mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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09ca5fa910
This was done in 0.10 to avoid conflict with the rtp elements in farsight, but the gst-prefixing is no longer needed in 0.11
91 lines
3.1 KiB
Python
Executable file
91 lines
3.1 KiB
Python
Executable file
#! /usr/bin/env python
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import gobject, pygst
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pygst.require("0.10")
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import gst
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#gst-launch -v rtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
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# rtpbin.send_rtp_src_0 ! udpsink port=10000 host=xxx.xxx.xxx.xxx \
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# rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=xxx.xxx.xxx.xxx sync=false async=false \
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# udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0
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DEST_HOST = '127.0.0.1'
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AUDIO_SRC = 'audiotestsrc'
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AUDIO_ENC = 'alawenc'
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AUDIO_PAY = 'rtppcmapay'
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RTP_SEND_PORT = 5002
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RTCP_SEND_PORT = 5003
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RTCP_RECV_PORT = 5007
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# the pipeline to hold everything
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pipeline = gst.Pipeline('rtp_server')
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# the pipeline to hold everything
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audiosrc = gst.element_factory_make(AUDIO_SRC, 'audiosrc')
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audioconv = gst.element_factory_make('audioconvert', 'audioconv')
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audiores = gst.element_factory_make('audioresample', 'audiores')
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# the pipeline to hold everything
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audioenc = gst.element_factory_make(AUDIO_ENC, 'audioenc')
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audiopay = gst.element_factory_make(AUDIO_PAY, 'audiopay')
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# add capture and payloading to the pipeline and link
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pipeline.add(audiosrc, audioconv, audiores, audioenc, audiopay)
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res = gst.element_link_many(audiosrc, audioconv, audiores, audioenc, audiopay)
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# the rtpbin element
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rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
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pipeline.add(rtpbin)
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# the udp sinks and source we will use for RTP and RTCP
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rtpsink = gst.element_factory_make('udpsink', 'rtpsink')
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rtpsink.set_property('port', RTP_SEND_PORT)
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rtpsink.set_property('host', DEST_HOST)
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rtcpsink = gst.element_factory_make('udpsink', 'rtcpsink')
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rtcpsink.set_property('port', RTCP_SEND_PORT)
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rtcpsink.set_property('host', DEST_HOST)
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# no need for synchronisation or preroll on the RTCP sink
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rtcpsink.set_property('async', False)
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rtcpsink.set_property('sync', False)
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rtcpsrc = gst.element_factory_make('udpsrc', 'rtcpsrc')
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rtcpsrc.set_property('port', RTCP_RECV_PORT)
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pipeline.add(rtpsink, rtcpsink, rtcpsrc)
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# now link all to the rtpbin, start by getting an RTP sinkpad for session 0
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sinkpad = gst.Element.get_request_pad(rtpbin, 'send_rtp_sink_0')
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srcpad = gst.Element.get_static_pad(audiopay, 'src')
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lres = gst.Pad.link(srcpad, sinkpad)
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# get the RTP srcpad that was created when we requested the sinkpad above and
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# link it to the rtpsink sinkpad
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srcpad = gst.Element.get_static_pad(rtpbin, 'send_rtp_src_0')
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sinkpad = gst.Element.get_static_pad(rtpsink, 'sink')
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lres = gst.Pad.link(srcpad, sinkpad)
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# get an RTCP srcpad for sending RTCP to the receiver
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srcpad = gst.Element.get_request_pad(rtpbin, 'send_rtcp_src_0')
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sinkpad = gst.Element.get_static_pad(rtcpsink, 'sink')
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lres = gst.Pad.link(srcpad, sinkpad)
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# we also want to receive RTCP, request an RTCP sinkpad for session 0 and
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# link it to the srcpad of the udpsrc for RTCP
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srcpad = gst.Element.get_static_pad(rtcpsrc, 'src')
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sinkpad = gst.Element.get_request_pad(rtpbin, 'recv_rtcp_sink_0')
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lres = gst.Pad.link(srcpad, sinkpad)
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# set the pipeline to playing
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gst.Element.set_state(pipeline, gst.STATE_PLAYING)
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# we need to run a GLib main loop to get the messages
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mainloop = gobject.MainLoop()
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mainloop.run()
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gst.Element.set_state(pipeline, gst.STATE_NULL)
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