[Ffmpeg-devel] Decoded raw audio buffer format
Fri Nov 18 17:41:24 CET 2005
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,
> 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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