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#! /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)
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