|
From: Wes M. <wm...@sd...> - 2010-10-19 14:18:31
|
Marco, Better, still not quite right. Removing audioconvert and audioresample on both sender and receiver seem to have little or no effect, so they are now out. Pulsesink is working on the receiver (my Linux workstation/host). I can use pulsesrc on the sender wince Ti/RidgeRun don't seem to include the pulse stuff in their ports of gst. I keep eading about alsa hardware on the Leopardboard...??? I used fakesink to get the sender caps (from fakesink0:Gstpad:sink) and I notice that the ssrc, clock-base and seqnum change every time I run the pipeline. If the clock-base is different each time I start the sender, how can the receiver ever actually match the sender? Is there a tcp-ish way to pass the caps to the receiver and insert them in the receiver pipeline? (sounds like a great, first, element writing project, doesn't it?) I've tried to find out what ssrc is/are and can't find a description. So what is it? Does it matter? As ever, many thanks, Wes -- View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Choppy-Audio-over-UDP-tp2997741p3002180.html Sent from the GStreamer-devel mailing list archive at Nabble.com. |