ffmpeg | branch: master | Paul B Mahol <one...@gmail.com> | Sun Nov 17 15:21:07 2019 +0100| [2e7ccd493aec98c19f788f2e330bfaf97ac9fbea] | committer: Paul B Mahol
avfilter/f_loop: fix pts handling when timebase and 1/samplerate differ > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2e7ccd493aec98c19f788f2e330bfaf97ac9fbea --- libavfilter/f_loop.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/libavfilter/f_loop.c b/libavfilter/f_loop.c index 5ec44d9da2..879db621ef 100644 --- a/libavfilter/f_loop.c +++ b/libavfilter/f_loop.c @@ -107,7 +107,7 @@ static int push_samples(AVFilterContext *ctx, int nb_samples) } out->pts = s->pts; out->nb_samples = ret; - s->pts += out->nb_samples; + s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); i += out->nb_samples; s->current_sample += out->nb_samples; @@ -145,7 +145,7 @@ static int afilter_frame(AVFilterLink *inlink, AVFrame *frame) drain = FFMAX(0, s->start - s->ignored_samples); s->pts = frame->pts; av_audio_fifo_drain(s->fifo, drain); - s->pts += s->start - s->ignored_samples; + s->pts += av_rescale_q(s->start - s->ignored_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); } s->nb_samples += ret - drain; drain = frame->nb_samples - written; @@ -158,7 +158,7 @@ static int afilter_frame(AVFilterLink *inlink, AVFrame *frame) av_audio_fifo_drain(s->left, drain); } frame->nb_samples = ret; - s->pts += ret; + s->pts += av_rescale_q(ret, (AVRational){1, outlink->sample_rate}, outlink->time_base); ret = ff_filter_frame(outlink, frame); } else { int nb_samples = frame->nb_samples; @@ -169,7 +169,7 @@ static int afilter_frame(AVFilterLink *inlink, AVFrame *frame) } else { s->ignored_samples += frame->nb_samples; frame->pts = s->pts; - s->pts += frame->nb_samples; + s->pts += av_rescale_q(frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); ret = ff_filter_frame(outlink, frame); } @@ -195,7 +195,7 @@ static int arequest_frame(AVFilterLink *outlink) return AVERROR(ENOMEM); av_audio_fifo_read(s->left, (void **)out->extended_data, nb_samples); out->pts = s->pts; - s->pts += nb_samples; + s->pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); ret = ff_filter_frame(outlink, out); if (ret < 0) return ret; _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog To unsubscribe, visit link above, or email ffmpeg-cvslog-requ...@ffmpeg.org with subject "unsubscribe".
