Work in progress...
[toast/stream2beamer.git] / lagarde.py
index 9cb9945e96c388c25c6e9341827ac5c5de03c29e..05b2fe008489ff19077277e06be98f9cadebd70b 100755 (executable)
@@ -1,5 +1,4 @@
 #!/usr/bin/python3
 #!/usr/bin/python3
-
 import argparse
 import asyncio
 import json
 import argparse
 import asyncio
 import json
@@ -26,13 +25,18 @@ from gi.repository import Gst, GstRtspServer, GObject, GLib
 log = logging.getLogger(__name__)
 
 
 log = logging.getLogger(__name__)
 
 
-class GstreamerRtspServer():
+class RtspServer:
     def __init__(self):
         server = GstRtspServer.RTSPServer()
         server.set_address("::")
         server.set_service('8554')  # port as string
         factory = GstRtspServer.RTSPMediaFactory()
     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 ! 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)
         factory.set_shared(True)
         mountPoints = server.get_mount_points()
         mountPoints.add_factory("/cug", factory)
@@ -40,27 +44,126 @@ class GstreamerRtspServer():
         self.server = server
 
 
         self.server = server
 
 
-class Lagarde:
+class Events:
     def __init__(self):
     def __init__(self):
-        self.sdp_offer: Optional[str] = None
-        self.websocket: Optional[websockets.client.WebSocketClientProtocol] = None
-        self.session_id = None
-        self.received_ice_candidates = queue.Queue()
+        self.sdp_offer = queue.Queue()
+        self.sdp_answer = queue.Queue()
         self.generated_ice_candidates = queue.Queue()
         self.generated_ice_candidates = queue.Queue()
-        self.user_fragments: Optional[List] = None
-        self.mids: Optional[List] = None
-        self.pipe = None
-        self.webrtcbin = None
+        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
+
+    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):
+        self.events = events
+        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):
 
     def on_negotiation_needed(self, element):
-        log.debug('on_negotiation_needed')
+        log.info('on_negotiation_needed')
 
     def on_ice_candidate(self, element, mlineindex, candidate):
 
     def on_ice_candidate(self, element, mlineindex, candidate):
-        log.debug('on_ice_candidate')
-        self.generated_ice_candidates.put_nowait((mlineindex, candidate))
+        log.info('on_ice_candidate')
+        self.events.generated_ice_candidates.put_nowait((mlineindex, candidate))
 
     def webrtcbin_pad_added(self, element, pad):
 
     def webrtcbin_pad_added(self, element, pad):
-        log.debug('webrtcbin_pad_added')
+        log.info('webrtcbin_pad_added')
         if pad.direction != Gst.PadDirection.SRC:
             return
         decodebin = Gst.ElementFactory.make('decodebin')
         if pad.direction != Gst.PadDirection.SRC:
             return
         decodebin = Gst.ElementFactory.make('decodebin')
@@ -70,20 +173,21 @@ class Lagarde:
         self.webrtcbin.link(decodebin)
 
     def decodebin_pad_added(self, element, pad):
         self.webrtcbin.link(decodebin)
 
     def decodebin_pad_added(self, element, pad):
-        log.debug('decodebin_pad_added')
+        log.info('decodebin_pad_added')
         if not pad.has_current_caps():
         if not pad.has_current_caps():
-            log.debug(pad, 'has no caps, ignoring')
+            log.info(pad, 'has no caps, ignoring')
             return
             return
-
         caps = pad.get_current_caps()
         padsize = caps.get_size()
         for i in range(padsize):
         caps = pad.get_current_caps()
         padsize = caps.get_size()
         for i in range(padsize):
-            s = caps.get_structure(i) # Gst.Structure
+            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')
             name = s.get_name()
             if name.startswith('video'):
                 q = Gst.ElementFactory.make('queue')
                 conv = Gst.ElementFactory.make('videoconvert')
-                sink = Gst.ElementFactory.make('intervideosink')
+                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'
                 self.pipe.add(q)
                 self.pipe.add(conv)
                 self.pipe.add(sink)
                 self.pipe.add(q)
                 self.pipe.add(conv)
                 self.pipe.add(sink)
@@ -91,6 +195,10 @@ class Lagarde:
                 pad.link(q.get_static_pad('sink'))
                 q.link(conv)
                 conv.link(sink)
                 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')
                 conv = Gst.ElementFactory.make('audioconvert')
             elif name.startswith('audio'):
                 q = Gst.ElementFactory.make('queue')
                 conv = Gst.ElementFactory.make('audioconvert')
@@ -106,14 +214,31 @@ class Lagarde:
                 conv.link(resample)
                 resample.link(sink)
 
                 conv.link(resample)
                 resample.link(sink)
 
-    async def listen_to_gstreamer_bus(self):
-        self.webrtcbin = Gst.ElementFactory.make('webrtcbin', 'laplace')
-        self.pipe = Gst.Pipeline.new("pipeline")
-        Gst.Bin.do_add_element(self.pipe, self.webrtcbin)
+    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')
+        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())]
+        self.events.sdp_info.put_nowait((mids, user_fragments))
+        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)
+
+    def set_local_description_done(self, gst_promise):
+        gst_promise.get_reply()
+
+    async def run(self):
         bus = Gst.Pipeline.get_bus(self.pipe)
         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:
         self.pipe.set_state(Gst.State.PLAYING)
         try:
             while True:
@@ -125,118 +250,31 @@ class Lagarde:
                     elif msg.type == Gst.MessageType.EOS:  # end of stream
                         log.info(f'Gstreamer message bus reports end of stream')
                         return
                     elif msg.type == Gst.MessageType.EOS:  # end of stream
                         log.info(f'Gstreamer message bus reports end of stream')
                         return
-                elif self.sdp_offer is not None:
+                elif self.events.sdp_offer.qsize() > 0:
+                    sdp_offer = self.events.sdp_offer.get_nowait()
                     res, sm = GstSdp.SDPMessage.new()
                     assert res == GstSdp.SDPResult.OK
                     res, sm = GstSdp.SDPMessage.new()
                     assert res == GstSdp.SDPResult.OK
-                    GstSdp.sdp_message_parse_buffer(bytes(self.sdp_offer.encode()), sm)
+                    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)
                     # 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)
                     self.webrtcbin.emit('set-remote-description', rd, gst_promise)
-                    await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
-                    self.sdp_offer = None
-
-                    gst_promise = Gst.Promise.new()
-                    self.webrtcbin.emit('create-answer', None, gst_promise)
-                    result = await asyncio.get_event_loop().run_in_executor(None, 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(f'Send SDP answer:\n{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)
-                    await asyncio.get_event_loop().run_in_executor(None, gst_promise.wait)
-                    gst_promise.get_reply()
-                    await self.websocket.send(sdp_answer_msg)
-
-                elif self.received_ice_candidates.qsize() > 0:
-                    ic = self.received_ice_candidates.get_nowait()
+
+                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'])
 
                     if ic['candidate'] != '':
                         self.webrtcbin.emit('add-ice-candidate', ic['sdpMLineIndex'], ic['candidate'])
 
-                elif self.generated_ice_candidates.qsize() > 0:
-                    mlineindex, candidate = self.generated_ice_candidates.get_nowait()
-                    icemsg_value = json.dumps({
-                        "candidate": candidate,
-                        "sdpMid": self.mids[mlineindex],
-                        "sdpMLineIndex": mlineindex,
-                        "usernameFragment": self.user_fragments[mlineindex],
-                    })
-                    icemsg = json.dumps({
-                        'SessionID': self.session_id,
-                        'Type': 'addCalleeIceCandidate',
-                        'Value': icemsg_value,
-                    })
-                    log.info(f'Send ICE candidate: {icemsg_value}')
-                    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)
 
 
                 else:
                     await asyncio.sleep(0.1)
         finally:
             self.pipe.set_state(Gst.State.NULL)
 
-    async def talk_to_websocket(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:\n{sdp}')
-                self.sdp_offer = sdp
-            elif msg_type == 'addCallerIceCandidate':
-                value_json = json.loads(msg_value)
-                log.info(f'Got ICE candidate: {value_json}')
-                self.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.session_id = None
-                return
-            else:
-                log.error(f'Unknown message type {msg_type}')
-
-    async def run(self, uri):
-        ssl_context = ssl.SSLContext()
-        ssl_context.check_hostname = False
-        ssl_context.verify_mode = ssl.CERT_NONE
-        try:
-            async with websockets.connect(uri, ssl=ssl_context) as self.websocket:
-                talk_to_websocket_task = asyncio.Task(self.talk_to_websocket(uri))
-                listen_to_gstreamer_bus_task = asyncio.Task(self.listen_to_gstreamer_bus())
-                main_loop = asyncio.Task(gstreamer_main_loop())
-                done, pending = await asyncio.wait(
-                    [talk_to_websocket_task, listen_to_gstreamer_bus_task, main_loop],
-                    return_when=asyncio.FIRST_COMPLETED)
-                for d in done:
-                    d.result()
-                for p in pending:
-                    p.cancel()
-        except OSError as e:
-            print(e)
-
 
 async def gstreamer_main_loop():
     """Does the equivalent of the following lines in an async friendly way:
 
 async def gstreamer_main_loop():
     """Does the equivalent of the following lines in an async friendly way:
@@ -250,17 +288,44 @@ async def gstreamer_main_loop():
         await asyncio.sleep(0. if events_dispatched else 0.01)
 
 
         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):
+    try:
+        events = Events()
+
+        # rtsp = RtspServer()
+        webrtc = WebRTCClient(events)
+        signaling = SignalingClient(events, uri)
+
+        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)
+
+        for task in done:
+            task.result()
+        for task in pending:
+            task.cancel()
+    except OSError as e:
+        print(e)
+
+
 def main():
 def main():
-    logging.basicConfig(level=logging.INFO, format='%(asctime)-15s %(message)s')
+    logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')
     parser = argparse.ArgumentParser()
     parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
                         help='Signalling server URI')
     args = parser.parse_args()
 
     Gst.init(None)
     parser = argparse.ArgumentParser()
     parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
                         help='Signalling server URI')
     args = parser.parse_args()
 
     Gst.init(None)
-    rtsp = GstreamerRtspServer()
-    lagarde = Lagarde()
-    asyncio.run(lagarde.run(args.uri), debug=True)
+    asyncio.run(run(args.uri), debug=True)
 
 
 if __name__ == '__main__':
 
 
 if __name__ == '__main__':