[Libav-user] Converting audio sample buffer format

[Brad O'Hearne]

On Feb 25, 2013, at 10:43 AM, "René J.V. Bertin" <rjvbertin at gmail.com> wrote:

> No need for that, as noted above. All you need to do is drag the FFmpeg folder (the source folder, not your build directory) into your project and make sure the code is not added to any of your projects. Xcode will index the code, and its gdb interface will use that information to give you access to the source code for the library functions.
> NB: gcc and clang accept -g with -O{,1,2,3,s}.

I'll give it a try and see if I can't get it to play ball. 

> Another option would be to run the analyser on your code (supposing Xcode still has that option).

I've already got the static analyzer in play -- 100% clean. 

> **) don't x86 CPUs have multiple settings that control the way calculations are rounded off, or am I confounding with instruction variants that round off differently?

I'm building for 64-bit. But anyway, I'm not sure that changing rounding necessarily solves the problem, as any approach could result in 0 -- perhaps not for a selected sample, but eventually, for other random samples, I would think it possible.

Brad