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

Benjamin Larsson banan
Mon Jan 21 11:34:42 CET 2008

Andreas ?man wrote:
> 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.

Well I'm not sure that would work with 24bits data. And I don't like the 
idea of codecs outputting scaled samples that something else has to 
rescale. What's wrong with codecs being able to output regular float and 
fast_float(range in 384 to 386) depending on the availability of SIMD?

Benjamin Larsson

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