[gst-devel] adder with multiple RTP multicast streams.

[Olivier Crête]

Hi,

On Thu, 2010-10-14 at 17:25 -0400, Herb Peyerl wrote:
> $ gst-launch-0.10 udpsrc multicast-group=224.1.1.1 port=5000 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)L16,channels=(int)1" !
> gstrtpbin ! rtpL16depay ! audioconvert ! adder ! alsasink

You can't do that using gst-launch. You need to write a small program
that first creates the first part of your pipeline (until gstrtpbin) and
then hooks up to the "pad-added" signal on gstrtpbin and when a new
recv_rtp_src_* pad appears, then you need to add the
depayloader/audioconvert/liveadder/alsasink. And then a second one
appears, you need to add a new depayloader/audioconvert and request a
new pad from liveadder.

To remove timed out sources, set the "autoremove" property on gstrtpbin
to TRUE and then listen for the "pad-removed" signal on gstrtpbin and
remove the following elements in that handler (you can also hook up to
the "unlinked" signal on those created pads, it might be easier to
handle).

You also want to use liveadder, I wrote it exactly for this case.

-- 
Olivier Crête
olivier.crete at collabora.co.uk
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