[FFmpeg-devel] AAC decoder round 9

[Robert Swain]

2008/8/21 madshi <dear at madshi.net>:
> Robert Swain schrieb:
>  > There may be some areas where we don't conform
>  > because the specification has a significantly poor design
>  > decision that we feel we wish to override
>
> So that means you hope to improve decoding quality
> over other decoders?

Actually, it's more of a code cleanliness/performance issue. If
handling window transitions in a sane way causes no audible artifacts
then we consider it better to do it that way than do to bizarre and
unusual things that are suggested in the specification. Even the
specification calls the unusual things 'meaningless'.

>  > There is work going on to allow various audio sample
>  > formats. AAC decodes to float as AC3, etc. but only
>  > supports s16le output at the moment. It would be
>  > easy enough to add 32-bit float output though I suppose.
>
> The MLP and MPEG audio decoders support (or at least
> used to support) the following switch to enable native
> bit depth/format output:
>
> #define CONFIG_AUDIO_NONSHORT 1
>
> Maybe you could add support for that, too? That would
> be very welcome (by me, at least).

A sample format support 'framework' has been added to the internal
libavcodec APIs. We should be able to check the requested sample
format using that. It shouldn't take massive editing of the code, but
it's not a priority. I'll see how it should be done.

Regards,
Rob