mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-08 18:39:54 +00:00
09ca5fa910
This was done in 0.10 to avoid conflict with the rtp elements in farsight, but the gst-prefixing is no longer needed in 0.11
90 lines
3.1 KiB
Python
Executable file
90 lines
3.1 KiB
Python
Executable file
#! /usr/bin/env python
|
|
|
|
import gobject, pygst
|
|
pygst.require("0.10")
|
|
import 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
|
|
|
|
# the pipeline to hold everything
|
|
pipeline = gst.Pipeline('rtp_server')
|
|
|
|
# the pipeline to hold everything
|
|
audiosrc = gst.element_factory_make(AUDIO_SRC, 'audiosrc')
|
|
audioconv = gst.element_factory_make('audioconvert', 'audioconv')
|
|
audiores = gst.element_factory_make('audioresample', 'audiores')
|
|
|
|
# the pipeline to hold everything
|
|
audioenc = gst.element_factory_make(AUDIO_ENC, 'audioenc')
|
|
audiopay = gst.element_factory_make(AUDIO_PAY, 'audiopay')
|
|
|
|
# add capture and payloading to the pipeline and link
|
|
pipeline.add(audiosrc, audioconv, audiores, audioenc, audiopay)
|
|
|
|
res = gst.element_link_many(audiosrc, audioconv, audiores, audioenc, audiopay)
|
|
|
|
# the rtpbin element
|
|
rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
|
|
|
|
pipeline.add(rtpbin)
|
|
|
|
# the udp sinks and source we will use for RTP and RTCP
|
|
rtpsink = gst.element_factory_make('udpsink', 'rtpsink')
|
|
rtpsink.set_property('port', RTP_SEND_PORT)
|
|
rtpsink.set_property('host', DEST_HOST)
|
|
|
|
rtcpsink = gst.element_factory_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.element_factory_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)
|