| #! /usr/bin/env python | |
| import gi | |
| import sys | |
| gi.require_version('Gst', '1.0') | |
| from gi.repository import GObject, Gst | |
| #gst-launch -v rtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \ | |
| # rtpbin.send_rtp_src_0 ! udpsink port=10000 host=xxx.xxx.xxx.xxx \ | |
| # rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=xxx.xxx.xxx.xxx sync=false async=false \ | |
| # udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0 | |
| DEST_HOST = '127.0.0.1' | |
| AUDIO_SRC = 'audiotestsrc' | |
| AUDIO_ENC = 'alawenc' | |
| AUDIO_PAY = 'rtppcmapay' | |
| RTP_SEND_PORT = 5002 | |
| RTCP_SEND_PORT = 5003 | |
| RTCP_RECV_PORT = 5007 | |
| GObject.threads_init() | |
| Gst.init(sys.argv) | |
| # the pipeline to hold everything | |
| pipeline = Gst.Pipeline('rtp_server') | |
| # the pipeline to hold everything | |
| audiosrc = Gst.ElementFactory.make(AUDIO_SRC, 'audiosrc') | |
| audioconv = Gst.ElementFactory.make('audioconvert', 'audioconv') | |
| audiores = Gst.ElementFactory.make('audioresample', 'audiores') | |
| # the pipeline to hold everything | |
| audioenc = Gst.ElementFactory.make(AUDIO_ENC, 'audioenc') | |
| audiopay = Gst.ElementFactory.make(AUDIO_PAY, 'audiopay') | |
| # add capture and payloading to the pipeline and link | |
| pipeline.add(audiosrc, audioconv, audiores, audioenc, audiopay) | |
| audiosrc.link(audioconv) | |
| audioconv.link(audiores) | |
| audiores.link(audioenc) | |
| audioenc.link(audiopay) | |
| # the rtpbin element | |
| rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin') | |
| pipeline.add(rtpbin) | |
| # the udp sinks and source we will use for RTP and RTCP | |
| rtpsink = Gst.ElementFactory.make('udpsink', 'rtpsink') | |
| rtpsink.set_property('port', RTP_SEND_PORT) | |
| rtpsink.set_property('host', DEST_HOST) | |
| rtcpsink = Gst.ElementFactory.make('udpsink', 'rtcpsink') | |
| rtcpsink.set_property('port', RTCP_SEND_PORT) | |
| rtcpsink.set_property('host', DEST_HOST) | |
| # no need for synchronisation or preroll on the RTCP sink | |
| rtcpsink.set_property('async', False) | |
| rtcpsink.set_property('sync', False) | |
| rtcpsrc = Gst.ElementFactory.make('udpsrc', 'rtcpsrc') | |
| rtcpsrc.set_property('port', RTCP_RECV_PORT) | |
| pipeline.add(rtpsink, rtcpsink, rtcpsrc) | |
| # now link all to the rtpbin, start by getting an RTP sinkpad for session 0 | |
| sinkpad = Gst.Element.get_request_pad(rtpbin, 'send_rtp_sink_0') | |
| srcpad = Gst.Element.get_static_pad(audiopay, 'src') | |
| lres = Gst.Pad.link(srcpad, sinkpad) | |
| # get the RTP srcpad that was created when we requested the sinkpad above and | |
| # link it to the rtpsink sinkpad | |
| srcpad = Gst.Element.get_static_pad(rtpbin, 'send_rtp_src_0') | |
| sinkpad = Gst.Element.get_static_pad(rtpsink, 'sink') | |
| lres = Gst.Pad.link(srcpad, sinkpad) | |
| # get an RTCP srcpad for sending RTCP to the receiver | |
| 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) | |
| # we also want to receive RTCP, request an RTCP sinkpad for session 0 and | |
| # link it to the srcpad of the udpsrc for RTCP | |
| 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) | |
| # set the pipeline to playing | |
| Gst.Element.set_state(pipeline, Gst.State.PLAYING) | |
| # we need to run a GLib main loop to get the messages | |
| mainloop = GObject.MainLoop() | |
| mainloop.run() | |
| Gst.Element.set_state(pipeline, Gst.State.NULL) |