laplace_client.py

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