Problem Streaming Live Audio Using RTP and UDPSink
wmetcalf at niftytv.com
Mon Jul 18 11:23:51 PDT 2011
I am trying to capture live audio from a capture card in a gstreamer
application, encode it into AAC format, and then send the audio over a
network using UDPSink and RTP. My method of accomplishing this task
works almost perfectly, except that occasionally the audio will become
very jumpy for a few seconds and then return to normal, and then after a
few seconds it will get jumpy, etc. I am not getting any errors when I
play the audio, so I am assuming that it must be some property I am not
setting correctly, or maybe there is an element I am missing which can
help solve the problem. My pipelines are as follows:
Server Pipeline: appsrc max-bytes=8000 is-live=true typefind=true !
audioparse ! faac bitrate=320000 ! rtpmp4apay host=192.168.42.68 port=52222
Client Pipeline: udpsrc port=52222 !
! rtpmp4adepay !
"audio/mpeg,mpegversion=(int)4,channels=(int)2,rate=(int)44100" ! faad !
I am very close to having this work, so any help that anyone can provide
will be greatly appreciated!
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