Add visualization of concept.

[toast/stream2beamer.git] / lagarde.py

diff --git a/lagarde.py b/lagarde.py
index 15ba5159961a5477e8bac56a0067a3224faa936e..f31bc2dbbfdcfb75c7c97de24bcf2c287711cfef 100755 (executable)
--- a/lagarde.py
+++ b/lagarde.py
@@ -5,6 +5,7 @@ import json
 import logging
 import ssl
 import queue
 import logging
 import ssl
 import queue

+from typing import List

 
 import gi
 import websockets
 
 import gi
 import websockets


@@ -169,16 +170,17 @@ class WebRTCClient:
                 enc.set_property('tune', 'zerolatency')
                 capsfilter = Gst.ElementFactory.make('capsfilter')
                 capsfilter.set_properties(Gst.Caps.from_string('video/x-h264,stream-format=(string)avc'))
                 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', self.rtmp_uri)
                 sink = Gst.ElementFactory.make('rtmpsink')
                 sink.set_property('location', self.rtmp_uri)

-                assert q and conv and enc and capsfilter and flmux and sink
+                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(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')
                 self.pipe.add(sink)
 
                 q_pad_sink = q.get_static_pad('sink')


@@ -192,9 +194,9 @@ class WebRTCClient:
                 assert ok
                 ok = enc.link(capsfilter)
                 assert ok
                 assert ok
                 ok = enc.link(capsfilter)
                 assert ok

-                ok = capsfilter.link(flmux)
+                ok = capsfilter.link(flvmux)

                 assert ok
                 assert ok

-                ok = flmux.link(sink)
+                ok = flvmux.link(sink)

                 assert ok
                 self.pipe.set_state(Gst.State.PLAYING)
                 self.pipe.sync_children_states()
                 assert ok
                 self.pipe.set_state(Gst.State.PLAYING)
                 self.pipe.sync_children_states()


@@ -221,22 +223,28 @@ class WebRTCClient:
     def create_answer_done(self, gst_promise):
         reply = gst_promise.get_reply()
         answer = reply.get_value('answer')
     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_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()
         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()
 
 
     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)
     async def run(self):
         bus = Gst.Pipeline.get_bus(self.pipe)
         self.pipe.set_state(Gst.State.PLAYING)


@@ -252,11 +260,8 @@ class WebRTCClient:
                         return
                 elif self.events.sdp_offer.qsize() > 0:
                     sdp_offer = self.events.sdp_offer.get_nowait()
                         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
                     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)
                     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)


@@ -276,7 +281,7 @@ class WebRTCClient:
             self.pipe.set_state(Gst.State.NULL)
 
 
             self.pipe.set_state(Gst.State.NULL)
 
 

-async def run(laplace_uri: str, rtmp_uri: str):
+async def run_room(laplace_uri: str, rtmp_uri: str):

     try:
         events = Events()
         webrtc = WebRTCClient(events, rtmp_uri)
     try:
         events = Events()
         webrtc = WebRTCClient(events, rtmp_uri)


@@ -295,30 +300,43 @@ async def run(laplace_uri: str, rtmp_uri: str):
         print(e)
 
 
         print(e)
 
 

-async def run_repeated(laplace_uri: str, rtmp_uri: str, sleep_time: float):
+async def run_room_repeated(laplace_uri: str, rtmp_uri: str, sleep_time: float):

     while True:
     while True:

-        await run(laplace_uri, rtmp_uri)
+        await run_room(laplace_uri, rtmp_uri)

         await asyncio.sleep(sleep_time)
 
 
         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')
 def main():
     logging.basicConfig(level=logging.DEBUG, format='%(asctime)-15s %(message)s')

-    default_source = 'wss://localhost:1234/ws_connect?id=cug'
-    default_dest = 'rtmp://localhost:1935/cug'
+    default_source = 'wss://localhost:1234/ws_connect?id='
+    default_dest = 'rtmp://localhost:1935/'
+    default_room = 'cug'

     parser = argparse.ArgumentParser()
     parser.add_argument('-s', '--source', default=default_source,
     parser = argparse.ArgumentParser()
     parser.add_argument('-s', '--source', default=default_source,

-                        help=f'Laplace signalling websocket URI, default: {default_source}')
+                        help=f'Laplace signalling websocket base URI, default: {default_source}')

     parser.add_argument('-d', '--destination', default=default_dest,
     parser.add_argument('-d', '--destination', default=default_dest,

-                        help=f'RTMP server URI, 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('-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)
     args = parser.parse_args()
 
     Gst.init(None)

-    if args.retry:
-        job = run_repeated(args.source, args.destination, 2.)
-    else:
-        job = run(args.source, args.destination)
-    asyncio.run(job)
+    rooms = args.room
+    if len(rooms) == 0:
+        rooms = [default_room]
+    asyncio.run(run_rooms(args.source, args.destination, rooms, args.retry))

 
 
 if __name__ == '__main__':
 
 
 if __name__ == '__main__':