[Libav-user] AAC decoding problems

[Kalileo]

On Dec 29, 2012, at 18:54 , Funky Factory Development wrote:

> Hi,
> 
> I'm having some issues with AAC decoding with recent version of ffmpeg:
> 
> I'm using this code to decode AAC Frames (from a stream).
> I had some troubles with the decoding - everything runs fine but the after a few seconds playing
> the decoded buffer (m_avAACFrame->data) only contains zeros.
> 
...
> memcpy(outbuffer,*(byte**)&m_avAACFrame->data,*outputsize);
> 
> So i switched to the latest version of ffmpeg (Zeranoe FFmpeg for Windows) and now i have the problem that the output sample format is always AV_SAMPLE_FMT_FLT instead of AV_SAMPLE_FMT_S16 ?
> 
> 
...
> 
> Is AAC not properly supported on ffmpeg ?
> 
> Thanks for your help
> 

The output sample format for aac has indeed changed (on November 26, 2012). 

If you need the old format, then you need to integrate resampling after decoding to convert it to the old format.