[FFmpeg-devel] who should set framesize?

Frank Barchard fbarchard
Thu Jun 25 11:08:53 CEST 2009

On Thu, Jun 25, 2009 at 12:11 AM, Luca Abeni <lucabe72 at email.it> wrote:

> Ronald S. Bultje wrote:
>> Hi Luca,
>> On Wed, Jun 24, 2009 at 10:55 AM, Luca Abeni <lucabe72 at email.it> wrote:
>>  Ronald S. Bultje wrote:
>>>  ffplay rtsp://qtss1.bgsu.edu/wfal08.sdp fails:
>>>> [rtsp @ 0x1032600]Could not find codec parameters (Audio: aac, 22050
>>>> Hz, 2 channels, s16)
>>>> rtsp://qtss1.bgsu.edu/wfal08.sdp: could not find codec parameters
>>>> because AVCodecContext->frame_size is 0. Who should set it? Demuxer?
>>>> how/where?
>>>>  AFAIK, for RTP the frame size is set by the decoder, not by the
>>> demuxer.
>> Aha, you're right, it just takes very long. OK, bugreport then: the file
>> plays twice as fast as it should (Quicktime does it OK), both with ffAAC
>> and
>> with faad. Care to look into this? Want a issue tracker entry?
> Well, this looks like an AAC decoder issue: vlc plays the stream correctly,
> and sees 2 channels at 44100Hz, but ffmpeg sees 2 channels at 22050Hz.
> The funny thing is that the server is also confused about this, because the
> SDP contains a line saying:
>        a=rtpmap:96 mpeg4-generic/22050/2
> I guess it's the AAC SBR thing (someone told me that is "fakes" the sample
> rate, doubling it... But I do not know this stuff, so I really have no
> idea if this is the problem or not). Is SBR supported by ffmpeg?
> Anyway, this is something for people that understand AAC better than me...
> Maybe opening an issue on roundup (with AAC mentioned in the subject) or
> starting a new thread that explicitly mentions AAC would help.

I've got a reproducible case where  ffmpeg produced SBR, but couldnt decode
it (garbled).
it was libfaac with ffmpeg v18867.
The newer ffmpeg v19134 doesnt accept -libfaac... haven't tried -faac; I'm
using the faac.exe tool directly to avoid the issue.
If someone is interested in pursuing the SBR weirdness I can provide files.

>                                Luca
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