import logging
import ssl
import queue
+from typing import List
import gi
import websockets
class WebRTCClient:
- def __init__(self, events: Events):
+ 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)
enc.set_property('tune', 'zerolatency')
capsfilter = Gst.ElementFactory.make('capsfilter')
capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
- flmux = Gst.ElementFactory.make('flvmux')
+ flvmux = Gst.ElementFactory.make('flvmux')
+ flvmux.set_property('streamable', True)
sink = Gst.ElementFactory.make('rtmpsink')
- sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa')
- assert q and conv and enc and capsfilter and flmux and sink
+ 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(flmux)
+ self.pipe.add(flvmux)
self.pipe.add(sink)
q_pad_sink = q.get_static_pad('sink')
assert ok
ok = enc.link(capsfilter)
assert ok
- ok = capsfilter.link(flmux)
+ ok = capsfilter.link(flvmux)
assert ok
- ok = flmux.link(sink)
+ ok = flvmux.link(sink)
assert ok
self.pipe.set_state(Gst.State.PLAYING)
self.pipe.sync_children_states()
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())]
+ 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())]
- self.events.sdp_info.put_nowait((mids, user_fragments))
+ for i in range(sdp_message.medias_len())]
sdp_answer = sdp_message.as_text()
- log.info(f'Send SDP answer')
- log.debug(f'SDP answer:\n{sdp_answer}')
- self.events.sdp_answer.put_nowait(sdp_answer)
- gst_promise = Gst.Promise.new_with_change_func(self.set_local_description_done)
- self.webrtcbin.emit('set-local-description', answer, gst_promise)
+ 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.pipe.set_state(Gst.State.PLAYING)
return
elif self.events.sdp_offer.qsize() > 0:
sdp_offer = self.events.sdp_offer.get_nowait()
- res, sm = GstSdp.SDPMessage.new()
+ res, sm = GstSdp.SDPMessage.new_from_text(sdp_offer)
assert res == GstSdp.SDPResult.OK
- GstSdp.sdp_message_parse_buffer(bytes(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_with_change_func(self.set_remote_desciption_done)
self.webrtcbin.emit('set-remote-description', rd, gst_promise)
self.pipe.set_state(Gst.State.NULL)
-async def run_repeated(task):
- while True:
- await task()
- await asyncio.sleep(0.1)
-
-
-async def run(uri):
+async def run_room(laplace_uri: str, rtmp_uri: str):
try:
events = Events()
- webrtc = WebRTCClient(events)
- signaling = SignalingClient(events, uri)
+ webrtc = WebRTCClient(events, rtmp_uri)
+ signaling = SignalingClient(events, laplace_uri)
webrtc_task = asyncio.Task(webrtc.run())
signaling_task = asyncio.Task(signaling.run())
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.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()
Gst.init(None)
- asyncio.run(run(args.uri), debug=True)
+ 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