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alsa.c File Reference

ALSA input and output: common code. More...

#include <alsa/asoundlib.h>
#include "avdevice.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "alsa.h"

Go to the source code of this file.

Macros

#define MAKE_REORDER_FUNC(NAME, TYPE, CHANNELS, LAYOUT, MAP)
 
#define MAKE_REORDER_FUNCS(CHANNELS, LAYOUT, MAP)
 
#define FORMAT_I8   0
 
#define FORMAT_I16   1
 
#define FORMAT_I32   2
 
#define FORMAT_F32   3
 
#define PICK_REORDER(layout)
 

Functions

static av_cold snd_pcm_format_t codec_id_to_pcm_format (int codec_id)
 
 MAKE_REORDER_FUNCS (5, out_50, out[0]=in[0];out[1]=in[1];out[2]=in[3];out[3]=in[4];out[4]=in[2];)
 
 MAKE_REORDER_FUNCS (6, out_51, out[0]=in[0];out[1]=in[1];out[2]=in[4];out[3]=in[5];out[4]=in[2];out[5]=in[3];)
 
 MAKE_REORDER_FUNCS (8, out_71, out[0]=in[0];out[1]=in[1];out[2]=in[4];out[3]=in[5];out[4]=in[2];out[5]=in[3];out[6]=in[6];out[7]=in[7];)
 
static av_cold int find_reorder_func (AlsaData *s, int codec_id, uint64_t layout, int out)
 
av_cold int ff_alsa_open (AVFormatContext *ctx, snd_pcm_stream_t mode, unsigned int *sample_rate, int channels, enum AVCodecID *codec_id)
 Open an ALSA PCM. More...
 
av_cold int ff_alsa_close (AVFormatContext *s1)
 Close the ALSA PCM. More...
 
int ff_alsa_xrun_recover (AVFormatContext *s1, int err)
 Try to recover from ALSA buffer underrun. More...
 
int ff_alsa_extend_reorder_buf (AlsaData *s, int min_size)
 
int ff_alsa_get_device_list (AVDeviceInfoList *device_list, snd_pcm_stream_t stream_type)
 

Detailed Description

ALSA input and output: common code.

Author
Luca Abeni ( lucabe72 email it )
Benoit Fouet ( benoit fouet free fr )
Nicolas George ( nicolas george normalesup org )

Definition in file alsa.c.

Macro Definition Documentation

#define MAKE_REORDER_FUNC (   NAME,
  TYPE,
  CHANNELS,
  LAYOUT,
  MAP 
)
Value:
static void alsa_reorder_ ## NAME ## _ ## LAYOUT(const void *in_v, \
void *out_v, \
int n) \
{ \
const TYPE *in = in_v; \
TYPE *out = out_v; \
\
while (n-- > 0) { \
MAP \
in += CHANNELS; \
out += CHANNELS; \
} \
}
_
#define _(x)
n
int n
Definition: avisynth_c.h:547
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Definition: audio_convert.c:194
out
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
Definition: audio_convert.c:194

Definition at line 65 of file alsa.c.

#define MAKE_REORDER_FUNCS (   CHANNELS,
  LAYOUT,
  MAP 
)
Value:
MAKE_REORDER_FUNC(int8, int8_t, CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(int16, int16_t, CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(int32, int32_t, CHANNELS, LAYOUT, MAP) \
MAKE_REORDER_FUNC(f32, float, CHANNELS, LAYOUT, MAP)
int32_t
int32_t
Definition: audio_convert.c:194
MAKE_REORDER_FUNC
#define MAKE_REORDER_FUNC(NAME, TYPE, CHANNELS, LAYOUT, MAP)
Definition: alsa.c:65

Definition at line 80 of file alsa.c.

#define FORMAT_I8   0

Definition at line 114 of file alsa.c.

Referenced by find_reorder_func().

#define FORMAT_I16   1

Definition at line 115 of file alsa.c.

Referenced by find_reorder_func().

#define FORMAT_I32   2

Definition at line 116 of file alsa.c.

Referenced by find_reorder_func().

#define FORMAT_F32   3

Definition at line 117 of file alsa.c.

Referenced by find_reorder_func().

#define PICK_REORDER (   layout)
Value:
switch(format) {\
case FORMAT_I8: s->reorder_func = alsa_reorder_int8_out_ ##layout; break;\
case FORMAT_I16: s->reorder_func = alsa_reorder_int16_out_ ##layout; break;\
case FORMAT_I32: s->reorder_func = alsa_reorder_int32_out_ ##layout; break;\
case FORMAT_F32: s->reorder_func = alsa_reorder_f32_out_ ##layout; break;\
}
s
const char * s
Definition: avisynth_c.h:631
FORMAT_F32
#define FORMAT_F32
Definition: alsa.c:117
FORMAT_I16
#define FORMAT_I16
Definition: alsa.c:115
FORMAT_I32
#define FORMAT_I32
Definition: alsa.c:116
FORMAT_I8
#define FORMAT_I8
Definition: alsa.c:114
layout
uint64_t layout
Definition: channel_layout.c:77

Definition at line 119 of file alsa.c.

Referenced by find_reorder_func().

Function Documentation

static av_cold snd_pcm_format_t codec_id_to_pcm_format ( int  codec_id)
static

Definition at line 38 of file alsa.c.

Referenced by ff_alsa_open().

MAKE_REORDER_FUNCS ( 5  ,
out_50  ,
out  [0] = in[0];out[1]=in[1];out[2]=in[3];out[3]=in[4];out[4]=in[2]; 
)
MAKE_REORDER_FUNCS ( 6  ,
out_51  ,
out  [0] = in[0];out[1]=in[1];out[2]=in[4];out[3]=in[5];out[4]=in[2];out[5]=in[3]; 
)
MAKE_REORDER_FUNCS ( 8  ,
out_71  ,
out  [0] = in[0];out[1]=in[1];out[2]=in[4];out[3]=in[5];out[4]=in[2];out[5]=in[3];out[6]=in[6];out[7]=in[7]; 
)
static av_cold int find_reorder_func ( AlsaData *  s,
int  codec_id,
uint64_t  layout,
int  out 
)
static

Definition at line 127 of file alsa.c.

Referenced by ff_alsa_open().

av_cold int ff_alsa_open ( AVFormatContext *  s,
snd_pcm_stream_t  mode,
unsigned int *  sample_rate,
int  channels,
enum AVCodecID *  codec_id 
)

Open an ALSA PCM.

Parameters
smedia file handle
modeeither SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
sample_ratein: requested sample rate; out: actually selected sample rate
channelsnumber of channels
codec_idin: requested AVCodecID or AV_CODEC_ID_NONE; out: actually selected AVCodecID, changed only if AV_CODEC_ID_NONE was requested
Returns
0 if OK, AVERROR_xxx on error

Definition at line 167 of file alsa.c.

Referenced by audio_read_header(), and audio_write_header().

av_cold int ff_alsa_close ( AVFormatContext *  s1)

Close the ALSA PCM.

Parameters
s1media file handle
Returns
0

Definition at line 299 of file alsa.c.

int ff_alsa_xrun_recover ( AVFormatContext *  s1,
int  err 
)

Try to recover from ALSA buffer underrun.

Parameters
s1media file handle
errerror code reported by the previous ALSA call
Returns
0 if OK, AVERROR_xxx on error

Definition at line 310 of file alsa.c.

Referenced by audio_read_packet(), and audio_write_packet().

int ff_alsa_extend_reorder_buf ( AlsaData *  s,
int  min_size 
)

Definition at line 331 of file alsa.c.

Referenced by audio_write_packet().

int ff_alsa_get_device_list ( AVDeviceInfoList *  device_list,
snd_pcm_stream_t  stream_type 
)

Definition at line 348 of file alsa.c.

Referenced by audio_get_device_list().

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