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Merge branch 'topic/misc' into for-linus

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This commit is contained in:
Takashi Iwai
2012-01-12 09:59:14 +01:00
parent 29abceb67f 8c3f5d8a9b
commit 627b79628f
172 changed files with 2777 additions and 915 deletions
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include/sound/Kbuild
+2
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@@ -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
include/sound/compress_driver.h
+167
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@@ -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
include/sound/compress_offload.h
+161
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@@ -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
include/sound/compress_params.h
+397
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@@ -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
include/sound/core.h
+1
View File
@@ -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;
include/sound/minors.h
+3 -1
View File
@@ -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 */