[Libav-user] Converting audio sample buffer format

Carl Eugen Hoyos cehoyos at ag.or.at
Mon Feb 25 10:29:33 CET 2013

Brad O'Hearne <brado at ...> writes:

> On Feb 18, 2013, at 3:50 PM, Carl Eugen Hoyos <cehoyos at ...> 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

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