[FFmpeg-devel] [PATCH] Handle MP3ADU in RealRTSP, restructure rtpdec/rtsp handling of AVStream time_base

Martin Storsjö martin
Sun Dec 5 12:59:52 CET 2010


Hi,

The attached patchset adds support for playing the 
rtsp://dl.lib.brown.edu:554/areserves/1093545294660883.mp3 stream, as 
added by Compn to the RTSP sample URL list a few days ago.

In order to play this back properly, I had to restructure the handling of 
AVStream time_base in rtsp/rtpdec. Earlier, time_base has always been set 
blindly in rtpdec to 90 kHz for video, and to the sample rate for audio. 
In this stream, the rtpmap says

a=rtpmap:101 X-MP3-draft-00/1000

(and RTP timestamps increase by 1000 per second) while the SDP a few lines 
later says

a=SampleRate:integer;44100

With the current code in rtpdec.c, where the AVStream time_base always is 
overwritten either with a default value or with the sample rate, we can't 
really handle it properly. The attached patchset changes this to set 
time_base within rtsp.c at the same time as sample_rate is set initially, 
so that the later SDP line can override only the sample rate, not the time 
base.

I've tested this change with quite a few different streams, and didn't see 
any regression anywhere, but please do check if you know of any weird 
stream that might break.

// Martin
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