[Ffmpeg-devel] aac adts input

Rich Felker dalias
Wed Aug 10 13:16:20 CEST 2005

On Wed, Aug 10, 2005 at 10:39:25AM +0200, Michael Niedermayer wrote:
> Hi
> On Tue, Aug 09, 2005 at 11:13:58PM -0400, Justin Ruggles wrote:
> > Justin Ruggles wrote:
> > > Patch coming soon.
> > 
> > And here it is.  This patch moves all the SDL audio initialization to
> > after the codec initialization.  I debated just detecting if the codec
> > init changes the samplerate or number of channels, and if so,
> > reinitializing SDL with new params.  But it seems that moving the entire
> > SDL portion doesn't do any harm.  If anyone disagrees, I can resubmit a
> > patch doing it the other way.
> well, if the file stores 22050hz and FAAD resamples it during decoding then
> the bug is in FAAD and should be fixed there
> if OTOH the file contains 44100hz and the mov headers say its 22050 then 
> maybe the mov demuxer should avoid using this header value for AAC or
> some other solution
> but "fixing" ffplay wont help, as ffplay isnt the only application which 
> uses libav*

the stuff faad does seems broken, but actually it's comparable to
lavc's "lowres" decoding option. i have the opposite problem with
faad: files with 44100 stored in the header, but faad outputs 22050
because sbr doesn't work with the speed-optimized mode i have to
compile faad with in order to make it use less than 90% cpu.


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