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Add libswresample.

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Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
pull/2/head
Michael Niedermayer 13 years ago
parent 53e37840bf
commit b5875b9111
18 changed files with 1634 additions and 67 deletions
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  1. 1
      Makefile
  2. 2
      common.mak
  3. 7
      configure
  4. 83
      ffmpeg.c
  5. 12
      libswresample/Makefile
  6. 113
      libswresample/audioconvert.c
  7. 65
      libswresample/audioconvert.h
  8. 271
      libswresample/rematrix.c
  9. 38
      libswresample/rematrix_template.c
  10. 352
      libswresample/resample2.c
  11. 432
      libswresample/swresample.c
  12. 79
      libswresample/swresample.h
  13. 77
      libswresample/swresample_internal.h
  14. 149
      libswresample/swresample_test.c
  15. 6
      tests/ref/acodec/g726
  16. 6
      tests/ref/acodec/pcm
  17. 4
      tests/ref/lavf/dv_fmt
  18. 4
      tests/ref/lavf/mxf_d10

1
Makefile
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@ -33,6 +33,7 @@ FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
FFLIBS := avutil

2
common.mak
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@ -20,7 +20,7 @@ $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/

7
configure vendored
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@ -89,6 +89,7 @@ Configuration options:
--disable-avdevice disable libavdevice build
--disable-avcodec disable libavcodec build
--disable-avformat disable libavformat build
--disable-swresample disable libswresample build
--disable-swscale disable libswscale build
--disable-postproc disable libpostproc build
--disable-avfilter disable video filter support [no]
@ -1037,6 +1038,7 @@ CONFIG_LIST="
small
sram
static
swresample
swscale
swscale_alpha
thumb
@ -1603,7 +1605,7 @@ avformat_deps="avcodec"
postproc_deps="gpl"
# programs
ffmpeg_deps="avcodec avformat swscale"
ffmpeg_deps="avcodec avformat swscale swresample"
ffmpeg_select="buffer_filter buffersink_filter"
avconv_deps="avcodec avformat swscale"
avconv_select="buffer_filter"
@ -1766,6 +1768,7 @@ enable postproc
enable protocols
enable static
enable stripping
enable swresample
enable swscale
enable swscale_alpha
@ -3143,7 +3146,7 @@ enabled extra_warnings && check_cflags -Winline
# add some linker flags
check_ldflags -Wl,--warn-common
check_ldflags -Wl,-rpath-link=libpostproc:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil
check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil
test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic
echo "X{};" > $TMPV

83
ffmpeg.c
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@ -45,6 +45,7 @@
#include "libavutil/avstring.h"
#include "libavutil/libm.h"
#include "libavformat/os_support.h"
#include "libswresample/swresample.h"
#include "libavformat/ffm.h" // not public API
@ -229,15 +230,14 @@ typedef struct OutputStream {
/* audio only */
int audio_resample;
ReSampleContext *resample; /* for audio resampling */
int resample_sample_fmt;
int resample_channels;
int resample_sample_rate;
int reformat_pair;
AVAudioConvert *reformat_ctx;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
FILE *logfile;
struct SwrContext *swr;
#if CONFIG_AVFILTER
AVFilterContext *output_video_filter;
AVFilterContext *input_video_filter;
@ -843,14 +843,15 @@ need_realloc:
exit_program(1);
}
if (enc->channels != dec->channels)
if (enc->channels != dec->channels
|| enc->sample_fmt != dec->sample_fmt)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->resample) || resample_changed) {
if ((ost->audio_resample && !ost->swr) || resample_changed) {
if (resample_changed) {
av_log(NULL, AV_LOG_INFO, "Input stream #%d.%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
ist->file_index, ist->st->index,
@ -859,24 +860,29 @@ need_realloc:
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_sample_rate = dec->sample_rate;
if (ost->resample)
audio_resample_close(ost->resample);
swr_free(&ost->swr);
}
/* if audio_sync_method is >1 the resampler is needed for audio drift compensation */
if (audio_sync_method <= 1 &&
ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_sample_rate == enc->sample_rate) {
ost->resample = NULL;
//ost->swr = NULL;
ost->audio_resample = 0;
} else {
if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
fprintf(stderr, "Warning, using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, dec->channels,
enc->sample_rate, dec->sample_rate,
enc->sample_fmt, dec->sample_fmt,
16, 10, 0, 0.8);
if (!ost->resample) {
ost->swr = swr_alloc2(ost->swr,
enc->channel_layout, enc->sample_fmt, enc->sample_rate,
dec->channel_layout, dec->sample_fmt, dec->sample_rate,
0, NULL);
av_set_int(ost->swr, "ich", dec->channels);
av_set_int(ost->swr, "och", enc->channels);
if(audio_sync_method>1) av_set_int(ost->swr, "flags", SWR_FLAG_RESAMPLE);
if(ost->swr && swr_init(ost->swr) < 0){
fprintf(stderr, "swr_init() failed\n");
swr_free(&ost->swr);
}
if (!ost->swr) {
fprintf(stderr, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
dec->channels, dec->sample_rate,
enc->channels, enc->sample_rate);
@ -885,21 +891,7 @@ need_realloc:
}
}
#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
if (ost->reformat_ctx)
av_audio_convert_free(ost->reformat_ctx);
ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1,
dec->sample_fmt, 1, NULL, 0);
if (!ost->reformat_ctx) {
fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
av_get_sample_fmt_name(dec->sample_fmt),
av_get_sample_fmt_name(enc->sample_fmt));
exit_program(1);
}
ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
}
av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt);
if(audio_sync_method){
double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
@ -941,7 +933,7 @@ need_realloc:
if(verbose > 2)
fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate);
// fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2));
av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
swr_compensate(ost->swr, comp, enc->sample_rate);
}
}
}else
@ -950,30 +942,15 @@ need_realloc:
if (ost->audio_resample) {
buftmp = audio_buf;
size_out = audio_resample(ost->resample,
(short *)buftmp, (short *)buf,
size / (dec->channels * isize));
size_out = swr_convert(ost->swr, ( uint8_t*[]){buftmp}, audio_buf_size / (enc->channels * osize),
(const uint8_t*[]){buf }, size / (dec->channels * isize));
size_out = size_out * enc->channels * osize;
} else {
buftmp = buf;
size_out = size;
}
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) {
const void *ibuf[6]= {buftmp};
void *obuf[6]= {audio_buf};
int istride[6]= {isize};
int ostride[6]= {osize};
int len= size_out/istride[0];
if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
printf("av_audio_convert() failed\n");
if (exit_on_error)
exit_program(1);
return;
}
buftmp = audio_buf;
size_out = len*osize;
}
av_assert0(ost->audio_resample || dec->sample_fmt==enc->sample_fmt);
/* now encode as many frames as possible */
if (enc->frame_size > 1) {
@ -2133,7 +2110,6 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
if (!ost->fifo) {
return AVERROR(ENOMEM);
}
ost->reformat_pair = MAKE_SFMT_PAIR(AV_SAMPLE_FMT_NONE,AV_SAMPLE_FMT_NONE);
if (!codec->sample_rate) {
codec->sample_rate = icodec->sample_rate;
}
@ -2149,6 +2125,8 @@ static int transcode_init(OutputFile *output_files, int nb_output_files,
if (av_get_channel_layout_nb_channels(codec->channel_layout) != codec->channels)
codec->channel_layout = 0;
ost->audio_resample = codec->sample_rate != icodec->sample_rate || audio_sync_method > 1;
ost->audio_resample |= codec->sample_fmt != icodec->sample_fmt
|| codec->channel_layout != icodec->channel_layout;
icodec->request_channels = codec->channels;
ist->decoding_needed = 1;
ost->encoding_needed = 1;
@ -2679,10 +2657,7 @@ static int transcode(OutputFile *output_files, int nb_output_files,
av_free(ost->forced_kf_pts);
if (ost->video_resample)
sws_freeContext(ost->img_resample_ctx);
if (ost->resample)
audio_resample_close(ost->resample);
if (ost->reformat_ctx)
av_audio_convert_free(ost->reformat_ctx);
swr_free(&ost->swr);
av_dict_free(&ost->opts);
}
}

12
libswresample/Makefile
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@ -0,0 +1,12 @@
include $(SUBDIR)../config.mak
NAME = swresample
FFLIBS = avutil
HEADERS = swresample.h
OBJS = swresample.o audioconvert.o resample2.o rematrix.o
TESTPROGS = swresample_test
include $(SUBDIR)../subdir.mak

113
libswresample/audioconvert.c
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@ -0,0 +1,113 @@
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avstring.h"
#include "libavutil/avassert.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "audioconvert.h"
struct AVAudioConvert {
int channels;
int fmt_pair;
};
AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int flags)
{
AVAudioConvert *ctx;
ctx = av_malloc(sizeof(AVAudioConvert));
if (!ctx)
return NULL;
ctx->channels = channels;
ctx->fmt_pair = out_fmt + AV_SAMPLE_FMT_NB*in_fmt;
return ctx;
}
void swr_audio_convert_free(AVAudioConvert **ctx)
{
av_freep(ctx);
}
int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData*in, int len)
{
int ch;
av_assert0(ctx->channels == out->ch_count);
//FIXME optimize common cases
for(ch=0; ch<ctx->channels; ch++){
const int is= (in ->planar ? 1 : in->ch_count) * in->bps;
const int os= (out->planar ? 1 :out->ch_count) *out->bps;
const uint8_t *pi= in ->ch[ch];
uint8_t *po= out->ch[ch];
uint8_t *end= po + os*len;
if(!po)
continue;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding ?
CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80)
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16)
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31)))
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;
}
return 0;
}

65
libswresample/audioconvert.h
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@ -0,0 +1,65 @@
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWR_AUDIOCONVERT_H
#define SWR_AUDIOCONVERT_H
/**
* @file
* Audio format conversion routines
*/
#include "swresample_internal.h"
#include "libavutil/cpu.h"
#include "libavutil/audioconvert.h"
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param in_fmt Input sample format
* @param channels Number of channels
* @param flags See AV_CPU_FLAG_xx
* @return NULL on error
*/
AVAudioConvert *swr_audio_convert_alloc(enum AVSampleFormat out_fmt,
enum AVSampleFormat in_fmt,
int channels, int flags);
/**
* Free audio sample format converter context.
* and set the pointer to NULL
*/
void swr_audio_convert_free(AVAudioConvert **ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] in array of input buffers for each channel
* @param len length of audio frame size (measured in samples)
*/
int swr_audio_convert(AVAudioConvert *ctx, AudioData *out, AudioData *in, int len);
#endif /* AVCODEC_AUDIOCONVERT_H */

271
libswresample/rematrix.c
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@ -0,0 +1,271 @@
/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "swresample_internal.h"
#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
#define SAMPLE float
#define RENAME(x) x ## _float
#include "rematrix_template.c"
#undef SAMPLE
#undef RENAME
#define SAMPLE int16_t
#define RENAME(x) x ## _s16
#include "rematrix_template.c"
#define FRONT_LEFT 0
#define FRONT_RIGHT 1
#define FRONT_CENTER 2
#define LOW_FREQUENCY 3
#define BACK_LEFT 4
#define BACK_RIGHT 5
#define FRONT_LEFT_OF_CENTER 6
#define FRONT_RIGHT_OF_CENTER 7
#define BACK_CENTER 8
#define SIDE_LEFT 9
#define SIDE_RIGHT 10
#define TOP_CENTER 11
#define TOP_FRONT_LEFT 12
#define TOP_FRONT_CENTER 13
#define TOP_FRONT_RIGHT 14
#define TOP_BACK_LEFT 15
#define TOP_BACK_CENTER 16
#define TOP_BACK_RIGHT 17
static int even(int64_t layout){
if(!layout) return 1;
if(layout&(layout-1)) return 1;
return 0;
}
static int sane_layout(int64_t layout){
if(!(layout & AV_CH_LAYOUT_SURROUND)) // at least 1 front speaker
return 0;
if(!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT))) // no asymetric front
return 0;
if(!even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT))) // no asymetric side
return 0;
if(!even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT)))
return 0;
if(!even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER)))
return 0;
if(av_get_channel_layout_nb_channels(layout) >= SWR_CH_MAX)
return 0;
return 1;
}
int swr_rematrix_init(SwrContext *s){
int i, j, in_i, out_i;
float matrix[64][64]={0};
int64_t unaccounted= s->in_ch_layout & ~s->out_ch_layout;
float maxcoef=0;
for(i=0; i<64; i++){
if(s->in_ch_layout & s->out_ch_layout & (1LL<<i))
matrix[i][i]= 1.0;
}
if(!sane_layout(s->in_ch_layout)){
av_log(s, AV_LOG_ERROR, "Input channel layout isnt supported\n");
return AVERROR(EINVAL);
}
if(!sane_layout(s->out_ch_layout)){
av_log(s, AV_LOG_ERROR, "Output channel layout isnt supported\n");
return AVERROR(EINVAL);
}
//FIXME implement dolby surround
//FIXME implement full ac3
if(unaccounted & AV_CH_FRONT_CENTER){
if((s->out_ch_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO){
matrix[ FRONT_LEFT][FRONT_CENTER]+= sqrt(0.5);
matrix[FRONT_RIGHT][FRONT_CENTER]+= sqrt(0.5);
}else
av_assert0(0);
}
if(unaccounted & AV_CH_LAYOUT_STEREO){
if(s->out_ch_layout & AV_CH_FRONT_CENTER){
matrix[FRONT_CENTER][ FRONT_LEFT]+= sqrt(0.5);
matrix[FRONT_CENTER][FRONT_RIGHT]+= sqrt(0.5);
if(s->in_ch_layout & AV_CH_FRONT_CENTER)
matrix[FRONT_CENTER][ FRONT_CENTER] = s->clev*sqrt(2);
}else
av_assert0(0);
}
if(unaccounted & AV_CH_BACK_CENTER){
if(s->out_ch_layout & AV_CH_BACK_LEFT){
matrix[ BACK_LEFT][BACK_CENTER]+= sqrt(0.5);
matrix[BACK_RIGHT][BACK_CENTER]+= sqrt(0.5);
}else if(s->out_ch_layout & AV_CH_SIDE_LEFT){
matrix[ SIDE_LEFT][BACK_CENTER]+= sqrt(0.5);
matrix[SIDE_RIGHT][BACK_CENTER]+= sqrt(0.5);
}else if(s->out_ch_layout & AV_CH_FRONT_LEFT){
matrix[ FRONT_LEFT][BACK_CENTER]+= s->slev*sqrt(0.5);
matrix[FRONT_RIGHT][BACK_CENTER]+= s->slev*sqrt(0.5);
}else if(s->out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][BACK_CENTER]+= s->slev*sqrt(0.5);
}else
av_assert0(0);
}
if(unaccounted & AV_CH_BACK_LEFT){
if(s->out_ch_layout & AV_CH_BACK_CENTER){
matrix[BACK_CENTER][ BACK_LEFT]+= sqrt(0.5);
matrix[BACK_CENTER][BACK_RIGHT]+= sqrt(0.5);
}else if(s->out_ch_layout & AV_CH_SIDE_LEFT){
if(s->in_ch_layout & AV_CH_SIDE_LEFT){
matrix[ SIDE_LEFT][ BACK_LEFT]+= sqrt(0.5);
matrix[SIDE_RIGHT][BACK_RIGHT]+= sqrt(0.5);
}else{
matrix[ SIDE_LEFT][ BACK_LEFT]+= 1.0;
matrix[SIDE_RIGHT][BACK_RIGHT]+= 1.0;
}
}else if(s->out_ch_layout & AV_CH_FRONT_LEFT){
matrix[ FRONT_LEFT][ BACK_LEFT]+= s->slev;
matrix[FRONT_RIGHT][BACK_RIGHT]+= s->slev;
}else if(s->out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][BACK_LEFT ]+= s->slev*sqrt(0.5);
matrix[ FRONT_CENTER][BACK_RIGHT]+= s->slev*sqrt(0.5);
}else
av_assert0(0);
}
if(unaccounted & AV_CH_SIDE_LEFT){
if(s->out_ch_layout & AV_CH_BACK_LEFT){
matrix[ BACK_LEFT][ SIDE_LEFT]+= 1.0;
matrix[BACK_RIGHT][SIDE_RIGHT]+= 1.0;
}else if(s->out_ch_layout & AV_CH_BACK_CENTER){
matrix[BACK_CENTER][ SIDE_LEFT]+= sqrt(0.5);
matrix[BACK_CENTER][SIDE_RIGHT]+= sqrt(0.5);
}else if(s->out_ch_layout & AV_CH_FRONT_LEFT){
matrix[ FRONT_LEFT][ SIDE_LEFT]+= s->slev;
matrix[FRONT_RIGHT][SIDE_RIGHT]+= s->slev;
}else if(s->out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][SIDE_LEFT ]+= s->slev*sqrt(0.5);
matrix[ FRONT_CENTER][SIDE_RIGHT]+= s->slev*sqrt(0.5);
}else
av_assert0(0);
}
if(unaccounted & AV_CH_FRONT_LEFT_OF_CENTER){
if(s->out_ch_layout & AV_CH_FRONT_LEFT){
matrix[ FRONT_LEFT][ FRONT_LEFT_OF_CENTER]+= 1.0;
matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER]+= 1.0;
}else if(s->out_ch_layout & AV_CH_FRONT_CENTER){
matrix[ FRONT_CENTER][ FRONT_LEFT_OF_CENTER]+= sqrt(0.5);
matrix[ FRONT_CENTER][FRONT_RIGHT_OF_CENTER]+= sqrt(0.5);
}else
av_assert0(0);
}
//FIXME quantize for integeres
for(out_i=i=0; i<64; i++){
double sum=0;
int in_i=0;
int ch_in=0;
for(j=0; j<64; j++){
s->matrix[out_i][in_i]= matrix[i][j];
if(matrix[i][j]){
s->matrix_ch[out_i][++ch_in]= in_i;
sum += fabs(matrix[i][j]);
}
if(s->in_ch_layout & (1ULL<<j))
in_i++;
}
s->matrix_ch[out_i][0]= ch_in;
maxcoef= FFMAX(maxcoef, sum);
if(s->out_ch_layout & (1ULL<<i))
out_i++;
}
if(( s->out_sample_fmt < AV_SAMPLE_FMT_FLT
|| s->int_sample_fmt < AV_SAMPLE_FMT_FLT) && maxcoef > 1.0){
for(i=0; i<SWR_CH_MAX; i++)
for(j=0; j<SWR_CH_MAX; j++)
s->matrix[i][j] /= maxcoef;
}
for(i=0; i<av_get_channel_layout_nb_channels(s->out_ch_layout); i++){
for(j=0; j<av_get_channel_layout_nb_channels(s->in_ch_layout); j++){
av_log(NULL, AV_LOG_ERROR, "%f ", s->matrix[i][j]);
}
av_log(NULL, AV_LOG_ERROR, "\n");
}
return 0;
}
int swr_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy){
int out_i, in_i, i, j;
av_assert0(out->ch_count == av_get_channel_layout_nb_channels(s->out_ch_layout));
av_assert0(in ->ch_count == av_get_channel_layout_nb_channels(s-> in_ch_layout));
for(out_i=0; out_i<out->ch_count; out_i++){
switch(s->matrix_ch[out_i][0]){
case 1:
in_i= s->matrix_ch[out_i][1];
if(mustcopy || s->matrix[out_i][in_i]!=1.0){
if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){
copy_float(out->ch[out_i], in->ch[in_i], s->matrix[out_i][in_i], len);
}else
copy_s16 (out->ch[out_i], in->ch[in_i], s->matrix[out_i][in_i], len);
}else{
out->ch[out_i]= in->ch[in_i];
}
break;
case 2:
if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){
sum2_float(out->ch[out_i], in->ch[ s->matrix_ch[out_i][1] ], in->ch[ s->matrix_ch[out_i][2] ],
s->matrix[out_i][ s->matrix_ch[out_i][1] ], s->matrix[out_i][ s->matrix_ch[out_i][2] ],
len);
}else{
sum2_s16 (out->ch[out_i], in->ch[ s->matrix_ch[out_i][1] ], in->ch[ s->matrix_ch[out_i][2] ],
s->matrix[out_i][ s->matrix_ch[out_i][1] ], s->matrix[out_i][ s->matrix_ch[out_i][2] ],
len);
}
break;
default:
if(s->int_sample_fmt == AV_SAMPLE_FMT_FLT){
for(i=0; i<len; i++){
float v=0;
for(j=0; j<s->matrix_ch[out_i][0]; j++){
in_i= s->matrix_ch[out_i][1+j];
v+= ((float*)in->ch[in_i])[i] * s->matrix[out_i][in_i];
}
((float*)out->ch[out_i])[i]= v;
}
}else{
for(i=0; i<len; i++){
int v=0;
for(j=0; j<s->matrix_ch[out_i][0]; j++){
in_i= s->matrix_ch[out_i][1+j];
v+= ((int16_t*)in->ch[in_i])[i] * s->matrix[out_i][in_i]; //FIXME use int16 coeffs
}
((int16_t*)out->ch[out_i])[i]= v;
}
}
}
}
return 0;
}

38
libswresample/rematrix_template.c
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/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
static void RENAME(sum2)(SAMPLE *out, const SAMPLE *in1, const SAMPLE *in2, float coeff1, float coeff2, int len){
int i;
for(i=0; i<len; i++)
out[i] = coeff1*in1[i] + coeff2*in2[i]; //FIXME better int16
}
static void RENAME(copy)(SAMPLE *out, const SAMPLE *in, float coeff, int len){
if(coeff == 1.0){
memcpy(out, in, sizeof(SAMPLE)*len);
}else{
int i;
for(i=0; i<len; i++)
out[i] = coeff*in[i]; //FIXME better int16
}
}

352
libswresample/resample2.c
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/*
* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/log.h"
#include "swresample_internal.h"
#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
#define FILTER_SHIFT 0
#define FELEM double
#define FELEM2 double
#define FELEML double
#define WINDOW_TYPE 24
#endif
typedef struct AVResampleContext{
const AVClass *av_class;
FELEM *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
double factor;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
static double bessel(double x){
double v=1;
double lastv=0;
double t=1;
int i;
static const double inv[100]={
1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),
1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),
1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),
1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),
1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),
1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),
1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),
1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),
1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),
1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)
};
x= x*x/4;
for(i=0; v != lastv; i++){
lastv=v;
t *= x*inv[i];
v += t;
}
return v;
}
/**
* builds a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
* @return 0 on success, negative on error
*/
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i;
double x, y, w;
double *tab = av_malloc(tap_count * sizeof(*tab));
const int center= (tap_count-1)/2;
if (!tab)
return AVERROR(ENOMEM);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
for(ph=0;ph<phase_count;ph++) {
double norm = 0;
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch(type){
case 0:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case 1:
w = 2.0*x / (factor*tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
break;
default:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
break;
}
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
#endif
}
}
#if 0
{
#define LEN 1024
int j,k;
double sine[LEN + tap_count];
double filtered[LEN];
double maxff=-2, minff=2, maxsf=-2, minsf=2;
for(i=0; i<LEN; i++){
double ss=0, sf=0, ff=0;
for(j=0; j<LEN+tap_count; j++)
sine[j]= cos(i*j*M_PI/LEN);
for(j=0; j<LEN; j++){
double sum=0;
ph=0;
for(k=0; k<tap_count; k++)
sum += filter[ph * tap_count + k] * sine[k+j];
filtered[j]= sum / (1<<FILTER_SHIFT);
ss+= sine[j + center] * sine[j + center];
ff+= filtered[j] * filtered[j];
sf+= sine[j + center] * filtered[j];
}
ss= sqrt(2*ss/LEN);
ff= sqrt(2*ff/LEN);
sf= 2*sf/LEN;
maxff= FFMAX(maxff, ff);
minff= FFMIN(minff, ff);
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
minff=minsf= 2;
maxff=maxsf= -2;
}
}
}
#endif
av_free(tab);
return 0;
}
AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
if(!c || c->phase_shift!=phase_shift || c->linear!=linear || c->factor != factor
|| c->filter_length!=FFMAX((int)ceil(filter_size/factor), 1)){
c= av_mallocz(sizeof(AVResampleContext));
if (!c)
return NULL;
c->phase_shift= phase_shift;
c->phase_mask= phase_count-1;
c->linear= linear;
c->factor= factor;
c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
if (!c->filter_bank)
goto error;
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
goto error;
memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
}
c->compensation_distance= 0;
c->src_incr= out_rate;
c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
c->index= -phase_count*((c->filter_length-1)/2);
c->frac= 0;
return c;
error:
av_free(c->filter_bank);
av_free(c);
return NULL;
}
void swr_resample_free(AVResampleContext **c){
if(!*c)
return;
av_freep(&(*c)->filter_bank);
av_freep(c);
}
void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
AVResampleContext *c= s->resample;
// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
c->compensation_distance= compensation_distance;
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
}
int swr_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int compensation_distance= c->compensation_distance;
if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
int64_t index2= ((int64_t)index)<<32;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
for(dst_index=0; dst_index < dst_size; dst_index++){
dst[dst_index] = src[index2>>32];
index2 += incr;
}
frac += dst_index * dst_incr_frac;
index += dst_index * dst_incr;
index += frac / c->src_incr;
frac %= c->src_incr;
}else{
for(dst_index=0; dst_index < dst_size; dst_index++){
FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
int sample_index= index >> c->phase_shift;
FELEM2 val=0;
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
}else if(c->linear){
FELEM2 v2=0;
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
}
val+=(v2-val)*(FELEML)frac / c->src_incr;
}else{
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
}
}
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
#endif
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
if(dst_index + 1 == compensation_distance){
compensation_distance= 0;
dst_incr_frac= c->ideal_dst_incr % c->src_incr;
dst_incr= c->ideal_dst_incr / c->src_incr;
}
}
}
*consumed= FFMAX(index, 0) >> c->phase_shift;
if(index>=0) index &= c->phase_mask;
if(compensation_distance){
compensation_distance -= dst_index;
assert(compensation_distance > 0);
}
if(update_ctx){
c->frac= frac;
c->index= index;
c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
c->compensation_distance= compensation_distance;
}
#if 0
if(update_ctx && !c->compensation_distance){
#undef rand
av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
}
#endif
return dst_index;
}
int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
int i, ret= -1;
for(i=0; i<dst->ch_count; i++){
ret= swr_resample(c, dst->ch[i], src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
}
return ret;
}

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/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "swresample_internal.h"
#include "audioconvert.h"
#include "libavutil/avassert.h"
#define C30DB M_SQRT2
#define C15DB 1.189207115
#define C__0DB 1.0
#define C_15DB 0.840896415
#define C_30DB M_SQRT1_2
#define C_45DB 0.594603558
#define C_60DB 0.5
//TODO split options array out?
#define OFFSET(x) offsetof(SwrContext,x)
static const AVOption options[]={
{"ich", "input channel count", OFFSET( in.ch_count ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
{"och", "output channel count", OFFSET(out.ch_count ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
{"isr", "input sample rate" , OFFSET( in_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
{"osr", "output sample rate" , OFFSET(out_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
{"ip" , "input planar" , OFFSET( in.planar ), FF_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
{"op" , "output planar" , OFFSET(out.planar ), FF_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
{"isf", "input sample format", OFFSET( in_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1, 0},
{"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1, 0},
{"tsf", "internal sample format", OFFSET(int_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
{"icl", "input channel layout" , OFFSET( in_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
{"ocl", "output channel layout", OFFSET(out_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
{"clev", "center mix level" , OFFSET(clev) , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
{"slev", "sourround mix level" , OFFSET(slev) , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
{"flags", NULL , OFFSET(flags) , FF_OPT_TYPE_FLAGS, {.dbl=0}, 0, UINT_MAX, 0, "flags"},
{"res", "force resampling", 0, FF_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
{0}
};
static const char* context_to_name(void* ptr) {
return "SWR";
}
static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
static int resample(SwrContext *s, AudioData *out_param, int out_count,
const AudioData * in_param, int in_count);
SwrContext *swr_alloc(void){
SwrContext *s= av_mallocz(sizeof(SwrContext));
if(s){
s->av_class= &av_class;
av_opt_set_defaults2(s, 0, 0);
}
return s;
}
SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx){
if(!s) s= swr_alloc();
if(!s) return NULL;
s->log_level_offset= log_offset;
s->log_ctx= log_ctx;
av_set_int(s, "ocl", out_ch_layout);
av_set_int(s, "osf", out_sample_fmt);
av_set_int(s, "osr", out_sample_rate);
av_set_int(s, "icl", in_ch_layout);
av_set_int(s, "isf", in_sample_fmt);
av_set_int(s, "isr", in_sample_rate);
s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
s->int_sample_fmt = AV_SAMPLE_FMT_S16;
return s;
}
static void free_temp(AudioData *a){
av_free(a->data);
memset(a, 0, sizeof(*a));
}
void swr_free(SwrContext **ss){
SwrContext *s= *ss;
if(s){
free_temp(&s->postin);
free_temp(&s->midbuf);
free_temp(&s->preout);
free_temp(&s->in_buffer);
swr_audio_convert_free(&s-> in_convert);
swr_audio_convert_free(&s->out_convert);
swr_resample_free(&s->resample);
}
av_freep(ss);
}
static int64_t guess_layout(int ch){
switch(ch){
case 1: return AV_CH_LAYOUT_MONO;
case 2: return AV_CH_LAYOUT_STEREO;
case 5: return AV_CH_LAYOUT_5POINT0;
case 6: return AV_CH_LAYOUT_5POINT1;
case 7: return AV_CH_LAYOUT_7POINT0;
case 8: return AV_CH_LAYOUT_7POINT1;
default: return 0;
}
}
int swr_init(SwrContext *s){
s->in_buffer_index= 0;
s->in_buffer_count= 0;
s->resample_in_constraint= 0;
free_temp(&s->postin);
free_temp(&s->midbuf);
free_temp(&s->preout);
free_temp(&s->in_buffer);
swr_audio_convert_free(&s-> in_convert);
swr_audio_convert_free(&s->out_convert);
//We assume AVOptions checked the various values and the defaults where allowed
if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
&&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
return AVERROR(EINVAL);
}
//FIXME should we allow/support using FLT on material that doesnt need it ?
if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
s->int_sample_fmt= AV_SAMPLE_FMT_S16;
}else
s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
s->resample = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
}else
swr_resample_free(&s->resample);
if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
return -1;
}
if(!s-> in_ch_layout)
s-> in_ch_layout= guess_layout(s->in.ch_count);
if(!s->out_ch_layout)
s->out_ch_layout= guess_layout(s->out.ch_count);
s->rematrix= s->out_ch_layout !=s->in_ch_layout;
#define RSC 1 //FIXME finetune
if(!s-> in.ch_count)
s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
if(!s->out.ch_count)
s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
av_assert0(s-> in.ch_count);
av_assert0(s->out.ch_count);
s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
s-> in.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
s-> in_sample_fmt, s-> in.ch_count, 0);
s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
s->int_sample_fmt, s->out.ch_count, 0);
s->postin= s->in;
s->preout= s->out;
s->midbuf= s->in;
s->in_buffer= s->in;
if(!s->resample_first){
s->midbuf.ch_count= s->out.ch_count;
s->in_buffer.ch_count = s->out.ch_count;
}
s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
if(s->rematrix && swr_rematrix_init(s)<0)
return -1;
return 0;
}
static int realloc_audio(AudioData *a, int count){
int i, countb;
AudioData old;
if(a->count >= count)
return 0;
count*=2;
countb= FFALIGN(count*a->bps, 32);
old= *a;
av_assert0(a->planar);
av_assert0(a->bps);
av_assert0(a->ch_count);
a->data= av_malloc(countb*a->ch_count);
if(!a->data)
return AVERROR(ENOMEM);
for(i=0; i<a->ch_count; i++){
a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
}
av_free(old.data);
a->count= count;
return 1;
}
static void copy(AudioData *out, AudioData *in,
int count){
av_assert0(out->planar == in->planar);
av_assert0(out->bps == in->bps);
av_assert0(out->ch_count == in->ch_count);
if(out->planar){
int ch;
for(ch=0; ch<out->ch_count; ch++)
memcpy(out->ch[ch], in->ch[ch], count*out->bps);
}else
memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
}
int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
const uint8_t *in_arg [SWR_CH_MAX], int in_count){
AudioData *postin, *midbuf, *preout;
int ret, i/*, in_max*/;
AudioData * in= &s->in;
AudioData *out= &s->out;
AudioData preout_tmp, midbuf_tmp;
if(!s->resample){
if(in_count > out_count)
return -1;
out_count = in_count;
}
av_assert0(in ->planar == 0);
av_assert0(out->planar == 0);
for(i=0; i<s-> in.ch_count; i++)
in ->ch[i]= in_arg[0] + i* in->bps;
for(i=0; i<s->out.ch_count; i++)
out->ch[i]= out_arg[0] + i*out->bps;
// in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
// in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
if((ret=realloc_audio(&s->postin, in_count))<0)
return ret;
if(s->resample_first){
av_assert0(s->midbuf.ch_count == s-> in.ch_count);
if((ret=realloc_audio(&s->midbuf, out_count))<0)
return ret;
}else{
av_assert0(s->midbuf.ch_count == s->out.ch_count);
if((ret=realloc_audio(&s->midbuf, in_count))<0)
return ret;
}
if((ret=realloc_audio(&s->preout, out_count))<0)
return ret;
postin= &s->postin;
midbuf_tmp= s->midbuf;
midbuf= &midbuf_tmp;
preout_tmp= s->preout;
preout= &preout_tmp;
if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
postin= in;
if(s->resample_first ? !s->resample : !s->rematrix)
midbuf= postin;
if(s->resample_first ? !s->rematrix : !s->resample)
preout= midbuf;
if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
if(preout==in){
out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
copy(out, in, out_count);
return out_count;
}
else if(preout==postin) preout= midbuf= postin= out;
else if(preout==midbuf) preout= midbuf= out;
else preout= out;
}
if(in != postin){
swr_audio_convert(s->in_convert, postin, in, in_count);
}
if(s->resample_first){
if(postin != midbuf)
out_count= resample(s, midbuf, out_count, postin, in_count);
if(midbuf != preout)
swr_rematrix(s, preout, midbuf, out_count, preout==out);
}else{
if(postin != midbuf)
swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
if(midbuf != preout)
out_count= resample(s, preout, out_count, midbuf, in_count);
}
if(preout != out){
//FIXME packed doesnt need more than 1 chan here!
swr_audio_convert(s->out_convert, out, preout, out_count);
}
return out_count;
}
/**
*
* out may be equal in.
*/
static void buf_set(AudioData *out, AudioData *in, int count){
if(in->planar){
int ch;
for(ch=0; ch<out->ch_count; ch++)
out->ch[ch]= in->ch[ch] + count*out->bps;
}else
out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
}
/**
*
* @return number of samples output per channel
*/
static int resample(SwrContext *s, AudioData *out_param, int out_count,
const AudioData * in_param, int in_count){
AudioData in, out, tmp;
int ret_sum=0;
int border=0;
int ch_count= s->resample_first ? s->in.ch_count : s->out.ch_count;
tmp=out=*out_param;
in = *in_param;
do{
int ret, size, consumed;
if(!s->resample_in_constraint && s->in_buffer_count){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
ret= swr_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
out_count -= ret;
ret_sum += ret;
buf_set(&out, &out, ret);
s->in_buffer_count -= consumed;
s->in_buffer_index += consumed;
if(!in_count)
break;
if(s->in_buffer_count <= border){
buf_set(&in, &in, -s->in_buffer_count);
in_count += s->in_buffer_count;
s->in_buffer_count=0;
s->in_buffer_index=0;
border = 0;
}
}
if(in_count && !s->in_buffer_count){
s->in_buffer_index=0;
ret= swr_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
out_count -= ret;
ret_sum += ret;
buf_set(&out, &out, ret);
in_count -= consumed;
buf_set(&in, &in, consumed);
}
//TODO is this check sane considering the advanced copy avoidance below
size= s->in_buffer_index + s->in_buffer_count + in_count;
if( size > s->in_buffer.count
&& s->in_buffer_count + in_count <= s->in_buffer_index){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
copy(&s->in_buffer, &tmp, s->in_buffer_count);
s->in_buffer_index=0;
}else
if((ret=realloc_audio(&s->in_buffer, size)) < 0)
return ret;
if(in_count){
int count= in_count;
if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
copy(&tmp, &in, /*in_*/count);
s->in_buffer_count += count;
in_count -= count;
border += count;
buf_set(&in, &in, count);
s->resample_in_constraint= 0;
if(s->in_buffer_count != count || in_count)
continue;
}
break;
}while(1);
s->resample_in_constraint= !!out_count;
return ret_sum;
}

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libswresample/swresample.h
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/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWR_H
#define SWR_H
#include <inttypes.h>
#include "libavutil/samplefmt.h"
#define LIBSWRESAMPLE_VERSION_MAJOR 0
#define LIBSWRESAMPLE_VERSION_MINOR 0
#define LIBSWRESAMPLE_VERSION_MICRO 0
#define SWR_CH_MAX 16
#define SWR_FLAG_RESAMPLE 1///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?
struct SwrContext;
/**
* Allocate SwrContext.
* @see swr_init(),swr_free()
* @return NULL on error
*/
struct SwrContext *swr_alloc(void);
/**
* Initialize context after user parameters have been set.
* @return negativo n error
*/
int swr_init(struct SwrContext *s);
/**
* Allocate SwrContext.
* @see swr_init(),swr_free()
* @return NULL on error
*/
struct SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx);
/**
* Free the given SwrContext.
* And set the pointer to NULL
*/
void swr_free(struct SwrContext **s);
/**
* Convert audio.
* @param in_count Number of input samples available in one channel.
* @param out_count Amount of space available for output in samples per channel.
* @return number of samples output per channel
*/
int swr_convert(struct SwrContext *s, uint8_t *out[SWR_CH_MAX], int out_count,
const uint8_t *in [SWR_CH_MAX], int in_count);
void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance);
#endif

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libswresample/swresample_internal.h
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/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWR_INTERNAL_H
#define SWR_INTERNAL_H
#include "swresample.h"
typedef struct AudioData{
uint8_t *ch[SWR_CH_MAX];
uint8_t *data;
int ch_count;
int bps;
int count;
int planar;
} AudioData;
typedef struct SwrContext { //FIXME find unused fields
AVClass *av_class;
int log_level_offset;
void *log_ctx;
enum AVSampleFormat in_sample_fmt;
enum AVSampleFormat int_sample_fmt; ///<AV_SAMPLE_FMT_FLT OR AV_SAMPLE_FMT_S16
enum AVSampleFormat out_sample_fmt;
int64_t in_ch_layout;
int64_t out_ch_layout;
int in_sample_rate;
int out_sample_rate;
int flags;
float slev, clev;
//below are private
int int_bps;
int resample_first;
int rematrix; ///< flag to indicate if rematrixing is used
AudioData in, postin, midbuf, preout, out, in_buffer;
int in_buffer_index;
int in_buffer_count;
int resample_in_constraint;
struct AVAudioConvert *in_convert;
struct AVAudioConvert *out_convert;
struct AVResampleContext *resample;
float matrix[SWR_CH_MAX][SWR_CH_MAX];
uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];
//TODO callbacks for asm optims
}SwrContext;
struct AVResampleContext *swr_resample_init(struct AVResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff);
void swr_resample_free(struct AVResampleContext **c);
int swr_multiple_resample(struct AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
void swr_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
int swr_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
int swr_rematrix_init(SwrContext *s);
int swr_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
#endif

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libswresample/swresample_test.c
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/*
* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/avassert.h"
#include "libavutil/common.h"
#include "libavutil/audioconvert.h"
#include "swresample.h"
#undef fprintf
#define SAMPLES 1000
#define ASSERT_LEVEL 2
static double get(const void *p, int index, enum AVSampleFormat f){
switch(f){
case AV_SAMPLE_FMT_U8 : return ((const uint8_t*)p)[index]/255.0*2-1.0;
case AV_SAMPLE_FMT_S16: return ((const int16_t*)p)[index]/32767.0;
case AV_SAMPLE_FMT_S32: return ((const int32_t*)p)[index]/2147483647.0;
case AV_SAMPLE_FMT_FLT: return ((const float *)p)[index];
case AV_SAMPLE_FMT_DBL: return ((const double *)p)[index];
default: av_assert2(0);
}
}
static void set(void *p, int index, enum AVSampleFormat f, double v){
switch(f){
case AV_SAMPLE_FMT_U8 : ((uint8_t*)p)[index]= (v+1.0)*255.0/2; break;
case AV_SAMPLE_FMT_S16: ((int16_t*)p)[index]= v*32767; break;
case AV_SAMPLE_FMT_S32: ((int32_t*)p)[index]= v*2147483647; break;
case AV_SAMPLE_FMT_FLT: ((float *)p)[index]= v; break;
case AV_SAMPLE_FMT_DBL: ((double *)p)[index]= v; break;
default: av_assert2(0);
}
}
uint64_t layouts[]={
AV_CH_LAYOUT_MONO ,
AV_CH_LAYOUT_STEREO ,
AV_CH_LAYOUT_2_1 ,
AV_CH_LAYOUT_SURROUND ,
AV_CH_LAYOUT_4POINT0 ,
AV_CH_LAYOUT_2_2 ,
AV_CH_LAYOUT_QUAD ,
AV_CH_LAYOUT_5POINT0 ,
AV_CH_LAYOUT_5POINT1 ,
AV_CH_LAYOUT_5POINT0_BACK ,
AV_CH_LAYOUT_5POINT1_BACK ,
AV_CH_LAYOUT_7POINT0 ,
AV_CH_LAYOUT_7POINT1 ,
AV_CH_LAYOUT_7POINT1_WIDE ,
0
};
int main(int argc, char **argv){
int in_sample_rate, out_sample_rate, ch ,i, in_ch_layout_index, out_ch_layout_index, osr;
uint64_t in_ch_layout, out_ch_layout;
enum AVSampleFormat in_sample_fmt, out_sample_fmt;
int sample_rates[]={8000,11025,16000,22050,32000};
uint8_t array_in[SAMPLES*8*8];
uint8_t array_mid[SAMPLES*8*8*3];
uint8_t array_out[SAMPLES*8*8+100];
struct SwrContext * forw_ctx= NULL;
struct SwrContext *backw_ctx= NULL;
in_sample_rate=16000;
for(osr=0; osr<5; osr++){
out_sample_rate= sample_rates[osr];
for(in_sample_fmt= AV_SAMPLE_FMT_U8; in_sample_fmt<=AV_SAMPLE_FMT_DBL; in_sample_fmt++){
for(out_sample_fmt= AV_SAMPLE_FMT_U8; out_sample_fmt<=AV_SAMPLE_FMT_DBL; out_sample_fmt++){
for(in_ch_layout_index=0; layouts[in_ch_layout_index]; in_ch_layout_index++){
in_ch_layout= layouts[in_ch_layout_index];
int in_ch_count= av_get_channel_layout_nb_channels(in_ch_layout);
for(out_ch_layout_index=0; layouts[out_ch_layout_index]; out_ch_layout_index++){
int out_count, mid_count;
out_ch_layout= layouts[out_ch_layout_index];
int out_ch_count= av_get_channel_layout_nb_channels(out_ch_layout);
fprintf(stderr, "ch %d->%d, rate:%5d->%5d, fmt:%s->%s",
in_ch_count, out_ch_count,
in_sample_rate, out_sample_rate,
av_get_sample_fmt_name(in_sample_fmt), av_get_sample_fmt_name(out_sample_fmt));
forw_ctx = swr_alloc2(forw_ctx, out_ch_layout, out_sample_fmt, out_sample_rate,
in_ch_layout, in_sample_fmt, in_sample_rate, 0, 0);
backw_ctx = swr_alloc2(backw_ctx,in_ch_layout, in_sample_fmt, in_sample_rate,
out_ch_layout, out_sample_fmt, out_sample_rate, 0, 0);
if(swr_init( forw_ctx) < 0)
fprintf(stderr, "swr_init(->) failed\n");
if(swr_init(backw_ctx) < 0)
fprintf(stderr, "swr_init(<-) failed\n");
if(!forw_ctx)
fprintf(stderr, "Failed to init forw_cts\n");
if(!backw_ctx)
fprintf(stderr, "Failed to init backw_ctx\n");
//FIXME test planar
for(ch=0; ch<in_ch_count; ch++){
for(i=0; i<SAMPLES; i++)
set(array_in, ch + i*in_ch_count, in_sample_fmt, sin(i*i*3/SAMPLES));
}
mid_count= swr_convert(forw_ctx, ( uint8_t*[]){array_mid}, 3*SAMPLES,
(const uint8_t*[]){array_in }, SAMPLES);
out_count= swr_convert(backw_ctx,( uint8_t*[]){array_out}, 3*SAMPLES,
(const uint8_t*[]){array_mid}, mid_count);
for(ch=0; ch<in_ch_count; ch++){
double sse, x, maxdiff=0;
double sum_a= 0;
double sum_b= 0;
double sum_aa= 0;
double sum_bb= 0;
double sum_ab= 0;
for(i=0; i<SAMPLES; i++){
double a= get(array_in , ch + i*in_ch_count, in_sample_fmt);
double b= get(array_out, ch + i*in_ch_count, in_sample_fmt);
sum_a += a;
sum_b += b;
sum_aa+= a*a;
sum_bb+= b*b;
sum_ab+= a*b;
maxdiff= FFMAX(maxdiff, FFABS(a-b));
}
x = sum_ab/sum_bb;
sse= sum_aa + sum_bb*x*x - 2*x*sum_ab;
fprintf(stderr, "[%f %f %f] len:%5d\n", sqrt(sse/SAMPLES), x, maxdiff, out_count);
}
fprintf(stderr, "\n");
}
}
}
}
}
return 0;
}

6
tests/ref/acodec/g726
Unescape View File

@ -1,4 +1,4 @@
fd090ddf05cc3401cc75c4a5ace1d05a *./tests/data/acodec/g726.wav
a76fc937faac62c5de057cd69191732a *./tests/data/acodec/g726.wav
24052 ./tests/data/acodec/g726.wav
74abea06027375111eeac1b2f8c7d3af *./tests/data/g726.acodec.out.wav
stddev: 8554.55 PSNR: 17.69 MAXDIFF:29353 bytes: 95984/ 1058400
124de13e6cb5af64ea8758aa49feb7fc *./tests/data/g726.acodec.out.wav
stddev: 8554.23 PSNR: 17.69 MAXDIFF:29353 bytes: 95984/ 1058400

6
tests/ref/acodec/pcm
Unescape View File

@ -66,7 +66,7 @@ stddev: 0.00 PSNR:999.99 MAXDIFF: 0 bytes: 1058400/ 1058400
529256 ./tests/data/acodec/pcm_zork.wav
864c8c866ac25642c29a13b122c70709 *./tests/data/pcm.acodec.out.wav
stddev: 633.11 PSNR: 40.30 MAXDIFF:32768 bytes: 1058400/ 1058400
8168a5c1343553ef027541830f2cb879 *./tests/data/acodec/pcm_s24daud.302
1b75d5198ae789ab3c48f7024e08f4a9 *./tests/data/acodec/pcm_s24daud.302
10368730 ./tests/data/acodec/pcm_s24daud.302
f552afadfdfcd6348a07095da6382de5 *./tests/data/pcm.acodec.out.wav
stddev: 9416.28 PSNR: 16.85 MAXDIFF:42744 bytes: 6911796/ 1058400
4708f86529c594e29404603c64bb208c *./tests/data/pcm.acodec.out.wav
stddev: 8967.92 PSNR: 17.28 MAXDIFF:42548 bytes: 6911796/ 1058400

4
tests/ref/lavf/dv_fmt
Unescape View File

@ -1,3 +1,3 @@
188f804bd2d10cd436c8a7b111bdcd2a *./tests/data/lavf/lavf.dv
6e716216d5f9e3819db8eb8796de9129 *./tests/data/lavf/lavf.dv
3600000 ./tests/data/lavf/lavf.dv
./tests/data/lavf/lavf.dv CRC=0x02c0af30
./tests/data/lavf/lavf.dv CRC=0x92d1e3f0

4
tests/ref/lavf/mxf_d10
Unescape View File

@ -1,3 +1,3 @@
b3174e2db508564c1cce0b5e3c1bc1bd *./tests/data/lavf/lavf.mxf_d10
8eb67301f72f2b5860fafab422b920ad *./tests/data/lavf/lavf.mxf_d10
5330989 ./tests/data/lavf/lavf.mxf_d10
./tests/data/lavf/lavf.mxf_d10 CRC=0xc3f4f92e
./tests/data/lavf/lavf.mxf_d10 CRC=0x96c02dfd

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