Found another thread with similar question: http://libav-users.943685.n4.nabble.com/Libav-user-PTS-values-and-writing-encoded-audio-and-video-frames-td4551001.html
Again, not answered. I do not think no one has ever considered video and audio clocks can slightly drift away from each other, so the lack of feedback is likely because the questions are wrong. I am therefore trying to rephrase my, and all these similar questions, as follows.. Using the sample code referred to, there is a direct connection between a video frame (lasting 1/fps) and an audio frame (fixed length lasting length/samplerate). This works fine as long as the audio and video clock are in perfect sync. However when has only the smallest difference from the other, after a while (could be hours) an error accumulates resulting in a slight AV sync problem. Obviously this problem can be fixed by monitoring this difference and resampling the audio data, but it uses resources. Changing the audio packet size dynamically seems a solution, but it does not seem possible. Is there another way, like "modulating" some time stamp and if so, how? Mike _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
