diff options
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/Kbuild | 2 | ||||
| -rw-r--r-- | include/sound/compress_driver.h | 167 | ||||
| -rw-r--r-- | include/sound/compress_offload.h | 161 | ||||
| -rw-r--r-- | include/sound/compress_params.h | 397 | ||||
| -rw-r--r-- | include/sound/control.h | 8 | ||||
| -rw-r--r-- | include/sound/core.h | 1 | ||||
| -rw-r--r-- | include/sound/info.h | 2 | ||||
| -rw-r--r-- | include/sound/minors.h | 4 | ||||
| -rw-r--r-- | include/sound/saif.h | 4 | ||||
| -rw-r--r-- | include/sound/sh_fsi.h | 12 | ||||
| -rw-r--r-- | include/sound/soc-dapm.h | 5 | ||||
| -rw-r--r-- | include/sound/soc.h | 27 | ||||
| -rw-r--r-- | include/sound/sta32x.h | 35 | ||||
| -rw-r--r-- | include/sound/wm8903.h | 7 |
14 files changed, 818 insertions, 14 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/info.h b/include/sound/info.h index 5492cc40dc57..9ca1a493d370 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 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/saif.h b/include/sound/saif.h index d0e0de7984ec..f22f3e16edf4 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 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 |