Hello,
When transcoding a movie, using libav, the audio starts drifting when
resampling 44100 to 48000 khz
I saw this piece of code in ffmpeg.c, but I don't really understand how
ffmpeg calculates the delta.
It is based on the state of input and output stream.
Also, I don't know exactly the value of audio_sync_method...
See this:
double delta = get_sync_ipts(ost) * enc->sample_rate -
ost->sync_opts
- av_fifo_size(&ost->fifo)/(ost->st->codec->channels * 2);
.......
int comp= av_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:%"PRId64"
ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts,
(int64_t)(get_sync_ipts(ost) * enc->sample_rate),
av_fifo_size(&ost->fifo)/(ost->st->codec->channels * 2));
av_resample_compensate(*(struct
AVResampleContext**)ost->resample, comp, enc->sample_rate);
}
Is there any simple way to calculate delta, for instance by comparing the
number of samples written or read?
Arthur
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