FFmpeg  4.3
Data Structures | Macros | Functions | Variables
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)
 
AudioMixff_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

◆ MIX_FUNC_NAME

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

Definition at line 99 of file audio_mix.c.

◆ MIX_FUNC_GENERIC

#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; \
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]; \
} \
}

Definition at line 101 of file audio_mix.c.

◆ GET_MATRIX_CONVERT

#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++; \
}

◆ CONVERT_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;

Function Documentation

◆ ff_audio_mix_set_func()

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()

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

Definition at line 120 of file audio_mix.c.

◆ mix_2_to_1_s16p_flt_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().

◆ mix_2_to_1_s16p_q8_c()

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().

◆ mix_1_to_2_fltp_flt_c()

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().

◆ mix_6_to_2_fltp_flt_c()

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().

◆ mix_2_to_6_fltp_flt_c()

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().

◆ 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().

◆ ff_audio_mix_alloc()

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().

◆ ff_audio_mix_free()

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().

◆ ff_audio_mix()

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().

◆ ff_audio_mix_get_matrix()

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().

◆ reduce_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().

◆ 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

◆ coeff_type_names

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().

stride
int stride
Definition: mace.c:144
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame 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 they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
out
FILE * out
Definition: movenc.c:54
AVRESAMPLE_MAX_CHANNELS
#define AVRESAMPLE_MAX_CHANNELS
Definition: avresample.h:104
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
MIX_FUNC_NAME
#define MIX_FUNC_NAME(fmt, cfmt)
Definition: audio_mix.c:99
type
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
Definition: writing_filters.txt:86
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
in
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=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;} 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_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);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_log(ac->avr, AV_LOG_TRACE, "%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-> in
Definition: audio_convert.c:326
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
len
int len
Definition: vorbis_enc_data.h:452
suffix
const char * suffix
Definition: checkasm.c:198
temp
else temp
Definition: vf_mcdeint.c:256