[Libav-user] # of audio samples, calculated vs. codec context

[Brad O'Hearne]

On May 20, 2013, at 8:39 AM, Brad O'Hearne <brado at bighillsoftware.com> 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