364 lignes
14 KiB
Python
Fichiers exécutables
364 lignes
14 KiB
Python
Fichiers exécutables
#!/usr/bin/python3
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import random
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import ssl
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import string
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import websockets
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import asyncio
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import os
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import sys
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import json
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import argparse
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import qrcode
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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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PIPELINE_START = 'webrtcbin name=sendrecv\n'
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PIPELINE_VIDEO_POSTFIX = ''' ! videoconvert ! queue !
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vp8enc deadline=1 ! rtpvp8pay !
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queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 !
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sendrecv.
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'''
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PIPELINE_AUDIO_POSTFIX = ''' ! audioconvert ! audioresample ! queue !
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opusenc ! rtpopuspay !
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queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 !
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sendrecv.
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'''
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class WebRTCClient:
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def __init__(self, args):
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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.url = args.url
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self.has_offer = False
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self.is_host = True
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self.args = args
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if args.roomName is None:
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# From https://stackoverflow.com/a/2030081
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self.roomName = ''.join(random.choice(string.ascii_lowercase)
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for i in range(6))
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else:
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self.roomName = args.roomName
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qr = qrcode.QRCode()
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client_url = '#'.join([self.url, self.roomName])
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print(client_url)
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qr.add_data(client_url)
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qr.print_ascii(tty=True)
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self.server = 'ws' + self.url[4:] + 'ws/'\
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+ ('host' if self.is_host else 'client') + '/'\
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+ self.roomName + '/'
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falseStrings = ['false', 'null', 'none', 'no']
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testStrings = ['test']
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audioPipeline = self.args.sendAudio
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if audioPipeline.lower() in falseStrings:
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self.sendAudio = False
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audioPipeline = 'audiotestsrc wave=silence'
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elif audioPipeline.lower() in testStrings:
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self.sendAudio = True
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audioPipeline = 'audiotestsrc wave=red-noise'
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else:
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self.sendAudio = True
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videoPipeline = self.args.sendVideo
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if videoPipeline.lower() in falseStrings:
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self.sendVideo = False
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videoPipeline = 'videotestsrc pattern=solid-color'
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elif videoPipeline.lower() in testStrings:
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self.sendVideo = True
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videoPipeline = 'videotestsrc pattern=ball'
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enableAudio = self.sendAudio or self.args.receiveAudio
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enableVideo = self.sendVideo or self.args.receiveVideo != 'false'
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if not (enableAudio or enableVideo):
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print('Must enable audio or video.')
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sys.exit()
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self.pipeline = PIPELINE_START
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if enableAudio:
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self.pipeline += audioPipeline + PIPELINE_AUDIO_POSTFIX
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if enableVideo:
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self.pipeline += videoPipeline + PIPELINE_VIDEO_POSTFIX
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async def connect(self):
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sslctx = ssl.create_default_context(purpose=ssl.Purpose.CLIENT_AUTH)
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self.conn = await websockets.connect(self.server, ssl=sslctx)
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if not self.is_host:
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await self.conn.send('{"ready": "separateIce"}')
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self.start_pipeline()
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def send_sdp_offer(self, offer):
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if not self.is_host and not self.has_offer:
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pass
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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({'description': {'type': 'offer', 'sdp': text}})
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loop = asyncio.new_event_loop()
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loop.run_until_complete(self.conn.send(msg))
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def on_offer_created(self, promise, _, __):
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print('In on_offer_created...')
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promise.wait()
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reply = promise.get_reply()
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offer = reply.get_value('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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self.send_sdp_offer(offer)
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def on_negotiation_needed(self, element):
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print('In on_negotiation_needed...')
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promise = Gst.Promise.new_with_change_func(self.on_offer_created,
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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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if not self.is_host and not self.has_offer:
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pass
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icemsg = json.dumps({'candidate': candidate,
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'sdpMLineIndex': mlineindex})
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loop = asyncio.new_event_loop()
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loop.run_until_complete(self.conn.send(icemsg))
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def on_incoming_decodebin_stream(self, _, pad):
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print('In on_incoming_decodebin_stream...')
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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 caps.get_size()
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s = caps.get_structure(0)
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name = s.get_name()
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if name.startswith('video'):
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print("Connecting incoming video stream...")
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q = Gst.ElementFactory.make('queue')
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conv = Gst.ElementFactory.make('videoconvert')
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if self.args.receiveVideoTo == 'auto':
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print('Displaying video to screen using autovideosink.')
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sink = Gst.ElementFactory.make('autovideosink')
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self.pipe.add(q)
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self.pipe.add(conv)
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self.pipe.add(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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else:
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print('Sending video to v4l2 device %s.'
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% self.args.receiveVideoTo)
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caps = Gst.Caps.from_string("video/x-raw,format=YUY2")
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capsfilter = Gst.ElementFactory.make("capsfilter", "vfilter")
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capsfilter.set_property("caps", caps)
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sink = Gst.ElementFactory.make('v4l2sink')
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sink.set_property('device', self.args.receiveVideoTo)
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self.pipe.add(q)
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self.pipe.add(conv)
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self.pipe.add(capsfilter)
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self.pipe.add(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(capsfilter)
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capsfilter.link(sink)
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elif name.startswith('audio'):
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print("Connecting incoming audio stream...")
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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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if self.args.receiveAudioTo == 'auto':
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print('Playing audio using autoaudiosink.')
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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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elif self.args.receiveAudioTo.startswith('device='):
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device = self.args.receiveAudioTo[len('device='):]
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print('Playing audio using pulseaudio device %s.' % device)
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sink = Gst.ElementFactory.make('pulsesink')
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sink.set_property('device', device)
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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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else:
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print('Sending audio to file %s.' % self.args.receiveAudioTo)
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caps = Gst.Caps.from_string(
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"audio/x-raw,format=S16LE,channels=1")
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capsfilter = Gst.ElementFactory.make("capsfilter", "afilter")
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capsfilter.set_property("caps", caps)
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sink = Gst.ElementFactory.make('filesink')
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sink.set_property('location', self.args.receiveAudioTo)
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sink.set_property('sync', 'true')
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self.pipe.add(q)
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self.pipe.add(conv)
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self.pipe.add(resample)
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self.pipe.add(capsfilter)
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self.pipe.add(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(capsfilter)
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capsfilter.link(sink)
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def on_incoming_stream(self, _, pad):
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print('In on_incoming_stream...')
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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_data_channel_open(self):
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print('In on_data_channel_open...')
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def on_data_channel_message(self, msg):
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print('In on_data_channel_message...')
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print('Data channel message: %s' % msg)
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def on_data_channel(self, channel):
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print('In on_data_channel...')
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self.data_channel = channel
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channel.connect('on-open', self.on_data_channel_open)
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channel.connect('on-message-string', self.on_data_channel_message)
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def start_pipeline(self):
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print('In start_pipeline...')
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self.pipe = Gst.parse_launch(self.pipeline)
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self.webrtc = self.pipe.get_by_name('sendrecv')
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self.webrtc.connect('on-negotiation-needed',
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self.on_negotiation_needed)
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self.webrtc.connect('on-ice-candidate',
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self.send_ice_candidate_message)
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self.webrtc.connect('on-data-channel',
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self.on_data_channel)
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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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async def handle_sdp(self, msg):
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if not self.webrtc:
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self.start_pipeline()
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assert (self.webrtc)
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if 'description' in msg:
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print('connection-state=%s'
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% self.webrtc.get_property('connection-state'))
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self.has_offer = True
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sdp = msg['description']
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assert(sdp['type'] == 'answer')
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sdp = sdp['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(
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GstWebRTC.WebRTCSDPType.ANSWER,
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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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elif 'candidate' in msg:
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candidate = msg['candidate']
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sdpmlineindex = msg['sdpMLineIndex']
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self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
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async def loop(self):
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assert self.conn
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async for message in self.conn:
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msg = json.loads(message)
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if 'ready' in msg:
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self.start_pipeline()
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await self.conn.send(json.dumps({'settings': {
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'separateIce': True,
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'serverless': False,
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'client-video': 'none' if self.args.receiveVideo == 'false' else self.args.receiveVideo,
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'client-audio': self.args.receiveAudio,
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'host-video': self.sendVideo,
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'host-audio': self.sendAudio,
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'debug': True,
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}}))
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else:
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await self.handle_sdp(msg)
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return 0
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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(
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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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if __name__ == '__main__':
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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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parser = argparse.ArgumentParser()
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parser.add_argument('--url', help='URL from minimal-webrtc',
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default='https://localhost/camera/')
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parser.add_argument('--roomName', help='room name to host')
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parser.add_argument('--sendAudio', default='test',
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help='GStreamer audio pipeline to send')
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parser.add_argument('--sendVideo', default='test',
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help='GStreamer video pipeline to send')
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parser.add_argument('--receiveAudio', action='store_true', default=None,
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help='Enable receiving audio')
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parser.add_argument('--receiveVideo', default=None,
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help='Set video to receive ("screen", '
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+ '"environment", "facing", "true", "false")')
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parser.add_argument('--receiveAudioTo', default=None,
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help='"auto" or file path or device=DEVICE '
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+ 'where DEVICE is a PulseAudio sink '
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+ 'to send received audio to ')
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parser.add_argument('--receiveVideoTo', default=None,
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help='"auto" or file path to send received video to')
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args = parser.parse_args()
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# Support only one of receiveAudio/receiveAudioTo or
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# receiveVideo/receiveVideoTo while setting reasonable defaults.
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if args.receiveAudio is not None and args.receiveAudioTo is not None:
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pass
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elif args.receiveAudio is None and args.receiveAudioTo is None:
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args.receiveAudio = False
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elif args.receiveAudio is None:
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args.receiveAudio = True
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elif args.receiveAudioTo is None:
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args.receiveAudioTo = 'auto'
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if args.receiveVideo is not None and args.receiveVideoTo is not None:
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pass
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elif args.receiveVideo is None and args.receiveVideoTo is None:
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args.receiveVideo = False
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elif args.receiveVideo is None:
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args.receiveVideo = True
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elif args.receiveVideoTo is None:
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args.receiveVideoTo = 'auto'
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c = WebRTCClient(args)
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asyncio.get_event_loop().run_until_complete(c.connect())
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res = asyncio.get_event_loop().run_until_complete(c.loop())
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sys.exit(res)
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