#!/usr/bin/python3
-
import argparse
import asyncio
-import datetime
import json
import logging
-import pathlib
import ssl
-import sys
-from typing import Optional, List
-
-import websockets
+import queue
+from typing import List
import gi
+import websockets
gi.require_version('Gst', '1.0')
from gi.repository import Gst
log = logging.getLogger(__name__)
-class Lagarde:
+class Events:
def __init__(self):
- self.sdp_offer: Optional[str] = None
- self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
+ self.sdp_offer = queue.Queue()
+ self.sdp_answer = queue.Queue()
+ self.generated_ice_candidates = queue.Queue()
+ self.received_ice_candidates = queue.Queue()
+ self.sdp_info = queue.Queue() # (sdp_mids, user_fragments)
+ self.room_left = queue.Queue()
+
+
+class SignalingClient:
+ def __init__(self, events: Events, uri):
+ self.events = events
+ self.uri = uri
+ self.ssl_context = ssl.SSLContext()
+ self.ssl_context.check_hostname = False
+ self.ssl_context.verify_mode = ssl.CERT_NONE
self.session_id = None
- self.received_ice_candidates = []
- self.generated_ice_candidates = []
- self.user_fragments: Optional[List] = None
- self.mids: Optional[List] = None
- self.pipe = None
- self.webrtcbin = None
+
+ async def receive(self, uri):
+ async for msg in self.websocket:
+ msg_json = json.loads(msg)
+ msg_type = msg_json['Type']
+ msg_value = msg_json['Value']
+ assert self.session_id is None or self.session_id == msg_json['SessionID']
+ if msg_type == 'newSession':
+ self.session_id = msg_json['SessionID']
+ log.info(f"New session {self.session_id}")
+ elif msg_type == 'gotOffer':
+ value_json = json.loads(msg_value)
+ sdp = value_json['sdp']
+ log.info(f'Got SDP offer')
+ log.debug(f'SDP offer:\n{sdp}')
+ self.events.sdp_offer.put_nowait(sdp)
+ elif msg_type == 'addCallerIceCandidate':
+ value_json = json.loads(msg_value)
+ log.info(f'Got ICE candidate')
+ log.debug(f'ICE candidate: {value_json}')
+ self.events.received_ice_candidates.put_nowait(value_json)
+ elif msg_type == 'roomNotFound':
+ log.error(f'The room was not found: {uri}')
+ return
+ elif msg_type == 'roomClosed':
+ log.info(f'Oh noes, the room went away (session {self.session_id})!')
+ self.events.room_left.put_nowait(True)
+ return
+ else:
+ log.error(f'Unknown message type {msg_type}')
+
+ async def send(self):
+ sdp_mids = None
+ user_fragments = None
+ while True:
+ if self.events.sdp_answer.qsize() > 0:
+ sdp_answer = self.events.sdp_answer.get_nowait()
+ sdp_answer_msg = json.dumps({
+ 'SessionID': self.session_id,
+ 'Type': "gotAnswer",
+ 'Value': json.dumps({
+ 'type': 'answer',
+ 'sdp': sdp_answer
+ })
+ })
+ await self.websocket.send(sdp_answer_msg)
+
+ elif self.events.sdp_info.qsize() > 0:
+ sdp_mids, user_fragments = self.events.sdp_info.get_nowait()
+
+ elif self.events.generated_ice_candidates.qsize() > 0 \
+ and sdp_mids is not None and user_fragments is not None:
+ mlineindex, candidate = self.events.generated_ice_candidates.get_nowait()
+ sdp_mid = sdp_mids[mlineindex]
+ user_fragment = user_fragments[mlineindex]
+ icemsg_value = json.dumps({
+ "candidate": candidate,
+ "sdpMid": sdp_mid,
+ "sdpMLineIndex": mlineindex,
+ "usernameFragment": user_fragment,
+ })
+ icemsg = json.dumps({
+ 'SessionID': self.session_id,
+ 'Type': 'addCalleeIceCandidate',
+ 'Value': icemsg_value,
+ })
+ log.info(f'Send ICE candidate')
+ log.debug(f'ICE candidate: {icemsg_value}')
+ await self.websocket.send(icemsg)
+
+ else:
+ await asyncio.sleep(0.2)
+
+ async def run(self):
+ self.session_id = None
+ async with websockets.connect(self.uri, ssl=self.ssl_context, close_timeout=0.5) as self.websocket:
+ receive_task = asyncio.Task(self.receive(self.uri))
+ send_task = asyncio.Task(self.send())
+ done, pending = await asyncio.wait([receive_task, send_task], return_when=asyncio.FIRST_COMPLETED)
+ for task in pending:
+ task.cancel()
+
+
+class WebRTCClient:
+ def __init__(self, events: Events, rtmp_uri: str):
+ self.events = events
+ self.rtmp_uri = rtmp_uri
+ self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
+ self.pipe = Gst.Pipeline.new("pipeline")
+ Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
+ self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
+ self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
+ self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
def on_negotiation_needed(self, element):
log.info('on_negotiation_needed')
def on_ice_candidate(self, element, mlineindex, candidate):
log.info('on_ice_candidate')
- self.generated_ice_candidates.append((mlineindex, candidate))
+ self.events.generated_ice_candidates.put_nowait((mlineindex, candidate))
def webrtcbin_pad_added(self, element, pad):
log.info('webrtcbin_pad_added')
if not pad.has_current_caps():
log.info(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)
-
- async def listen_to_gstreamer_bus(self):
- Gst.init(None)
- self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
- self.pipe = Gst.Pipeline.new("pipeline")
- Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
+ padsize = caps.get_size()
+
+ for i in range(padsize):
+ s = caps.get_structure(i) # Gst.Structure
+ name = s.get_name()
+ if name.startswith('video'):
+ q = Gst.ElementFactory.make('queue')
+ conv = Gst.ElementFactory.make('videoconvert')
+ enc = Gst.ElementFactory.make('x264enc')
+ enc.set_property('bitrate', 1000)
+ enc.set_property('tune', 'zerolatency')
+ capsfilter = Gst.ElementFactory.make('capsfilter')
+ capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
+ flvmux = Gst.ElementFactory.make('flvmux')
+ flvmux.set_property('streamable', True)
+ sink = Gst.ElementFactory.make('rtmpsink')
+ sink.set_property('location', self.rtmp_uri)
+ assert q and conv and enc and capsfilter and flvmux and sink
+
+ self.pipe.add(q)
+ self.pipe.add(conv)
+ self.pipe.add(enc)
+ self.pipe.add(capsfilter)
+ self.pipe.add(flvmux)
+ self.pipe.add(sink)
+
+ q_pad_sink = q.get_static_pad('sink')
+ assert q_pad_sink
+ pad_link_return = pad.link(q_pad_sink)
+ assert pad_link_return == Gst.PadLinkReturn.OK
+
+ ok = q.link(conv)
+ assert ok
+ ok = conv.link(enc)
+ assert ok
+ ok = enc.link(capsfilter)
+ assert ok
+ ok = capsfilter.link(flvmux)
+ assert ok
+ ok = flvmux.link(sink)
+ assert ok
+ self.pipe.set_state(Gst.State.PLAYING)
+ self.pipe.sync_children_states()
+
+ 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)
+ self.pipe.add(conv)
+ self.pipe.add(resample)
+ self.pipe.add(sink)
+ self.pipe.sync_children_states()
+ pad.link(q.get_static_pad('sink'))
+ q.link(conv)
+ conv.link(resample)
+ resample.link(sink)
+
+ def set_remote_desciption_done(self, gst_promise):
+ gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done)
+ self.webrtcbin.emit('create-answer', None, gst_promise)
+
+ def create_answer_done(self, gst_promise):
+ reply = gst_promise.get_reply()
+ answer = reply.get_value('answer')
+ gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
+ self.webrtcbin.emit('set-local-description', answer, gst_promise)
+
+ sdp_message = answer.sdp
+ mids = [sdp_message.get_media(i).get_attribute_val('mid')
+ for i in range(sdp_message.medias_len())]
+ user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
+ for i in range(sdp_message.medias_len())]
+ sdp_answer = sdp_message.as_text()
+ self.mids_uf = mids, user_fragments
+ self.answer = sdp_answer
+
+ def set_local_description_done(self, gst_promise):
+ gst_promise.get_reply()
+
+ sdp_answer = self.answer
+ log.info(f'Send SDP answer')
+ log.debug(f'SDP answer:\n{sdp_answer}')
+ self.events.sdp_answer.put_nowait(sdp_answer)
+ mids, user_fragments = self.mids_uf
+ self.events.sdp_info.put_nowait((mids, user_fragments))
+
+ async def run(self):
bus = Gst.Pipeline.get_bus(self.pipe)
- self.webrtcbin.connect('on-negotiation-needed', self.on_negotiation_needed)
- self.webrtcbin.connect('on-ice-candidate', self.on_ice_candidate)
- self.webrtcbin.connect('pad-added', self.webrtcbin_pad_added)
self.pipe.set_state(Gst.State.PLAYING)
try:
while True:
if bus.have_pending():
- msg = bus.pop() # Gst.Message, has to be unref'ed.
- if msg.type != Gst.MessageType.STATE_CHANGED:
- # log.info(f'Receive Gst.Message: {msg.type}, {msg.seqnum}, {msg.get_structure()}')
- # log.info(f'{webrtcbin.props.signaling_state} {webrtcbin.props.ice_gathering_state} {webrtcbin.props.ice_connection_state}')
- # Gst.Message.unref(msg)
- pass
- elif self.sdp_offer is not None:
- res, sm = GstSdp.SDPMessage.new()
+ msg = bus.pop()
+ if msg.type == Gst.MessageType.ERROR:
+ log.error(f'Error from gstreamer message bus: {msg.get_structure()}')
+ return
+ elif msg.type == Gst.MessageType.EOS: # end of stream
+ log.info(f'Gstreamer message bus reports end of stream')
+ return
+ elif self.events.sdp_offer.qsize() > 0:
+ sdp_offer = self.events.sdp_offer.get_nowait()
+ res, sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
assert res == GstSdp.SDPResult.OK
- GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
- # the three lines above can also be done this way in new versions of GStreamer:
- # sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
rd = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sm)
- gst_promise = Gst.Promise.new()
+ gst_promise = Gst.Promise.new_with_change_func(self.set_remote_desciption_done)
self.webrtcbin.emit('set-remote-description', rd, gst_promise)
- gst_promise.wait()
- self.sdp_offer = None
-
- log.info('create-answer')
- gst_promise = Gst.Promise.new()
- self.webrtcbin.emit('create-answer', None, gst_promise)
- result = gst_promise.wait()
- assert result == Gst.PromiseResult.REPLIED
- reply = gst_promise.get_reply()
- answer = reply.get_value('answer')
- sdp_message = answer.sdp
- self.user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
- for i in range(sdp_message.medias_len())]
- self.mids = [sdp_message.get_media(i).get_attribute_val('mid')
- for i in range(sdp_message.medias_len())]
- sdp_answer = sdp_message.as_text()
- log.info(sdp_answer)
- sdp_answer_msg = json.dumps({
- 'SessionID': self.session_id,
- 'Type': "gotAnswer",
- 'Value': json.dumps({
- 'type': 'answer',
- 'sdp': sdp_answer
- })
- })
- gst_promise = Gst.Promise.new()
- self.webrtcbin.emit('set-local-description', answer, gst_promise)
- gst_promise.wait()
- gst_promise.get_reply()
- await self.websocket.send(sdp_answer_msg)
-
- elif len(self.received_ice_candidates) > 0:
- ic = self.received_ice_candidates.pop(0)
+
+ elif self.events.received_ice_candidates.qsize() > 0:
+ ic = self.events.received_ice_candidates.get_nowait()
if ic['candidate'] != '':
self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
- elif len(self.generated_ice_candidates) > 0:
- mlineindex, candidate = self.generated_ice_candidates.pop(0)
- icemsg = json.dumps({
- 'SessionID': self.session_id,
- 'Type': 'addCalleeIceCandidate',
- 'Value': json.dumps({
- "candidate": candidate,
- "sdpMid": self.mids[mlineindex],
- "sdpMLineIndex": mlineindex,
- "usernameFragment": self.user_fragments[mlineindex],
- })
- })
- log.info(f'send_ice_candidate_message with {icemsg}')
- await self.websocket.send(icemsg)
+ elif self.events.room_left.qsize() > 0:
+ self.events.room_left.get_nowait()
+ return
else:
await asyncio.sleep(0.1)
finally:
self.pipe.set_state(Gst.State.NULL)
- async def talk_to_websocket(self, uri):
- ssl_context = ssl.SSLContext()
- ssl_context.check_hostname = False
- ssl_context.verify_mode = ssl.CERT_NONE
- async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
- async for msg in self.websocket:
- msg_json = json.loads(msg)
- msg_type = msg_json['Type']
- msg_value = msg_json['Value']
- self.session_id = msg_json['SessionID']
- log.info(f"receive for session {self.session_id} type {msg_type}")
- if msg_type == 'newSession':
- pass
- elif msg_type == 'gotOffer':
- value_json = json.loads(msg_value)
- sdp = value_json['sdp']
- log.info(f'SDP: {sdp}')
- self.sdp_offer = sdp
- elif msg_type == 'addCallerIceCandidate':
- value_json = json.loads(msg_value)
- log.info(f'ICE: {value_json}')
- self.received_ice_candidates.append(value_json)
- elif msg_type == 'roomClosed':
- log.info('Oh noes, the room went away!')
- # and here we should clean up
- else:
- log.error(f'Unknown message type {msg_type}')
- async def run(self, uri):
- talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
- listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
- done, pending = await asyncio.wait(
- [talk_to_websocket_task, listen_to_gstreamer_bus_task],
- return_when=asyncio.FIRST_COMPLETED)
- for d in done:
- d.result()
- for p in pending:
- p.cancel()
+async def run_room(laplace_uri: str, rtmp_uri: str):
+ try:
+ events = Events()
+ webrtc = WebRTCClient(events, rtmp_uri)
+ signaling = SignalingClient(events, laplace_uri)
+
+ webrtc_task = asyncio.Task(webrtc.run())
+ signaling_task = asyncio.Task(signaling.run())
+
+ done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED)
+
+ for task in done:
+ task.result()
+ for task in pending:
+ task.cancel()
+ except OSError as e:
+ print(e)
+
+
+async def run_room_repeated(laplace_uri: str, rtmp_uri: str, sleep_time: float):
+ while True:
+ await run_room(laplace_uri, rtmp_uri)
+ await asyncio.sleep(sleep_time)
+
+
+async def run_rooms(laplace_base_uri: str, rtmp_base_uri: str, rooms: List[str], retry: bool):
+ tasks = []
+ for room in rooms:
+ laplace_uri = laplace_base_uri + room # TODO: encode
+ rtmp_uri = rtmp_base_uri + room # TODO: encode
+ if retry:
+ tasks.append(run_room_repeated(laplace_uri, rtmp_uri, 2.))
+ else:
+ tasks.append(run_room(laplace_uri, rtmp_uri))
+ await asyncio.gather(*tasks)
def main():
- logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
+ logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
+ default_source = 'wss://localhost:1234/ws_connect?id='
+ default_dest = 'rtmp://localhost:1935/'
+ default_room = 'cug'
parser = argparse.ArgumentParser()
- parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
- help='Signalling server URI')
+ parser.add_argument('-s', '--source', default=default_source,
+ help=f'Laplace signalling websocket base URI, default: {default_source}')
+ parser.add_argument('-d', '--destination', default=default_dest,
+ help=f'RTMP server base URI, default: {default_dest}')
+ parser.add_argument('-r', '--retry', action='store_true', help=f'Retry forever if room not found or closed')
+ parser.add_argument('room', nargs='*', help=f'Room names to be used, "{default_room}" if omitted')
args = parser.parse_args()
- lagarde = Lagarde()
- asyncio.run(lagarde.run(args.uri), debug=True)
+
+ Gst.init(None)
+ rooms = args.room
+ if len(rooms) == 0:
+ rooms = [default_room]
+ asyncio.run(run_rooms(args.source, args.destination, rooms, args.retry))
if __name__ == '__main__':
ToastFreeware / toast / stream2beamer.git / blobdiff