From b50ce27b14cee327c942c7bf54c335aac81c75cf Mon Sep 17 00:00:00 2001
From: Wim Taymans <wim.taymans@collabora.co.uk>
Date: Mon, 4 Oct 2010 17:52:22 +0200
Subject: [PATCH] tests: Add python RTP client example

Add a python version of the PCMA client app.

Ported by Sp4rc on IRC.
---
 tests/examples/rtp/client-PCMA.py | 115 ++++++++++++++++++++++++++++++
 1 file changed, 115 insertions(+)
 create mode 100755 tests/examples/rtp/client-PCMA.py

diff --git a/tests/examples/rtp/client-PCMA.py b/tests/examples/rtp/client-PCMA.py
new file mode 100755
index 0000000000..f6230f969f
--- /dev/null
+++ b/tests/examples/rtp/client-PCMA.py
@@ -0,0 +1,115 @@
+#! /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) 
+
+