[FFmpeg-devel] who should set framesize?

Luca Abeni lucabe72
Thu Jun 25 09:11:28 CEST 2009

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.


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