[Ffmpeg-devel] Decoded raw audio buffer format

Thanos Kyritsis djart
Sun Nov 6 23:13:38 CET 2005

I have a question regarding the outcome of calling the 
avcodec_decode_audio() function.

For example:
len = avcodec_decode_audio(c, (short *)outbuf, &got_audio, pkt.data, 

What is the format of the raw audio data being put in outbuf ? 
Is it always uncompressed PCM Signed 16 Bit Little Endian containing the 
same number of channels  at the same sample rate as the original 
(input) stream ??

What if I run the code on MacOSX or Windows ? Is it still Little 
endian ?

Is there any way to change/define the format of outbuf's raw contents or 
will I have to recode the samples in such a case ?

Thanos Kyritsis <djart at linux.gr>

- What's your ONE purpose in life ?
- To explode, of course! ;-)

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