Ви не можете вибрати більше 25 тем Теми мають розпочинатися з літери або цифри, можуть містити дефіси (-) і не повинні перевищувати 35 символів.

364 рядки
14 KiB
Python

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