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

[Ronald S. Bultje]

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

Ronald