On 2/25/13, Rene J.V. Bertin <[email protected]> wrote: > > > Carl Eugen Hoyos <[email protected]> wrote: > >>Brad O'Hearne <brado@...> writes: >> >>> On Feb 18, 2013, at 3:50 PM, Carl Eugen Hoyos <cehoyos@...> wrote: >>> >>> > While I have _no_ idea what the "flv audio codec" could >>> > be, please use either the aconvert filter or libswresample >>> > directly to convert from one audio format to another. >>> >>> This has turned out to be much more difficult than expected. >> >>Before you start debugging (the cast to sourceData looks >>suspicious): Did you look at doc/examples/filtering_audio.c >>and doc/examples/resampling_audio.c ? >>I suspect using the aconvert filter has the advantage that >>you can do other changes to the audio without additional >>code (and bugs). >> >>In any case, using gdb should quickly show you were the >>problem lies. >> >>Carl Eugen >> >>_______________________________________________ >>Libav-user mailing list >>[email protected] >>http://ffmpeg.org/mailman/listinfo/libav-user > > > Exactly, but you'd need to build the libav libs yourself, with debugging > info. > > There's another thing that's nagging me. IIUC, the goal here is to convert a > buffer of (C) floats into signed shorts. I have some difficulty believing > that doing this through a generic workhouse function can be more efficient > than writing a simple loop and let a good optimising compiler create the > best assembly out of it ...
Why not do it? And than compare speed. You will learn a lot. > > R. > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user > _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
