[Libav-user] Converting audio sample buffer format
Carl Eugen Hoyos
cehoyos at ag.or.at
Mon Feb 25 12:00:26 CET 2013
René J.V. Bertin <rjvbertin at ...> writes:
> The SIMD version ran twice as fast as the scalar version
> until I used gcc 4.7, which has auto-vectorisation
Unfortunately, turning auto-vectorisation on triggers
bugs in gcc and is therefore no option;-(
> > One could argue that on this mailing list, only self-compiled
> > FFmpeg is supported
> One could, but that argument would not be supported by
> the mailing list's own title:
> "This list is about using libavcodec, libavformat,
> libavutil, libavdevice and libavfilter. <libav-user at ...>"
> The concept "using libav*" doesn't imply anything about
> how you obtained the libraries
We only provide sources and therefore only support
And since this is a mailing list for developers, it
makes absolutely no sense to argue "compilation is
so difficult", especially as it is only a question
of doing "./configure && make" if you are not cross-
> and indeed, who built the libraries you're using
> (as opposed to which and how) should be irrelevant.
> It does however constitute a clear invitation to
> post questions like "how do I convert an audio format".
And such questions are welcome here!
As said, I think a cast should not be necessary when
calling swr_convert(), so this is a good start, but
gdb will probably tell you more exactly.
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