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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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115 lines
4.2 KiB
Python
Executable file
115 lines
4.2 KiB
Python
Executable file
#! /usr/bin/env python
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import pygst
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pygst.require("0.10")
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import gst
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import gobject
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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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AUDIO_CAPS = 'application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA'
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AUDIO_DEPAY = 'rtppcmadepay'
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AUDIO_DEC = 'alawdec'
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AUDIO_SINK = 'autoaudiosink'
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DEST = '127.0.0.1'
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RTP_RECV_PORT = 5002
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RTCP_RECV_PORT = 5003
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RTCP_SEND_PORT = 5007
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#gst-launch -v gstrtpbin name=rtpbin \
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# udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0 \
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# rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
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# udpsrc port=$RTCP_RECV_PORT ! rtpbin.recv_rtcp_sink_0 \
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# rtpbin.send_rtcp_src_0 ! udpsink port=$RTCP_SEND_PORT host=$DEST sync=false async=false
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def pad_added_cb(rtpbin, new_pad, depay):
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sinkpad = gst.Element.get_static_pad(depay, 'sink')
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lres = gst.Pad.link(new_pad, sinkpad)
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# the pipeline to hold eveything
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pipeline = gst.Pipeline('rtp_client')
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# the udp src and source we will use for RTP and RTCP
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rtpsrc = gst.element_factory_make('udpsrc', 'rtpsrc')
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rtpsrc.set_property('port', RTP_RECV_PORT)
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# we need to set caps on the udpsrc for the RTP data
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caps = gst.caps_from_string(AUDIO_CAPS)
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rtpsrc.set_property('caps', caps)
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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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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)
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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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pipeline.add(rtpsrc, rtcpsrc, rtcpsink)
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# the depayloading and decoding
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audiodepay = gst.element_factory_make(AUDIO_DEPAY, 'audiodepay')
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audiodec = gst.element_factory_make(AUDIO_DEC, 'audiodec')
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# the audio playback and format conversion
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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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audiosink = gst.element_factory_make(AUDIO_SINK, 'audiosink')
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# add depayloading and playback to the pipeline and link
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pipeline.add(audiodepay, audiodec, audioconv, audiores, audiosink)
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res = gst.element_link_many(audiodepay, audiodec, audioconv, audiores, audiosink)
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# the rtpbin element
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rtpbin = gst.element_factory_make('gstrtpbin', 'rtpbin')
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pipeline.add(rtpbin)
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# now link all to the rtpbin, start by getting an RTP sinkpad for session 0
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srcpad = gst.Element.get_static_pad(rtpsrc, 'src')
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sinkpad = gst.Element.get_request_pad(rtpbin, 'recv_rtp_sink_0')
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lres = gst.Pad.link(srcpad, sinkpad)
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# get an RTCP sinkpad in session 0
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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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# get an RTCP srcpad for sending RTCP back to the sender
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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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rtpbin.connect('pad-added', pad_added_cb, audiodepay)
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gst.Element.set_state(pipeline, gst.STATE_PLAYING)
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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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