On May 21, 2013, at 20:40 , Brad O'Hearne wrote: > 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?
That is an assumption, nothing but an assumption. I would not base any action on such assumptions. I also think that it is not a good style to argue like that. Can you convert your audio using ffmpeg command line? If yes I would check the sources of ffmpeg's resampling code. _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
