[FFmpeg-devel] [PATCH] Get rid of SAMPLE_FMT_S24

Andreas Öman andreas
Sun Nov 18 14:45:29 CET 2007


Baptiste Coudurier wrote:
> Hi
> 
> Andreas ?man a ?crit :
>> hi,
>>
>> see $subj
>>
> 
> Some codecs support 24 bit (alac, flac, aes3, dvd) samples, decoders 
> should output what is stored, ideally, so I don't see why SAMPLE_FMT_S24 
> should be dropped.

Indeed true, but the audio mixing stuff only works with native
endianess. Personally I'm not convinced that the audioconvert API
should deal with endian swapping (just look at the current RGB/BGR mess)

Thus, this would require the codecs to convert the data to native
endianess and when doing that it's probably easier to scale up to
32 bit.

Just my two cents.
I dunno what Michaels reasons are.






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