Hi,

recent ffmpeg/libav versions doesn't decode aac (youtube for instance) to S16
sample format anymore, they switched to FLTP (Planar floating point).
Current gnash implementation expects S16 samples only and resampling from planar
formats doesn't work.
My changes introduce libswresample(ffmpeg)/libavresample(libav).
You can find them in resample branch at [email protected]:gg0/gnash.git

Thanks for reviewing.

--
Gabriele

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