[FFmpeg-devel] [PATCH] audio conversion clipping/overflows

Michael Niedermayer michaelni
Mon Mar 15 22:40:38 CET 2010


On Wed, Mar 03, 2010 at 01:54:37PM -0500, Ronald S. Bultje wrote:
> Hi,
> 
> On Tue, Mar 2, 2010 at 6:36 AM, Michael Niedermayer <michaelni at gmx.at> wrote:
> > On Mon, Mar 01, 2010 at 09:18:58AM -0500, Ronald S. Bultje wrote:
> >> ?libavcodec/audioconvert.c | ? 18 ++++++++++++------
> >> ?libavutil/common.h ? ? ? ?| ? 11 +++++++++++
> >> ?2 files changed, 23 insertions(+), 6 deletions(-)
> >> 3cbfbc44e4362ec7017ebdc358b3021319023bf1 ?aconv.patch
> >
> > ok but you know this needs optimizations
> 
> So let's talk optimization then (since removing it from decoders is
> something I can probably do without screwing up multiple times)... How
> do you suggest I optimize this? Or would you prefer / allow / accept
> if I do that separately afterwards?

iam ok with optims afterwards


> I'm guessing you'd like
> pix_fmt-style declarations of individual from/to-smp_fmt functions,
> which can then individually be SIMD'ified for those used often (as I
> said before: float <-> int16) or so?

possible
but do this in small steps so you dont have to redo alot if someone
has a better idea for a design


> Or is there a simpler way that
> I'm missing?

runtime generated code :)

[...]
-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

I do not agree with what you have to say, but I'll defend to the death your
right to say it. -- Voltaire
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