#! /usr/bin/env python import pygst pygst.require("0.10") import gst import gobject # # A simple RTP receiver # # receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003. # the receiver RTCP reports are sent to port 5007 # # .-------. .----------. .---------. .-------. .--------. # RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink| # port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink | # '-------' | | '---------' '-------' '--------' # | | # | | .-------. # | | |udpsink| RTCP # | send_rtcp->sink | port=5007 # .-------. | | '-------' sync=false # RTCP |udpsrc | | | async=false # port=5003 | src->recv_rtcp | # '-------' '----------' AUDIO_CAPS = 'application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA' AUDIO_DEPAY = 'rtppcmadepay' AUDIO_DEC = 'alawdec' AUDIO_SINK = 'autoaudiosink' DEST = '127.0.0.1' RTP_RECV_PORT = 5002 RTCP_RECV_PORT = 5003 RTCP_SEND_PORT = 5007 #gst-launch -v gstrtpbin name=rtpbin \ # udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0 \ # rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \ # udpsrc port=$RTCP_RECV_PORT ! rtpbin.recv_rtcp_sink_0 \ # rtpbin.send_rtcp_src_0 ! udpsink port=$RTCP_SEND_PORT host=$DEST sync=false async=false def pad_added_cb(rtpbin, new_pad, depay): sinkpad = gst.Element.get_static_pad(depay, 'sink') lres = gst.Pad.link(new_pad, sinkpad) # the pipeline to hold eveything pipeline = gst.Pipeline('rtp_client') # the udp src and source we will use for RTP and RTCP rtpsrc = gst.element_factory_make('udpsrc', 'rtpsrc') rtpsrc.set_property('port', RTP_RECV_PORT) # we need to set caps on the udpsrc for the RTP data caps = gst.caps_from_string(AUDIO_CAPS) rtpsrc.set_property('caps', caps) rtcpsrc = gst.element_factory_make('udpsrc', 'rtcpsrc') rtcpsrc.set_property('port', RTCP_RECV_PORT) rtcpsink = gst.element_factory_make('udpsink', 'rtcpsink') rtcpsink.set_property('port', RTCP_SEND_PORT) rtcpsink.set_property('host', DEST) # no need for synchronisation or preroll on the RTCP sink rtcpsink.set_property('async', False) rtcpsink.set_property('sync', False) pipeline.add(rtpsrc, rtcpsrc, rtcpsink) # the depayloading and decoding audiodepay = gst.element_factory_make(AUDIO_DEPAY, 'audiodepay') audiodec = gst.element_factory_make(AUDIO_DEC, 'audiodec') # the audio playback and format conversion audioconv = gst.element_factory_make('audioconvert', 'audioconv') audiores = gst.element_factory_make('audioresample', 'audiores') audiosink = gst.element_factory_make(AUDIO_SINK, 'audiosink') # add depayloading and playback to the pipeline and link pipeline.add(audiodepay, audiodec, audioconv, audiores, audiosink) res = gst.element_link_many(audiodepay, audiodec, audioconv, audiores, audiosink) # the rtpbin element rtpbin = gst.element_factory_make('gstrtpbin', 'rtpbin') pipeline.add(rtpbin) # now link all to the rtpbin, start by getting an RTP sinkpad for session 0 srcpad = gst.Element.get_static_pad(rtpsrc, 'src') sinkpad = gst.Element.get_request_pad(rtpbin, 'recv_rtp_sink_0') lres = gst.Pad.link(srcpad, sinkpad) # get an RTCP sinkpad in session 0 srcpad = gst.Element.get_static_pad(rtcpsrc, 'src') sinkpad = gst.Element.get_request_pad(rtpbin, 'recv_rtcp_sink_0') lres = gst.Pad.link(srcpad, sinkpad) # get an RTCP srcpad for sending RTCP back to the sender srcpad = gst.Element.get_request_pad(rtpbin, 'send_rtcp_src_0') sinkpad = gst.Element.get_static_pad(rtcpsink, 'sink') lres = gst.Pad.link(srcpad, sinkpad) rtpbin.connect('pad-added', pad_added_cb, audiodepay) gst.Element.set_state(pipeline, gst.STATE_PLAYING) mainloop = gobject.MainLoop() mainloop.run() gst.Element.set_state(pipeline, gst.STATE_NULL)