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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 <input_filename> -shortest -apad 512 -af asetnsamples=n=512 -b:a 1058400 -ac 1 -ar 44100 -async 0 -acodec pcm_s16le -sn -f matroska -y <output_file>

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 <michaelni@gmx.at>
This commit is contained in:
Alex Sukhanov 2013-11-14 17:56:36 -08:00 committed by Michael Niedermayer
parent 995f450b44
commit 86b3435fc0
1 changed files with 7 additions and 8 deletions
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15
libavfilter/af_aresample.c
View File

@ -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);
}