subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py

   1 import random
   2 import ssl
   3 import websockets
   4 import asyncio
   5 import os
   6 import sys
   7 import json
   8 import argparse
   9
  10 import gi
  11 gi.require_version('Gst', '1.0')
  12 from gi.repository import Gst
  13 gi.require_version('GstWebRTC', '1.0')
  14 from gi.repository import GstWebRTC
  15 gi.require_version('GstSdp', '1.0')
  16 from gi.repository import GstSdp
  17
  18 PIPELINE_DESC = '''
  19 webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
  20  videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
  21  queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
  22  audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
  23  queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
  24 '''
  25
  26 from websockets.version import version as wsv
  27
  28 class WebRTCClient:
  29     def __init__(self, id_, peer_id, server):
  30         self.id_ = id_
  31         self.conn = None
  32         self.pipe = None
  33         self.webrtc = None
  34         self.peer_id = peer_id
  35         self.server = server or 'wss://webrtc.nirbheek.in:8443'
  36
  37
  38     async def connect(self):
  39         sslctx = ssl.create_default_context(purpose=ssl.Purpose.CLIENT_AUTH)
  40         self.conn = await websockets.connect(self.server, ssl=sslctx)
  41         await self.conn.send('HELLO %d' % self.id_)
  42
  43     async def setup_call(self):
  44         await self.conn.send('SESSION {}'.format(self.peer_id))
  45
  46     def send_sdp_offer(self, offer):
  47         text = offer.sdp.as_text()
  48         print ('Sending offer:\n%s' % text)
  49         msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
  50         loop = asyncio.new_event_loop()
  51         loop.run_until_complete(self.conn.send(msg))
  52         loop.close()
  53
  54     def on_offer_created(self, promise, _, __):
  55         promise.wait()
  56         reply = promise.get_reply()
  57         offer = reply['offer']
  58         promise = Gst.Promise.new()
  59         self.webrtc.emit('set-local-description', offer, promise)
  60         promise.interrupt()
  61         self.send_sdp_offer(offer)
  62
  63     def on_negotiation_needed(self, element):
  64         promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
  65         element.emit('create-offer', None, promise)
  66
  67     def send_ice_candidate_message(self, _, mlineindex, candidate):
  68         icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
  69         loop = asyncio.new_event_loop()
  70         loop.run_until_complete(self.conn.send(icemsg))
  71         loop.close()
  72
  73     def on_incoming_decodebin_stream(self, _, pad):
  74         if not pad.has_current_caps():
  75             print (pad, 'has no caps, ignoring')
  76             return
  77
  78         caps = pad.get_current_caps()
  79         assert (len(caps))
  80         s = caps[0]
  81         name = s.get_name()
  82         if name.startswith('video'):
  83             q = Gst.ElementFactory.make('queue')
  84             conv = Gst.ElementFactory.make('videoconvert')
  85             sink = Gst.ElementFactory.make('autovideosink')
  86             self.pipe.add(q, conv, sink)
  87             self.pipe.sync_children_states()
  88             pad.link(q.get_static_pad('sink'))
  89             q.link(conv)
  90             conv.link(sink)
  91         elif name.startswith('audio'):
  92             q = Gst.ElementFactory.make('queue')
  93             conv = Gst.ElementFactory.make('audioconvert')
  94             resample = Gst.ElementFactory.make('audioresample')
  95             sink = Gst.ElementFactory.make('autoaudiosink')
  96             self.pipe.add(q, conv, resample, sink)
  97             self.pipe.sync_children_states()
  98             pad.link(q.get_static_pad('sink'))
  99             q.link(conv)
 100             conv.link(resample)
 101             resample.link(sink)
 102
 103     def on_incoming_stream(self, _, pad):
 104         if pad.direction != Gst.PadDirection.SRC:
 105             return
 106
 107         decodebin = Gst.ElementFactory.make('decodebin')
 108         decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
 109         self.pipe.add(decodebin)
 110         decodebin.sync_state_with_parent()
 111         self.webrtc.link(decodebin)
 112
 113     def start_pipeline(self):
 114         self.pipe = Gst.parse_launch(PIPELINE_DESC)
 115         self.webrtc = self.pipe.get_by_name('sendrecv')
 116         self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
 117         self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
 118         self.webrtc.connect('pad-added', self.on_incoming_stream)
 119         self.pipe.set_state(Gst.State.PLAYING)
 120
 121     def handle_sdp(self, message):
 122         assert (self.webrtc)
 123         msg = json.loads(message)
 124         if 'sdp' in msg:
 125             sdp = msg['sdp']
 126             assert(sdp['type'] == 'answer')
 127             sdp = sdp['sdp']
 128             print ('Received answer:\n%s' % sdp)
 129             res, sdpmsg = GstSdp.SDPMessage.new()
 130             GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
 131             answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
 132             promise = Gst.Promise.new()
 133             self.webrtc.emit('set-remote-description', answer, promise)
 134             promise.interrupt()
 135         elif 'ice' in msg:
 136             ice = msg['ice']
 137             candidate = ice['candidate']
 138             sdpmlineindex = ice['sdpMLineIndex']
 139             self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
 140
 141     def close_pipeline(self):
 142         self.pipe.set_state(Gst.State.NULL)
 143         self.pipe = None
 144         self.webrtc = None
 145
 146     async def loop(self):
 147         assert self.conn
 148         async for message in self.conn:
 149             if message == 'HELLO':
 150                 await self.setup_call()
 151             elif message == 'SESSION_OK':
 152                 self.start_pipeline()
 153             elif message.startswith('ERROR'):
 154                 print (message)
 155                 self.close_pipeline()
 156                 return 1
 157             else:
 158                 self.handle_sdp(message)
 159         self.close_pipeline()
 160         return 0
 161
 162     async def stop(self):
 163         if self.conn:
 164             await self.conn.close()
 165         self.conn = None
 166
 167
 168 def check_plugins():
 169     needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
 170               "rtpmanager", "videotestsrc", "audiotestsrc"]
 171     missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
 172     if len(missing):
 173         print('Missing gstreamer plugins:', missing)
 174         return False
 175     return True
 176
 177
 178 if __name__=='__main__':
 179     Gst.init(None)
 180     if not check_plugins():
 181         sys.exit(1)
 182     parser = argparse.ArgumentParser()
 183     parser.add_argument('peerid', help='String ID of the peer to connect to')
 184     parser.add_argument('--server', help='Signalling server to connect to, eg "wss://127.0.0.1:8443"')
 185     args = parser.parse_args()
 186     our_id = random.randrange(10, 10000)
 187     c = WebRTCClient(our_id, args.peerid, args.server)
 188     loop = asyncio.get_event_loop()
 189     loop.run_until_complete(c.connect())
 190     res = loop.run_until_complete(c.loop())
 191     sys.exit(res)