[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 
self-compiled versions.
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.

Carl Eugen

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