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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-21 10:55:51 +02:00

swr: More flexible and convenient buffering

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-03-25 21:04:48 +02:00
parent e479013ae4
commit 1b0fcf33b8
2 changed files with 105 additions and 41 deletions

View File

@ -240,6 +240,7 @@ av_assert0(s->out.ch_count);
s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);
s->in_buffer= s->in;
if(!s->resample && !s->rematrix && !s->channel_map){
s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
@ -256,20 +257,26 @@ av_assert0(s->out.ch_count);
s->postin= s->in;
s->preout= s->out;
s->midbuf= s->in;
s->in_buffer= s->in;
if(s->channel_map){
s->postin.ch_count=
s->midbuf.ch_count=
s->in_buffer.ch_count= s->used_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;
s->in_buffer.ch_count = s->out.ch_count;
if(s->resample)
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;
s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
if(s->resample){
s->in_buffer.bps = s->int_bps;
s->in_buffer.planar = 1;
}
if(s->rematrix)
return swri_rematrix_init(s);
@ -421,45 +428,12 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count,
return ret_sum;
}
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){
static int swr_convert_internal(struct SwrContext *s, AudioData *out[SWR_CH_MAX], int out_count,
AudioData *in [SWR_CH_MAX], int in_count){
AudioData *postin, *midbuf, *preout;
int ret/*, 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;
}
if(!in_arg){
if(s->in_buffer_count){
if (!s->flushed) {
AudioData *a= &s->in_buffer;
int i, j, ret;
if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
return ret;
av_assert0(a->planar);
for(i=0; i<a->ch_count; i++){
for(j=0; j<s->in_buffer_count; j++){
memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
}
}
s->in_buffer_count += (s->in_buffer_count+1)/2;
s->resample_in_constraint = 0;
s->flushed = 1;
}
}else{
return 0;
}
}else
fill_audiodata(in , (void*)in_arg);
fill_audiodata(out, out_arg);
if(s->full_convert){
av_assert0(!s->resample);
swri_audio_convert(s->full_convert, out, in, in_count);
@ -534,3 +508,89 @@ int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_coun
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(!in_arg){
if(s->in_buffer_count){
if (s->resample && !s->flushed) {
AudioData *a= &s->in_buffer;
int i, j, ret;
if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
return ret;
av_assert0(a->planar);
for(i=0; i<a->ch_count; i++){
for(j=0; j<s->in_buffer_count; j++){
memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
}
}
s->in_buffer_count += (s->in_buffer_count+1)/2;
s->resample_in_constraint = 0;
s->flushed = 1;
}
}else{
return 0;
}
}else
fill_audiodata(in , (void*)in_arg);
fill_audiodata(out, out_arg);
if(s->resample){
return swr_convert_internal(s, out, out_count, in, in_count);
}else{
AudioData tmp= *in;
int ret2=0;
int ret, size;
int in_buffer_count= s->in_buffer_count;
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=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);
copy(&tmp, in, in_count);
s->in_buffer_count += in_count;
}
}
return ret2;
}
}

View File

@ -102,6 +102,10 @@ void swr_free(struct SwrContext **s);
* in and in_count can be set to 0 to flush the last few samples out at the
* end.
*
* If more input is provided than output space then the input will be buffered.
* You can avoid this buffering by providing more output space than input.
* Convertion will run directly without copying whenever possible.
*
* @param s allocated Swr context, with parameters set
* @param out output buffers, only the first one need be set in case of packed audio
* @param out_count amount of space available for output in samples per channel