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         self.userfragments = []
  35
  36     def send_sdp_offer(self, offer):
  37         text = offer.sdp.as_text()
  38         log.info(f'send_sdp_offer with {text}')
  39         msg = json.dumps({
  40             'SessionID': self.session_id,
  41             'Type': "gotAnswer",
  42             'Value': json.dumps({
  43                 'type': 'answer',
  44                 'sdp': text
  45             })
  46         })
  47         loop = asyncio.new_event_loop()
  48         loop.run_until_complete(self.websocket.send(msg))
  49         loop.close()
  50
  51     def on_offer_created(self, promise, _, __):
  52         log.info('on_offer_created')
  53         promise.wait()
  54         reply = promise.get_reply()
  55         offer = reply.get_value('offer')
  56         promise = Gst.Promise.new()
  57         self.webrtc.emit('set-local-description', offer, promise)
  58         promise.interrupt()
  59         self.send_sdp_offer(offer)
  60
  61         sdp = offer.sdp
  62         self.userfragments = [sdp.get_media(i).get_attribute_val('ice-ufrag') for i in range(sdp.medias_len())]
  63
  64     def on_negotiation_needed(self, element):
  65         log.info('on_negotiation_needed')
  66         promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
  67         element.emit('create-offer', None, promise)
  68
  69     def send_ice_candidate_message(self, _, mlineindex, candidate):
  70         icemsg = json.dumps({
  71             'SessionID': self.session_id,
  72             'Type': 'addCalleeIceCandidate',
  73             'Value': json.dumps({
  74                 "candidate": candidate,
  75                 "sdpMid": f"{mlineindex}",
  76                 "sdpMLineIndex": mlineindex,
  77                 "usernameFragment": self.userfragments[mlineindex],
  78                 })
  79             })
  80         log.info(f'send_ice_candidate_message with {icemsg}')
  81         loop = asyncio.new_event_loop()
  82         loop.run_until_complete(self.websocket.send(icemsg))
  83         loop.close()
  84
  85     def on_incoming_decodebin_stream(self, _, pad):
  86         log.info('on_incoming_decodebin_stream')
  87         if not pad.has_current_caps():
  88             log.info(pad, 'has no caps, ignoring')
  89             return
  90
  91         caps = pad.get_current_caps()
  92         assert (len(caps))
  93         s = caps[0]
  94         name = s.get_name()
  95         if name.startswith('video'):
  96             q = Gst.ElementFactory.make('queue')
  97             conv = Gst.ElementFactory.make('videoconvert')
  98             sink = Gst.ElementFactory.make('autovideosink')
  99             self.pipe.add(q, conv, sink)
 100             self.pipe.sync_children_states()
 101             pad.link(q.get_static_pad('sink'))
 102             q.link(conv)
 103             conv.link(sink)
 104         elif name.startswith('audio'):
 105             q = Gst.ElementFactory.make('queue')
 106             conv = Gst.ElementFactory.make('audioconvert')
 107             resample = Gst.ElementFactory.make('audioresample')
 108             sink = Gst.ElementFactory.make('autoaudiosink')
 109             self.pipe.add(q, conv, resample, sink)
 110             self.pipe.sync_children_states()
 111             pad.link(q.get_static_pad('sink'))
 112             q.link(conv)
 113             conv.link(resample)
 114             resample.link(sink)
 115
 116     def on_incoming_stream(self, _, pad):
 117         log.info('on_incoming_stream')
 118         if pad.direction != Gst.PadDirection.SRC:
 119             return
 120         decodebin = Gst.ElementFactory.make('decodebin')
 121         decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
 122         self.pipe.add(decodebin)
 123         decodebin.sync_state_with_parent()
 124         self.webrtc.link(decodebin)
 125
 126     def start_pipeline(self):
 127         self.webrtc = Gst.ElementFactory.make('webrtcbin', 'laplace')
 128         # self.webrtc.set_property("bundle-policy", 3)
 129         direction = GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
 130         video_caps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
 131         audio_caps = Gst.caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,clock-rate=48000,payload=111")
 132         self.webrtc.emit('add-transceiver', direction, video_caps)
 133         self.webrtc.emit('add-transceiver', direction, audio_caps)
 134         self.pipe = Gst.Pipeline.new("pipeline")
 135         Gst.Bin.do_add_element(self.pipe, self.webrtc)
 136         self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
 137         self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
 138         self.webrtc.connect('pad-added', self.on_incoming_stream)
 139         self.pipe.set_state(Gst.State.PLAYING)
 140         self.webrtc.emit('create-data-channel', 'laplace', None)
 141
 142     def close_pipeline(self):
 143         self.pipe.set_state(Gst.State.NULL)
 144         self.pipe = None
 145         self.webrtc = None
 146
 147     def handle_sdp(self, sdp):
 148         log.info(f'handle_sdp: {sdp}')
 149         res, sdpmsg = GstSdp.SDPMessage.new()
 150         GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
 151         answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
 152         promise = Gst.Promise.new()
 153         self.webrtc.emit('set-remote-description', answer, promise)
 154         promise.interrupt()
 155
 156     def handle_ice(self, ice):
 157         log.info(f'handle_ice: {ice}')
 158         candidate = ice['candidate']
 159         sdpmlineindex = ice['sdpMLineIndex']
 160         self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
 161
 162     async def run(self):
 163         async with websockets.connect(self.uri, ssl=self.ssl_context) as websocket:
 164             self.websocket = websocket
 165             self.start_pipeline()
 166             async for msg in websocket:
 167                 msg_json = json.loads(msg)
 168                 msg_type = msg_json['Type']
 169                 msg_value = msg_json['Value']
 170                 session_id = msg_json['SessionID']
 171                 log.info(f"receive for session {session_id} type {msg_type}")
 172                 if msg_type == 'newSession':
 173                     self.session_id = session_id
 174                 elif msg_type == 'gotOffer':
 175                     value_json = json.loads(msg_value)
 176                     sdp = value_json['sdp']
 177                     self.handle_sdp(sdp)
 178                 elif msg_type == 'addCallerIceCandidate':
 179                     value_json = json.loads(msg_value)
 180                     self.handle_ice(value_json)
 181         self.close_pipeline()
 182         self.websocket = None
 183         self.session_id = None
 184
 185
 186 def check_plugins():
 187     for plugin in ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
 188                    "rtpmanager", "videotestsrc", "audiotestsrc"]:
 189         if Gst.Registry.get().find_plugin(plugin) is None:
 190             print('Missing gstreamer plugin:', plugin)
 191             return False
 192     return True
 193
 194
 195 def main():
 196     logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
 197     Gst.init(None)
 198     if not check_plugins():
 199         sys.exit(1)
 200     parser = argparse.ArgumentParser()
 201     parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
 202         help='Signalling server URI')
 203     args = parser.parse_args()
 204     c = WebRTCClient(args.uri)
 205     loop = asyncio.get_event_loop()
 206     loop.run_until_complete(c.run())
 207
 208
 209 if __name__=='__main__':
 210     main()