ffmpeg | branch: master | Andreas Unterweger <dustsi...@gmail.com> | Tue Jan 27 09:03:08 2015 +0100| [3a70c0c95feacb3844d05eebd579fc8189a77eee] | committer: Anton Khirnov
examples/transcode_aac: generate proper PTS and set the muxer timebase Signed-off-by: Anton Khirnov <an...@khirnov.net> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3a70c0c95feacb3844d05eebd579fc8189a77eee --- doc/examples/transcode_aac.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index e0efe2d..60698f7 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -182,6 +182,10 @@ static int open_output_file(const char *filename, /** Allow the use of the experimental AAC encoder */ (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL; + /** Set the sample rate for the container. */ + stream->time_base.den = input_codec_context->sample_rate; + stream->time_base.num = 1; + /** * Some container formats (like MP4) require global headers to be present * Mark the encoder so that it behaves accordingly. @@ -553,6 +557,9 @@ static int init_output_frame(AVFrame **frame, return 0; } +/** Global timestamp for the audio frames */ +static int64_t pts = 0; + /** Encode one frame worth of audio to the output file. */ static int encode_audio_frame(AVFrame *frame, AVFormatContext *output_format_context, @@ -564,6 +571,12 @@ static int encode_audio_frame(AVFrame *frame, int error; init_packet(&output_packet); + /** Set a timestamp based on the sample rate for the container. */ + if (frame) { + frame->pts = pts; + pts += frame->nb_samples; + } + /** * Encode the audio frame and store it in the temporary packet. * The output audio stream encoder is used to do this. _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog
