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

[Ronald S. Bultje]

Hi,

On Wed, Mar 3, 2010 at 2:54 PM, Ronald S. Bultje <rsbultje at gmail.com> wrote:
> 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?

Ping? Or should I leave optimizations for later and first fix the bug
(= apply the patch mentioned earlier, and remove related workarounds
in some - particularly voice-based - decoders)?

Ronald