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audio_mix.c File Reference
#include <stdint.h>
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avresample.h"
#include "internal.h"
#include "audio_data.h"
#include "audio_mix.h"

Go to the source code of this file.

Data Structures

struct  AudioMix
 

Macros

#define MIX_FUNC_NAME(fmt, cfmt)   mix_any_ ## fmt ##_## cfmt ##_c
 
#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr)
 
#define GET_MATRIX_CONVERT(suffix, scale)
 
#define CONVERT_MATRIX(type, expr)
 

Functions

void ff_audio_mix_set_func (AudioMix *am, enum AVSampleFormat fmt, enum AVMixCoeffType coeff_type, int in_channels, int out_channels, int ptr_align, int samples_align, const char *descr, void *mix_func)
 Set mixing function if the parameters match. More...
 
 MIX_FUNC_GENERIC (S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum)))
 
static void mix_2_to_1_s16p_flt_c (int16_t **samples, float **matrix, int len, int out_ch, int in_ch)
 
static void mix_2_to_1_s16p_q8_c (int16_t **samples, int16_t **matrix, int len, int out_ch, int in_ch)
 
static void mix_1_to_2_fltp_flt_c (float **samples, float **matrix, int len, int out_ch, int in_ch)
 
static void mix_6_to_2_fltp_flt_c (float **samples, float **matrix, int len, int out_ch, int in_ch)
 
static void mix_2_to_6_fltp_flt_c (float **samples, float **matrix, int len, int out_ch, int in_ch)
 
static av_cold int mix_function_init (AudioMix *am)
 
AudioMix * ff_audio_mix_alloc (AVAudioResampleContext *avr)
 Allocate and initialize an AudioMix context. More...
 
void ff_audio_mix_free (AudioMix **am_p)
 Free an AudioMix context. More...
 
int ff_audio_mix (AudioMix *am, AudioData *src)
 Apply channel mixing to audio data using the current mixing matrix. More...
 
int ff_audio_mix_get_matrix (AudioMix *am, double *matrix, int stride)
 Get the current mixing matrix. More...
 
static void reduce_matrix (AudioMix *am, const double *matrix, int stride)
 
int ff_audio_mix_set_matrix (AudioMix *am, const double *matrix, int stride)
 Set the current mixing matrix. More...
 

Variables

static const char *const coeff_type_names [] = { "q8", "q15", "flt" }
 

Macro Definition Documentation

#define MIX_FUNC_NAME (   fmt,
  cfmt 
)    mix_any_ ## fmt ##_## cfmt ##_c

Definition at line 99 of file audio_mix.c.

Referenced by mix_function_init().

#define MIX_FUNC_GENERIC (   fmt,
  cfmt,
  stype,
  ctype,
  sumtype,
  expr 
)
Value:
static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \
int len, int out_ch, int in_ch) \
{ \
int i, in, out; \
stype temp[AVRESAMPLE_MAX_CHANNELS]; \
for (i = 0; i < len; i++) { \
for (out = 0; out < out_ch; out++) { \
sumtype sum = 0; \
for (in = 0; in < in_ch; in++) \
sum += samples[in][i] * matrix[out][in]; \
temp[out] = expr; \
} \
for (out = 0; out < out_ch; out++) \
samples[out][i] = temp[out]; \
} \
}
MIX_FUNC_NAME
#define MIX_FUNC_NAME(fmt, cfmt)
Definition: audio_mix.c:99
fmt
const char * fmt
Definition: avisynth_c.h:632
temp
else temp
Definition: vf_mcdeint.c:257
AVRESAMPLE_MAX_CHANNELS
#define AVRESAMPLE_MAX_CHANNELS
Definition: avresample.h:104
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
len
int len
Definition: vorbis_enc_data.h:452
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
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469

Definition at line 101 of file audio_mix.c.

#define GET_MATRIX_CONVERT (   suffix,
  scale 
)
Value:
if (!am->matrix_ ## suffix[0]) { \
av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \
return AVERROR(EINVAL); \
} \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
if (am->input_skip[i] || am->output_zero[o]) \
matrix[o * stride + i] = 0.0; \
else \
matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \
(scale); \
if (!am->input_skip[i]) \
i0++; \
} \
if (!am->output_zero[o]) \
o0++; \
}
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
suffix
const char * suffix
Definition: checkasm.c:75
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVERROR
#define AVERROR(e)
Definition: error.h:43
return
return
Definition: audio_convert.c:415
stride
GLint GLenum GLboolean GLsizei stride
Definition: opengl_enc.c:105
if
if(ret< 0)
Definition: vf_mcdeint.c:280
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469

Referenced by ff_audio_mix_get_matrix().

#define CONVERT_MATRIX (   type,
  expr 
)
Value:
am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \
am->in_matrix_channels * \
sizeof(*am->matrix_## type[0])); \
if (!am->matrix_## type[0]) \
return AVERROR(ENOMEM); \
for (o = 0, o0 = 0; o < am->out_channels; o++) { \
if (am->output_zero[o] || am->output_skip[o]) \
continue; \
if (o0 > 0) \
am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \
am->in_matrix_channels; \
for (i = 0, i0 = 0; i < am->in_channels; i++) { \
double v; \
if (am->input_skip[i] || am->output_zero[i]) \
continue; \
v = matrix[o * stride + i]; \
am->matrix_## type[o0][i0] = expr; \
i0++; \
} \
o0++; \
} \
am->matrix = (void **)am->matrix_## type;
v
float v
Definition: avf_showspectrum.c:96
AVERROR
#define AVERROR(e)
Definition: error.h:43
return
return
Definition: audio_convert.c:415
type
GLint GLenum type
Definition: opengl_enc.c:105
stride
GLint GLenum GLboolean GLsizei stride
Definition: opengl_enc.c:105
if
if(ret< 0)
Definition: vf_mcdeint.c:280
av_mallocz
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469

Referenced by ff_audio_mix_set_matrix().

Function Documentation

void ff_audio_mix_set_func ( AudioMix *  am,
enum AVSampleFormat  fmt,
enum AVMixCoeffType  coeff_type,
int  in_channels,
int  out_channels,
int  ptr_align,
int  samples_align,
const char *  descr,
void *  mix_func 
)

Set mixing function if the parameters match.

This compares the parameters of the mixing function to the parameters in the AudioMix context. If the parameters do not match, no changes are made to the active functions. If the parameters do match and the alignment is not constrained, the function is set as the generic mixing function. If the parameters match and the alignment is constrained, the function is set as the optimized mixing function.

Parameters
amAudioMix context
fmtinput/output sample format
coeff_typemixing coefficient type
in_channelsnumber of input channels, or 0 for any number of channels
out_channelsnumber of output channels, or 0 for any number of channels
ptr_alignbuffer pointer alignment, in bytes
samples_alignbuffer size alignment, in samples
descrfunction type description (e.g. "C" or "SSE")
mix_funcmixing function pointer

Definition at line 61 of file audio_mix.c.

Referenced by ff_audio_mix_init_x86(), and mix_function_init().

MIX_FUNC_GENERIC ( S16P  ,
FLT  ,
int16_t  ,
float  ,
float  ,
av_clip_int16(lrintf(sum))   
)

Definition at line 120 of file audio_mix.c.

static void mix_2_to_1_s16p_flt_c ( int16_t **  samples,
float **  matrix,
int  len,
int  out_ch,
int  in_ch 
)
static

Definition at line 148 of file audio_mix.c.

Referenced by mix_function_init().

static void mix_2_to_1_s16p_q8_c ( int16_t **  samples,
int16_t **  matrix,
int  len,
int  out_ch,
int  in_ch 
)
static

Definition at line 170 of file audio_mix.c.

Referenced by mix_function_init().

static void mix_1_to_2_fltp_flt_c ( float **  samples,
float **  matrix,
int  len,
int  out_ch,
int  in_ch 
)
static

Definition at line 192 of file audio_mix.c.

Referenced by mix_function_init().

static void mix_6_to_2_fltp_flt_c ( float **  samples,
float **  matrix,
int  len,
int  out_ch,
int  in_ch 
)
static

Definition at line 225 of file audio_mix.c.

Referenced by mix_function_init().

static void mix_2_to_6_fltp_flt_c ( float **  samples,
float **  matrix,
int  len,
int  out_ch,
int  in_ch 
)
static

Definition at line 259 of file audio_mix.c.

Referenced by mix_function_init().

static av_cold int mix_function_init ( AudioMix *  am)
static

Definition at line 285 of file audio_mix.c.

Referenced by ff_audio_mix_set_matrix().

AudioMix* ff_audio_mix_alloc ( AVAudioResampleContext *  avr)

Allocate and initialize an AudioMix context.

The parameters in the AVAudioResampleContext are used to initialize the AudioMix context.

Parameters
avrAVAudioResampleContext
Returns
newly-allocated AudioMix context.

Definition at line 341 of file audio_mix.c.

Referenced by avresample_open().

void ff_audio_mix_free ( AudioMix **  am_p)

Free an AudioMix context.

Definition at line 409 of file audio_mix.c.

Referenced by avresample_close().

int ff_audio_mix ( AudioMix *  am,
AudioData *  src 
)

Apply channel mixing to audio data using the current mixing matrix.

Definition at line 428 of file audio_mix.c.

Referenced by avresample_convert().

int ff_audio_mix_get_matrix ( AudioMix *  am,
double *  matrix,
int  stride 
)

Get the current mixing matrix.

Definition at line 483 of file audio_mix.c.

Referenced by avresample_get_matrix().

static void reduce_matrix ( AudioMix *  am,
const double *  matrix,
int  stride 
)
static

Definition at line 530 of file audio_mix.c.

Referenced by ff_audio_mix_set_matrix().

int ff_audio_mix_set_matrix ( AudioMix *  am,
const double *  matrix,
int  stride 
)

Set the current mixing matrix.

Definition at line 653 of file audio_mix.c.

Referenced by avresample_set_matrix(), and ff_audio_mix_alloc().

Variable Documentation

const char* const coeff_type_names[] = { "q8", "q15", "flt" }
static

Definition at line 31 of file audio_mix.c.

Referenced by ff_audio_mix_set_func(), and mix_function_init().

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