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ffmpeg/libswresample/swresample.c
Alexander Strasser ac25b31ede lswr: Improve default resampler's default parameters 
After making some blind tests on a small collection of music
samples for home usage. It turned out that the default cutoff
was too low.

The impact of filter_size was not clearly distinguishable (the
results were on the edge) with the music samples but turned out
to be clearly audible in some synthetic samples.

Thanks to Daniel for helping out with the listening tests.

Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
2013-01-04 16:47:57 +01:00

852 lines
36 KiB
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/*
* Copyright (C) 2011-2012 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"
#include "libavutil/channel_layout.h"
#include <float.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
#define ALIGN 32
//TODO split options array out?
#define OFFSET(x) offsetof(SwrContext,x)
#define PARAM AV_OPT_FLAG_AUDIO_PARAM
static const AVOption options[]={
{"ich" , "set input channel count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"in_channel_count" , "set input channel count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"och" , "set output channel count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"out_channel_count" , "set output channel count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"uch" , "set used channel count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"used_channel_count" , "set used channel count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
{"isr" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"in_sample_rate" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"osr" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"out_sample_rate" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
{"isf" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM},
{"in_sample_fmt" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM},
{"osf" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM},
{"out_sample_fmt" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM},
{"tsf" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM},
{"internal_sample_fmt" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM},
{"icl" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
{"in_channel_layout" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
{"ocl" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
{"out_channel_layout" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
{"clev" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"center_mix_level" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"slev" , "set surround mix level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"surround_mix_level" , "set surround mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
{"lfe_mix_level" , "set LFE mix level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
{"rmvol" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
{"rematrix_volume" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
{"flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
{"swr_flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
{"res" , "force resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
{"dither_scale" , "set dither scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
{"dither_method" , "set dither method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
{"rectangular" , "select rectangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
{"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
{"triangular_hp" , "select triangular dither with high pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
{"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM },
{"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},
{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},
{"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"},
{"precision" , "set soxr resampling precision (in bits)"
, OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
{"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
, OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
{"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
, OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
{"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
, OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
{"comp_duration" , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
, OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
{"max_soft_comp" , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
, OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
{"async" , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
, OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
{ "matrix_encoding" , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
{ "none", "select none", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "filter_type" , "select swr filter type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
{ "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser_beta" , "set swr Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
{0}
};
static const char* context_to_name(void* ptr) {
return "SWR";
}
static const AVClass av_class = {
.class_name = "SWResampler",
.item_name = context_to_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.log_level_offset_offset = OFFSET(log_level_offset),
.parent_log_context_offset = OFFSET(log_ctx),
.category = AV_CLASS_CATEGORY_SWRESAMPLER,
};
unsigned swresample_version(void)
{
av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
return LIBSWRESAMPLE_VERSION_INT;
}
const char *swresample_configuration(void)
{
return FFMPEG_CONFIGURATION;
}
const char *swresample_license(void)
{
#define LICENSE_PREFIX "libswresample license: "
return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
}
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
if(!s || s->in_convert) // s needs to be allocated but not initialized
return AVERROR(EINVAL);
s->channel_map = channel_map;
return 0;
}
const AVClass *swr_get_class(void)
{
return &av_class;
}
av_cold struct SwrContext *swr_alloc(void){
SwrContext *s= av_mallocz(sizeof(SwrContext));
if(s){
s->av_class= &av_class;
av_opt_set_defaults(s);
}
return s;
}
struct SwrContext *swr_alloc_set_opts(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_opt_set_int(s, "ocl", out_ch_layout, 0);
av_opt_set_int(s, "osf", out_sample_fmt, 0);
av_opt_set_int(s, "osr", out_sample_rate, 0);
av_opt_set_int(s, "icl", in_ch_layout, 0);
av_opt_set_int(s, "isf", in_sample_fmt, 0);
av_opt_set_int(s, "isr", in_sample_rate, 0);
av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
av_opt_set_int(s, "uch", 0, 0);
return s;
}
static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
a->fmt = fmt;
a->bps = av_get_bytes_per_sample(fmt);
a->planar= av_sample_fmt_is_planar(fmt);
}
static void free_temp(AudioData *a){
av_free(a->data);
memset(a, 0, sizeof(*a));
}
av_cold 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);
free_temp(&s->dither);
swri_audio_convert_free(&s-> in_convert);
swri_audio_convert_free(&s->out_convert);
swri_audio_convert_free(&s->full_convert);
if (s->resampler)
s->resampler->free(&s->resample);
swri_rematrix_free(s);
}
av_freep(ss);
}
av_cold int swr_init(struct 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);
free_temp(&s->dither);
memset(s->in.ch, 0, sizeof(s->in.ch));
memset(s->out.ch, 0, sizeof(s->out.ch));
swri_audio_convert_free(&s-> in_convert);
swri_audio_convert_free(&s->out_convert);
swri_audio_convert_free(&s->full_convert);
swri_rematrix_free(s);
s->flushed = 0;
if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
return AVERROR(EINVAL);
}
if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
return AVERROR(EINVAL);
}
if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
}else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
}else{
av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
}
}
if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
&&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
&&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
&&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
return AVERROR(EINVAL);
}
switch(s->engine){
#if CONFIG_LIBSOXR
extern struct Resampler const soxr_resampler;
case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
#endif
case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
default:
av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
return AVERROR(EINVAL);
}
set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
set_audiodata_fmt(&s->out, s->out_sample_fmt);
if (s->async) {
if (s->min_compensation >= FLT_MAX/2)
s->min_compensation = 0.001;
if (s->async > 1.0001) {
s->max_soft_compensation = s->async / (double) s->in_sample_rate;
}
}
if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby);
}else
s->resampler->free(&s->resample);
if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
&& s->int_sample_fmt != AV_SAMPLE_FMT_S32P
&& s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
&& s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
&& s->resample){
av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
return -1;
}
if(!s->used_ch_count)
s->used_ch_count= s->in.ch_count;
if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
s-> in_ch_layout= 0;
}
if(!s-> in_ch_layout)
s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
if(!s->out_ch_layout)
s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
s->rematrix_custom;
#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->used_ch_count)
s->used_ch_count= s->in.ch_count;
if(!s->out.ch_count)
s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
if(!s-> in.ch_count){
av_assert0(!s->in_ch_layout);
av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
return -1;
}
if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
char l1[1024], l2[1024];
av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
"but there is not enough information to do it\n", l1, l2);
return -1;
}
av_assert0(s->used_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_buffer= s->in;
if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
return 0;
}
s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
s->int_sample_fmt, s->out.ch_count, NULL, 0);
s->postin= s->in;
s->preout= s->out;
s->midbuf= s->in;
if(s->channel_map){
s->postin.ch_count=
s->midbuf.ch_count= s->used_ch_count;
if(s->resample)
s->in_buffer.ch_count= s->used_ch_count;
}
if(!s->resample_first){
s->midbuf.ch_count= s->out.ch_count;
if(s->resample)
s->in_buffer.ch_count = s->out.ch_count;
}
set_audiodata_fmt(&s->postin, s->int_sample_fmt);
set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
set_audiodata_fmt(&s->preout, s->int_sample_fmt);
if(s->resample){
set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
}
s->dither = s->preout;
if(s->rematrix || s->dither_method)
return swri_rematrix_init(s);
return 0;
}
int swri_realloc_audio(AudioData *a, int count){
int i, countb;
AudioData old;
if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
return AVERROR(EINVAL);
if(a->count >= count)
return 0;
count*=2;
countb= FFALIGN(count*a->bps, ALIGN);
old= *a;
av_assert0(a->bps);
av_assert0(a->ch_count);
a->data= av_mallocz(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);
}
if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_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);
}
static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
int i;
if(!in_arg){
memset(out->ch, 0, sizeof(out->ch));
}else if(out->planar){
for(i=0; i<out->ch_count; i++)
out->ch[i]= in_arg[i];
}else{
for(i=0; i<out->ch_count; i++)
out->ch[i]= in_arg[0] + i*out->bps;
}
}
static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
int i;
if(out->planar){
for(i=0; i<out->ch_count; i++)
in_arg[i]= out->ch[i];
}else{
in_arg[0]= out->ch[0];
}
}
/**
*
* out may be equal in.
*/
static void buf_set(AudioData *out, AudioData *in, int count){
int ch;
if(in->planar){
for(ch=0; ch<out->ch_count; ch++)
out->ch[ch]= in->ch[ch] + count*out->bps;
}else{
for(ch=out->ch_count-1; ch>=0; ch--)
out->ch[ch]= in->ch[0] + (ch + 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;
av_assert1(s->in_buffer.ch_count == in_param->ch_count);
av_assert1(s->in_buffer.planar == in_param->planar);
av_assert1(s->in_buffer.fmt == in_param->fmt);
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= s->resampler->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((s->flushed || in_count) && !s->in_buffer_count){
s->in_buffer_index=0;
ret= s->resampler->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=swri_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;
}
static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
AudioData *in , int in_count){
AudioData *postin, *midbuf, *preout;
int ret/*, in_max*/;
AudioData preout_tmp, midbuf_tmp;
if(s->full_convert){
av_assert0(!s->resample);
swri_audio_convert(s->full_convert, out, in, in_count);
return out_count;
}
// 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=swri_realloc_audio(&s->postin, in_count))<0)
return ret;
if(s->resample_first){
av_assert0(s->midbuf.ch_count == s->used_ch_count);
if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
return ret;
}else{
av_assert0(s->midbuf.ch_count == s->out.ch_count);
if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
return ret;
}
if((ret=swri_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 && !s->channel_map)
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 the 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){
swri_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)
swri_rematrix(s, preout, midbuf, out_count, preout==out);
}else{
if(postin != midbuf)
swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
if(midbuf != preout)
out_count= resample(s, preout, out_count, midbuf, in_count);
}
if(preout != out && out_count){
if(s->dither_method){
int ch;
int dither_count= FFMAX(out_count, 1<<16);
av_assert0(preout != in);
if((ret=swri_realloc_audio(&s->dither, dither_count))<0)
return ret;
if(ret)
for(ch=0; ch<s->dither.ch_count; ch++)
swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
av_assert0(s->dither.ch_count == preout->ch_count);
if(s->dither_pos + out_count > s->dither.count)
s->dither_pos = 0;
for(ch=0; ch<preout->ch_count; ch++)
s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
s->dither_pos += out_count;
}
//FIXME packed doesnt need more than 1 chan here!
swri_audio_convert(s->out_convert, out, preout, out_count);
}
return out_count;
}
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 * in= &s->in;
AudioData *out= &s->out;
if(s->drop_output > 0){
int ret;
AudioData tmp = s->out;
uint8_t *tmp_arg[SWR_CH_MAX];
tmp.count = 0;
tmp.data = NULL;
if((ret=swri_realloc_audio(&tmp, s->drop_output))<0)
return ret;
reversefill_audiodata(&tmp, tmp_arg);
s->drop_output *= -1; //FIXME find a less hackish solution
ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
s->drop_output *= -1;
if(ret>0)
s->drop_output -= ret;
av_freep(&tmp.data);
if(s->drop_output || !out_arg)
return 0;
in_count = 0;
}
if(!in_arg){
if(s->resample){
if (!s->flushed)
s->resampler->flush(s);
s->resample_in_constraint = 0;
s->flushed = 1;
}else if(!s->in_buffer_count){
return 0;
}
}else
fill_audiodata(in , (void*)in_arg);
fill_audiodata(out, out_arg);
if(s->resample){
int ret = swr_convert_internal(s, out, out_count, in, in_count);
if(ret>0 && !s->drop_output)
s->outpts += ret * (int64_t)s->in_sample_rate;
return ret;
}else{
AudioData tmp= *in;
int ret2=0;
int ret, size;
size = FFMIN(out_count, s->in_buffer_count);
if(size){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
ret= swr_convert_internal(s, out, size, &tmp, size);
if(ret<0)
return ret;
ret2= ret;
s->in_buffer_count -= ret;
s->in_buffer_index += ret;
buf_set(out, out, ret);
out_count -= ret;
if(!s->in_buffer_count)
s->in_buffer_index = 0;
}
if(in_count){
size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
if(in_count > out_count) { //FIXME move after swr_convert_internal
if( size > s->in_buffer.count
&& s->in_buffer_count + in_count - out_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=swri_realloc_audio(&s->in_buffer, size)) < 0)
return ret;
}
if(out_count){
size = FFMIN(in_count, out_count);
ret= swr_convert_internal(s, out, size, in, size);
if(ret<0)
return ret;
buf_set(in, in, ret);
in_count -= ret;
ret2 += ret;
}
if(in_count){
buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
copy(&tmp, in, in_count);
s->in_buffer_count += in_count;
}
}
if(ret2>0 && !s->drop_output)
s->outpts += ret2 * (int64_t)s->in_sample_rate;
return ret2;
}
}
int swr_drop_output(struct SwrContext *s, int count){
s->drop_output += count;
if(s->drop_output <= 0)
return 0;
av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
return swr_convert(s, NULL, s->drop_output, NULL, 0);
}
int swr_inject_silence(struct SwrContext *s, int count){
int ret, i;
AudioData silence = s->in;
uint8_t *tmp_arg[SWR_CH_MAX];
if(count <= 0)
return 0;
silence.count = 0;
silence.data = NULL;
if((ret=swri_realloc_audio(&silence, count))<0)
return ret;
if(silence.planar) for(i=0; i<silence.ch_count; i++) {
memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
} else
memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
reversefill_audiodata(&silence, tmp_arg);
av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
av_freep(&silence.data);
return ret;
}
int64_t swr_get_delay(struct SwrContext *s, int64_t base){
if (s->resampler && s->resample){
return s->resampler->get_delay(s, base);
}else{
return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
}
}
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
int ret;
if (!s || compensation_distance < 0)
return AVERROR(EINVAL);
if (!compensation_distance && sample_delta)
return AVERROR(EINVAL);
if (!s->resample) {
s->flags |= SWR_FLAG_RESAMPLE;
ret = swr_init(s);
if (ret < 0)
return ret;
}
if (!s->resampler->set_compensation){
return AVERROR(EINVAL);
}else{
return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
}
}
int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
if(pts == INT64_MIN)
return s->outpts;
if(s->min_compensation >= FLT_MAX) {
return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
} else {
int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
if(fabs(fdelta) > s->min_compensation) {
if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
int ret;
if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
if(ret<0){
av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
}
} else if(s->soft_compensation_duration && s->max_soft_compensation) {
int duration = s->out_sample_rate * s->soft_compensation_duration;
double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
swr_set_compensation(s, comp, duration);
}
}
return s->outpts;
}
}
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