Bob, you can simply multiply number of samples and audio time_base. Yurii
вт, 2 окт. 2018 г. в 18:15, Bob Kirnum <[email protected]>: > We are using the libavformat APIs in order to support various containers > including WebM, MKV, MP4, and MOV. We do not use libavcodec APIs for > encoding or decoding, we have our own implementations. Our implementation > is a real-time media server which can play from or record to these > containers. We also offer DVR like controls for skipping ahead or behind. > I am having some difficulty finding a consistent implementation for all > containers to determine the frame times for the audio and video frames we > read. When using some MP4 files, the cur_dts value is always > AV_NOPTS_VALUE so this can't be used. When playing some MKV files, the > audio and video codec context time_base values are inconsistent. For video > the time_base is (1001 / 24000) which can be used to determine the frame > rate and duration (23.976 fps or 41.7 msec). However, the audio values are > (1 / 8000) which is the sample rate not the frame size. It is certainly > possible that I am not understanding this correctly. Can someone recommend > a consistent calculation that will work for audio and video in these > various containers? > > Thanks, > Bob > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user >
_______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
