alsa-audio-common.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-audio.h"
Include dependency graph for alsa-audio-common.c:

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)
 

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-audio-common.c.

Macro Definition Documentation

#define FORMAT_F32   3

Definition at line 117 of file alsa-audio-common.c.

Referenced by find_reorder_func().

#define FORMAT_I16   1

Definition at line 115 of file alsa-audio-common.c.

Referenced by find_reorder_func().

#define FORMAT_I32   2

Definition at line 116 of file alsa-audio-common.c.

Referenced by find_reorder_func().

#define FORMAT_I8   0

Definition at line 114 of file alsa-audio-common.c.

Referenced by find_reorder_func().

#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) { \
in += CHANNELS; \
out += CHANNELS; \
} \
}
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
#define _(x)
Synth Windw Norm while(pin< pend)%Until the end...%---Analysis x_w
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;}return ac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> out

Definition at line 65 of file alsa-audio-common.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
#define MAKE_REORDER_FUNC(NAME, TYPE, CHANNELS, LAYOUT, MAP)

Definition at line 80 of file alsa-audio-common.c.

#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;\
}
const char * s
Definition: avisynth_c.h:668
#define FORMAT_I8
#define FORMAT_I32
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample format(the sample packing is implied by the sample format) and sample rate.The lists are not just lists
#define FORMAT_I16
#define FORMAT_F32
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFilterBuffer structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel layout

Definition at line 119 of file alsa-audio-common.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-audio-common.c.

Referenced by ff_alsa_open().

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-audio-common.c.

int ff_alsa_extend_reorder_buf ( AlsaData s,
int  min_size 
)

Definition at line 331 of file alsa-audio-common.c.

Referenced by audio_write_packet().

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-audio-common.c.

Referenced by audio_read_header(), and audio_write_header().

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-audio-common.c.

Referenced by audio_read_packet(), and audio_write_packet().

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-audio-common.c.

Referenced by ff_alsa_open().

MAKE_REORDER_FUNCS ( ,
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 ( ,
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 ( ,
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]; 
)