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503e5e9db5
FFmpeg/libswresample/swresample.c
Michael Niedermayer 503e5e9db5 libswresample: support directly converting sampleformats and packed/planar in a single pass. 
Previously a intermediate planar format was used when both input and output where packed.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2011-10-06 00:07:03 +02:00

466 lines
16 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+256, 0},
{"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 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_audio_convert_free(&s->full_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);
swr_audio_convert_free(&s->full_convert);
s-> in.planar= s-> in_sample_fmt >= 0x100;
s->out.planar= s->out_sample_fmt >= 0x100;
s-> in_sample_fmt &= 0xFF;
s->out_sample_fmt &= 0xFF;
if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->in_sample_fmt));
return AVERROR(EINVAL);
}
if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->out_sample_fmt));
return AVERROR(EINVAL);
}
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;
if(!s->resample && !s->rematrix){
s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
s-> in_sample_fmt, s-> in.ch_count, 0);
return 0;
}
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);
}
static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
int i;
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;
}
}
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;
}
fill_audiodata(in , in_arg);
fill_audiodata(out, out_arg);
if(s->full_convert){
av_assert0(!s->resample);
swr_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=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;
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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