On May 20, 2013, at 8:39 AM, Brad O'Hearne <[email protected]> wrote:
> I take it by sound of crickets (no response) to my question above that either > I've done a bad job communicating the issue, or it is indeed a real stumper. > In the event that it is the former, I'm going to take another stab at this by > distilling it all down to a very simple question: > > How does one encode decompressed audio received where source data sample > buffers have 512 samples each and a sample rate of 16000, and encode it to a > sample rate of 44100? Given no answer, is it safe to conclude that FFmpeg is unable to deal with this? A solution would be great, but if none exists, confirmation that FFmpeg can't cope with such a scenario is helpful too. Thanks for your help. Brad _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
