[gst-devel] Problem with clockrates in rtp stream

[Wim Taymans]

On Wed, 2010-05-05 at 09:28 -0400, Morris Ford wrote:
> 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. )"
> 

Don't use rtpmux here, it's not needed and it confuses the server
because it expects elements with names pay%d to be unlinked elements.

Wim

> 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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