adder with multiple RTP multicast streams.

[Herb Peyerl]

So I'm struggling with this ... Imagine a network of devices with
microphones and speakers.  Each device is running gstreamer.
Devices can come and go over time and the number of devices is not
known in advance... Each device transmits its audio on 224.1.1.1:5000.
Each device receives multiple streams on 224.1.1.1:5000 and uses
gstrtpbin to demux them based on SSRC and wants to feed them into
an adder and sink them on an alsasink.

I understand there's a problem with 'adder' in that it wants to
collect data from all pads before adding anything which violates
my usage model.  I see also that 'liveadder' can accomplish this
in theory.

So as a test, my rtp sources look like this:

gst-launch audiotestsrc is-live=true freq=660 ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=8000 ! rtpL16pay ! udpsink host=224.1.1.1 port=5000
gst-launch audiotestsrc is-live=true freq=330 ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=8000 ! rtpL16pay ! udpsink host=224.1.1.1 port=5000

And I'm using 'adder' in this manner:

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

my problem is that it only 'works' if I start one of the sources
first, then launch the adder pipeline. Then I get a tone. If I
launch the adder pipeline first and then launch a single audiotestsrc
pipeline next, I get a short 50ms 'bleep' and then silence; presumably
because adder is confused about the sources...

I can't actually seem to add/remove RTP streams dynamically...

The same sort of thing happens with liveadder except that it doesn't
matter the order in which I start the audiotestsrc stream.. But as
soon as I add the second stream, I get the following:

$ 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
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2507): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)
Execution ended after 245538038 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

or more precisely:

0:00:09.701670986 23790 0xb4d11818 LOG           GST_SCHEDULING gstpad.c:4408:gst_pad_push_data:<rtpbin0:recv_rtp_src_0_4055336445_96> pushing, but it was not linked

So I speculate that what I need to do is build a new container that
can catch on-new-ssrc and payload-type-change which will allow me
to construct the recv_rtp_src_%d_%d_%d pad and link it to something...
But I don't know what ...

Anyway, I feel like I'm barking up the wrong tree but it's a big forest
and I can't make sense of the map..

Does anyone have a better pipeline or approach to solving my problem?

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[Olivier Crête-2]

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
[hidden email]

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[ej]

Hello Olivier,

I'm trying to code the small program as you described in your reply but I do not succeed.

Perhaps you can help me.

The elements in my pipeline are
udpsrc, gstrtpbin, rtpL16depay, audioconv, audioresample, liveadder, alsasink sync=false

I have tried two options one works but is not what I need and the other doesn't work.

Option 1
I create all elements in the main program, create links between all elements except between gstrtpbin and rtpL16depay. Then set the pipeline to playing.

In the "on-pad-added" function I link the pads of the gstrtpbin and the rtpL16depay.
I have a working pipeline but I can not dynamically create (rtpL16depay, audioconv, audioresample) on a new rtp stream.

Option 2
I create udpsrc, gstrtpbin, liveadder, alsasink in the main program, create links between udpsrc, gstrtpbin and between liveadder and alsasink. then connect a signal to gstrtpbin with liveadder as data. Then set the pipeline to playing.

In the "on-pad-added" function I create the rtpL16depay, audioconv, audioresample elements and link them together. Then I link audioresample to liveadder (given via data) and I link the new gstrtpbin pad to a static sink pad of the new rtpL16depay element. I get no errors but also no sound in this option. I've tried to set the pipeline to PLAYING again in the "on-pad-added" function but that didn't help.

Do you see my problem or do you have an example program for me?

Thanks,
Ernst-Jan

[Biloute]

Hi.

I'm not saying I have the answer to your problem, but have a look at the rtp example code in the test/examples/rtp folder in your gst-plugins-good folder.

You need to have an rtp server for each one of your sources (assuming they are on different machines). If

Good luck
Biloute