import logging
import ssl
import queue
-from typing import Optional, List
+from typing import List
import gi
import websockets
gi.require_version('GstSdp', '1.0')
from gi.repository import GstSdp
-gi.require_version('GstRtspServer', '1.0')
-from gi.repository import Gst, GstRtspServer, GObject, GLib
-
log = logging.getLogger(__name__)
-class RtspServer:
- def __init__(self):
- server = GstRtspServer.RTSPServer()
- server.set_address("::")
- server.set_service('8554') # port as string
- factory = GstRtspServer.RTSPMediaFactory()
- # factory.set_launch("intervideosrc ! decodebin ! theoraenc ! queue ! rtptheorapay name=pay0")
- # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! theoraenc ! queue ! rtptheorapay name=pay0")
- # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,format=I420,framerate=10/1 ! theoraenc ! queue ! rtptheorapay name=pay0")
- # factory.set_launch("intervideosrc ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
- factory.set_launch("intervideosrc ! decodebin ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
- # factory.set_launch("intervideosrc ! decodebin ! videorate ! videoconvert ! video/x-raw,framerate=10/1 ! x264enc ! queue ! rtph264pay pt=96 name=pay0")
- factory.set_shared(True)
- mountPoints = server.get_mount_points()
- mountPoints.add_factory("/cug", factory)
- server.attach()
- self.server = server
-
-
class Events:
def __init__(self):
self.sdp_offer = queue.Queue()
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)
return
caps = pad.get_current_caps()
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.props.location = 'rtmp://127.0.0.1:1935/cug'
- # sink.props.location = 'rtmp://127.0.0.1:1936/cug'
+ 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()
- pad.link(q.get_static_pad('sink'))
- q.link(conv)
- conv.link(sink)
- # self.pipe.set_state(Gst.State.PLAYING)
- print(dir(Gst.DebugGraphDetails))
- Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL)
elif name.startswith('audio'):
q = Gst.ElementFactory.make('queue')
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 gstreamer_main_loop():
- """Does the equivalent of the following lines in an async friendly way:
- loop = GLib.MainLoop()
- loop.run()
- """
- gst_loop = GLib.MainLoop()
- context = gst_loop.get_context()
- while True:
- events_dispatched = context.iteration(False)
- await asyncio.sleep(0. if events_dispatched else 0.01)
-
-
-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, rtmp_uri)
+ signaling = SignalingClient(events, laplace_uri)
- # rtsp = RtspServer()
- webrtc = WebRTCClient(events)
- signaling = SignalingClient(events, uri)
+ webrtc_task = asyncio.Task(webrtc.run())
+ signaling_task = asyncio.Task(signaling.run())
- main_loop_task = asyncio.Task(gstreamer_main_loop())
- webrtc_task = asyncio.Task(run_repeated(webrtc.run))
- signaling_task = asyncio.Task(run_repeated(signaling.run))
-
- done, pending = await asyncio.wait([main_loop_task, webrtc_task, signaling_task],
- return_when=asyncio.FIRST_COMPLETED)
+ done, pending = await asyncio.wait([webrtc_task, signaling_task], return_when=asyncio.FIRST_COMPLETED)
for task in done:
task.result()
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