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Now RTSP is used without re-encoding video (however, some assumptions are made).
[toast/stream2beamer.git] / lagarde.py
1 #!/usr/bin/python3
2 import argparse
3 import asyncio
4 import json
5 import logging
6 import ssl
7 import queue
8
9 import gi
10 import websockets
11
12 gi.require_version('Gst', '1.0')
13 from gi.repository import Gst
14
15 gi.require_version('GstWebRTC', '1.0')
16 from gi.repository import GstWebRTC
17
18 gi.require_version('GstSdp', '1.0')
19 from gi.repository import GstSdp
20
21 log = logging.getLogger(__name__)
22
23
24 class Events:
25     def __init__(self):
26         self.sdp_offer = queue.Queue()
27         self.sdp_answer = queue.Queue()
28         self.generated_ice_candidates = queue.Queue()
29         self.received_ice_candidates = queue.Queue()
30         self.sdp_info = queue.Queue()  # (sdp_mids, user_fragments)
31         self.room_left = queue.Queue()
32
33
34 class SignalingClient:
35     def __init__(self, events: Events, uri):
36         self.events = events
37         self.uri = uri
38         self.ssl_context = ssl.SSLContext()
39         self.ssl_context.check_hostname = False
40         self.ssl_context.verify_mode = ssl.CERT_NONE
41         self.session_id = None
42
43     async def receive(self, uri):
44         async for msg in self.websocket:
45             msg_json = json.loads(msg)
46             msg_type = msg_json['Type']
47             msg_value = msg_json['Value']
48             assert self.session_id is None or self.session_id == msg_json['SessionID']
49             if msg_type == 'newSession':
50                 self.session_id = msg_json['SessionID']
51                 log.info(f"New session {self.session_id}")
52             elif msg_type == 'gotOffer':
53                 value_json = json.loads(msg_value)
54                 sdp = value_json['sdp']
55                 log.info(f'Got SDP offer')
56                 log.debug(f'SDP offer:\n{sdp}')
57                 self.events.sdp_offer.put_nowait(sdp)
58             elif msg_type == 'addCallerIceCandidate':
59                 value_json = json.loads(msg_value)
60                 log.info(f'Got ICE candidate')
61                 log.debug(f'ICE candidate: {value_json}')
62                 self.events.received_ice_candidates.put_nowait(value_json)
63             elif msg_type == 'roomNotFound':
64                 log.error(f'The room was not found: {uri}')
65                 return
66             elif msg_type == 'roomClosed':
67                 log.info(f'Oh noes, the room went away (session {self.session_id})!')
68                 self.events.room_left.put_nowait(True)
69                 return
70             else:
71                 log.error(f'Unknown message type {msg_type}')
72
73     async def send(self):
74         sdp_mids = None
75         user_fragments = None
76         while True:
77             if self.events.sdp_answer.qsize() > 0:
78                 sdp_answer = self.events.sdp_answer.get_nowait()
79                 sdp_answer_msg = json.dumps({
80                     'SessionID': self.session_id,
81                     'Type': "gotAnswer",
82                     'Value': json.dumps({
83                         'type': 'answer',
84                         'sdp': sdp_answer
85                     })
86                 })
87                 await self.websocket.send(sdp_answer_msg)
88
89             elif self.events.sdp_info.qsize() > 0:
90                 sdp_mids, user_fragments = self.events.sdp_info.get_nowait()
91
92             elif self.events.generated_ice_candidates.qsize() > 0 \
93                     and sdp_mids is not None and user_fragments is not None:
94                 mlineindex, candidate = self.events.generated_ice_candidates.get_nowait()
95                 sdp_mid = sdp_mids[mlineindex]
96                 user_fragment = user_fragments[mlineindex]
97                 icemsg_value = json.dumps({
98                     "candidate": candidate,
99                     "sdpMid": sdp_mid,
100                     "sdpMLineIndex": mlineindex,
101                     "usernameFragment": user_fragment,
102                 })
103                 icemsg = json.dumps({
104                     'SessionID': self.session_id,
105                     'Type': 'addCalleeIceCandidate',
106                     'Value': icemsg_value,
107                 })
108                 log.info(f'Send ICE candidate')
109                 log.debug(f'ICE candidate: {icemsg_value}')
110                 await self.websocket.send(icemsg)
111
112             else:
113                 await asyncio.sleep(0.2)
114
115     async def run(self):
116         self.session_id = None
117         async with websockets.connect(self.uri, ssl=self.ssl_context, close_timeout=0.5) as self.websocket:
118             receive_task = asyncio.Task(self.receive(self.uri))
119             send_task = asyncio.Task(self.send())
120             done, pending = await asyncio.wait([receive_task, send_task], return_when=asyncio.FIRST_COMPLETED)
121             for task in pending:
122                 task.cancel()
123
124
125 class WebRTCClient:
126     def __init__(self, events: Events, rtmp_uri: str):
127         self.events = events
128         self.rtmp_uri = rtmp_uri
129         self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
130         self.pipe = Gst.Pipeline.new("pipeline")
131         Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
132         self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
133         self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
134         self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
135
136     def on_negotiation_needed(self, element):
137         log.info('on_negotiation_needed')
138
139     def on_ice_candidate(self, element, mlineindex, candidate):
140         log.info('on_ice_candidate')
141         self.events.generated_ice_candidates.put_nowait((mlineindex, candidate))
142
143     def webrtcbin_pad_added(self, element, pad):
144         log.info('webrtcbin_pad_added')
145         if pad.direction != Gst.PadDirection.SRC:
146             return
147         demux = Gst.ElementFactory.make('rtpptdemux')
148         assert demux
149         self.pipe.add(demux)
150         demux.connect('pad-added', self.demux_pad_added)
151         demux.sync_state_with_parent()
152         ok = self.webrtcbin.link(demux)
153         assert ok
154
155     def demux_pad_added(self, element, pad):
156         log.info('demux_pad_added')
157         if pad.direction != Gst.PadDirection.SRC:
158             return
159         depay = Gst.ElementFactory.make('rtpvp8depay')
160         assert depay
161         self.pipe.add(depay)
162         ok = element.link(depay)
163         assert ok
164
165         q = Gst.ElementFactory.make('queue')
166         sink = Gst.ElementFactory.make('rtspclientsink')
167         sink.set_property('location', self.rtmp_uri)
168         sink.set_property('debug', True)
169         assert q and sink
170
171         self.pipe.add(q)
172         self.pipe.add(sink)
173         ok = depay.link(q)
174         assert ok
175         ok = q.link(sink)
176         assert ok
177
178         self.pipe.sync_children_states()
179         self.pipe.set_state(Gst.State.PLAYING)
180         self.pipe.sync_children_states()
181
182     def set_remote_desciption_done(self, gst_promise):
183         gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done)
184         self.webrtcbin.emit('create-answer', None, gst_promise)
185
186     def create_answer_done(self, gst_promise):
187         reply = gst_promise.get_reply()
188         answer = reply.get_value('answer')
189         sdp_message = answer.sdp
190         mids = [sdp_message.get_media(i).get_attribute_val('mid')
191                 for i in range(sdp_message.medias_len())]
192         user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
193                           for i in range(sdp_message.medias_len())]
194         self.events.sdp_info.put_nowait((mids, user_fragments))
195         sdp_answer = sdp_message.as_text()
196         log.info(f'Send SDP answer')
197         log.debug(f'SDP answer:\n{sdp_answer}')
198         self.events.sdp_answer.put_nowait(sdp_answer)
199         gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
200         self.webrtcbin.emit('set-local-description', answer, gst_promise)
201
202     def set_local_description_done(self, gst_promise):
203         gst_promise.get_reply()
204
205     async def run(self):
206         bus = Gst.Pipeline.get_bus(self.pipe)
207         self.pipe.set_state(Gst.State.PLAYING)
208         try:
209             while True:
210                 if bus.have_pending():
211                     msg = bus.pop()
212                     if msg.type == Gst.MessageType.ERROR:
213                         log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
214                         return
215                     elif msg.type == Gst.MessageType.EOS:  # end of stream
216                         log.info(f'Gstreamer message bus reports end of stream')
217                         return
218                 elif self.events.sdp_offer.qsize() > 0:
219                     sdp_offer = self.events.sdp_offer.get_nowait()
220                     res, sm = GstSdp.SDPMessage.new()
221                     assert res == GstSdp.SDPResult.OK
222                     GstSdp.sdp_message_parse_buffer(bytes(sdp_offer.encode()), sm)
223                     # the three lines above can also be done this way in new versions of GStreamer:
224                     # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
225                     rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
226                     gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done)
227                     self.webrtcbin.emit('set-remote-description', rd, gst_promise)
228
229                 elif self.events.received_ice_candidates.qsize() > 0:
230                     ic = self.events.received_ice_candidates.get_nowait()
231                     if ic['candidate'] != '':
232                         self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
233
234                 elif self.events.room_left.qsize() > 0:
235                     self.events.room_left.get_nowait()
236                     return
237
238                 else:
239                     await asyncio.sleep(0.1)
240         finally:
241             self.pipe.set_state(Gst.State.NULL)
242
243
244 async def run_repeated(task):
245     while True:
246         await task()
247         await asyncio.sleep(0.1)
248
249
250 async def run(laplace_uri: str, rtmp_uri: str):
251     try:
252         events = Events()
253         webrtc = WebRTCClient(events, rtmp_uri)
254         signaling = SignalingClient(events, laplace_uri)
255
256         webrtc_task = asyncio.Task(webrtc.run())
257         signaling_task = asyncio.Task(signaling.run())
258
259         done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED)
260
261         for task in done:
262             task.result()
263         for task in pending:
264             task.cancel()
265     except OSError as e:
266         print(e)
267
268
269 def main():
270     logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
271     default_source = 'wss://localhost:1234/ws_connect?id=cug'
272     default_dest = 'rtsp://localhost:8554/cug'
273     parser = argparse.ArgumentParser()
274     parser.add_argument('-s', '--source', default=default_source,
275                         help=f'Laplace signalling websocket URI, default: {default_source}')
276     parser.add_argument('-d', '--destination', default=default_dest,
277                         help=f'RTMP server URI, default: {default_dest}')
278     args = parser.parse_args()
279
280     Gst.init(None)
281     asyncio.run(run(args.source, args.destination))
282
283
284 if __name__ == '__main__':
285     main()