8 from typing import Optional, List
13 gi.require_version('Gst', '1.0')
14 from gi.repository import Gst
16 gi.require_version('GstWebRTC', '1.0')
17 from gi.repository import GstWebRTC
19 gi.require_version('GstSdp', '1.0')
20 from gi.repository import GstSdp
22 gi.require_version('GstRtspServer', '1.0')
23 from gi.repository import Gst, GstRtspServer, GObject, GLib
25 log = logging.getLogger(__name__)
30 server = GstRtspServer.RTSPServer()
31 server.set_address("::")
32 server.set_service('8554') # port as string
33 factory = GstRtspServer.RTSPMediaFactory()
34 # factory.set_launch("intervideosrc ! decodebin ! theoraenc ! queue ! rtptheorapay name=pay0")
35 # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! theoraenc ! queue ! rtptheorapay name=pay0")
36 # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,format=I420,framerate=10/1 ! theoraenc ! queue ! rtptheorapay name=pay0")
37 # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
38 factory.set_launch("intervideosrc ! decodebin ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
39 # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,framerate=10/1 ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
40 factory.set_shared(True)
41 mountPoints = server.get_mount_points()
42 mountPoints.add_factory("/cug", factory)
49 self.sdp_offer = queue.Queue()
50 self.sdp_answer = queue.Queue()
51 self.generated_ice_candidates = queue.Queue()
52 self.received_ice_candidates = queue.Queue()
53 self.sdp_info = queue.Queue() # (sdp_mids, user_fragments)
54 self.room_left = queue.Queue()
57 class SignalingClient:
58 def __init__(self, events: Events, uri):
61 self.ssl_context = ssl.SSLContext()
62 self.ssl_context.check_hostname = False
63 self.ssl_context.verify_mode = ssl.CERT_NONE
64 self.session_id = None
66 async def receive(self, uri):
67 async for msg in self.websocket:
68 msg_json = json.loads(msg)
69 msg_type = msg_json['Type']
70 msg_value = msg_json['Value']
71 assert self.session_id is None or self.session_id == msg_json['SessionID']
72 if msg_type == 'newSession':
73 self.session_id = msg_json['SessionID']
74 log.info(f"New session {self.session_id}")
75 elif msg_type == 'gotOffer':
76 value_json = json.loads(msg_value)
77 sdp = value_json['sdp']
78 log.info(f'Got SDP offer')
79 log.debug(f'SDP offer:\n{sdp}')
80 self.events.sdp_offer.put_nowait(sdp)
81 elif msg_type == 'addCallerIceCandidate':
82 value_json = json.loads(msg_value)
83 log.info(f'Got ICE candidate')
84 log.debug(f'ICE candidate: {value_json}')
85 self.events.received_ice_candidates.put_nowait(value_json)
86 elif msg_type == 'roomNotFound':
87 log.error(f'The room was not found: {uri}')
89 elif msg_type == 'roomClosed':
90 log.info(f'Oh noes, the room went away (session {self.session_id})!')
91 self.events.room_left.put_nowait(True)
94 log.error(f'Unknown message type {msg_type}')
100 if self.events.sdp_answer.qsize() > 0:
101 sdp_answer = self.events.sdp_answer.get_nowait()
102 sdp_answer_msg = json.dumps({
103 'SessionID': self.session_id,
105 'Value': json.dumps({
110 await self.websocket.send(sdp_answer_msg)
112 elif self.events.sdp_info.qsize() > 0:
113 sdp_mids, user_fragments = self.events.sdp_info.get_nowait()
115 elif self.events.generated_ice_candidates.qsize() > 0 \
116 and sdp_mids is not None and user_fragments is not None:
117 mlineindex, candidate = self.events.generated_ice_candidates.get_nowait()
118 sdp_mid = sdp_mids[mlineindex]
119 user_fragment = user_fragments[mlineindex]
120 icemsg_value = json.dumps({
121 "candidate": candidate,
123 "sdpMLineIndex": mlineindex,
124 "usernameFragment": user_fragment,
126 icemsg = json.dumps({
127 'SessionID': self.session_id,
128 'Type': 'addCalleeIceCandidate',
129 'Value': icemsg_value,
131 log.info(f'Send ICE candidate')
132 log.debug(f'ICE candidate: {icemsg_value}')
133 await self.websocket.send(icemsg)
136 await asyncio.sleep(0.2)
139 self.session_id = None
140 async with websockets.connect(self.uri, ssl=self.ssl_context, close_timeout=0.5) as self.websocket:
141 receive_task = asyncio.Task(self.receive(self.uri))
142 send_task = asyncio.Task(self.send())
143 done, pending = await asyncio.wait([receive_task, send_task], return_when=asyncio.FIRST_COMPLETED)
149 def __init__(self, events: Events):
151 self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
152 self.pipe = Gst.Pipeline.new("pipeline")
153 Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
154 self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
155 self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
156 self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
158 def on_negotiation_needed(self, element):
159 log.info('on_negotiation_needed')
161 def on_ice_candidate(self, element, mlineindex, candidate):
162 log.info('on_ice_candidate')
163 self.events.generated_ice_candidates.put_nowait((mlineindex, candidate))
165 def webrtcbin_pad_added(self, element, pad):
166 log.info('webrtcbin_pad_added')
167 if pad.direction != Gst.PadDirection.SRC:
169 decodebin = Gst.ElementFactory.make('decodebin')
170 decodebin.connect('pad-added', self.decodebin_pad_added)
171 self.pipe.add(decodebin)
172 decodebin.sync_state_with_parent()
173 self.webrtcbin.link(decodebin)
175 def decodebin_pad_added(self, element, pad):
176 log.info('decodebin_pad_added')
177 if not pad.has_current_caps():
178 log.info(pad, 'has no caps, ignoring')
180 caps = pad.get_current_caps()
181 padsize = caps.get_size()
183 log.info(f'>>>> {padsize} {caps}')
185 for i in range(padsize):
186 s = caps.get_structure(i) # Gst.Structure
188 log.info(f'###### {name}')
189 if name.startswith('video'):
190 q = Gst.ElementFactory.make('queue')
191 conv = Gst.ElementFactory.make('videoconvert')
192 enc = Gst.ElementFactory.make('x264enc')
193 enc.set_property('bitrate', 1000)
194 enc.set_property('tune', 'zerolatency')
195 capsfilter = Gst.ElementFactory.make('capsfilter')
196 capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
197 flmux = Gst.ElementFactory.make('flvmux')
198 sink = Gst.ElementFactory.make('rtmpsink')
199 sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa')
200 # sink.set_property('location', 'rtmp://bla:1936/gregoa')
201 print(sink.props.location, dir(sink.props))
202 assert q and conv and enc and capsfilter and flmux and sink
207 self.pipe.add(capsfilter)
210 self.pipe.sync_children_states()
212 q_pad_sink = q.get_static_pad('sink')
214 pad_link_return = pad.link(q_pad_sink)
215 assert pad_link_return == Gst.PadLinkReturn.OK
217 # ok = element.link(q)
224 ok = enc.link(capsfilter)
226 ok = capsfilter.link(flmux)
228 ok = flmux.link(sink)
230 self.pipe.set_state(Gst.State.PLAYING)
231 #print(dir(Gst.DebugGraphDetails))
232 #Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL)
234 elif name.startswith('audio'):
235 q = Gst.ElementFactory.make('queue')
236 conv = Gst.ElementFactory.make('audioconvert')
237 resample = Gst.ElementFactory.make('audioresample')
238 sink = Gst.ElementFactory.make('autoaudiosink')
241 self.pipe.add(resample)
243 self.pipe.sync_children_states()
244 pad.link(q.get_static_pad('sink'))
249 def set_remote_desciption_done(self, gst_promise):
250 gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done)
251 self.webrtcbin.emit('create-answer', None, gst_promise)
253 def create_answer_done(self, gst_promise):
254 reply = gst_promise.get_reply()
255 answer = reply.get_value('answer')
256 sdp_message = answer.sdp
257 mids = [sdp_message.get_media(i).get_attribute_val('mid')
258 for i in range(sdp_message.medias_len())]
259 user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
260 for i in range(sdp_message.medias_len())]
261 self.events.sdp_info.put_nowait((mids, user_fragments))
262 sdp_answer = sdp_message.as_text()
263 log.info(f'Send SDP answer')
264 log.debug(f'SDP answer:\n{sdp_answer}')
265 self.events.sdp_answer.put_nowait(sdp_answer)
266 gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
267 self.webrtcbin.emit('set-local-description', answer, gst_promise)
269 def set_local_description_done(self, gst_promise):
270 gst_promise.get_reply()
273 bus = Gst.Pipeline.get_bus(self.pipe)
274 self.pipe.set_state(Gst.State.PLAYING)
277 if bus.have_pending():
279 if msg.type == Gst.MessageType.ERROR:
280 log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
282 elif msg.type == Gst.MessageType.EOS: # end of stream
283 log.info(f'Gstreamer message bus reports end of stream')
285 elif self.events.sdp_offer.qsize() > 0:
286 sdp_offer = self.events.sdp_offer.get_nowait()
287 res, sm = GstSdp.SDPMessage.new()
288 assert res == GstSdp.SDPResult.OK
289 GstSdp.sdp_message_parse_buffer(bytes(sdp_offer.encode()), sm)
290 # the three lines above can also be done this way in new versions of GStreamer:
291 # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
292 rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
293 gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done)
294 self.webrtcbin.emit('set-remote-description', rd, gst_promise)
296 elif self.events.received_ice_candidates.qsize() > 0:
297 ic = self.events.received_ice_candidates.get_nowait()
298 if ic['candidate'] != '':
299 self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
301 elif self.events.room_left.qsize() > 0:
302 self.events.room_left.get_nowait()
306 await asyncio.sleep(0.1)
308 self.pipe.set_state(Gst.State.NULL)
311 async def gstreamer_main_loop():
312 """Does the equivalent of the following lines in an async friendly way:
313 loop = GLib.MainLoop()
316 gst_loop = GLib.MainLoop()
317 context = gst_loop.get_context()
319 events_dispatched = context.iteration(False)
320 await asyncio.sleep(0. if events_dispatched else 0.01)
323 async def run_repeated(task):
326 await asyncio.sleep(0.1)
333 # rtsp = RtspServer()
334 webrtc = WebRTCClient(events)
335 signaling = SignalingClient(events, uri)
337 main_loop_task = asyncio.Task(gstreamer_main_loop())
338 webrtc_task = asyncio.Task(run_repeated(webrtc.run))
339 signaling_task = asyncio.Task(run_repeated(signaling.run))
341 done, pending = await asyncio.wait([main_loop_task, webrtc_task, signaling_task],
342 return_when=asyncio.FIRST_COMPLETED)
353 logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
354 parser = argparse.ArgumentParser()
355 parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
356 help='Signalling server URI')
357 args = parser.parse_args()
360 asyncio.run(run(args.uri), debug=True)
363 if __name__ == '__main__':