[FFmpeg-devel] suggestion for expanding audio bitdepth support in libav/ffmpeg

Reimar Döffinger Reimar.Doeffinger
Mon Jan 21 11:45:24 CET 2008

On Mon, Jan 21, 2008 at 11:34:55AM +0100, Michel Bardiaux wrote:
> Andreas ?man a ?crit :
> > Ian Caulfield wrote:
> >> I think the idea was that the codec interface would be changed and a
> >> new function would be added (avcodec_decode_audio3?) to use the new
> >> interface, while avcodec_decode_audio2 would wrap the changes to
> >> maintain the current API...
> > 
> > Yes, but the real bugger is ff_float_to_int16_c() which, as of today,
> > is embedded in the decoders (and thus, the decoders may prescale
> > their coefficients to speed up the conversion). This needs some thought
> > as well. Personally, i would think it would be cleanest (API-wise) if
> > the SAMPLE_FMT_FLT would be data in the range of -1 to +1. I'm gonna
> > make some speed tests on this to check the effect of doing such a
> > change.
> >
> Oh, please! I admit speed must be a priority where video is concerned, 
> but certainly not for sound!

You commit the error of thinking only of totally overblown
desktop/server systems. For embedded systems things like this make the
difference between being able to support and audio format or not (though
admittedly for most of those systems nothing that uses float at all will
In addition I have also heard about servers that decrypt 50 or more
audio streams at once, I guess there things like these will matter as

Reimar D?ffinger

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