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(f'send_sdp_offer with {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         log.info('on_offer_created')
  52         promise.wait()
  53         reply = promise.get_reply()
  54         offer = reply.get_value('offer')
  55         promise = Gst.Promise.new()
  56         self.webrtc.emit('set-local-description', offer, promise)
  57         promise.interrupt()
  58         self.send_sdp_offer(offer)
  59
  60     def on_negotiation_needed(self, element):
  61         log.info('on_negotiation_needed')
  62         promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
  63         element.emit('create-offer', None, promise)
  64
  65     def send_ice_candidate_message(self, _, mlineindex, candidate):
  66         icemsg = json.dumps({
  67             'SessionID': self.session_id,
  68             'Type': 'addCalleeIceCandidate',
  69             'Value': json.dumps({
  70                 "candidate": candidate,
  71                 "sdpMid": "0",
  72                 "sdpMLineIndex": mlineindex,
  73                 })
  74             })
  75         log.info(f'send_ice_candidate_message with {icemsg}')
  76         loop = asyncio.new_event_loop()
  77         loop.run_until_complete(self.websocket.send(icemsg))
  78         loop.close()
  79
  80     def on_incoming_decodebin_stream(self, _, pad):
  81         log.info('on_incoming_decodebin_stream')
  82         if not pad.has_current_caps():
  83             log.info(pad, 'has no caps, ignoring')
  84             return
  85
  86         caps = pad.get_current_caps()
  87         assert (len(caps))
  88         s = caps[0]
  89         name = s.get_name()
  90         if name.startswith('video'):
  91             q = Gst.ElementFactory.make('queue')
  92             conv = Gst.ElementFactory.make('videoconvert')
  93             sink = Gst.ElementFactory.make('autovideosink')
  94             self.pipe.add(q, conv, sink)
  95             self.pipe.sync_children_states()
  96             pad.link(q.get_static_pad('sink'))
  97             q.link(conv)
  98             conv.link(sink)
  99         elif name.startswith('audio'):
 100             q = Gst.ElementFactory.make('queue')
 101             conv = Gst.ElementFactory.make('audioconvert')
 102             resample = Gst.ElementFactory.make('audioresample')
 103             sink = Gst.ElementFactory.make('autoaudiosink')
 104             self.pipe.add(q, conv, resample, sink)
 105             self.pipe.sync_children_states()
 106             pad.link(q.get_static_pad('sink'))
 107             q.link(conv)
 108             conv.link(resample)
 109             resample.link(sink)
 110
 111     def on_incoming_stream(self, _, pad):
 112         log.info('on_incoming_stream')
 113         if pad.direction != Gst.PadDirection.SRC:
 114             return
 115         decodebin = Gst.ElementFactory.make('decodebin')
 116         decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
 117         self.pipe.add(decodebin)
 118         decodebin.sync_state_with_parent()
 119         self.webrtc.link(decodebin)
 120
 121     def start_pipeline(self):
 122         self.webrtc = Gst.ElementFactory.make('webrtcbin', 'laplace')
 123         # self.webrtc.set_property("bundle-policy", 3)
 124         direction = GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
 125         video_caps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
 126         audio_caps = Gst.caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,clock-rate=48000,payload=111")
 127         self.webrtc.emit('add-transceiver', direction, video_caps)
 128         self.webrtc.emit('add-transceiver', direction, audio_caps)
 129         self.pipe = Gst.Pipeline.new("pipeline")
 130         Gst.Bin.do_add_element(self.pipe, self.webrtc)
 131         self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
 132         self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
 133         self.webrtc.connect('pad-added', self.on_incoming_stream)
 134         self.pipe.set_state(Gst.State.PLAYING)
 135         self.webrtc.emit('create-data-channel', 'laplace', None)
 136
 137     def close_pipeline(self):
 138         self.pipe.set_state(Gst.State.NULL)
 139         self.pipe = None
 140         self.webrtc = None
 141
 142     def handle_sdp(self, sdp):
 143         log.info(f'handle_sdp: {sdp}')
 144         res, sdpmsg = GstSdp.SDPMessage.new()
 145         GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
 146         answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
 147         promise = Gst.Promise.new()
 148         self.webrtc.emit('set-remote-description', answer, promise)
 149         promise.interrupt()
 150
 151     def handle_ice(self, ice):
 152         log.info(f'handle_ice: {ice}')
 153         candidate = ice['candidate']
 154         sdpmlineindex = ice['sdpMLineIndex']
 155         self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
 156
 157     async def run(self):
 158         async with websockets.connect(self.uri, ssl=self.ssl_context) as websocket:
 159             self.websocket = websocket
 160             self.start_pipeline()
 161             async for msg in websocket:
 162                 msg_json = json.loads(msg)
 163                 msg_type = msg_json['Type']
 164                 msg_value = msg_json['Value']
 165                 session_id = msg_json['SessionID']
 166                 log.info(f"receive for session {session_id} type {msg_type}")
 167                 if msg_type == 'newSession':
 168                     self.session_id = session_id
 169                 elif msg_type == 'gotOffer':
 170                     value_json = json.loads(msg_value)
 171                     sdp = value_json['sdp']
 172                     self.handle_sdp(sdp)
 173                 elif msg_type == 'addCallerIceCandidate':
 174                     value_json = json.loads(msg_value)
 175                     self.handle_ice(value_json)
 176         self.close_pipeline()
 177         self.websocket = None
 178         self.session_id = None
 179
 180
 181 def check_plugins():
 182     for plugin in ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
 183                    "rtpmanager", "videotestsrc", "audiotestsrc"]:
 184         if Gst.Registry.get().find_plugin(plugin) is None:
 185             print('Missing gstreamer plugin:', plugin)
 186             return False
 187     return True
 188
 189
 190 def main():
 191     logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
 192     Gst.init(None)
 193     if not check_plugins():
 194         sys.exit(1)
 195     parser = argparse.ArgumentParser()
 196     parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
 197         help='Signalling server URI')
 198     args = parser.parse_args()
 199     c = WebRTCClient(args.uri)
 200     loop = asyncio.get_event_loop()
 201     loop.run_until_complete(c.run())
 202
 203
 204 if __name__=='__main__':
 205     main()