[Ffmpeg-devel] aac adts input
Tue Aug 9 14:04:47 CEST 2005
Michael Niedermayer wrote:
>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
Also to be taken into consideration is if the file is using SBR, eg
doubbling the frequency. I would guess this is the case here.
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