9 from typing import Optional, List
14 gi.require_version('Gst', '1.0')
15 from gi.repository import Gst
17 gi.require_version('GstWebRTC', '1.0')
18 from gi.repository import GstWebRTC
20 gi.require_version('GstSdp', '1.0')
21 from gi.repository import GstSdp
23 log = logging.getLogger(__name__)
28 self.sdp_offer: Optional[str] = None
29 self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
30 self.session_id = None
31 self.received_ice_candidates = queue.Queue()
32 self.generated_ice_candidates = queue.Queue()
33 self.user_fragments: Optional[List] = None
34 self.mids: Optional[List] = None
38 def on_negotiation_needed(self, element):
39 log.debug('on_negotiation_needed')
41 def on_ice_candidate(self, element, mlineindex, candidate):
42 log.debug('on_ice_candidate')
43 self.generated_ice_candidates.put_nowait((mlineindex, candidate))
45 def webrtcbin_pad_added(self, element, pad):
46 log.debug('webrtcbin_pad_added')
47 if pad.direction != Gst.PadDirection.SRC:
49 decodebin = Gst.ElementFactory.make('decodebin')
50 decodebin.connect('pad-added', self.decodebin_pad_added)
51 self.pipe.add(decodebin)
52 decodebin.sync_state_with_parent()
53 self.webrtcbin.link(decodebin)
55 def decodebin_pad_added(self, element, pad):
56 log.debug('decodebin_pad_added')
57 if not pad.has_current_caps():
58 log.debug(pad, 'has no caps, ignoring')
61 caps = pad.get_current_caps()
62 padsize = caps.get_size()
63 for i in range(padsize):
64 s = caps.get_structure(i) # Gst.Structure
66 if name.startswith('video'):
67 q = Gst.ElementFactory.make('queue')
68 conv = Gst.ElementFactory.make('videoconvert')
69 # sink = Gst.ElementFactory.make('autovideosink') # needs XDG_RUNTIME_DIR
70 sink = Gst.ElementFactory.make('xvimagesink')
74 self.pipe.sync_children_states()
75 pad.link(q.get_static_pad('sink'))
78 elif name.startswith('audio'):
79 q = Gst.ElementFactory.make('queue')
80 conv = Gst.ElementFactory.make('audioconvert')
81 resample = Gst.ElementFactory.make('audioresample')
82 sink = Gst.ElementFactory.make('autoaudiosink')
85 self.pipe.add(resample)
87 self.pipe.sync_children_states()
88 pad.link(q.get_static_pad('sink'))
93 async def listen_to_gstreamer_bus(self):
95 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
96 self.pipe = Gst.Pipeline.new("pipeline")
97 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
98 bus = Gst.Pipeline.get_bus(self.pipe)
99 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
100 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
101 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
102 self.pipe.set_state(Gst.State.PLAYING)
105 if bus.have_pending():
107 if msg.type == Gst.MessageType.ERROR:
108 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
110 elif msg.type == Gst.MessageType.EOS: # end of stream
111 log.info(f'Gstreamer message bus reports end of stream')
113 elif self.sdp_offer is not None:
114 res, sm = GstSdp.SDPMessage.new()
115 assert res == GstSdp.SDPResult.OK
116 GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
117 # the three lines above can also be done this way in new versions of GStreamer:
118 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
119 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
120 gst_promise = Gst.Promise.new()
121 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
122 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
123 self.sdp_offer = None
125 gst_promise = Gst.Promise.new()
126 self.webrtcbin.emit('create-answer', None, gst_promise)
127 result = await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
128 assert result == Gst.PromiseResult.REPLIED
129 reply = gst_promise.get_reply()
130 answer = reply.get_value('answer')
131 sdp_message = answer.sdp
132 self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
133 for i in range(sdp_message.medias_len())]
134 self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
135 for i in range(sdp_message.medias_len())]
136 sdp_answer = sdp_message.as_text()
137 log.info(f'Send SDP answer:\n{sdp_answer}')
138 sdp_answer_msg = json.dumps({
139 'SessionID': self.session_id,
141 'Value': json.dumps({
146 gst_promise = Gst.Promise.new()
147 self.webrtcbin.emit('set-local-description', answer, gst_promise)
148 await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
149 gst_promise.get_reply()
150 await self.websocket.send(sdp_answer_msg)
152 elif self.received_ice_candidates.qsize() > 0:
153 ic = self.received_ice_candidates.get_nowait()
154 if ic['candidate'] != '':
155 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
157 elif self.generated_ice_candidates.qsize() > 0:
158 mlineindex, candidate = self.generated_ice_candidates.get_nowait()
159 icemsg_value = json.dumps({
160 "candidate": candidate,
161 "sdpMid": self.mids[mlineindex],
162 "sdpMLineIndex": mlineindex,
163 "usernameFragment": self.user_fragments[mlineindex],
165 icemsg = json.dumps({
166 'SessionID': self.session_id,
167 'Type': 'addCalleeIceCandidate',
168 'Value': icemsg_value,
170 log.info(f'Send ICE candidate: {icemsg_value}')
171 await self.websocket.send(icemsg)
174 await asyncio.sleep(0.1)
176 self.pipe.set_state(Gst.State.NULL)
178 async def talk_to_websocket(self, uri):
179 async for msg in self.websocket:
180 msg_json = json.loads(msg)
181 msg_type = msg_json['Type']
182 msg_value = msg_json['Value']
183 assert self.session_id is None or self.session_id == msg_json['SessionID']
184 if msg_type == 'newSession':
185 self.session_id = msg_json['SessionID']
186 log.info(f"New session {self.session_id}")
187 elif msg_type == 'gotOffer':
188 value_json = json.loads(msg_value)
189 sdp = value_json['sdp']
190 log.info(f'Got SDP offer:\n{sdp}')
192 elif msg_type == 'addCallerIceCandidate':
193 value_json = json.loads(msg_value)
194 log.info(f'Got ICE candidate: {value_json}')
195 self.received_ice_candidates.put_nowait(value_json)
196 elif msg_type == 'roomNotFound':
197 log.error(f'The room was not found: {uri}')
199 elif msg_type == 'roomClosed':
200 log.info(f'Oh noes, the room went away (session {self.session_id})!')
201 self.session_id = None
204 log.error(f'Unknown message type {msg_type}')
206 async def run(self, uri):
207 ssl_context = ssl.SSLContext()
208 ssl_context.check_hostname = False
209 ssl_context.verify_mode = ssl.CERT_NONE
211 async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
212 talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
213 listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
214 done, pending = await asyncio.wait(
215 [talk_to_websocket_task, listen_to_gstreamer_bus_task],
216 return_when=asyncio.FIRST_COMPLETED)
226 logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
227 parser = argparse.ArgumentParser()
228 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
229 help='Signalling server URI')
230 args = parser.parse_args()
232 asyncio.run(lagarde.run(args.uri), debug=True)
235 if __name__ == '__main__':