[Ffmpeg-devel] Decoded raw audio buffer format
Sun Nov 6 23:13:38 CET 2005
I have a question regarding the outcome of calling the
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
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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