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examples: port python rtp PCMA client/server tests to 1.0
https://bugzilla.gnome.org/show_bug.cgi?id=739930
This commit is contained in:
Henning Heinold
Tim-Philipp Müller
parent
0b36fe59e1
commit
8aa2630068
2 changed files with 90 additions and 77 deletions
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@ -1,12 +1,12 @@
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#! /usr/bin/env python
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#! /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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import gi
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import sys
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gi.require_version('Gst', '1.0')
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from gi.repository import GObject, Gst
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#
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# A simple RTP receiver
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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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@ -15,14 +15,14 @@ import gobject
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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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# | | .-------.
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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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# 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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@ -33,7 +33,10 @@ 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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RTCP_SEND_PORT = 5007
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GObject.threads_init()
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Gst.init(sys.argv)
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#gst-launch -v rtpbin name=rtpbin \
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# udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0 \
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@ -42,74 +45,76 @@ RTCP_SEND_PORT = 5007
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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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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 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 = Gst.ElementFactory.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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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 = Gst.ElementFactory.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 = Gst.ElementFactory.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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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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audiodepay = Gst.ElementFactory.make(AUDIO_DEPAY, 'audiodepay')
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audiodec = Gst.ElementFactory.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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audioconv = Gst.ElementFactory.make('audioconvert', 'audioconv')
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audiores = Gst.ElementFactory.make('audioresample', 'audiores')
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audiosink = Gst.ElementFactory.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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audiodepay.link(audiodec)
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audiodec.link(audioconv)
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audioconv.link(audiores)
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audiores.link(audiosink)
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# the rtpbin element
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rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
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rtpbin = Gst.ElementFactory.make('rtpbin', '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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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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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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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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rtpbin.connect('pad-added', pad_added_cb, audiodepay)
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gst.Element.set_state(pipeline, gst.STATE_PLAYING)
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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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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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@ -1,13 +1,15 @@
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#! /usr/bin/env python
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#! /usr/bin/env python
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import gi
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import sys
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gi.require_version('Gst', '1.0')
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from gi.repository import GObject, Gst
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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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# udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0
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DEST_HOST = '127.0.0.1'
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@ -17,74 +19,80 @@ 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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RTCP_RECV_PORT = 5007
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GObject.threads_init()
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Gst.init(sys.argv)
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# the pipeline to hold everything
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pipeline = gst.Pipeline('rtp_server')
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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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audiosrc = Gst.ElementFactory.make(AUDIO_SRC, 'audiosrc')
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audioconv = Gst.ElementFactory.make('audioconvert', 'audioconv')
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audiores = Gst.ElementFactory.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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audioenc = Gst.ElementFactory.make(AUDIO_ENC, 'audioenc')
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audiopay = Gst.ElementFactory.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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audiosrc.link(audioconv)
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audioconv.link(audiores)
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audiores.link(audioenc)
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audioenc.link(audiopay)
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# the rtpbin element
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rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
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rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin')
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pipeline.add(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 = Gst.ElementFactory.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 = Gst.ElementFactory.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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rtcpsink.set_property('sync', False)
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rtcpsrc = gst.element_factory_make('udpsrc', 'rtcpsrc')
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rtcpsrc = Gst.ElementFactory.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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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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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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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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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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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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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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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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Gst.Element.set_state(pipeline, Gst.State.NULL)
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