[FFmpeg-user] libavcodec aac encoding with AV_SAMPLE_FMT_FLT

[Harald Jordan]

Thanks rodger, i didnt know that there is a libavcodec list too.
Anyway, your comment that 1024 may be just some default brought me on the 
right way again! It seems that one can set the framesize but you may not 
vary with it, depending on the capabilities of your encoder.
I will solve the problem by having some audio queue that is able to deliver 
as much samples as the default frame_size of the codec wants to have.

cheers,
harald

-----Ursprüngliche Nachricht----- 
From: Roger Pack
Sent: Friday, August 31, 2012 3:11 PM
To: FFmpeg user questions
Subject: Re: [FFmpeg-user] libavcodec aac encoding with AV_SAMPLE_FMT_FLT

> I am using libavcodec within a cpp project. Basically the program runs
> without any problem, as long as i can use AV_SAMPLE_FMT_S16 for the audio
> encoding part.
> Now when it comes to aac, the aac codec only supports the FLT Sample 
> format
> and here comes the problem: at S16, the frame_size of the audio codec is
> automatically set to bitrate/framerate.
> At 8000 Samples/sec input this means for S16 a framesize of 320. (dec)
> The initial frame_size for AV_SAMPLE_FMT_FLT a framesize of 1024.
> I tried setting the framesize to 320, but this didnt work. So the question
> is: why is the initial framesize of FMT_FLT 1024, i mean 1024 has nothing 
> to
> do with my Sample Rate of 8000, nor with any other known sample rate 
> usually
> used.

I think it's just some default.

> After Resampling the S16 to FLT, the number of samples did not increase, 
> so
> what is the catch here?

You may get more help from the libav-user group.
As a note I recently used the asetnsamples filter to force a specific
sample size, if that helps at all.
Cheers!
-r
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