gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
2022-03-01 16:33:28 +00:00

200 lines
7.1 KiB
Python

import random
import ssl
import websockets
import asyncio
import os
import sys
import json
import argparse
import gi
gi.require_version('Gst', '1.0')
from gi.repository import Gst
gi.require_version('GstWebRTC', '1.0')
from gi.repository import GstWebRTC
gi.require_version('GstSdp', '1.0')
from gi.repository import GstSdp
# Ensure that gst-python is installed
try:
from gi.overrides import Gst as _
except ImportError:
print('gstreamer-python binding overrides aren\'t available, please install them')
raise
# These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations
PIPELINE_DESC = '''
webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay !
queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
'''
from websockets.version import version as wsv
class WebRTCClient:
def __init__(self, id_, peer_id, server):
self.id_ = id_
self.conn = None
self.pipe = None
self.webrtc = None
self.peer_id = peer_id
self.server = server or 'wss://webrtc.nirbheek.in:8443'
async def connect(self):
self.conn = await websockets.connect(self.server)
await self.conn.send('HELLO %d' % self.id_)
async def setup_call(self):
await self.conn.send('SESSION {}'.format(self.peer_id))
def send_sdp_offer(self, offer):
text = offer.sdp.as_text()
print('Sending offer:\n%s' % text)
msg = json.dumps({'sdp': {'type': 'offer', 'sdp': text}})
loop = asyncio.new_event_loop()
loop.run_until_complete(self.conn.send(msg))
loop.close()
def on_offer_created(self, promise, _, __):
promise.wait()
reply = promise.get_reply()
offer = reply['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):
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):
icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
loop = asyncio.new_event_loop()
loop.run_until_complete(self.conn.send(icemsg))
loop.close()
def on_incoming_decodebin_stream(self, _, pad):
if not pad.has_current_caps():
print(pad, 'has no caps, ignoring')
return
caps = pad.get_current_caps()
assert (len(caps))
s = caps[0]
name = s.get_name()
if name.startswith('video'):
q = Gst.ElementFactory.make('queue')
conv = Gst.ElementFactory.make('videoconvert')
sink = Gst.ElementFactory.make('autovideosink')
self.pipe.add(q, conv, sink)
self.pipe.sync_children_states()
pad.link(q.get_static_pad('sink'))
q.link(conv)
conv.link(sink)
elif name.startswith('audio'):
q = Gst.ElementFactory.make('queue')
conv = Gst.ElementFactory.make('audioconvert')
resample = Gst.ElementFactory.make('audioresample')
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)
def on_incoming_stream(self, _, pad):
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)
def start_pipeline(self):
self.pipe = Gst.parse_launch(PIPELINE_DESC)
self.webrtc = self.pipe.get_by_name('sendrecv')
self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
self.webrtc.connect('pad-added', self.on_incoming_stream)
self.pipe.set_state(Gst.State.PLAYING)
def handle_sdp(self, message):
assert (self.webrtc)
msg = json.loads(message)
if 'sdp' in msg:
sdp = msg['sdp']
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)
answer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
promise = Gst.Promise.new()
self.webrtc.emit('set-remote-description', answer, promise)
promise.interrupt()
elif 'ice' in msg:
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.webrtc.emit('add-ice-candidate', sdpmlineindex, candidate)
def close_pipeline(self):
if self.pipe:
self.pipe.set_state(Gst.State.NULL)
self.pipe = None
self.webrtc = None
async def loop(self):
assert self.conn
async for message in self.conn:
if message == 'HELLO':
await self.setup_call()
elif message == 'SESSION_OK':
self.start_pipeline()
elif message.startswith('ERROR'):
print(message)
self.close_pipeline()
return 1
else:
self.handle_sdp(message)
self.close_pipeline()
return 0
async def stop(self):
if self.conn:
await self.conn.close()
self.conn = None
def check_plugins():
needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
"rtpmanager", "videotestsrc", "audiotestsrc"]
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
if __name__ == '__main__':
Gst.init(None)
if not check_plugins():
sys.exit(1)
parser = argparse.ArgumentParser()
parser.add_argument('peerid', help='String ID of the peer to connect to')
parser.add_argument('--server', help='Signalling server to connect to, eg "wss://127.0.0.1:8443"')
args = parser.parse_args()
our_id = random.randrange(10, 10000)
c = WebRTCClient(our_id, args.peerid, args.server)
loop = asyncio.new_event_loop()
loop.run_until_complete(c.connect())
res = loop.run_until_complete(c.loop())
sys.exit(res)