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 # Ensure that gst-python is installed
  19 try:
  20     from gi.overrides import Gst as _
  21 except ImportError:
  22     print('gstreamer-python binding overrides aren\'t available, please install them')
  23     raise
  24
  25 # These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations
  26 PIPELINE_DESC = '''
  27 webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
  28  videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
  29   vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay !
  30   queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
  31  audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
  32   queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
  33 '''
  34
  35 from websockets.version import version as wsv
  36
  37
  38 class WebRTCClient:
  39     def __init__(self, loop, id_, peer_id, server):
  40         self.event_loop = loop
  41         self.id_ = id_
  42         self.conn = None
  43         self.pipe = None
  44         self.webrtc = None
  45         self.peer_id = peer_id
  46         self.server = server
  47
  48     async def send(self, msg):
  49         assert self.conn
  50         print(f'>>> Sending {msg}')
  51         await self.conn.send(msg)
  52
  53     async def connect(self):
  54         self.conn = await websockets.connect(self.server)
  55         await self.send('HELLO %d' % self.id_)
  56
  57     async def setup_call(self):
  58         await self.send('SESSION {}'.format(self.peer_id))
  59
  60     def send_soon(self, msg):
  61         asyncio.run_coroutine_threadsafe(self.send(msg), self.event_loop)
  62
  63     def send_sdp_offer(self, offer):
  64         text = offer.sdp.as_text()
  65         print('Sending offer:\n%s' % text)
  66         msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
  67         self.send_soon(msg)
  68
  69     def on_offer_created(self, promise, _, __):
  70         promise.wait()
  71         reply = promise.get_reply()
  72         offer = reply['offer']
  73         promise = Gst.Promise.new()
  74         print('Offer created, setting local description')
  75         self.webrtc.emit('set-local-description', offer, promise)
  76         promise.interrupt()
  77         self.send_sdp_offer(offer)
  78
  79     def on_negotiation_needed(self, element):
  80         promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
  81         element.emit('create-offer', None, promise)
  82
  83     def send_ice_candidate_message(self, _, mlineindex, candidate):
  84         icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
  85         self.send_soon(icemsg)
  86
  87     def on_incoming_decodebin_stream(self, _, pad):
  88         if not pad.has_current_caps():
  89             print(pad, 'has no caps, ignoring')
  90             return
  91
  92         caps = pad.get_current_caps()
  93         assert (len(caps))
  94         s = caps[0]
  95         name = s.get_name()
  96         if name.startswith('video'):
  97             q = Gst.ElementFactory.make('queue')
  98             conv = Gst.ElementFactory.make('videoconvert')
  99             sink = Gst.ElementFactory.make('autovideosink')
 100             self.pipe.add(q, conv, sink)
 101             self.pipe.sync_children_states()
 102             pad.link(q.get_static_pad('sink'))
 103             q.link(conv)
 104             conv.link(sink)
 105         elif name.startswith('audio'):
 106             q = Gst.ElementFactory.make('queue')
 107             conv = Gst.ElementFactory.make('audioconvert')
 108             resample = Gst.ElementFactory.make('audioresample')
 109             sink = Gst.ElementFactory.make('autoaudiosink')
 110             self.pipe.add(q, conv, resample, sink)
 111             self.pipe.sync_children_states()
 112             pad.link(q.get_static_pad('sink'))
 113             q.link(conv)
 114             conv.link(resample)
 115             resample.link(sink)
 116
 117     def on_incoming_stream(self, _, pad):
 118         if pad.direction != Gst.PadDirection.SRC:
 119             return
 120
 121         decodebin = Gst.ElementFactory.make('decodebin')
 122         decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
 123         self.pipe.add(decodebin)
 124         decodebin.sync_state_with_parent()
 125         self.webrtc.link(decodebin)
 126
 127     def start_pipeline(self):
 128         self.pipe = Gst.parse_launch(PIPELINE_DESC)
 129         self.webrtc = self.pipe.get_by_name('sendrecv')
 130         self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
 131         self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
 132         self.webrtc.connect('pad-added', self.on_incoming_stream)
 133         self.pipe.set_state(Gst.State.PLAYING)
 134
 135     def handle_sdp(self, message):
 136         assert (self.webrtc)
 137         msg = json.loads(message)
 138         if 'sdp' in msg:
 139             sdp = msg['sdp']
 140             assert(sdp['type'] == 'answer')
 141             sdp = sdp['sdp']
 142             print('Received answer:\n%s' % sdp)
 143             res, sdpmsg = GstSdp.SDPMessage.new()
 144             GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
 145             answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
 146             promise = Gst.Promise.new()
 147             self.webrtc.emit('set-remote-description', answer, promise)
 148             promise.interrupt()
 149         elif 'ice' in msg:
 150             ice = msg['ice']
 151             candidate = ice['candidate']
 152             sdpmlineindex = ice['sdpMLineIndex']
 153             self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
 154
 155     def close_pipeline(self):
 156         if self.pipe:
 157             self.pipe.set_state(Gst.State.NULL)
 158             self.pipe = None
 159         self.webrtc = None
 160
 161     async def loop(self):
 162         assert self.conn
 163         async for message in self.conn:
 164             if message == 'HELLO':
 165                 await self.setup_call()
 166             elif message == 'SESSION_OK':
 167                 self.start_pipeline()
 168             elif message.startswith('ERROR'):
 169                 print(message)
 170                 self.close_pipeline()
 171                 return 1
 172             else:
 173                 self.handle_sdp(message)
 174         self.close_pipeline()
 175         return 0
 176
 177     async def stop(self):
 178         if self.conn:
 179             await self.conn.close()
 180         self.conn = None
 181
 182
 183 def check_plugins():
 184     needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
 185               "rtpmanager", "videotestsrc", "audiotestsrc"]
 186     missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
 187     if len(missing):
 188         print('Missing gstreamer plugins:', missing)
 189         return False
 190     return True
 191
 192
 193 if __name__ == '__main__':
 194     Gst.init(None)
 195     if not check_plugins():
 196         sys.exit(1)
 197     parser = argparse.ArgumentParser()
 198     parser.add_argument('peerid', help='String ID of the peer to connect to')
 199     parser.add_argument('--server', default='wss://webrtc.nirbheek.in:8443',
 200                         help='Signalling server to connect to, eg "wss://127.0.0.1:8443"')
 201     args = parser.parse_args()
 202     our_id = random.randrange(10, 10000)
 203     loop = asyncio.new_event_loop()
 204     c = WebRTCClient(loop, our_id, args.peerid, args.server)
 205     loop.run_until_complete(c.connect())
 206     res = loop.run_until_complete(c.loop())
 207     sys.exit(res)