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ffmpeg/libswresample/swresample.c
Michael Niedermayer b5875b9111 Add libswresample. 
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>
2011-09-19 07:04:17 +02:00

433 lines
15 KiB
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
*/
#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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