[Ffmpeg-devel] aac adts input

[Benjamin Larsson]

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*
>
>[...]
>
>  
>
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.

MvH
Benjamin Larsson