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gst/avi/gstavidemux.c: Fix typo in comments.
Original commit message from CVS: * gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows): Fix typo in comments. * tests/examples/rtp/client-H263p-PCMA.sdp: * tests/examples/rtp/client-H263p-PCMA.sh: * tests/examples/rtp/client-H264-PCMA.sdp: * tests/examples/rtp/client-H264-PCMA.sh: * tests/examples/rtp/client-H264.sdp: * tests/examples/rtp/client-H264.sh: * tests/examples/rtp/client-PCMA.sdp: * tests/examples/rtp/client-PCMA.sh: * tests/examples/rtp/server-alsasrc-PCMA.sh: * tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh: * tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh: Add some more docs and fix examples.
This commit is contained in:
parent
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commit
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13 changed files with 273 additions and 52 deletions
18
ChangeLog
18
ChangeLog
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@ -1,3 +1,21 @@
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2008-04-25 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows):
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Fix typo in comments.
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* tests/examples/rtp/client-H263p-PCMA.sdp:
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* tests/examples/rtp/client-H263p-PCMA.sh:
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* tests/examples/rtp/client-H264-PCMA.sdp:
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* tests/examples/rtp/client-H264-PCMA.sh:
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* tests/examples/rtp/client-H264.sdp:
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* tests/examples/rtp/client-H264.sh:
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* tests/examples/rtp/client-PCMA.sdp:
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* tests/examples/rtp/client-PCMA.sh:
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* tests/examples/rtp/server-alsasrc-PCMA.sh:
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* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
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* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
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Add some more docs and fix examples.
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2008-04-25 Sebastian Dröge <slomo@circular-chaos.org>
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* tests/check/elements/multifile.c:
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@ -3361,7 +3361,7 @@ gst_avi_demux_combine_flows (GstAviDemux * avi, avi_stream_context * stream,
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/* if we get here, all other pads were unlinked and we return
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* NOT_LINKED then */
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done:
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GST_LOG_OBJECT (avi, "cobined return %s", gst_flow_get_name (ret));
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GST_LOG_OBJECT (avi, "combined return %s", gst_flow_get_name (ret));
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return ret;
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}
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@ -1,12 +1,12 @@
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v=0
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o=- 1188340656180883 1 IN IP4 192.168.1.110
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o=- 1188340656180883 1 IN IP4 127.0.0.1
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s=Session streamed by GStreamer
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i=server.sh
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t=0 0
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a=tool:GStreamer
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a=type:broadcast
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m=video 5000 RTP/AVP 96
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c=IN IP4 0.0.0.0
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c=IN IP4 127.0.0.1
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a=rtpmap:96 H263-1998/90000
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m=audio 5002 RTP/AVP 8
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c=IN IP4 0.0.0.0
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c=IN IP4 127.0.0.1
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@ -6,12 +6,17 @@
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VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998"
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AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
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gst-launch -v gstrtpbin name=rtpbin \
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#DEST=192.168.1.126
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DEST=localhost
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LATENCY=100
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gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \
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udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
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rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
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udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
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rtpbin.send_rtcp_src_0 ! udpsink host=$DEST port=5005 sync=false async=false \
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udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \
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rtpbin. ! rtppcmadepay ! alawdec ! alsasink \
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rtpbin. ! rtppcmadepay ! alawdec ! alsasink buffer-time=10000 \
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udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
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rtpbin.send_rtcp_src_1 ! udpsink host=$DEST port=5007 sync=false async=false
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@ -1,12 +1,12 @@
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v=0
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o=- 1188340656180883 1 IN IP4 192.168.1.110
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o=- 1188340656180883 1 IN IP4 127.0.0.1
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s=Session streamed by GStreamer
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i=server.sh
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t=0 0
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a=tool:GStreamer
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a=type:broadcast
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m=video 5000 RTP/AVP 96
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c=IN IP4 0.0.0.0
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c=IN IP4 127.0.0.1
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a=rtpmap:96 H264/90000
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m=audio 5002 RTP/AVP 8
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c=IN IP4 0.0.0.0
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c=IN IP4 127.0.0.1
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@ -2,16 +2,57 @@
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#
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# A simple RTP receiver
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#
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# receives H264 encoded RTP video on port 5000, RTCP is received on port 5001.
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# the receiver RTCP reports are sent to port 5006
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# receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003.
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# the receiver RTCP reports are sent to port 5007
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#
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# .-------. .----------. .---------. .-------. .-----------.
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# RTP |udpsrc | | rtpbin | |h264depay| |h264dec| |xvimagesink|
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# port=5000 | src->recv_rtp recv_rtp->sink src->sink src->sink |
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# '-------' | | '---------' '-------' '-----------'
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# | |
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# | | .-------.
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# | | |udpsink| RTCP
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# | send_rtcp->sink | port=5005
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# .-------. | | '-------' sync=false
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# RTCP |udpsrc | | | async=false
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# port=5001 | src->recv_rtcp |
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# '-------' | |
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# | |
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# .-------. | | .---------. .-------. .--------.
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# RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink|
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# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink |
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# '-------' | | '---------' '-------' '--------'
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# | |
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# | | .-------.
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# | | |udpsink| RTCP
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# | send_rtcp->sink | port=5007
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# .-------. | | '-------' sync=false
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# RTCP |udpsrc | | | async=false
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# port=5003 | src->recv_rtcp |
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# '-------' '----------'
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# the destination machine to send RTCP to. This is the address of the sender and
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# is used to send back the RTCP reports of this receiver. If the data is sent
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# from another machine, change this address.
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DEST=127.0.0.1
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# this adjusts the latency in the receiver
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LATENCY=200
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# the caps of the sender RTP stream. This is usually negotiated out of band with
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# SDP or RTSP. normally these caps will also include SPS and PPS but we don't
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# have a mechanism to get this from the sender with a -launch line.
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VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
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AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
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gst-launch -v gstrtpbin name=rtpbin latency=100 \
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gst-launch -v gstrtpbin name=rtpbin latency=$LATENCY \
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udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
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rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \
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udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
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rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false async=false \
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udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \
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rtpbin. ! rtppcmadepay ! alawdec ! alsasink slave-method=1 \
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rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
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udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
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rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=$DEST sync=false async=false
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10
tests/examples/rtp/client-H264.sdp
Normal file
10
tests/examples/rtp/client-H264.sdp
Normal file
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v=0
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o=- 1188340656180883 1 IN IP4 127.0.0.1
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s=Session streamed by GStreamer
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i=server.sh
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t=0 0
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a=tool:GStreamer
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a=type:broadcast
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m=video 5000 RTP/AVP 96
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c=IN IP4 127.0.0.1
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a=rtpmap:96 H264/90000
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36
tests/examples/rtp/client-H264.sh
Executable file
36
tests/examples/rtp/client-H264.sh
Executable file
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#!/bin/sh
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#
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# A simple RTP receiver
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#
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# receives H264 encoded RTP video on port 5000, RTCP is received on port 5001.
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# the receiver RTCP reports are sent to port 5006
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#
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# .-------. .----------. .---------. .-------. .-----------.
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# RTP |udpsrc | | rtpbin | |h264depay| |h264dec| |xvimagesink|
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# port=5000 | src->recv_rtp recv_rtp->sink src->sink src->sink |
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# '-------' | | '---------' '-------' '-----------'
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# | |
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# | | .-------.
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# | | |udpsink| RTCP
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# | send_rtcp->sink | port=5005
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# .-------. | | '-------' sync=false
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# RTCP |udpsrc | | | async=false
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# port=5001 | src->recv_rtcp |
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# '-------' '----------'
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# the caps of the sender RTP stream. This is usually negotiated out of band with
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# SDP or RTSP. normally these caps will also include SPS and PPS but we don't
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# have a mechanism to get this from the sender with a -launch line.
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VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
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# the destination machine to send RTCP to. This is the address of the sender and
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# is used to send back the RTCP reports of this receiver. If the data is sent
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# from another machine, change this address.
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DEST=127.0.0.1
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gst-launch -v gstrtpbin name=rtpbin \
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udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
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rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \
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udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false async=false
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9
tests/examples/rtp/client-PCMA.sdp
Normal file
9
tests/examples/rtp/client-PCMA.sdp
Normal file
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v=0
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o=- 1188340656180883 1 IN IP4 127.0.0.1
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s=Session streamed by GStreamer
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i=server.sh
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t=0 0
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a=tool:GStreamer
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a=type:broadcast
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m=audio 5002 RTP/AVP 8
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c=IN IP4 127.0.0.1
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@ -2,11 +2,34 @@
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#
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# A simple RTP receiver
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#
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# receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003.
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# the receiver RTCP reports are sent to port 5007
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#
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# .-------. .----------. .---------. .-------. .--------.
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# RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink|
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# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink |
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# '-------' | | '---------' '-------' '--------'
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# | |
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# | | .-------.
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# | | |udpsink| RTCP
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# | send_rtcp->sink | port=5007
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# .-------. | | '-------' sync=false
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# RTCP |udpsrc | | | async=false
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# port=5003 | src->recv_rtcp |
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# '-------' '----------'
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# the caps of the sender RTP stream. This is usually negotiated out of band with
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# SDP or RTSP.
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AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
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# the destination machine to send RTCP to. This is the address of the sender and
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# is used to send back the RTCP reports of this receiver. If the data is sent
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# from another machine, change this address.
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DEST=127.0.0.1
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gst-launch -v gstrtpbin name=rtpbin \
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udpsrc caps=$AUDIO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \
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rtpbin. ! rtppcmadepay ! alawdec ! alsasink \
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udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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rtpbin.send_rtcp_src_0 ! udpsink port=5003 sync=false async=false
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udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
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rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
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udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
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rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
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@ -1,10 +1,35 @@
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#!/bin/sh
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#
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# A simple RTP server
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# sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
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# port 5003. The destination is 127.0.0.1.
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# the receiver RTCP reports are received on port 5007
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#
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# .-------. .-------. .-------. .----------. .-------.
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# |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
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# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
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# '-------' '-------' '-------' | | '-------'
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# | |
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# | | .-------.
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# | | |udpsink| RTCP
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# | send_rtcp->sink | port=5003
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# .-------. | | '-------' sync=false
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# RTCP |udpsrc | | | async=false
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# port=5007 | src->recv_rtcp |
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# '-------' '----------'
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# change this to send the RTP data and RTCP to another host
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DEST=127.0.0.1
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#AELEM=alsasrc
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AELEM=audiotestsrc
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# PCMA encode from an the source
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ASOURCE="$AELEM ! audioconvert"
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AENC="alawenc ! rtppcmapay"
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gst-launch -v gstrtpbin name=rtpbin \
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alsasrc ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
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rtpbin.send_rtp_src_0 ! udpsink port=5000 \
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rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
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udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0
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$ASOURCE ! $AENC ! rtpbin.send_rtp_sink_0 \
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rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
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rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
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udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0
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@ -11,12 +11,14 @@ VOFFSET=0
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VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1"
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DEST=192.168.1.126
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gst-launch -v gstrtpbin name=rtpbin \
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v4l2src ! $VCAPS ! videorate ! ffmpegcolorspace ! ffenc_h263p ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
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rtpbin.send_rtp_src_0 ! queue ! udpsink port=5000 ts-offset=$AOFFSET \
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rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
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rtpbin.send_rtp_src_0 ! queue ! udpsink host=$HOST port=5000 ts-offset=$AOFFSET \
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rtpbin.send_rtcp_src_0 ! udpsink host=$HOST port=5001 sync=false async=false \
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udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
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alsasrc ! audioconvert ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
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rtpbin.send_rtp_src_1 ! queue ! udpsink port=5002 ts-offset=$VOFFSET \
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rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
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rtpbin.send_rtp_src_1 ! queue ! udpsink host=$HOST port=5002 ts-offset=$VOFFSET \
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rtpbin.send_rtcp_src_1 ! udpsink host=$HOST port=5003 sync=false async=false \
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udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
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@ -1,26 +1,78 @@
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#!/bin/sh
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#
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# A simple RTP server
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# sends the output of v4l2src as h264 encoded RTP on port 5000, RTCP is sent on
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# port 5001. The destination is 127.0.0.1.
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# the video receiver RTCP reports are received on port 5005
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# sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
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# port 5003. The destination is 127.0.0.1.
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# the receiver RTCP reports are received on port 5007
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#
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# .-------. .-------. .-------. .----------. .-------.
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# |v4lssrc| |h264enc| |h264pay| | rtpbin | |udpsink| RTP
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# | src->sink src->sink src->send_rtp send_rtp->sink | port=5000
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# '-------' '-------' '-------' | | '-------'
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# | |
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# | | .-------.
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# | | |udpsink| RTCP
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# | send_rtcp->sink | port=5001
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# .-------. | | '-------' sync=false
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# RTCP |udpsrc | | | async=false
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# port=5005 | src->recv_rtcp |
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# '-------' | |
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# | |
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# .-------. .-------. .-------. | | .-------.
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# |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
|
||||
# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
|
||||
# '-------' '-------' '-------' | | '-------'
|
||||
# | |
|
||||
# | | .-------.
|
||||
# | | |udpsink| RTCP
|
||||
# | send_rtcp->sink | port=5003
|
||||
# .-------. | | '-------' sync=false
|
||||
# RTCP |udpsrc | | | async=false
|
||||
# port=5007 | src->recv_rtcp |
|
||||
# '-------' '----------'
|
||||
#
|
||||
# ideally we should transport the properties on the RTP udpsink pads to the
|
||||
# receiver in order to transmit the SPS and PPS earlier.
|
||||
|
||||
# change this to send the RTP data and RTCP to another host
|
||||
DEST=127.0.0.1
|
||||
|
||||
# tuning parameters to make the sender send the streams out of sync. Can be used
|
||||
# ot test the client RTCP synchronisation.
|
||||
#VOFFSET=500000000
|
||||
VOFFSET=900000000
|
||||
AOFFSET=0
|
||||
VOFFSET=0
|
||||
|
||||
# H264 encode from a v4l2src
|
||||
# H264 encode from the source
|
||||
VELEM="v4l2src"
|
||||
#VELEM="videotestsrc is-live=1"
|
||||
VCAPS="video/x-raw-yuv,width=352,height=288,framerate=15/1"
|
||||
VSOURCE="v4l2src ! $VCAPS ! videorate ! ffmpegcolorspace"
|
||||
VSOURCE="$VELEM ! $VCAPS ! queue ! videorate ! ffmpegcolorspace"
|
||||
VENC="x264enc byte-stream=true bitrate=300 ! rtph264pay"
|
||||
|
||||
# PCMA encode from an alsasrc
|
||||
ASOURCE="alsasrc ! audioconvert"
|
||||
VRTPSINK="udpsink port=5000 host=$DEST ts-offset=$VOFFSET name=vrtpsink"
|
||||
VRTCPSINK="udpsink port=5001 host=$DEST sync=false async=false name=vrtcpsink"
|
||||
VRTCPSRC="udpsrc port=5005 name=vrtpsrc"
|
||||
|
||||
# PCMA encode from an the source
|
||||
AELEM="alsasrc"
|
||||
#AELEM="audiotestsrc is-live=1"
|
||||
ASOURCE="$AELEM ! queue ! audioconvert"
|
||||
AENC="alawenc ! rtppcmapay"
|
||||
|
||||
ARTPSINK="udpsink port=5002 host=$DEST ts-offset=$AOFFSET name=artpsink"
|
||||
ARTCPSINK="udpsink port=5003 host=$DEST sync=false async=false name=artcpsink"
|
||||
ARTCPSRC="udpsrc port=5007 name=artpsrc"
|
||||
|
||||
gst-launch -v gstrtpbin name=rtpbin \
|
||||
$VSOURCE ! $VENC ! rtpbin.send_rtp_sink_0 \
|
||||
rtpbin.send_rtp_src_0 ! queue ! udpsink port=5000 ts-offset=$AOFFSET \
|
||||
rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
|
||||
udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
|
||||
rtpbin.send_rtp_src_0 ! $VRTPSINK \
|
||||
rtpbin.send_rtcp_src_0 ! $VRTCPSINK \
|
||||
$VRTCPSRC ! rtpbin.recv_rtcp_sink_0 \
|
||||
$ASOURCE ! $AENC ! rtpbin.send_rtp_sink_1 \
|
||||
rtpbin.send_rtp_src_1 ! queue ! udpsink port=5002 ts-offset=$VOFFSET \
|
||||
rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
|
||||
udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
|
||||
rtpbin.send_rtp_src_1 ! $ARTPSINK \
|
||||
rtpbin.send_rtcp_src_1 ! $ARTCPSINK \
|
||||
$ARTCPSRC ! rtpbin.recv_rtcp_sink_1
|
||||
|
|
Loading…
Reference in a new issue