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** Copyright 2010, The Android Open-Source Project
** Copyright (c) 2011-2013, The Linux Foundation. All rights reserved.
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** See the License for the specific language governing permissions and
** limitations under the License.
#ifndef _AUDIO_H_
#define _AUDIO_H_
#include <sound/asound.h>
#define PCM_ERROR_MAX 128
struct pcm {
int fd;
int timer_fd;
unsigned rate;
unsigned channels;
unsigned flags;
unsigned format;
unsigned running:1;
int underruns;
unsigned buffer_size;
unsigned period_size;
unsigned period_cnt;
char error[PCM_ERROR_MAX];
struct snd_pcm_hw_params *hw_p;
struct snd_pcm_sw_params *sw_p;
struct snd_pcm_sync_ptr *sync_ptr;
struct snd_pcm_channel_info ch[2];
void *addr;
int card_no;
int device_no;
int start;
#define PCM_OUT 0x00000000
#define PCM_IN 0x10000000
#define PCM_STEREO 0x00000000
#define PCM_MONO 0x01000000
#define PCM_QUAD 0x02000000
#define PCM_5POINT1 0x04000000
#define PCM_7POINT1 0x08000000
#define PCM_44100HZ 0x00000000
#define PCM_48000HZ 0x00100000
#define PCM_8000HZ 0x00200000
#define PCM_RATE_MASK 0x00F00000
#define PCM_MMAP 0x00010000
#define PCM_NMMAP 0x00000000
#define DEBUG_ON 0x00000001
#define DEBUG_OFF 0x00000000
#define PCM_PERIOD_SZ_MIN 128
/* Acquire/release a pcm channel.
* Returns non-zero on error
struct mixer_ctl {
struct mixer *mixer;
struct snd_ctl_elem_info *info;
char **ename;
#define __snd_alloca(ptr,type) do { *ptr = (type *) alloca(sizeof(type)); memset(*ptr, 0, sizeof(type)); } while (0)
#define snd_ctl_elem_id_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_id)
#define snd_ctl_card_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_card_info)
#define snd_ctl_event_alloca(ptr) __snd_alloca(ptr, snd_ctl_event)
#define snd_ctl_elem_list_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_list)
#define snd_ctl_elem_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_info)
#define snd_ctl_elem_value_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_value)
enum snd_pcm_stream_t {
/** Playback stream */
/** Capture stream */
enum _snd_ctl_elem_iface {
/** Card level */
/** Hardware dependent device */
/** Mixer */
/** PCM */
/** RawMidi */
/** Timer */
/** Sequencer */
struct mixer {
int fd;
struct snd_ctl_elem_info *info;
struct mixer_ctl *ctl;
unsigned count;
int get_format(const char* name);
const char *get_format_name(int format);
const char *get_format_desc(int format);
struct pcm *pcm_open(unsigned flags, char *device);
int pcm_close(struct pcm *pcm);
int pcm_ready(struct pcm *pcm);
int mmap_buffer(struct pcm *pcm);
u_int8_t *dst_address(struct pcm *pcm);
int sync_ptr(struct pcm *pcm);
void param_init(struct snd_pcm_hw_params *p);
void param_set_mask(struct snd_pcm_hw_params *p, int n, unsigned bit);
void param_set_min(struct snd_pcm_hw_params *p, int n, unsigned val);
void param_set_int(struct snd_pcm_hw_params *p, int n, unsigned val);
void param_set_max(struct snd_pcm_hw_params *p, int n, unsigned val);
int param_set_hw_refine(struct pcm *pcm, struct snd_pcm_hw_params *params);
int param_set_hw_params(struct pcm *pcm, struct snd_pcm_hw_params *params);
int param_set_sw_params(struct pcm *pcm, struct snd_pcm_sw_params *sparams);
int get_compressed_format(const char *format);
void param_dump(struct snd_pcm_hw_params *p);
int pcm_prepare(struct pcm *pcm);
long pcm_avail(struct pcm *pcm);
int pcm_set_channel_map(struct pcm *pcm, struct mixer *mixer,
int max_channels, char *chmap);
/* Returns a human readable reason for the last error. */
const char *pcm_error(struct pcm *pcm);
/* Returns the buffer size (int bytes) that should be used for pcm_write.
* This will be 1/2 of the actual fifo size.
int pcm_buffer_size(struct snd_pcm_hw_params *params);
int pcm_period_size(struct snd_pcm_hw_params *params);
/* Write data to the fifo.
* Will start playback on the first write or on a write that
* occurs after a fifo underrun.
int pcm_write(struct pcm *pcm, void *data, unsigned count);
int pcm_read(struct pcm *pcm, void *data, unsigned count);
struct mixer;
struct mixer_ctl;
struct mixer *mixer_open(const char *device);
void mixer_close(struct mixer *mixer);
void mixer_dump(struct mixer *mixer);
struct mixer_ctl *mixer_get_control(struct mixer *mixer,
const char *name, unsigned index);
struct mixer_ctl *mixer_get_nth_control(struct mixer *mixer, unsigned n);
int mixer_ctl_set(struct mixer_ctl *ctl, unsigned percent);
int mixer_ctl_select(struct mixer_ctl *ctl, const char *value);
void mixer_ctl_get(struct mixer_ctl *ctl, unsigned *value);
void mixer_ctl_get_mulvalues(struct mixer_ctl *ctl, unsigned **value, unsigned *count);
int mixer_ctl_set_value(struct mixer_ctl *ctl, int count, char ** argv);
#define MAX_NUM_CODECS 32
/* compressed audio support */
#define MAX_NUM_CODECS 32
/* compressed TX */
#define Q6_AC3_DECODER 0x00010BF6
#define Q6_EAC3_DECODER 0x00010C3C
#define Q6_DTS 0x00010D88
#define Q6_DTS_LBR 0x00010DBB
/* 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)
#define SND_AUDIOCODEC_AC3 ((__u32) 0x0000000E)
#define SND_AUDIOCODEC_DTS ((__u32) 0x0000000F)
#define SND_AUDIOCODEC_AC3_PASS_THROUGH ((__u32) 0x00000010)
#define SND_AUDIOCODEC_WMA_PRO ((__u32) 0x00000011)
#define SND_AUDIOCODEC_DTS_PASS_THROUGH ((__u32) 0x00000012)
#define SND_AUDIOCODEC_DTS_LBR ((__u32) 0x00000013)
#define SND_AUDIOCODEC_DTS_TRANSCODE_LOOPBACK ((__u32) 0x00000014)
#define SND_AUDIOCODEC_PASS_THROUGH ((__u32) 0x00000015)
#define SND_AUDIOCODEC_MP2 ((__u32) 0x00000016)
#define SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH ((__u32) 0x00000017)
* 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 */
/* Encoder options */
struct snd_enc_wma {
__u32 super_block_align; /* WMA Type-specific data */
__u32 bits_per_sample;
__u32 channelmask;
__u32 encodeopt;
__u32 encodeopt1;
__u32 encodeopt2;
* 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
* 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];
struct snd_dec_dts {
__u32 modelIdLength;
__u8 *modelId;
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_dec_dts dts;
/** 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.
* @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;
__u32 transcode_dts;
struct snd_dec_dts dts;
union snd_codec_options options;
__u32 reserved[3];
* 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;
uint64_t timestamp;
* 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 {
* 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];
* struct snd_compr_audio_info: compressed input audio information
* @frame_size: legth of the encoded frame with valid data
* @reserved: reserved for furture use
struct snd_compr_audio_info {
uint32_t frame_size;
uint32_t reserved[15];
* 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
#define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x00, struct snd_compr_caps *)
#define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x01, struct snd_compr_codec_caps *)
#define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x02, struct snd_compr_params *)
#define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x03, struct snd_compr_params *)
#define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x10, struct snd_compr_tstamp *)
#define SNDRV_COMPRESS_AVAIL _IOR('C', 0x11, struct snd_compr_avail *)
#define SNDRV_COMPRESS_PAUSE _IO('C', 0x20)
#define SNDRV_COMPRESS_RESUME _IO('C', 0x21)
#define SNDRV_COMPRESS_START _IO('C', 0x22)
#define SNDRV_COMPRESS_STOP _IO('C', 0x23)
#define SNDRV_COMPRESS_DRAIN _IO('C', 0x24)
* 1. add mmap support
#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */