[gst-devel] Problem with clockrates in rtp stream

[Morris Ford]

I am trying to stream an mp4v / mp4a rtp stream. The pipeline is at the
moment:

./test-launch --gst-debug-level=0 "( ximagesrc use-damage=false endx=640
endy=640 startx=100 starty=100 ! video/x-raw-rgb,framerate=10/1 ! queue
max-size-buffers=5 ! ffmpegcolorspace ! videoscale !
video/x-raw-yuv,width=640,height=640 ! ffenc_mpeg4 ! queue ! rtpmp4vpay
name=pay0 pt=96 ! rtpmux name=muxer name=pay0 pt=96  audiotestsrc
is-live=true ! audio/x-raw-int,rate=(int)90000 ! faac ! audio/mpegversion=4
! queue ! rtpmp4apay name=pay1 pt=97 ! muxer. muxer. )"

The problem I am having is that the audio part is missing. If I remove the
caps after the faac plugin, the audio is in the stream. When I look at the
startup log I see a clock-rate of 44100 for the mp4a stream and a clock rate
of 90000 for the mp4v stream. I am pretty sure this is what is wrong. If I
stream with the caps for mpegversion=4 removed I see audio and video
clockrates both = 90000.

How do I get the clockrates to match up?

I tried to increase the audio rate to 90000 but it increased to 88200. I
tried to reduce the video clock rate but I could not get it to change.
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