gst-launch-1.0 rtpbin problem

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gst-launch-1.0 rtpbin problem

killerrats
This post was updated on .
I setup a pipeline to receive audio and video but seems to stop at the rtpbin receive it doesn't continue. It shows the caps that i entered but seems like it doesn't find the or what. If i remove the rtpbin it will receive but i need to have the audio and video to be in the one file. Anyone have an answer?

gst-launch-1.0 -e -v rtpbin name=gstrtpbin

udpsrc uri=[video IP] caps=application/x-rtp,media=video,clock-rate=90000 ! gstrtpbin.recv_rtp_sink_0 gstrtpbin.send_rtp_src_0 ! rtpmp4vdepay ! mpeg4videoparse ! mpegpsmux name=mux ! filesink location=test.mp4

udpsrc uri=[audio_IP] caps=application/x-rtp,media=audio ! gstrtpbin.recv_rtp_sink_1 gstrtpbin.send_rtp_src_1 ! rtppcmadepay ! audioparse raw-format=4 rate=32000 channels=1 ! audioconvert dithering=0 ! avenc_mp2 ! mux.

OUTPUT:

WARNING: no real random source present!
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
Setting pipeline to PLAYING ...
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
New clock: GstSystemClock
/GstPipeline:pipeline0/GstRtpBin:gstrtpbin.GstGhostPad:recv_rtp_sink_0.GstProxyPad:proxypad0: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
/GstPipeline:pipeline0/GstRtpBin:gstrtpbin/GstRtpSession:rtpsession0.GstPad:recv_rtp_src: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
/GstPipeline:pipeline0/GstRtpBin:gstrtpbin/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:sink: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
/GstPipeline:pipeline0/GstRtpBin:gstrtpbin/GstRtpSession:rtpsession0.GstPad:recv_rtp_sink: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
/GstPipeline:pipeline0/GstRtpBin:gstrtpbin.GstGhostPad:recv_rtp_sink_0: caps = "application/x-rtp\,\ media\=\(string\)video\,\ clock-rate\=\(int\)90000"
handling interrupt.
Interrupt: Stopping pipeline ...
EOS on shutdown enabled -- Forcing EOS on the pipeline
Waiting for EOS...
handling interrupt.
Interrupt: Stopping pipeline ...
Interrupt while waiting for EOS - stopping pipeline...
Execution ended after 0:00:07.911399224
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
------------------------------
Gstreamer 1.14.3
------------------------------
Windows
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Re: gst-launch-1.0 rtpbin problem

Sebastian Dröge-3
On Mo, 2016-03-14 at 08:37 -0700, killerrats wrote:
> I setup a pipeline to receive audio and video but seems to stop at the rtpbin
> receive it doesn't continue. It shows the caps that i entered but seems like
> it doesn't find the or what. If i remove the rtpbin it will receive but i
> need to have the audio and video to be in the one file. Anyone have an
> answer?

You mean that if you omit the rtpbin and directly link the udpsrc and
depayloader everything works for you? And with rtpbin, it never ever
provides a buffer to the depayloader?

Check if putting rtpjitterbuffer between udpsrc and depayloader still
works then, and if not provide a debug log with GST_DEBUG=3,*rtp*:6 .
Otherwise you'll have to check in the same log where exactly in rtpbin
it stops processing and why.


Note that you don't *have* to use rtpbin to have it all in one
pipeline, or one file for that matter. Except for the rtpjitterbuffer
(which you really should use), rtpbin has no further advantage in your
case here as your pipeline does not use RTCP or anything else that
rtpbin would be helpful for.

--
Sebastian Dröge, Centricular Ltd · http://www.centricular.com


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Re: gst-launch-1.0 rtpbin problem

killerrats
I used the rtpjitterbuffer and it works good i used the following pipeline

gst-launch-1.0 -e -v udpsrc address=[ADDRESS] port=[PORT]  caps=application/x-rtp,media=audio ! rtpjitterbuffer ! queue ! rtppcmadepay ! audioparse raw-format=4 rate=32000 channels=1 ! audioconvert dithering=0 ! lamemp3enc ! mux. udpsrc address=[ADDRESS] port=[PORT] caps=application/x-rtp,media=video ! rtpjitterbuffer ! queue ! rtpmp4vdepay ! mpeg4videoparse ! queue ! mpegpsmux name=mux ! queue name=EOSsink ! filesink buffer-size=80000000 location=[SAVING LOCATION]
------------------------------
Gstreamer 1.14.3
------------------------------
Windows
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Re: gst-launch-1.0 rtpbin problem

Sebastian Dröge-3
On Mi, 2016-03-16 at 09:56 -0700, killerrats wrote:

> I used the rtpjitterbuffer and it works good i used the following
> pipeline
>
> gst-launch-1.0 -e -v udpsrc address=[ADDRESS] port=[PORT] 
> caps=application/x-rtp,media=audio ! rtpjitterbuffer ! queue !
> rtppcmadepay
> ! audioparse raw-format=4 rate=32000 channels=1 ! audioconvert
> dithering=0 !
> lamemp3enc ! mux. udpsrc address=[ADDRESS] port=[PORT]
> caps=application/x-rtp,media=video ! rtpjitterbuffer ! queue !
> rtpmp4vdepay
> ! mpeg4videoparse ! queue ! mpegpsmux name=mux ! queue name=EOSsink !
> filesink buffer-size=80000000 location=[SAVING LOCATION]
So why doesn't rtpbin work then? In your case it's basically doing the
same thing. Did you check from the debug logs?

--
Sebastian Dröge, Centricular Ltd · http://www.centricular.com


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