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

[Måns Rullgård]

Benjamin Larsson wrote:
> 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?

SIMD availability is irrelevant.  That said, being able to select scale
and bias would make sense.

-- 
M?ns Rullg?rd
mans at mansr.com