ffmpeg audiodecoders after version 1.1 deliver audio samples in planar
formats by default instead of packed as before. Additionally, the AC3
decoder delivers now in planar float.
Use libswresample for sample format conversion, this will work with old
(where it hopefully does nothing) and new ffmpeg versions.
Later on, also sample rate and channel layout conversion could be
implemented if desired.
Tested with ffmpeg versions 1.0.6 and 1.2.1.
This is mostly a dummy implementation except for the dmx class which
should be working. It is intended for testing on PCs with budget
DVB cards which don't have a decoder anyway.
