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Hi list,

I try to use libavcodec to decode a MP3-Stream, similar to what you get
when you connect to a shoutcast/icecast (MP3 source) stream.
I didn't find much documentation but I read the api-example.c (which
seems to be outdated:
http://lists.mplayerhq.hu/pipermail/ffmpeg-user/2007-September/011258.html)
and I had a look at the "howtos" by Martin Böhme and Stephen Dranger.
When using the "decode audio"-part from the api-example.c on a normal
MP3-file, I also get "Incorrect frame size"-errors/warnings.
In my application, the MP3-data lies in a buffer with a fixed size. The
size can be set at compile time and is static at runtime. Using code
similar to the one used in api-example.c throws many "Incorrect frame
size"-messages. Does the codec expect that I feed exactly one frame to
it at a time? Is libavformat the way to go then (though I have a raw
MP3-stream not encapsulated in a container)?
Sure, I could use libmad or some other codec that have a lighter API but
I expect to extend to other formats beside MP3 so it seems wise to use a
multi-codec-library from the beginning.
My ffmpeg version is 0.6

Best regards,
        Jannis
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