videolan/ffmpeg
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better audio drift compensation

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Originally committed as revision 3275 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer 2004-06-30 17:53:05 +00:00
parent 0ff7199f59
commit ff4905a524
1 changed files with 44 additions and 13 deletions
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57
ffmpeg.c
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@ -276,6 +276,7 @@ typedef struct AVInputStream {
int64_t next_pts; /* synthetic pts for cases where pkt.pts
is not defined */
int64_t pts; /* current pts */
int is_start; /* is 1 at the start and after a discontinuity */
} AVInputStream;
typedef struct AVInputFile {
@ -421,7 +422,7 @@ static void do_audio_out(AVFormatContext *s,
const int audio_out_size= 4*MAX_AUDIO_PACKET_SIZE;
int size_out, frame_bytes, ret;
AVCodecContext *enc;
AVCodecContext *enc= &ost->st->codec;
/* SC: dynamic allocation of buffers */
if (!audio_buf)
@ -431,21 +432,49 @@ static void do_audio_out(AVFormatContext *s,
if (!audio_buf || !audio_out)
return; /* Should signal an error ! */
enc = &ost->st->codec;
if(audio_sync_method){
double delta = ost->sync_ipts * enc->sample_rate - ost->sync_opts
- fifo_size(&ost->fifo, ost->fifo.rptr)/(ost->st->codec.channels * 2);
double idelta= delta*ist->st->codec.sample_rate / enc->sample_rate;
int byte_delta= ((int)idelta)*2*ist->st->codec.channels;
//FIXME resample delay
if(fabs(delta) > 50){
int comp= clip(delta, -audio_sync_method, audio_sync_method);
assert(ost->audio_resample);
if(verbose > 2)
fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate);
// fprintf(stderr, "drift:%f len:%d opts:%lld ipts:%lld fifo:%d\n", delta, len/4, ost->sync_opts, (int64_t)(ost->sync_ipts * enc->sample_rate), fifo_size(&ost->fifo, ost->fifo.rptr)/(ost->st->codec.channels * 2));
av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
}
if(ist->is_start){
if(byte_delta < 0){
byte_delta= FFMIN(byte_delta, size);
size += byte_delta;
buf -= byte_delta;
if(verbose > 2)
fprintf(stderr, "discarding %d audio samples\n", (int)-delta);
if(!size)
return;
ist->is_start=0;
}else{
static uint8_t *input_tmp= NULL;
input_tmp= av_realloc(input_tmp, byte_delta + size);
if(byte_delta + size <= MAX_AUDIO_PACKET_SIZE)
ist->is_start=0;
else
byte_delta= MAX_AUDIO_PACKET_SIZE - size;
memset(input_tmp, 0, byte_delta);
memcpy(input_tmp + byte_delta, buf, size);
buf= input_tmp;
size += byte_delta;
if(verbose > 2)
fprintf(stderr, "adding %d audio samples of silence\n", (int)delta);
}
}else if(audio_sync_method>1){
int comp= clip(delta, -audio_sync_method, audio_sync_method);
assert(ost->audio_resample);
if(verbose > 2)
fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate);
fprintf(stderr, "drift:%f len:%d opts:%lld ipts:%lld fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(ost->sync_ipts * enc->sample_rate), fifo_size(&ost->fifo, ost->fifo.rptr)/(ost->st->codec.channels * 2));
av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
}
}
}else
ost->sync_opts= lrintf(ost->sync_ipts * enc->sample_rate)
- fifo_size(&ost->fifo, ost->fifo.rptr)/(ost->st->codec.channels * 2); //FIXME wrong
@ -1040,7 +1069,7 @@ static int output_packet(AVInputStream *ist, int ist_index,
AVFrame picture;
short samples[AVCODEC_MAX_AUDIO_FRAME_SIZE / 2];
void *buffer_to_free;
//fprintf(stderr, "output_packet %d, dts:%lld\n", pkt->stream_index, pkt->dts);
if (pkt && pkt->dts != AV_NOPTS_VALUE) { //FIXME seems redundant, as libavformat does this too
ist->next_pts = ist->pts = pkt->dts;
} else {
@ -1487,7 +1516,7 @@ static int av_encode(AVFormatContext **output_files,
ost->audio_resample = 1;
}
}
if(audio_sync_method)
if(audio_sync_method>1)
ost->audio_resample = 1;
if(ost->audio_resample){
@ -1676,6 +1705,7 @@ static int av_encode(AVFormatContext **output_files,
is = input_files[ist->file_index];
ist->pts = 0;
ist->next_pts = 0;
ist->is_start = 1;
}
/* compute buffer size max (should use a complete heuristic) */
@ -1788,6 +1818,7 @@ static int av_encode(AVFormatContext **output_files,
for(i=0; i<file_table[file_index].nb_streams; i++){
int index= file_table[file_index].ist_index + i;
ist_table[index]->next_pts += delta;
ist_table[index]->is_start=1;
}
}
}