laplace_client.py

   1 #!/usr/bin/python3
   2
   3 import argparse
   4 import asyncio
   5 import json
   6 import logging
   7 import pathlib
   8 import ssl
   9 import sys
  10
  11 import websockets
  12
  13 import gi
  14 gi.require_version('Gst', '1.0')
  15 from gi.repository import Gst
  16 gi.require_version('GstWebRTC', '1.0')
  17 from gi.repository import GstWebRTC
  18 gi.require_version('GstSdp', '1.0')
  19 from gi.repository import GstSdp
  20
  21
  22 log = logging.getLogger(__name__)
  23
  24
  25 class WebRTCClient:
  26
  27     def __init__(self, uri: str):
  28         self.uri = uri
  29         self.ssl_context = ssl.SSLContext()
  30         self.ssl_context.check_hostname = False
  31         self.ssl_context.verify_mode = ssl.CERT_NONE
  32         self.websocket = None
  33         self.session_id = None
  34
  35     def send_sdp_offer(self, offer):
  36         text = offer.sdp.as_text()
  37         log.info('Sending offer:\n%s' % text)
  38         msg = json.dumps({
  39             'SessionID': self.session_id,
  40             'Type': "gotAnswer",
  41             'Value': json.dumps({
  42                 'type': 'answer',
  43                 'sdp': text
  44             })
  45         })
  46         loop = asyncio.new_event_loop()
  47         loop.run_until_complete(self.websocket.send(msg))
  48         loop.close()
  49
  50     def on_offer_created(self, promise, _, __):
  51         promise.wait()
  52         reply = promise.get_reply()
  53         offer = reply['offer']
  54         promise = Gst.Promise.new()
  55         self.webrtc.emit('set-local-description', offer, promise)
  56         promise.interrupt()
  57         self.send_sdp_offer(offer)
  58
  59     def on_negotiation_needed(self, element):
  60         promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
  61         element.emit('create-offer', None, promise)
  62
  63     def send_ice_candidate_message(self, _, mlineindex, candidate):
  64         icemsg = json.dumps({
  65             'SessionID': self.session_id,
  66             'Type': 'addCalleeIceCandidate',
  67             'Value': json.dumps({
  68                 "candidate": candidate,
  69                 "sdpMid": "0",
  70                 "sdpMLineIndex": mlineindex,
  71                 })
  72             })
  73         loop = asyncio.new_event_loop()
  74         loop.run_until_complete(self.websocket.send(icemsg))
  75         loop.close()
  76
  77     def on_incoming_decodebin_stream(self, _, pad):
  78         if not pad.has_current_caps():
  79             log.info(pad, 'has no caps, ignoring')
  80             return
  81
  82         caps = pad.get_current_caps()
  83         assert (len(caps))
  84         s = caps[0]
  85         name = s.get_name()
  86         if name.startswith('video'):
  87             q = Gst.ElementFactory.make('queue')
  88             conv = Gst.ElementFactory.make('videoconvert')
  89             sink = Gst.ElementFactory.make('autovideosink')
  90             self.pipe.add(q, conv, sink)
  91             self.pipe.sync_children_states()
  92             pad.link(q.get_static_pad('sink'))
  93             q.link(conv)
  94             conv.link(sink)
  95         elif name.startswith('audio'):
  96             q = Gst.ElementFactory.make('queue')
  97             conv = Gst.ElementFactory.make('audioconvert')
  98             resample = Gst.ElementFactory.make('audioresample')
  99             sink = Gst.ElementFactory.make('autoaudiosink')
 100             self.pipe.add(q, conv, resample, sink)
 101             self.pipe.sync_children_states()
 102             pad.link(q.get_static_pad('sink'))
 103             q.link(conv)
 104             conv.link(resample)
 105             resample.link(sink)
 106
 107     def on_incoming_stream(self, _, pad):
 108         if pad.direction != Gst.PadDirection.SRC:
 109             return
 110         decodebin = Gst.ElementFactory.make('decodebin')
 111         decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
 112         self.pipe.add(decodebin)
 113         decodebin.sync_state_with_parent()
 114         self.webrtc.link(decodebin)
 115
 116     def start_pipeline(self):
 117         self.webrtc = Gst.ElementFactory.make('webrtcbin', 'laplace')
 118         self.webrtc.set_property("bundle-policy", 3)
 119         direction = GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
 120         caps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
 121         self.webrtc.emit('add-transceiver', direction, caps)
 122         self.pipe = Gst.Pipeline.new("pipeline")
 123         Gst.Bin.do_add_element(self.pipe, self.webrtc)
 124         self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
 125         self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
 126         self.webrtc.connect('pad-added', self.on_incoming_stream)
 127         self.pipe.set_state(Gst.State.PLAYING)
 128
 129     def close_pipeline(self):
 130         self.pipe.set_state(Gst.State.NULL)
 131         self.pipe = None
 132         self.webrtc = None
 133
 134     def handle_sdp(self, sdp):
 135         res, sdpmsg = GstSdp.SDPMessage.new()
 136         GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
 137         answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
 138         promise = Gst.Promise.new()
 139         self.webrtc.emit('set-remote-description', answer, promise)
 140         promise.interrupt()
 141
 142     def handle_ice(self, ice):
 143         candidate = ice['candidate']
 144         sdpmlineindex = ice['sdpMLineIndex']
 145         self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
 146
 147     async def run(self):
 148         async with websockets.connect(self.uri, ssl=self.ssl_context) as websocket:
 149             self.websocket = websocket
 150             self.start_pipeline()
 151             async for msg in websocket:
 152                 msg_json = json.loads(msg)
 153                 msg_type = msg_json['Type']
 154                 msg_value = msg_json['Value']
 155                 session_id = msg_json['SessionID']
 156                 log.info(f"receive for session {session_id} type {msg_type}")
 157                 if msg_type == 'newSession':
 158                     self.session_id = session_id
 159                 elif msg_type == 'gotOffer':
 160                     value_json = json.loads(msg_value)
 161                     sdp = value_json['sdp']
 162                     self.handle_sdp(sdp)
 163                 elif msg_type == 'addCallerIceCandidate':
 164                     value_json = json.loads(msg_value)
 165                     self.handle_ice(value_json)
 166         self.close_pipeline()
 167         self.websocket = None
 168         self.session_id = None
 169
 170
 171 def check_plugins():
 172     for plugin in ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
 173                    "rtpmanager", "videotestsrc", "audiotestsrc"]:
 174         if Gst.Registry.get().find_plugin(plugin) is None:
 175             print('Missing gstreamer plugin:', plugin)
 176             return False
 177     return True
 178
 179
 180 def main():
 181     logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
 182     Gst.init(None)
 183     if not check_plugins():
 184         sys.exit(1)
 185     parser = argparse.ArgumentParser()
 186     parser.add_argument('--uri', default='wss://localhost:2222/ws_connect?id=cug',
 187         help='Signalling server URI')
 188     args = parser.parse_args()
 189     c = WebRTCClient(args.uri)
 190     loop = asyncio.get_event_loop()
 191     loop.run_until_complete(c.run())
 192
 193
 194 if __name__=='__main__':
 195     main()