On 5/21/13, Brad O'Hearne <[email protected]> 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? A solution would be great, but if none exists, confirmation that > FFmpeg can't cope with such a scenario is helpful too. >
I'm safe to conclude you are troller. > Thanks for your help. > > Brad > _______________________________________________ > 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
