Open trevorirwin opened 4 years ago
The best approach to this would be to measure the standard deviation of elapsed time between packets arriving and set the buffer to some multiple of that.
This would be really nice for both testing and more exact timing (we're configuring everything with wall clock time, but really we don't care about that, just how many samples of time have elapsed in the audio system).
This is surprisingly tricky to do. To expand or shrink the buffer you basically have two options:
The very simple jitter buffer implemented in #54 is functional but very simple. There are many improvements that could still be made. Suggestions so far include: