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| author | Stefano Sabatini <stefasab@gmail.com> | 2014-01-08 15:43:25 +0100 |
|---|---|---|
| committer | Stefano Sabatini <stefasab@gmail.com> | 2014-01-09 10:54:38 +0100 |
| commit | 094c500caf0c2a67792f2ee8238f065926aebfc4 (patch) | |
| tree | a933131ab08fec7d6f2b6861472ec8307817374f | |
| parent | 8d22d37e548eea57747005aecbdfa2f76e59f19a (diff) | |
| download | ffmpeg-094c500caf0c2a67792f2ee8238f065926aebfc4.tar.gz | |
examples/muxing: set timestamps in output audio packet
In particular, fix trac ticket #3231.
| -rw-r--r-- | doc/examples/muxing.c | 7 |
1 files changed, 7 insertions, 0 deletions
diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c index 7d80e1c39e..bbc66d693d 100644 --- a/doc/examples/muxing.c +++ b/doc/examples/muxing.c @@ -132,6 +132,7 @@ static int max_dst_nb_samples; uint8_t **dst_samples_data; int dst_samples_linesize; int dst_samples_size; +int samples_count; struct SwrContext *swr_ctx = NULL; @@ -269,8 +270,10 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st) } audio_frame->nb_samples = dst_nb_samples; + audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base); avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt, dst_samples_data[0], dst_samples_size, 0); + samples_count += dst_nb_samples; ret = avcodec_encode_audio2(c, &pkt, audio_frame, &got_packet); if (ret < 0) { @@ -281,6 +284,10 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st) if (!got_packet) return; + /* rescale output packet timestamp values from codec to stream timebase */ + pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); + pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); + pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base); pkt.stream_index = st->index; /* Write the compressed frame to the media file. */ |