream {
+	const char *name;
+	struct snd_compr_ops *ops;
+	struct snd_compr_runtime *runtime;
+	struct snd_compr *device;
+	enum snd_compr_direction direction;
+	void *private_data;
+};
+
+/**
+ * struct snd_compr_ops: compressed path DSP operations
+ * @open: Open the compressed stream
+ * This callback is mandatory and shall keep dsp ready to receive the stream
+ * parameter
+ * @free: Close the compressed stream, mandatory
+ * @set_params: Sets the compressed stream parameters, mandatory
+ * This can be called in during stream creation only to set codec params
+ * and the stream properties
+ * @get_params: retrieve the codec parameters, mandatory
+ * @trigger: Trigger operations like start, pause, resume, drain, stop.
+ * This callback is mandatory
+ * @pointer: Retrieve current h/w pointer information. Mandatory
+ * @copy: Copy the compressed data to/from userspace, Optional
+ * Can't be implemented if DSP supports mmap
+ * @mmap: DSP mmap method to mmap DSP memory
+ * @ack: Ack for DSP when data is written to audio buffer, Optional
+ * Not valid if copy is implemented
+ * @get_caps: Retrieve DSP capabilities, mandatory
+ * @get_codec_caps: Retrieve capabilities for a specific codec, mandatory
+ */
+struct snd_compr_ops {
+	int (*open)(struct snd_compr_stream *stream);
+	int (*free)(struct snd_compr_stream *stream);
+	int (*set_params)(struct snd_compr_stream *stream,
+			struct snd_compr_params *params);
+	int (*get_params)(struct snd_compr_stream *stream,
+			struct snd_codec *params);
+	int (*trigger)(struct snd_compr_stream *stream, int cmd);
+	int (*pointer)(struct snd_compr_stream *stream,
+			struct snd_compr_tstamp *tstamp);
+	int (*copy)(struct snd_compr_stream *stream, const char __user *buf,
+		       size_t count);
+	int (*mmap)(struct snd_compr_stream *stream,
+			struct vm_area_struct *vma);
+	int (*ack)(struct snd_compr_stream *stream, size_t bytes);
+	int (*get_caps) (struct snd_compr_stream *stream,
+			struct snd_compr_caps *caps);
+	int (*get_codec_caps) (struct snd_compr_stream *stream,
+			struct snd_compr_codec_caps *codec);
+};
+
+/**
+ * struct snd_compr: Compressed device
+ * @name: DSP device name
+ * @dev: Device pointer
+ * @ops: pointer to DSP callbacks
+ * @private_data: pointer to DSP pvt data
+ * @card: sound card pointer
+ * @direction: Playback or capture direction
+ * @lock: device lock
+ * @device: device id
+ */
+struct snd_compr {
+	const char *name;
+	struct device *dev;
+	struct snd_compr_ops *ops;
+	void *private_data;
+	struct snd_card *card;
+	unsigned int direction;
+	struct mutex lock;
+	int device;
+};
+
+/* compress device register APIs */
+int snd_compress_register(struct snd_compr *device);
+int snd_compress_deregister(struct snd_compr *device);
+int snd_compress_new(struct snd_card *card, int device,
+			int type, struct snd_compr *compr);
+
+/* dsp driver callback apis
+ * For playback: driver should call snd_compress_fragment_elapsed() to let the
+ * framework know that a fragment has been consumed from the ring buffer
+ *
+ * For recording: we want to know when a frame is available or when
+ * at least one frame is available so snd_compress_frame_elapsed()
+ * callback should be called when a encodeded frame is available
+ */
+static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream)
+{
+	wake_up(&stream->runtime->sleep);
+}
+
+#endif
diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h
new file mode 100644
index 00000000000..05341a43fed
--- /dev/null
+++ b/include/sound/compress_offload.h
@@ -0,0 +1,161 @@
+/*
+ *  compress_offload.h - compress offload header definations
+ *
+ *  Copyright (C) 2011 Intel Corporation
+ *  Authors:	Vinod Koul <vinod.koul@linux.intel.com>
+ *		Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ */
+#ifndef __COMPRESS_OFFLOAD_H
+#define __COMPRESS_OFFLOAD_H
+
+#include <linux/types.h>
+#include <sound/asound.h>
+#include <sound/compress_params.h>
+
+
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
+/**
+ * struct snd_compressed_buffer: compressed buffer
+ * @fragment_size: size of buffer fragment in bytes
+ * @fragments: number of such fragments
+ */
+struct snd_compressed_buffer {
+	__u32 fragment_size;
+	__u32 fragments;
+};
+
+/**
+ * struct snd_compr_params: compressed stream params
+ * @buffer: buffer description
+ * @codec: codec parameters
+ * @no_wake_mode: dont wake on fragment elapsed
+ */
+struct snd_compr_params {
+	struct snd_compressed_buffer buffer;
+	struct snd_codec codec;
+	__u8 no_wake_mode;
+};
+
+/**
+ * struct snd_compr_tstamp: timestamp descriptor
+ * @byte_offset: Byte offset in ring buffer to DSP
+ * @copied_total: Total number of bytes copied from/to ring buffer to/by DSP
+ * @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by
+ *	large steps and should only be used to monitor encoding/decoding
+ *	progress. It shall not be used for timing estimates.
+ * @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio
+ * output/input. This field should be used for A/V sync or time estimates.
+ * @sampling_rate: sampling rate of audio
+ */
+struct snd_compr_tstamp {
+	__u32 byte_offset;
+	__u32 copied_total;
+	snd_pcm_uframes_t pcm_frames;
+	snd_pcm_uframes_t pcm_io_frames;
+	__u32 sampling_rate;
+};
+
+/**
+ * struct snd_compr_avail: avail descriptor
+ * @avail: Number of bytes available in ring buffer for writing/reading
+ * @tstamp: timestamp infomation
+ */
+struct snd_compr_avail {
+	__u64 avail;
+	struct snd_compr_tstamp tstamp;
+};
+
+enum snd_compr_direction {
+	SND_COMPRESS_PLAYBACK = 0,
+	SND_COMPRESS_CAPTURE
+};
+
+/**
+ * struct snd_compr_caps: caps descriptor
+ * @codecs: pointer to array of codecs
+ * @direction: direction supported. Of type snd_compr_direction
+ * @min_fragment_size: minimum fragment supported by DSP
+ * @max_fragment_size: maximum fragment supported by DSP
+ * @min_fragments: min fragments supported by DSP
+ * @max_fragments: max fragments supported by DSP
+ * @num_codecs: number of codecs supported
+ * @reserved: reserved field
+ */
+struct snd_compr_caps {
+	__u32 num_codecs;
+	__u32 direction;
+	__u32 min_fragment_size;
+	__u32 max_fragment_size;
+	__u32 min_fragments;
+	__u32 max_fragments;
+	__u32 codecs[MAX_NUM_CODECS];
+	__u32 reserved[11];
+};
+
+/**
+ * struct snd_compr_codec_caps: query capability of codec
+ * @codec: codec for which capability is queried
+ * @num_descriptors: number of codec descriptors
+ * @descriptor: array of codec capability descriptor
+ */
+struct snd_compr_codec_caps {
+	__u32 codec;
+	__u32 num_descriptors;
+	struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS];
+};
+
+/**
+ * compress path ioctl definitions
+ * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
+ * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec
+ * SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters
+ * Note: only codec params can be changed runtime and stream params cant be
+ * SNDRV_COMPRESS_GET_PARAMS: Query codec params
+ * SNDRV_COMPRESS_TSTAMP: get the current timestamp value
+ * SNDRV_COMPRESS_AVAIL: get the current buffer avail value.
+ * This also queries the tstamp properties
+ * SNDRV_COMPRESS_PAUSE: Pause the running stream
+ * SNDRV_COMPRESS_RESUME: resume a paused stream
+ * SNDRV_COMPRESS_START: Start a stream
+ * SNDRV_COMPRESS_STOP: stop a running stream, discarding ring buffer content
+ * and the buffers currently with DSP
+ * SNDRV_COMPRESS_DRAIN: Play till end of buffers and stop after that
+ * SNDRV_COMPRESS_IOCTL_VERSION: Query the API version
+ */
+#define SNDRV_COMPRESS_IOCTL_VERSION	_IOR('C', 0x00, int)
+#define SNDRV_COMPRESS_GET_CAPS		_IOWR('C', 0x10, struct snd_compr_caps)
+#define SNDRV_COMPRESS_GET_CODEC_CAPS	_IOWR('C', 0x11,\
+						struct snd_compr_codec_caps)
+#define SNDRV_COMPRESS_SET_PARAMS	_IOW('C', 0x12, struct snd_compr_params)
+#define SNDRV_COMPRESS_GET_PARAMS	_IOR('C', 0x13, struct snd_codec)
+#define SNDRV_COMPRESS_TSTAMP		_IOR('C', 0x20, struct snd_compr_tstamp)
+#define SNDRV_COMPRESS_AVAIL		_IOR('C', 0x21, struct snd_compr_avail)
+#define SNDRV_COMPRESS_PAUSE		_IO('C', 0x30)
+#define SNDRV_COMPRESS_RESUME		_IO('C', 0x31)
+#define SNDRV_COMPRESS_START		_IO('C', 0x32)
+#define SNDRV_COMPRESS_STOP		_IO('C', 0x33)
+#define SNDRV_COMPRESS_DRAIN		_IO('C', 0x34)
+/*
+ * TODO
+ * 1. add mmap support
+ *
+ */
+#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */
+#endif
diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
new file mode 100644
index 00000000000..d97d69f81a7
--- /dev/null
+++ b/include/sound/compress_params.h
@@ -0,0 +1,397 @@
+/*
+ *  compress_params.h - codec types and parameters for compressed data
+ *  streaming interface
+ *
+ *  Copyright (C) 2011 Intel Corporation
+ *  Authors:	Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *              Vinod Koul <vinod.koul@linux.intel.com>
+ *
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * The definitions in this file are derived from the OpenMAX AL version 1.1
+ * and OpenMAX IL v 1.1.2 header files which contain the copyright notice below.
+ *
+ * Copyright (c) 2007-2010 The Khronos Group Inc.
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and/or associated documentation files (the
+ * "Materials "), to deal in the Materials without restriction, including
+ * without limitation the rights to use, copy, modify, merge, publish,
+ * distribute, sublicense, and/or sell copies of the Materials, and to
+ * permit persons to whom the Materials are furnished to do so, subject to
+ * the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included
+ * in all copies or substantial portions of the Materials.
+ *
+ * THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
+ * OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
+ * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
+ * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
+ * MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS.
+ *
+ */
+#ifndef __SND_COMPRESS_PARAMS_H
+#define __SND_COMPRESS_PARAMS_H
+
+/* AUDIO CODECS SUPPORTED */
+#define MAX_NUM_CODECS 32
+#define MAX_NUM_CODEC_DESCRIPTORS 32
+#define MAX_NUM_BITRATES 32
+
+/* Codecs are listed linearly to allow for extensibility */
+#define SND_AUDIOCODEC_PCM                   ((__u32) 0x00000001)
+#define SND_AUDIOCODEC_MP3                   ((__u32) 0x00000002)
+#define SND_AUDIOCODEC_AMR                   ((__u32) 0x00000003)
+#define SND_AUDIOCODEC_AMRWB                 ((__u32) 0x00000004)
+#define SND_AUDIOCODEC_AMRWBPLUS             ((__u32) 0x00000005)
+#define SND_AUDIOCODEC_AAC                   ((__u32) 0x00000006)
+#define SND_AUDIOCODEC_WMA                   ((__u32) 0x00000007)
+#define SND_AUDIOCODEC_REAL                  ((__u32) 0x00000008)
+#define SND_AUDIOCODEC_VORBIS                ((__u32) 0x00000009)
+#define SND_AUDIOCODEC_FLAC                  ((__u32) 0x0000000A)
+#define SND_AUDIOCODEC_IEC61937              ((__u32) 0x0000000B)
+#define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
+#define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
+
+/*
+ * Profile and modes are listed with bit masks. This allows for a
+ * more compact representation of fields that will not evolve
+ * (in contrast to the list of codecs)
+ */
+
+#define SND_AUDIOPROFILE_PCM                 ((__u32) 0x00000001)
+
+/* MP3 modes are only useful for encoders */
+#define SND_AUDIOCHANMODE_MP3_MONO           ((__u32) 0x00000001)
+#define SND_AUDIOCHANMODE_MP3_STEREO         ((__u32) 0x00000002)
+#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO    ((__u32) 0x00000004)
+#define SND_AUDIOCHANMODE_MP3_DUAL           ((__u32) 0x00000008)
+
+#define SND_AUDIOPROFILE_AMR                 ((__u32) 0x00000001)
+
+/* AMR modes are only useful for encoders */
+#define SND_AUDIOMODE_AMR_DTX_OFF            ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AMR_VAD1               ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AMR_VAD2               ((__u32) 0x00000004)
+
+#define SND_AUDIOSTREAMFORMAT_UNDEFINED	     ((__u32) 0x00000000)
+#define SND_AUDIOSTREAMFORMAT_CONFORMANCE    ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_IF1            ((__u32) 0x00000002)
+#define SND_AUDIOSTREAMFORMAT_IF2            ((__u32) 0x00000004)
+#define SND_AUDIOSTREAMFORMAT_FSF            ((__u32) 0x00000008)
+#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD     ((__u32) 0x00000010)
+#define SND_AUDIOSTREAMFORMAT_ITU            ((__u32) 0x00000020)
+
+#define SND_AUDIOPROFILE_AMRWB               ((__u32) 0x00000001)
+
+/* AMRWB modes are only useful for encoders */
+#define SND_AUDIOMODE_AMRWB_DTX_OFF          ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AMRWB_VAD1             ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AMRWB_VAD2             ((__u32) 0x00000004)
+
+#define SND_AUDIOPROFILE_AMRWBPLUS           ((__u32) 0x00000001)
+
+#define SND_AUDIOPROFILE_AAC                 ((__u32) 0x00000001)
+
+/* AAC modes are required for encoders and decoders */
+#define SND_AUDIOMODE_AAC_MAIN               ((__u32) 0x00000001)
+#define SND_AUDIOMODE_AAC_LC                 ((__u32) 0x00000002)
+#define SND_AUDIOMODE_AAC_SSR                ((__u32) 0x00000004)
+#define SND_AUDIOMODE_AAC_LTP                ((__u32) 0x00000008)
+#define SND_AUDIOMODE_AAC_HE                 ((__u32) 0x00000010)
+#define SND_AUDIOMODE_AAC_SCALABLE           ((__u32) 0x00000020)
+#define SND_AUDIOMODE_AAC_ERLC               ((__u32) 0x00000040)
+#define SND_AUDIOMODE_AAC_LD                 ((__u32) 0x00000080)
+#define SND_AUDIOMODE_AAC_HE_PS              ((__u32) 0x00000100)
+#define SND_AUDIOMODE_AAC_HE_MPS             ((__u32) 0x00000200)
+
+/* AAC formats are required for encoders and decoders */
+#define SND_AUDIOSTREAMFORMAT_MP2ADTS        ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_MP4ADTS        ((__u32) 0x00000002)
+#define SND_AUDIOSTREAMFORMAT_MP4LOAS        ((__u32) 0x00000004)
+#define SND_AUDIOSTREAMFORMAT_MP4LATM        ((__u32) 0x00000008)
+#define SND_AUDIOSTREAMFORMAT_ADIF           ((__u32) 0x00000010)
+#define SND_AUDIOSTREAMFORMAT_MP4FF          ((__u32) 0x00000020)
+#define SND_AUDIOSTREAMFORMAT_RAW            ((__u32) 0x00000040)
+
+#define SND_AUDIOPROFILE_WMA7                ((__u32) 0x00000001)
+#define SND_AUDIOPROFILE_WMA8                ((__u32) 0x00000002)
+#define SND_AUDIOPROFILE_WMA9                ((__u32) 0x00000004)
+#define SND_AUDIOPROFILE_WMA10               ((__u32) 0x00000008)
+
+#define SND_AUDIOMODE_WMA_LEVEL1             ((__u32) 0x00000001)
+#define SND_AUDIOMODE_WMA_LEVEL2             ((__u32) 0x00000002)
+#define SND_AUDIOMODE_WMA_LEVEL3             ((__u32) 0x00000004)
+#define SND_AUDIOMODE_WMA_LEVEL4             ((__u32) 0x00000008)
+#define SND_AUDIOMODE_WMAPRO_LEVELM0         ((__u32) 0x00000010)
+#define SND_AUDIOMODE_WMAPRO_LEVELM1         ((__u32) 0x00000020)
+#define SND_AUDIOMODE_WMAPRO_LEVELM2         ((__u32) 0x00000040)
+#define SND_AUDIOMODE_WMAPRO_LEVELM3         ((__u32) 0x00000080)
+
+#define SND_AUDIOSTREAMFORMAT_WMA_ASF        ((__u32) 0x00000001)
+/*
+ * Some implementations strip the ASF header and only send ASF packets
+ * to the DSP
+ */
+#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR  ((__u32) 0x00000002)
+
+#define SND_AUDIOPROFILE_REALAUDIO           ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_REALAUDIO_G2           ((__u32) 0x00000001)
+#define SND_AUDIOMODE_REALAUDIO_8            ((__u32) 0x00000002)
+#define SND_AUDIOMODE_REALAUDIO_10           ((__u32) 0x00000004)
+#define SND_AUDIOMODE_REALAUDIO_SURROUND     ((__u32) 0x00000008)
+
+#define SND_AUDIOPROFILE_VORBIS              ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_VORBIS                 ((__u32) 0x00000001)
+
+#define SND_AUDIOPROFILE_FLAC                ((__u32) 0x00000001)
+
+/*
+ * Define quality levels for FLAC encoders, from LEVEL0 (fast)
+ * to LEVEL8 (best)
+ */
+#define SND_AUDIOMODE_FLAC_LEVEL0            ((__u32) 0x00000001)
+#define SND_AUDIOMODE_FLAC_LEVEL1            ((__u32) 0x00000002)
+#define SND_AUDIOMODE_FLAC_LEVEL2            ((__u32) 0x00000004)
+#define SND_AUDIOMODE_FLAC_LEVEL3            ((__u32) 0x00000008)
+#define SND_AUDIOMODE_FLAC_LEVEL4            ((__u32) 0x00000010)
+#define SND_AUDIOMODE_FLAC_LEVEL5            ((__u32) 0x00000020)
+#define SND_AUDIOMODE_FLAC_LEVEL6            ((__u32) 0x00000040)
+#define SND_AUDIOMODE_FLAC_LEVEL7            ((__u32) 0x00000080)
+#define SND_AUDIOMODE_FLAC_LEVEL8            ((__u32) 0x00000100)
+
+#define SND_AUDIOSTREAMFORMAT_FLAC           ((__u32) 0x00000001)
+#define SND_AUDIOSTREAMFORMAT_FLAC_OGG       ((__u32) 0x00000002)
+
+/* IEC61937 payloads without CUVP and preambles */
+#define SND_AUDIOPROFILE_IEC61937            ((__u32) 0x00000001)
+/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */
+#define SND_AUDIOPROFILE_IEC61937_SPDIF      ((__u32) 0x00000002)
+
+/*
+ * IEC modes are mandatory for decoders. Format autodetection
+ * will only happen on the DSP side with mode 0. The PCM mode should
+ * not be used, the PCM codec should be used instead.
+ */
+#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER  ((__u32) 0x00000000)
+#define SND_AUDIOMODE_IEC_LPCM		     ((__u32) 0x00000001)
+#define SND_AUDIOMODE_IEC_AC3		     ((__u32) 0x00000002)
+#define SND_AUDIOMODE_IEC_MPEG1		     ((__u32) 0x00000004)
+#define SND_AUDIOMODE_IEC_MP3		     ((__u32) 0x00000008)
+#define SND_AUDIOMODE_IEC_MPEG2		     ((__u32) 0x00000010)
+#define SND_AUDIOMODE_IEC_AACLC		     ((__u32) 0x00000020)
+#define SND_AUDIOMODE_IEC_DTS		     ((__u32) 0x00000040)
+#define SND_AUDIOMODE_IEC_ATRAC		     ((__u32) 0x00000080)
+#define SND_AUDIOMODE_IEC_SACD		     ((__u32) 0x00000100)
+#define SND_AUDIOMODE_IEC_EAC3		     ((__u32) 0x00000200)
+#define SND_AUDIOMODE_IEC_DTS_HD	     ((__u32) 0x00000400)
+#define SND_AUDIOMODE_IEC_MLP		     ((__u32) 0x00000800)
+#define SND_AUDIOMODE_IEC_DST		     ((__u32) 0x00001000)
+#define SND_AUDIOMODE_IEC_WMAPRO	     ((__u32) 0x00002000)
+#define SND_AUDIOMODE_IEC_REF_CXT            ((__u32) 0x00004000)
+#define SND_AUDIOMODE_IEC_HE_AAC	     ((__u32) 0x00008000)
+#define SND_AUDIOMODE_IEC_HE_AAC2	     ((__u32) 0x00010000)
+#define SND_AUDIOMODE_IEC_MPEG_SURROUND	     ((__u32) 0x00020000)
+
+#define SND_AUDIOPROFILE_G723_1              ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_G723_1_ANNEX_A         ((__u32) 0x00000001)
+#define SND_AUDIOMODE_G723_1_ANNEX_B         ((__u32) 0x00000002)
+#define SND_AUDIOMODE_G723_1_ANNEX_C         ((__u32) 0x00000004)
+
+#define SND_AUDIOPROFILE_G729                ((__u32) 0x00000001)
+
+#define SND_AUDIOMODE_G729_ANNEX_A           ((__u32) 0x00000001)
+#define SND_AUDIOMODE_G729_ANNEX_B           ((__u32) 0x00000002)
+
+/* <FIXME: multichannel encoders aren't supported for now. Would need
+   an additional definition of channel arrangement> */
+
+/* VBR/CBR definitions */
+#define SND_RATECONTROLMODE_CONSTANTBITRATE  ((__u32) 0x00000001)
+#define SND_RATECONTROLMODE_VARIABLEBITRATE  ((__u32) 0x00000002)
+
+/* Encoder options */
+
+struct snd_enc_wma {
+	__u32 super_block_align; /* WMA Type-specific data */
+};
+
+
+/**
+ * struct snd_enc_vorbis
+ * @quality: Sets encoding quality to n, between -1 (low) and 10 (high).
+ * In the default mode of operation, the quality level is 3.
+ * Normal quality range is 0 - 10.
+ * @managed: Boolean. Set  bitrate  management  mode. This turns off the
+ * normal VBR encoding, but allows hard or soft bitrate constraints to be
+ * enforced by the encoder. This mode can be slower, and may also be
+ * lower quality. It is primarily useful for streaming.
+ * @max_bit_rate: Enabled only if managed is TRUE
+ * @min_bit_rate: Enabled only if managed is TRUE
+ * @downmix: Boolean. Downmix input from stereo to mono (has no effect on
+ * non-stereo streams). Useful for lower-bitrate encoding.
+ *
+ * These options were extracted from the OpenMAX IL spec and Gstreamer vorbisenc
+ * properties
+ *
+ * For best quality users should specify VBR mode and set quality levels.
+ */
+
+struct snd_enc_vorbis {
+	__s32 quality;
+	__u32 managed;
+	__u32 max_bit_rate;
+	__u32 min_bit_rate;
+	__u32 downmix;
+};
+
+
+/**
+ * struct snd_enc_real
+ * @quant_bits: number of coupling quantization bits in the stream
+ * @start_region: coupling start region in the stream
+ * @num_regions: number of regions value
+ *
+ * These options were extracted from the OpenMAX IL spec
+ */
+
+struct snd_enc_real {
+	__u32 quant_bits;
+	__u32 start_region;
+	__u32 num_regions;
+};
+
+/**
+ * struct snd_enc_flac
+ * @num: serial number, valid only for OGG formats
+ *	needs to be set by application
+ * @gain: Add replay gain tags
+ *
+ * These options were extracted from the FLAC online documentation
+ * at http://flac.sourceforge.net/documentation_tools_flac.html
+ *
+ * To make the API simpler, it is assumed that the user will select quality
+ * profiles. Additional options that affect encoding quality and speed can
+ * be added at a later stage if needed.
+ *
+ * By default the Subset format is used by encoders.
+ *
+ * TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are
+ * not supported in this API.
+ */
+
+struct snd_enc_flac {
+	__u32 num;
+	__u32 gain;
+};
+
+struct snd_enc_generic {
+	__u32 bw;	/* encoder bandwidth */
+	__s32 reserved[15];
+};
+
+union snd_codec_options {
+	struct snd_enc_wma wma;
+	struct snd_enc_vorbis vorbis;
+	struct snd_enc_real real;
+	struct snd_enc_flac flac;
+	struct snd_enc_generic generic;
+};
+
+/** struct snd_codec_desc - description of codec capabilities
+ * @max_ch: Maximum number of audio channels
+ * @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this
+ * @bit_rate: Indexed array containing supported bit rates
+ * @num_bitrates: Number of valid values in bit_rate array
+ * @rate_control: value is specified by SND_RATECONTROLMODE defines.
+ * @profiles: Supported profiles. See SND_AUDIOPROFILE defines.
+ * @modes: Supported modes. See SND_AUDIOMODE defines
+ * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines
+ * @min_buffer: Minimum buffer size handled by codec implementation
+ * @reserved: reserved for future use
+ *
+ * This structure provides a scalar value for profiles, modes and stream
+ * format fields.
+ * If an implementation supports multiple combinations, they will be listed as
+ * codecs with different descriptors, for example there would be 2 descriptors
+ * for AAC-RAW and AAC-ADTS.
+ * This entails some redundancy but makes it easier to avoid invalid
+ * configurations.
+ *
+ */
+
+struct snd_codec_desc {
+	__u32 max_ch;
+	__u32 sample_rates;
+	__u32 bit_rate[MAX_NUM_BITRATES];
+	__u32 num_bitrates;
+	__u32 rate_control;
+	__u32 profiles;
+	__u32 modes;
+	__u32 formats;
+	__u32 min_buffer;
+	__u32 reserved[15];
+};
+
+/** struct snd_codec
+ * @id: Identifies the supported audio encoder/decoder.
+ *		See SND_AUDIOCODEC macros.
+ * @ch_in: Number of input audio channels
+ * @ch_out: Number of output channels. In case of contradiction between
+ *		this field and the channelMode field, the channelMode field
+ *		overrides.
+ * @sample_rate: Audio sample rate of input data
+ * @bit_rate: Bitrate of encoded data. May be ignored by decoders
+ * @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines.
+ *               Encoders may rely on profiles for quality levels.
+ *		 May be ignored by decoders.
+ * @profile: Mandatory for encoders, can be mandatory for specific
+ *		decoders as well. See SND_AUDIOPROFILE defines.
+ * @level: Supported level (Only used by WMA at the moment)
+ * @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines
+ * @format: Format of encoded bistream. Mandatory when defined.
+ *		See SND_AUDIOSTREAMFORMAT defines.
+ * @align: Block alignment in bytes of an audio sample.
+ *		Only required for PCM or IEC formats.
+ * @options: encoder-specific settings
+ * @reserved: reserved for future use
+ */
+
+struct snd_codec {
+	__u32 id;
+	__u32 ch_in;
+	__u32 ch_out;
+	__u32 sample_rate;
+	__u32 bit_rate;
+	__u32 rate_control;
+	__u32 profile;
+	__u32 level;
+	__u32 ch_mode;
+	__u32 format;
+	__u32 align;
+	union snd_codec_options options;
+	__u32 reserved[3];
+};
+
+#endif
diff --git a/include/sound/control.h b/include/sound/control.h
index 1a94a216ed9..b2796e83c7a 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -227,4 +227,12 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
 	return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
 }
 
+/*
+ * Helper functions for jack-detection controls
+ */
+struct snd_kcontrol *
+snd_kctl_jack_new(const char *name, int idx, void *private_data);
+void snd_kctl_jack_report(struct snd_card *card,
+			  struct snd_kcontrol *kctl, bool status);
+
 #endif	/* __SOUND_CONTROL_H */
diff --git a/include/sound/core.h b/include/sound/core.h
index 3be5ab782b9..5ab255f196c 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -62,6 +62,7 @@ typedef int __bitwise snd_device_type_t;
 #define	SNDRV_DEV_BUS		((__force snd_device_type_t) 0x1007)
 #define	SNDRV_DEV_CODEC		((__force snd_device_type_t) 0x1008)
 #define	SNDRV_DEV_JACK          ((__force snd_device_type_t) 0x1009)
+#define	SNDRV_DEV_COMPRESS	((__force snd_device_type_t) 0x100A)
 #define	SNDRV_DEV_LOWLEVEL	((__force snd_device_type_t) 0x2000)
 
 typedef int __bitwise snd_device_state_t;
diff --git a/include/sound/info.h b/include/sound/info.h
index 5492cc40dc5..9ca1a493d37 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -72,7 +72,7 @@ struct snd_info_entry_ops {
 
 struct snd_info_entry {
 	const char *name;
-	mode_t mode;
+	umode_t mode;
 	long size;
 	unsigned short content;
 	union {
diff --git a/include/sound/minors.h b/include/sound/minors.h
index 8f764204a85..5978f9a8c8b 100644
--- a/include/sound/minors.h
+++ b/include/sound/minors.h
@@ -35,7 +35,7 @@
 #define SNDRV_MINOR_TIMER		33	/* SNDRV_MINOR_GLOBAL + 1 * 32 */
 
 #ifndef CONFIG_SND_DYNAMIC_MINORS
-						/* 2 - 3 (reserved) */
+#define SNDRV_MINOR_COMPRESS		2	/* 2 - 3 */
 #define SNDRV_MINOR_HWDEP		4	/* 4 - 7 */
 #define SNDRV_MINOR_RAWMIDI		8	/* 8 - 15 */
 #define SNDRV_MINOR_PCM_PLAYBACK	16	/* 16 - 23 */
@@ -49,6 +49,7 @@
 #define SNDRV_DEVICE_TYPE_PCM_CAPTURE	SNDRV_MINOR_PCM_CAPTURE
 #define SNDRV_DEVICE_TYPE_SEQUENCER	SNDRV_MINOR_SEQUENCER
 #define SNDRV_DEVICE_TYPE_TIMER		SNDRV_MINOR_TIMER
+#define SNDRV_DEVICE_TYPE_COMPRESS	SNDRV_MINOR_COMPRESS
 
 #else /* CONFIG_SND_DYNAMIC_MINORS */
 
@@ -60,6 +61,7 @@ enum {
 	SNDRV_DEVICE_TYPE_RAWMIDI,
 	SNDRV_DEVICE_TYPE_PCM_PLAYBACK,
 	SNDRV_DEVICE_TYPE_PCM_CAPTURE,
+	SNDRV_DEVICE_TYPE_COMPRESS,
 };
 
 #endif /* CONFIG_SND_DYNAMIC_MINORS */
diff --git a/include/sound/saif.h b/include/sound/saif.h
index d0e0de7984e..f22f3e16edf 100644
--- a/include/sound/saif.h
+++ b/include/sound/saif.h
@@ -10,7 +10,7 @@
 #define __SOUND_SAIF_H__
 
 struct mxs_saif_platform_data {
-	int (*init) (void);
-	int (*get_master_id) (unsigned int saif_id);
+	bool master_mode;	/* if true use master mode */
+	int master_id;		/* id of the master if in slave mode */
 };
 #endif
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index 9a155f9d0a1..9b1aacaa82f 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -78,4 +78,16 @@ struct sh_fsi_platform_info {
 	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 };
 
+/*
+ * for fsi-ak4642
+ */
+struct fsi_ak4642_info {
+	const char *name;
+	const char *card;
+	const char *cpu_dai;
+	const char *codec;
+	const char *platform;
+	int id;
+};
+
 #endif /* __SOUND_FSI_H */
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 17a4c17f19f..d26a9b78477 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -43,6 +43,9 @@
 	.num_kcontrols = 0}
 
 /* platform domain */
+#define SND_SOC_DAPM_SIGGEN(wname) \
+{	.id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
+	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
 #define SND_SOC_DAPM_INPUT(wname) \
 {	.id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
@@ -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 11cfb5953e0..0992dff5595 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 00000000000..8d93b0357a1
--- /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 cf7ccb76a8d..b310c5a3a95 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -11,8 +11,11 @@
 #ifndef __LINUX_SND_WM8903_H
 #define __LINUX_SND_WM8903_H
 
-/* Used to enable configuration of a GPIO to all zeros */
-#define WM8903_GPIO_NO_CONFIG 0x8000
+/*
+ * Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of
+ * zero in platform data means "don't touch this pin".
+ */
+#define WM8903_GPIO_CONFIG_ZERO 0x8000
 
 /*
  * R6 (0x06) - Mic Bias Control 0