lagarde.py

   1 #!/usr/bin/python3
   2
   3 import argparse
   4 import asyncio
   5 import json
   6 import logging
   7 import ssl
   8 import queue
   9 from typing import Optional, List
  10
  11 import gi
  12 import websockets
  13
  14 gi.require_version('Gst', '1.0')
  15 from gi.repository import Gst
  16
  17 gi.require_version('GstWebRTC', '1.0')
  18 from gi.repository import GstWebRTC
  19
  20 gi.require_version('GstSdp', '1.0')
  21 from gi.repository import GstSdp
  22
  23 gi.require_version('GstRtspServer', '1.0')
  24 from gi.repository import Gst, GstRtspServer, GObject, GLib
  25
  26 log = logging.getLogger(__name__)
  27
  28
  29 class GstreamerRtspServer():
  30     def __init__(self):
  31         server = GstRtspServer.RTSPServer()
  32         server.set_address("::")
  33         server.set_service('8554')  # port as string
  34         factory = GstRtspServer.RTSPMediaFactory()
  35         factory.set_launch("intervideosrc ! decodebin ! theoraenc ! queue ! rtptheorapay name=pay0")
  36         factory.set_shared(True)
  37         mountPoints = server.get_mount_points()
  38         mountPoints.add_factory("/cug", factory)
  39         server.attach()
  40         self.server = server
  41
  42
  43 class Lagarde:
  44     def __init__(self):
  45         self.sdp_offer: Optional[str] = None
  46         self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
  47         self.session_id = None
  48         self.received_ice_candidates = queue.Queue()
  49         self.generated_ice_candidates = queue.Queue()
  50         self.user_fragments: Optional[List] = None
  51         self.mids: Optional[List] = None
  52         self.pipe = None
  53         self.webrtcbin = None
  54
  55     def on_negotiation_needed(self, element):
  56         log.debug('on_negotiation_needed')
  57
  58     def on_ice_candidate(self, element, mlineindex, candidate):
  59         log.debug('on_ice_candidate')
  60         self.generated_ice_candidates.put_nowait((mlineindex, candidate))
  61
  62     def webrtcbin_pad_added(self, element, pad):
  63         log.debug('webrtcbin_pad_added')
  64         if pad.direction != Gst.PadDirection.SRC:
  65             return
  66         decodebin = Gst.ElementFactory.make('decodebin')
  67         decodebin.connect('pad-added', self.decodebin_pad_added)
  68         self.pipe.add(decodebin)
  69         decodebin.sync_state_with_parent()
  70         self.webrtcbin.link(decodebin)
  71
  72     def decodebin_pad_added(self, element, pad):
  73         log.debug('decodebin_pad_added')
  74         if not pad.has_current_caps():
  75             log.debug(pad, 'has no caps, ignoring')
  76             return
  77
  78         caps = pad.get_current_caps()
  79         padsize = caps.get_size()
  80         for i in range(padsize):
  81             s = caps.get_structure(i) # Gst.Structure
  82             name = s.get_name()
  83             if name.startswith('video'):
  84                 q = Gst.ElementFactory.make('queue')
  85                 conv = Gst.ElementFactory.make('videoconvert')
  86                 sink = Gst.ElementFactory.make('intervideosink')
  87                 self.pipe.add(q)
  88                 self.pipe.add(conv)
  89                 self.pipe.add(sink)
  90                 self.pipe.sync_children_states()
  91                 pad.link(q.get_static_pad('sink'))
  92                 q.link(conv)
  93                 conv.link(sink)
  94
  95     async def listen_to_gstreamer_bus(self):
  96         self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
  97         self.pipe = Gst.Pipeline.new("pipeline")
  98         Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
  99         bus = Gst.Pipeline.get_bus(self.pipe)
 100         self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
 101         self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
 102         self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
 103         self.pipe.set_state(Gst.State.PLAYING)
 104         try:
 105             while True:
 106                 if bus.have_pending():
 107                     msg = bus.pop()
 108                     if msg.type == Gst.MessageType.ERROR:
 109                         log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
 110                         return
 111                     elif msg.type == Gst.MessageType.EOS:  # end of stream
 112                         log.info(f'Gstreamer message bus reports end of stream')
 113                         return
 114                 elif self.sdp_offer is not None:
 115                     res, sm = GstSdp.SDPMessage.new()
 116                     assert res == GstSdp.SDPResult.OK
 117                     GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
 118                     # the three lines above can also be done this way in new versions of GStreamer:
 119                     # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
 120                     rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
 121                     gst_promise = Gst.Promise.new()
 122                     self.webrtcbin.emit('set-remote-description', rd, gst_promise)
 123                     await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
 124                     self.sdp_offer = None
 125
 126                     gst_promise = Gst.Promise.new()
 127                     self.webrtcbin.emit('create-answer', None, gst_promise)
 128                     result = await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
 129                     assert result == Gst.PromiseResult.REPLIED
 130                     reply = gst_promise.get_reply()
 131                     answer = reply.get_value('answer')
 132                     sdp_message = answer.sdp
 133                     self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
 134                                           for i in range(sdp_message.medias_len())]
 135                     self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
 136                                           for i in range(sdp_message.medias_len())]
 137                     sdp_answer = sdp_message.as_text()
 138                     log.info(f'Send SDP answer:\n{sdp_answer}')
 139                     sdp_answer_msg = json.dumps({
 140                         'SessionID': self.session_id,
 141                         'Type': "gotAnswer",
 142                         'Value': json.dumps({
 143                             'type': 'answer',
 144                             'sdp': sdp_answer
 145                         })
 146                     })
 147                     gst_promise = Gst.Promise.new()
 148                     self.webrtcbin.emit('set-local-description', answer, gst_promise)
 149                     await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
 150                     gst_promise.get_reply()
 151                     await self.websocket.send(sdp_answer_msg)
 152
 153                 elif self.received_ice_candidates.qsize() > 0:
 154                     ic = self.received_ice_candidates.get_nowait()
 155                     if ic['candidate'] != '':
 156                         self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
 157
 158                 elif self.generated_ice_candidates.qsize() > 0:
 159                     mlineindex, candidate = self.generated_ice_candidates.get_nowait()
 160                     icemsg_value = json.dumps({
 161                         "candidate": candidate,
 162                         "sdpMid": self.mids[mlineindex],
 163                         "sdpMLineIndex": mlineindex,
 164                         "usernameFragment": self.user_fragments[mlineindex],
 165                     })
 166                     icemsg = json.dumps({
 167                         'SessionID': self.session_id,
 168                         'Type': 'addCalleeIceCandidate',
 169                         'Value': icemsg_value,
 170                     })
 171                     log.info(f'Send ICE candidate: {icemsg_value}')
 172                     await self.websocket.send(icemsg)
 173
 174                 else:
 175                     await asyncio.sleep(0.1)
 176         finally:
 177             self.pipe.set_state(Gst.State.NULL)
 178
 179     async def talk_to_websocket(self, uri):
 180         async for msg in self.websocket:
 181             msg_json = json.loads(msg)
 182             msg_type = msg_json['Type']
 183             msg_value = msg_json['Value']
 184             assert self.session_id is None or self.session_id == msg_json['SessionID']
 185             if msg_type == 'newSession':
 186                 self.session_id = msg_json['SessionID']
 187                 log.info(f"New session {self.session_id}")
 188             elif msg_type == 'gotOffer':
 189                 value_json = json.loads(msg_value)
 190                 sdp = value_json['sdp']
 191                 log.info(f'Got SDP offer:\n{sdp}')
 192                 self.sdp_offer = sdp
 193             elif msg_type == 'addCallerIceCandidate':
 194                 value_json = json.loads(msg_value)
 195                 log.info(f'Got ICE candidate: {value_json}')
 196                 self.received_ice_candidates.put_nowait(value_json)
 197             elif msg_type == 'roomNotFound':
 198                 log.error(f'The room was not found: {uri}')
 199                 return
 200             elif msg_type == 'roomClosed':
 201                 log.info(f'Oh noes, the room went away (session {self.session_id})!')
 202                 self.session_id = None
 203                 return
 204             else:
 205                 log.error(f'Unknown message type {msg_type}')
 206
 207     async def run(self, uri):
 208         ssl_context = ssl.SSLContext()
 209         ssl_context.check_hostname = False
 210         ssl_context.verify_mode = ssl.CERT_NONE
 211         try:
 212             async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
 213                 talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
 214                 listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
 215                 main_loop = asyncio.Task(gstreamer_main_loop())
 216                 done, pending = await asyncio.wait(
 217                     [talk_to_websocket_task, listen_to_gstreamer_bus_task, main_loop],
 218                     return_when=asyncio.FIRST_COMPLETED)
 219                 for d in done:
 220                     d.result()
 221                 for p in pending:
 222                     p.cancel()
 223         except OSError as e:
 224             print(e)
 225
 226
 227 async def gstreamer_main_loop():
 228     """Does the equivalent of the following lines in an async friendly way:
 229         loop = GLib.MainLoop()
 230         loop.run()
 231     """
 232     gst_loop = GLib.MainLoop()
 233     context = gst_loop.get_context()
 234     while True:
 235         events_dispatched = context.iteration(False)
 236         await asyncio.sleep(0. if events_dispatched else 0.01)
 237
 238
 239 def main():
 240     logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
 241     parser = argparse.ArgumentParser()
 242     parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
 243                         help='Signalling server URI')
 244     args = parser.parse_args()
 245
 246     Gst.init(None)
 247     rtsp = GstreamerRtspServer()
 248     lagarde = Lagarde()
 249     asyncio.run(lagarde.run(args.uri), debug=True)
 250
 251
 252 if __name__ == '__main__':
 253     main()