From 86b3435fc01049cfa9dc0257ddbf1f531f7cb67a Mon Sep 17 00:00:00 2001 From: Alex Sukhanov Date: Thu, 14 Nov 2013 17:56:36 -0800 Subject: af_aresample: Fix timestamp of first padded PCM audio packet Problem: ffmpeg generated video file which had two audio packets with the same timestamp: last original audio packet and first padded audio packet. Timestamp of first added audio packet by 'apad' fitler had the same value as last original audio packet. The problem was in 'aresample' fitler, which used next pts instead of current one. As long as 'apad' and 'aresample' filters have separate mechanisms of timestamp calculation, they got the same values. Command line: ffmpeg -i -shortest -apad 512 -af asetnsamples=n=512 -b:a 1058400 -ac 1 -ar 44100 -async 0 -acodec pcm_s16le -sn -f matroska -y Fix: Call swr_next_pts() function before swr_convert() Tested: FATE tests passed. Fix has been tested in our Transcoder regression framework on ~10k test videos. It's about ~500k transcodes. Signed-off-by: Michael Niedermayer --- libavfilter/af_aresample.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'libavfilter') diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c index e21b3e4a04..e05c0a179d 100644 --- a/libavfilter/af_aresample.c +++ b/libavfilter/af_aresample.c @@ -230,10 +230,15 @@ static int request_frame(AVFilterLink *outlink) if (ret == AVERROR_EOF) { AVFrame *outsamplesref; int n_out = 4096; + int64_t pts; outsamplesref = ff_get_audio_buffer(outlink, n_out); if (!outsamplesref) return AVERROR(ENOMEM); + + pts = swr_next_pts(aresample->swr, INT64_MIN); + pts = ROUNDED_DIV(pts, inlink->sample_rate); + n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0); if (n_out <= 0) { av_frame_free(&outsamplesref); @@ -242,14 +247,8 @@ static int request_frame(AVFilterLink *outlink) outsamplesref->sample_rate = outlink->sample_rate; outsamplesref->nb_samples = n_out; -#if 0 - outsamplesref->pts = aresample->next_pts; - if(aresample->next_pts != AV_NOPTS_VALUE) - aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base); -#else - outsamplesref->pts = swr_next_pts(aresample->swr, INT64_MIN); - outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate); -#endif + + outsamplesref->pts = pts; return ff_filter_frame(outlink, outsamplesref); } -- cgit v1.2.3