[Ffmpeg-devel] aac adts input

Michael Niedermayer michaelni
Wed Aug 10 10:39:25 CEST 2005


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*



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