def _create_webrtc_element_for_new_connection(self, ws):
'''
We make a new webrtc element, and queue to feed into it, for every new peer (client).
That way, multiple clients can connect at once.
'''
self.peers[ws]['webrtcbin'] = Gst.ElementFactory.make('webrtcbin')
self.pipeline.add(self.peers[ws]['webrtcbin'])
self.peers[ws]['webrtcbin'].add_property_notify_watch(None, True)
if config.stun_server():
self.peers[ws]['webrtcbin'].set_property('stun-server', 'stun://' + config.stun_server())
if config.turn_server():
self.peers[ws]['webrtcbin'].set_property('turn-server', 'turn://' + config.turn_server())
if config.enable_video():
self.peers[ws]['video_queue'] = Gst.ElementFactory.make('queue')
self.pipeline.add(self.peers[ws]['video_queue'])
self.webrtc_video_tee.link(self.peers[ws]['video_queue'])
self.peers[ws]['video_queue'].link(self.peers[ws]['webrtcbin'])
if config.enable_audio():
self.peers[ws]['audio_queue'] = Gst.ElementFactory.make('queue')
self.pipeline.add(self.peers[ws]['audio_queue'])
self.webrtc_audio_tee.link(self.peers[ws]['audio_queue'])
self.peers[ws]['audio_queue'].link(self.peers[ws]['webrtcbin'])
def _create_webrtc_element_for_new_connection(self, ws):
'''
We make a new webrtc element, and queue to feed into it, for every new peer (client).
That way, multiple clients can connect at once.
'''
self.peers[ws]['webrtcbin'] = Gst.ElementFactory.make('webrtcbin')
self.pipeline.add(self.peers[ws]['webrtcbin'])
self.peers[ws]['webrtcbin'].set_property('bundle-policy', 'max-bundle')
self.peers[ws]['webrtcbin'].add_property_notify_watch(None, True)
self.peers[ws]['webrtcbin'].set_state(Gst.State.READY)
if config.stun_server():
self.peers[ws]['webrtcbin'].set_property('stun-server', 'stun://' + config.stun_server())
if config.turn_server():
self.peers[ws]['webrtcbin'].set_property('turn-server', 'turn://' + config.turn_server())
if config.enable_video():
self.peers[ws]['video_queue'] = Gst.ElementFactory.make('queue')
self.peers[ws]['video_queue'].set_property('leaky', 'upstream')
self.pipeline.add(self.peers[ws]['video_queue'])
self.webrtc_video_tee.link(self.peers[ws]['video_queue'])
self.peers[ws]['video_queue'].link(self.peers[ws]['webrtcbin'])
self.peers[ws]['video_queue'].set_state(Gst.State.READY)
if config.enable_audio():
self.peers[ws]['audio_queue'] = Gst.ElementFactory.make('queue')
self.peers[ws]['audio_queue'].set_property('leaky', 'upstream')
self.pipeline.add(self.peers[ws]['audio_queue'])
self.webrtc_audio_tee.link(self.peers[ws]['audio_queue'])
self.peers[ws]['audio_queue'].link(self.peers[ws]['webrtcbin'])
self.peers[ws]['audio_queue'].set_state(Gst.State.READY)
# We set the three elements above to READY first.
# We now move them to PLAYING
# This appears to prevent a race-condition that can
# intermittently cause _on_negotiation_needed to not be called.
self.peers[ws]['webrtcbin'].set_state(Gst.State.PLAYING)
if config.enable_video():
self.peers[ws]['video_queue'].set_state(Gst.State.PLAYING)
if config.enable_audio():
self.peers[ws]['audio_queue'].set_state(Gst.State.PLAYING)