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

[Vitor Sessak]

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:
>>> float 1.0 rounds to int INT_MIN (whatever int size). This leads to
>>> considerable audio clipping artifacts.
>>>
>>> Attached patch is a lame attempt of mine to fix this. It probably
>>> makes the code slower, so sue me. :-).
>>>  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
> 
> Well, yeah. I was wondering why the whole thing is constructed as:
> 
> for (n=0;n<n_samples;n++) {
>   if (from==.. && to=..) CONV...;
>   else if .. etc.
> }
> 
> Instead of the more logical pix_fmt way, i.e. a table of conversion
> functions, some of which could optionally be SIMD'ified (e.g.
> float<->int16)?
> 
>> also if you apply this, make sure you remove the cliping from every decoder
>> that matches SAMPLE_FMT_FLT
> 
> OK, separate commit OK or same? I know this affects SIPRO and QCELP at
> the very least. Vitor had some concerns about this also...

I was concerned about making codecs that does not need clipping slower, 
but since there is no such codec using SAMPLE_FMT_FLT ATM, I have 
nothing against it...

-Vitor