[Ffmpeg-devel] Decoded raw audio buffer format

[Thanos Kyritsis]

I'm wondering what's so strange or "newbish" about my question that I 
don't deserve an answer ....

If there was enough and/or efficient documentation for the ffmpeg 
library/code, I wouldn't have to ask, that's for sure.


On Monday 07 November 2005 00:13, Thanos Kyritsis wrote:
> 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,
> pkt.size);
>
> 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! ;-)