diff options
Diffstat (limited to 'tests/examples/rtp/client-PCMA.py')
| -rwxr-xr-x | tests/examples/rtp/client-PCMA.py | 87 |
1 files changed, 46 insertions, 41 deletions
diff --git a/tests/examples/rtp/client-PCMA.py b/tests/examples/rtp/client-PCMA.py index 5830b1081..9d2fd4291 100755 --- a/tests/examples/rtp/client-PCMA.py +++ b/tests/examples/rtp/client-PCMA.py @@ -1,12 +1,12 @@ -#! /usr/bin/env python +#! /usr/bin/env python -import pygst -pygst.require("0.10") -import gst -import gobject +import gi +import sys +gi.require_version('Gst', '1.0') +from gi.repository import GObject, Gst # -# A simple RTP receiver +# 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 @@ -15,14 +15,14 @@ import gobject # 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 | -# '-------' '----------' +# port=5003 | src->recv_rtcp | +# '-------' '----------' AUDIO_CAPS = 'application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA' AUDIO_DEPAY = 'rtppcmadepay' @@ -33,7 +33,10 @@ DEST = '127.0.0.1' RTP_RECV_PORT = 5002 RTCP_RECV_PORT = 5003 -RTCP_SEND_PORT = 5007 +RTCP_SEND_PORT = 5007 + +GObject.threads_init() +Gst.init(sys.argv) #gst-launch -v rtpbin name=rtpbin \ # udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0 \ @@ -42,74 +45,76 @@ RTCP_SEND_PORT = 5007 # 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) + 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 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 = Gst.ElementFactory.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) +caps = Gst.caps_from_string(AUDIO_CAPS) rtpsrc.set_property('caps', caps) -rtcpsrc = gst.element_factory_make('udpsrc', 'rtcpsrc') +rtcpsrc = Gst.ElementFactory.make('udpsrc', 'rtcpsrc') rtcpsrc.set_property('port', RTCP_RECV_PORT) -rtcpsink = gst.element_factory_make('udpsink', 'rtcpsink') +rtcpsink = Gst.ElementFactory.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) +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') +audiodepay = Gst.ElementFactory.make(AUDIO_DEPAY, 'audiodepay') +audiodec = Gst.ElementFactory.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') +audioconv = Gst.ElementFactory.make('audioconvert', 'audioconv') +audiores = Gst.ElementFactory.make('audioresample', 'audiores') +audiosink = Gst.ElementFactory.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) +audiodepay.link(audiodec) +audiodec.link(audioconv) +audioconv.link(audiores) +audiores.link(audiosink) # the rtpbin element -rtpbin = gst.element_factory_make('rtpbin', 'rtpbin') +rtpbin = Gst.ElementFactory.make('rtpbin', '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) +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) +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) +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) -gst.Element.set_state(pipeline, gst.STATE_PLAYING) +rtpbin.connect('pad-added', pad_added_cb, audiodepay) -mainloop = gobject.MainLoop() -mainloop.run() +Gst.Element.set_state(pipeline, Gst.State.PLAYING) -gst.Element.set_state(pipeline, gst.STATE_NULL) +mainloop = GObject.MainLoop() +mainloop.run() +Gst.Element.set_state(pipeline, Gst.State.NULL) |