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 ... R. _______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
