mirror of https://git.ffmpeg.org/ffmpeg.git
swr: More flexible and convenient buffering
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
parent
e479013ae4
commit
1b0fcf33b8
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@ -240,6 +240,7 @@ av_assert0(s->out.ch_count);
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s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
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s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
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s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
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s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
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s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);
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s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);
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s->in_buffer= s->in;
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if(!s->resample && !s->rematrix && !s->channel_map){
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if(!s->resample && !s->rematrix && !s->channel_map){
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s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
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s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
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@ -256,20 +257,26 @@ av_assert0(s->out.ch_count);
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s->postin= s->in;
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s->postin= s->in;
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s->preout= s->out;
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s->preout= s->out;
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s->midbuf= s->in;
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s->midbuf= s->in;
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s->in_buffer= s->in;
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if(s->channel_map){
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if(s->channel_map){
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s->postin.ch_count=
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s->postin.ch_count=
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s->midbuf.ch_count=
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s->midbuf.ch_count= s->used_ch_count;
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s->in_buffer.ch_count= s->used_ch_count;
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if(s->resample)
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s->in_buffer.ch_count= s->used_ch_count;
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}
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}
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if(!s->resample_first){
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if(!s->resample_first){
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s->midbuf.ch_count= s->out.ch_count;
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s->midbuf.ch_count= s->out.ch_count;
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s->in_buffer.ch_count = s->out.ch_count;
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if(s->resample)
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s->in_buffer.ch_count = s->out.ch_count;
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}
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}
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s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
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s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
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s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
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s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
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if(s->resample){
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s->in_buffer.bps = s->int_bps;
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s->in_buffer.planar = 1;
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}
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if(s->rematrix)
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if(s->rematrix)
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return swri_rematrix_init(s);
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return swri_rematrix_init(s);
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@ -421,45 +428,12 @@ static int resample(SwrContext *s, AudioData *out_param, int out_count,
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return ret_sum;
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return ret_sum;
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}
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}
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int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
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static int swr_convert_internal(struct SwrContext *s, AudioData *out[SWR_CH_MAX], int out_count,
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const uint8_t *in_arg [SWR_CH_MAX], int in_count){
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AudioData *in [SWR_CH_MAX], int in_count){
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AudioData *postin, *midbuf, *preout;
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AudioData *postin, *midbuf, *preout;
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int ret/*, in_max*/;
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int ret/*, in_max*/;
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AudioData * in= &s->in;
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AudioData *out= &s->out;
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AudioData preout_tmp, midbuf_tmp;
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AudioData preout_tmp, midbuf_tmp;
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if(!s->resample){
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if(in_count > out_count)
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return -1;
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out_count = in_count;
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}
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if(!in_arg){
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if(s->in_buffer_count){
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if (!s->flushed) {
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AudioData *a= &s->in_buffer;
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int i, j, ret;
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if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
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return ret;
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av_assert0(a->planar);
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for(i=0; i<a->ch_count; i++){
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for(j=0; j<s->in_buffer_count; j++){
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memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
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a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
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}
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}
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s->in_buffer_count += (s->in_buffer_count+1)/2;
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s->resample_in_constraint = 0;
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s->flushed = 1;
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}
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}else{
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return 0;
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}
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}else
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fill_audiodata(in , (void*)in_arg);
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fill_audiodata(out, out_arg);
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if(s->full_convert){
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if(s->full_convert){
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av_assert0(!s->resample);
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av_assert0(!s->resample);
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swri_audio_convert(s->full_convert, out, in, in_count);
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swri_audio_convert(s->full_convert, out, in, in_count);
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@ -534,3 +508,89 @@ int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_coun
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return out_count;
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return out_count;
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}
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}
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int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
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const uint8_t *in_arg [SWR_CH_MAX], int in_count){
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AudioData * in= &s->in;
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AudioData *out= &s->out;
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if(!in_arg){
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if(s->in_buffer_count){
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if (s->resample && !s->flushed) {
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AudioData *a= &s->in_buffer;
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int i, j, ret;
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if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
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return ret;
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av_assert0(a->planar);
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for(i=0; i<a->ch_count; i++){
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for(j=0; j<s->in_buffer_count; j++){
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memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
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a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
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}
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}
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s->in_buffer_count += (s->in_buffer_count+1)/2;
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s->resample_in_constraint = 0;
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s->flushed = 1;
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}
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}else{
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return 0;
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}
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}else
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fill_audiodata(in , (void*)in_arg);
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fill_audiodata(out, out_arg);
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if(s->resample){
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return swr_convert_internal(s, out, out_count, in, in_count);
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}else{
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AudioData tmp= *in;
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int ret2=0;
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int ret, size;
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int in_buffer_count= s->in_buffer_count;
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size = FFMIN(out_count, s->in_buffer_count);
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if(size){
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buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
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ret= swr_convert_internal(s, out, size, &tmp, size);
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if(ret<0)
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return ret;
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ret2= ret;
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s->in_buffer_count -= ret;
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s->in_buffer_index += ret;
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buf_set(out, out, ret);
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out_count -= ret;
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if(!s->in_buffer_count)
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s->in_buffer_index = 0;
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}
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if(in_count){
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size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
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if(in_count > out_count) { //FIXME move after swr_convert_internal
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if( size > s->in_buffer.count
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&& s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
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buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
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copy(&s->in_buffer, &tmp, s->in_buffer_count);
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s->in_buffer_index=0;
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}else
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if((ret=realloc_audio(&s->in_buffer, size)) < 0)
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return ret;
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}
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if(out_count){
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size = FFMIN(in_count, out_count);
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ret= swr_convert_internal(s, out, size, in, size);
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if(ret<0)
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return ret;
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buf_set(in, in, ret);
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in_count -= ret;
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ret2 += ret;
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}
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if(in_count){
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buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
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copy(&tmp, in, in_count);
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s->in_buffer_count += in_count;
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}
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}
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return ret2;
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}
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}
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@ -102,6 +102,10 @@ void swr_free(struct SwrContext **s);
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* in and in_count can be set to 0 to flush the last few samples out at the
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* in and in_count can be set to 0 to flush the last few samples out at the
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* end.
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* end.
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*
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*
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* If more input is provided than output space then the input will be buffered.
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* You can avoid this buffering by providing more output space than input.
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* Convertion will run directly without copying whenever possible.
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*
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* @param s allocated Swr context, with parameters set
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* @param s allocated Swr context, with parameters set
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* @param out output buffers, only the first one need be set in case of packed audio
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* @param out output buffers, only the first one need be set in case of packed audio
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* @param out_count amount of space available for output in samples per channel
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* @param out_count amount of space available for output in samples per channel
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