150 lines
5.4 KiB
150 lines
5.4 KiB
import os
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import sys
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import gi
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gi.require_version('Gst', '1.0')
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from gi.repository import Gst
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gi.require_version('GstWebRTC', '1.0')
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from gi.repository import GstWebRTC
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gi.require_version('GstSdp', '1.0')
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from gi.repository import GstSdp
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import json
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PIPELINE_DESC = '''
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webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
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videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
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queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
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audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc !rtpopuspay !
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queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
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'''
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class WebRTCCamera:
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def __init__(self, id_, peer_id, server=None):
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self.id_ = id_
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self.conn = None
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self.pipe = None
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self.webrtc = None
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self.socketio = None
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self.peer_id = peer_id
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self.room = 'default'
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self.sid = 'gstwebrtc1000'
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self.server = server or 'wss://localhost:5000/webrtc'
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self.connected = False
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Gst.init(None)
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if not check_plugins():
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sys.exit(1)
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def connect(self, socketio, room, sid):
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self.socketio = socketio
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self.room = room
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self.sid = sid
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self.start_pipeline()
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self.connected = True
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def disconnect(self):
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self.connected = False
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self.close_pipeline()
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self.socketio = None
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def start_pipeline(self):
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self.pipe = Gst.parse_launch(PIPELINE_DESC)
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self.webrtc = self.pipe.get_by_name('sendrecv')
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self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
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self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
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self.webrtc.connect('pad-added', self.on_incoming_stream)
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self.pipe.set_state(Gst.State.PLAYING)
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def close_pipeline(self):
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self.pipe.set_state(Gst.State.NULL)
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self.pipe = None
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self.webrtc = None
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def handle_sdp_answer(self, sdp):
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print ('Received answer:\n%s' % sdp)
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res, sdpmsg = GstSdp.SDPMessage.new()
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GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
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answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
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promise = Gst.Promise.new()
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self.webrtc.emit('set-remote-description', answer, promise)
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promise.interrupt()
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def handle_ice(self, ice):
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candidate = ice['candidate']
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sdpmlineindex = ice['sdpMLineIndex']
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self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
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def on_negotiation_needed(self, element):
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promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
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element.emit('create-offer', None, promise)
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def send_ice_candidate_message(self, _, mlineindex, candidate):
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icemsg = json.dumps({'type': 'candidate', 'candidate': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
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self.socketio.emit('data', type='candidate', data=icemsg, room=self.room, namespace='/webrtc', skip_sid=self.sid)
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def on_incoming_stream(self, _, pad):
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if pad.direction != Gst.PadDirection.SRC:
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return
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decodebin = Gst.ElementFactory.make('decodebin')
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decodebin.connect('pad-added', self.on_incoming_decodebin_stream)
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self.pipe.add(decodebin)
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decodebin.sync_state_with_parent()
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self.webrtc.link(decodebin)
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def on_incoming_decodebin_stream(self, _, pad):
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if not pad.has_current_caps():
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print (pad, 'has no caps, ignoring')
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return
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caps = pad.get_current_caps()
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assert (len(caps))
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s = caps[0]
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name = s.get_name()
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if name.startswith('video'):
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q = Gst.ElementFactory.make('queue')
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conv = Gst.ElementFactory.make('videoconvert')
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sink = Gst.ElementFactory.make('autovideosink')
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self.pipe.add(q, conv, sink)
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self.pipe.sync_children_states()
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pad.link(q.get_static_pad('sink'))
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q.link(conv)
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conv.link(sink)
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elif name.startswith('audio'):
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q = Gst.ElementFactory.make('queue')
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conv = Gst.ElementFactory.make('audioconvert')
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resample = Gst.ElementFactory.make('audioresample')
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sink = Gst.ElementFactory.make('autoaudiosink')
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self.pipe.add(q, conv, resample, sink)
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self.pipe.sync_children_states()
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pad.link(q.get_static_pad('sink'))
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q.link(conv)
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conv.link(resample)
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resample.link(sink)
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def on_offer_created(self, promise, _, __):
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promise.wait()
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reply = promise.get_reply()
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offer = reply['offer']
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promise = Gst.Promise.new()
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self.webrtc.emit('set-local-description', offer, promise)
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promise.interrupt()
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text = offer.sdp.as_text()
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print ('Sending offer:\n%s' % text)
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msg = json.dumps({'type': 'offer', 'sdp': text})
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self.socketio.emit('data', type='offer', data=msg, room=self.room, namespace='/webrtc', skip_sid=self.sid)
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def check_plugins():
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needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
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"rtpmanager", "videotestsrc", "audiotestsrc"]
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missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
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if len(missing):
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print('Missing gstreamer plugins:', missing)
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return False
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return True
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