11 from typing import Optional, List
17 gi.require_version('Gst', '1.0')
18 from gi.repository import Gst
20 gi.require_version('GstWebRTC', '1.0')
21 from gi.repository import GstWebRTC
23 gi.require_version('GstSdp', '1.0')
24 from gi.repository import GstSdp
26 log = logging.getLogger(__name__)
31 self.sdp_offer: Optional[str] = None
32 self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
33 self.session_id = None
34 self.received_ice_candidates = []
35 self.generated_ice_candidates = []
36 self.user_fragments: Optional[List] = None
37 self.mids: Optional[List] = None
41 def on_negotiation_needed(self, element):
42 log.info('on_negotiation_needed')
44 def on_ice_candidate(self, element, mlineindex, candidate):
45 log.info('on_ice_candidate')
46 self.generated_ice_candidates.append((mlineindex, candidate))
48 def webrtcbin_pad_added(self, element, pad):
49 log.info('webrtcbin_pad_added')
50 if pad.direction != Gst.PadDirection.SRC:
52 decodebin = Gst.ElementFactory.make('decodebin')
53 decodebin.connect('pad-added', self.decodebin_pad_added)
54 self.pipe.add(decodebin)
55 decodebin.sync_state_with_parent()
56 self.webrtcbin.link(decodebin)
58 def decodebin_pad_added(self, element, pad):
59 log.info('decodebin_pad_added')
60 if not pad.has_current_caps():
61 log.info(pad, 'has no caps, ignoring')
64 caps = pad.get_current_caps()
68 if name.startswith('video'):
69 q = Gst.ElementFactory.make('queue')
70 conv = Gst.ElementFactory.make('videoconvert')
71 sink = Gst.ElementFactory.make('autovideosink')
72 self.pipe.add(q, conv, sink)
73 self.pipe.sync_children_states()
74 pad.link(q.get_static_pad('sink'))
77 elif name.startswith('audio'):
78 q = Gst.ElementFactory.make('queue')
79 conv = Gst.ElementFactory.make('audioconvert')
80 resample = Gst.ElementFactory.make('audioresample')
81 sink = Gst.ElementFactory.make('autoaudiosink')
82 self.pipe.add(q, conv, resample, sink)
83 self.pipe.sync_children_states()
84 pad.link(q.get_static_pad('sink'))
89 async def listen_to_gstreamer_bus(self):
91 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
92 self.pipe = Gst.Pipeline.new("pipeline")
93 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
94 bus = Gst.Pipeline.get_bus(self.pipe)
95 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
96 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
97 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
98 self.pipe.set_state(Gst.State.PLAYING)
101 if bus.have_pending():
102 msg = bus.pop() # Gst.Message, has to be unref'ed.
103 if msg.type != Gst.MessageType.STATE_CHANGED:
104 # log.info(f'Receive Gst.Message: {msg.type}, {msg.seqnum}, {msg.get_structure()}')
105 # log.info(f'{webrtcbin.props.signaling_state} {webrtcbin.props.ice_gathering_state} {webrtcbin.props.ice_connection_state}')
106 # Gst.Message.unref(msg)
108 elif self.sdp_offer is not None:
109 res, sm = GstSdp.SDPMessage.new()
110 assert res == GstSdp.SDPResult.OK
111 GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
112 # the three lines above can also be done this way in new versions of GStreamer:
113 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
114 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
115 gst_promise = Gst.Promise.new()
116 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
118 self.sdp_offer = None
120 log.info('create-answer')
121 gst_promise = Gst.Promise.new()
122 self.webrtcbin.emit('create-answer', None, gst_promise)
123 result = gst_promise.wait()
124 assert result == Gst.PromiseResult.REPLIED
125 reply = gst_promise.get_reply()
126 answer = reply.get_value('answer')
127 sdp_message = answer.sdp
128 self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
129 for i in range(sdp_message.medias_len())]
130 self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
131 for i in range(sdp_message.medias_len())]
132 sdp_answer = sdp_message.as_text()
134 sdp_answer_msg = json.dumps({
135 'SessionID': self.session_id,
137 'Value': json.dumps({
142 gst_promise = Gst.Promise.new()
143 self.webrtcbin.emit('set-local-description', answer, gst_promise)
145 gst_promise.get_reply()
146 await self.websocket.send(sdp_answer_msg)
148 elif len(self.received_ice_candidates) > 0:
149 ic = self.received_ice_candidates.pop(0)
150 if ic['candidate'] != '':
151 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
153 elif len(self.generated_ice_candidates) > 0:
154 mlineindex, candidate = self.generated_ice_candidates.pop(0)
155 icemsg = json.dumps({
156 'SessionID': self.session_id,
157 'Type': 'addCalleeIceCandidate',
158 'Value': json.dumps({
159 "candidate": candidate,
160 "sdpMid": self.mids[mlineindex],
161 "sdpMLineIndex": mlineindex,
162 "usernameFragment": self.user_fragments[mlineindex],
165 log.info(f'send_ice_candidate_message with {icemsg}')
166 await self.websocket.send(icemsg)
169 await asyncio.sleep(0.1)
171 self.pipe.set_state(Gst.State.NULL)
173 async def talk_to_websocket(self, uri):
174 ssl_context = ssl.SSLContext()
175 ssl_context.check_hostname = False
176 ssl_context.verify_mode = ssl.CERT_NONE
177 async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
178 async for msg in self.websocket:
179 msg_json = json.loads(msg)
180 msg_type = msg_json['Type']
181 msg_value = msg_json['Value']
182 self.session_id = msg_json['SessionID']
183 log.info(f"receive for session {self.session_id} type {msg_type}")
184 if msg_type == 'newSession':
186 elif msg_type == 'gotOffer':
187 value_json = json.loads(msg_value)
188 sdp = value_json['sdp']
189 log.info(f'SDP: {sdp}')
191 elif msg_type == 'addCallerIceCandidate':
192 value_json = json.loads(msg_value)
193 log.info(f'ICE: {value_json}')
194 self.received_ice_candidates.append(value_json)
195 elif msg_type == 'roomClosed':
196 log.info('Oh noes, the room went away!')
197 # and here we should clean up
199 log.error(f'Unknown message type {msg_type}')
201 async def run(self, uri):
202 talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
203 listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
204 done, pending = await asyncio.wait(
205 [talk_to_websocket_task, listen_to_gstreamer_bus_task],
206 return_when=asyncio.FIRST_COMPLETED)
214 logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
215 parser = argparse.ArgumentParser()
216 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
217 help='Signalling server URI')
218 args = parser.parse_args()
220 asyncio.run(lagarde.run(args.uri), debug=True)
223 if __name__ == '__main__':