I want to decode some audio streams in the format aac. I already found an example of MP3 and MP2 decoding here: http://ffmpeg.org/doxygen/trunk/decoding__encoding_8c-source.html at line 236. But this code doesn't work for files in the format aac. I only want the raw data, so the decoding from aac into raw data, but i don't find anything. Anyone here with help? With the demuxing.c code: http://www.ffmpeg.org/doxygen/trunk/doc_2examples_2demuxing_8c-example.html the aac data decodes, but I can't play it. I often read that it don't decode in the pcm s16le format?
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It would probably be easier to use libavformat to configure your codecContext for you rather than manually.
If it decodes to something than s16le, you can use swresample to change the sample format to anything you want.
If you are struggling with libav*, you should probably now be looking at the examples you provided. They assume you know a lot about media decoding already. The http://dranger.com/ffmpeg tutorial may be better for you.