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

Kalileo kalileo at universalx.net
Tue May 21 15:58:51 CEST 2013

On May 21, 2013, at 20:40 , Brad O'Hearne wrote:

> 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? 

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.

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