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PL286/hardware/qcom/audio/legacy/libalsa-intf/alsa_audio.h

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/*
** Copyright 2010, The Android Open-Source Project
** Copyright (c) 2011-2012, Code Aurora Forum. 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
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** 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;
};
enum decoder_alias {
FORMAT_MP3,
FORMAT_AC3_PASS_THROUGH = 2,
};
#define FORMAT(v) SNDRV_PCM_FORMAT_##v
#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_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_CNT_MIN 2
#define PCM_PERIOD_CNT_SHIFT 16
#define PCM_PERIOD_CNT_MASK (0xF << PCM_PERIOD_CNT_SHIFT)
#define PCM_PERIOD_SZ_MIN 128
#define PCM_PERIOD_SZ_SHIFT 12
#define PCM_PERIOD_SZ_MASK (0xF << PCM_PERIOD_SZ_SHIFT)
#define TIMEOUT_INFINITE -1
/* 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 */
SND_PCM_STREAM_PLAYBACK = 0,
/** Capture stream */
SND_PCM_STREAM_CAPTURE,
SND_PCM_STREAM_LAST = SND_PCM_STREAM_CAPTURE
};
enum _snd_ctl_elem_iface {
/** Card level */
SND_CTL_ELEM_IFACE_CARD = 0,
/** Hardware dependent device */
SND_CTL_ELEM_IFACE_HWDEP,
/** Mixer */
SND_CTL_ELEM_IFACE_MIXER,
/** PCM */
SND_CTL_ELEM_IFACE_PCM,
/** RawMidi */
SND_CTL_ELEM_IFACE_RAWMIDI,
/** Timer */
SND_CTL_ELEM_IFACE_TIMER,
/** Sequencer */
SND_CTL_ELEM_IFACE_SEQUENCER,
SND_CTL_ELEM_IFACE_LAST = SND_CTL_ELEM_IFACE_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);
void param_dump(struct snd_pcm_hw_params *p);
int pcm_prepare(struct pcm *pcm);
long pcm_avail(struct pcm *pcm);
/* 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);
int mixer_ctl_set_value(struct mixer_ctl *ctl, int count, char ** argv);
#define MAX_NUM_CODECS 32
/* compressed audio support */
#ifdef QCOM_COMPRESSED_AUDIO_ENABLED
struct snd_compr_caps {
__u32 num_codecs;
__u32 min_fragment_size;
__u32 max_fragment_size;
__u32 min_fragments;
__u32 max_fragments;
__u32 codecs[MAX_NUM_CODECS];
__u32 reserved[11];
};
struct snd_enc_wma {
__u32 super_block_align; /* WMA Type-specific data */
__u32 bits_per_sample;
__u32 channelmask;
__u32 encodeopt;
};
struct snd_enc_vorbis {
int quality;
__u32 managed;
__u32 max_bit_rate;
__u32 min_bit_rate;
__u32 downmix;
};
struct snd_enc_real {
__u32 quant_bits;
__u32 start_region;
__u32 num_regions;
};
struct snd_enc_flac {
__u32 num;
__u32 gain;
};
struct snd_enc_generic {
__u32 bw; /* encoder bandwidth */
int 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 {
__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];
};
struct snd_compressed_buffer {
size_t fragment_size;
int fragments;
};
/* */
struct snd_compr_params {
struct snd_compressed_buffer buffer;
struct snd_codec codec;
};
struct snd_compr_tstamp {
size_t copied_bytes;
size_t copied_total;
size_t decoded;
size_t rendered;
__u32 sampling_rate;
uint64_t timestamp;
};
#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)
#endif
#endif