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

[Ronald S. Bultje]

Hi,

On Tue, Mar 2, 2010 at 5:57 PM, Michael Niedermayer <michaelni at gmx.at> wrote:
> On Tue, Mar 02, 2010 at 05:51:30PM -0500, Ronald S. Bultje 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:
>> >> 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.
>> }
>>
>
> its not constructed like that

I stand corrected, it's num_channels instead of n_samples. Why is
that? Is there a case where the output would differ from the
channel-agnostic way?

Ronald