Allow list of rooms.

[toast/stream2beamer.git] / lagarde.py

diff --git a/lagarde.py b/lagarde.py
index e9fca23f68b9ee02c66211575c3b662f2c31ffa6..94ea57e64ca76442a8516df8b8984b4ee40beb65 100755 (executable)
--- a/lagarde.py
+++ b/lagarde.py
@@ -5,7 +5,7 @@ import json
 import logging
 import ssl
 import queue
 import logging
 import ssl
 import queue

-from typing import Optional, List
+from typing import List

 
 import gi
 import websockets
 
 import gi
 import websockets


@@ -19,31 +19,9 @@ from gi.repository import GstWebRTC
 gi.require_version('GstSdp', '1.0')
 from gi.repository import GstSdp
 
 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__)
 
 
 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 Events:
     def __init__(self):
         self.sdp_offer = queue.Queue()


@@ -146,8 +124,9 @@ class SignalingClient:
 
 
 class WebRTCClient:
 
 
 class WebRTCClient:

-    def __init__(self, events: Events):
+    def __init__(self, events: Events, rtmp_uri: str):

         self.events = events
         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 = Gst.ElementFactory.make('webrtcbin', 'laplace')
         self.pipe = Gst.Pipeline.new("pipeline")
         Gst.Bin.do_add_element(self.pipe, self.webrtcbin)


@@ -177,24 +156,64 @@ class WebRTCClient:
         if not pad.has_current_caps():
             log.info(pad, 'has no caps, ignoring')
             return
         if not pad.has_current_caps():
             log.info(pad, 'has no caps, ignoring')
             return

-

         caps = pad.get_current_caps()
         padsize = caps.get_size()
         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')
         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')

-                sink = Gst.ElementFactory.make('intervideosink')
+                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'))
+                flmux = Gst.ElementFactory.make('flvmux')
+                sink = Gst.ElementFactory.make('rtmpsink')
+                sink.set_property('location', self.rtmp_uri)
+                assert q and conv and enc and capsfilter and flmux 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(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(flmux)
+                assert ok
+                ok = flmux.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(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)
                 self.pipe.add(sink)
                 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)
+                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)
 
     def set_remote_desciption_done(self, gst_promise):
         gst_promise = Gst.Promise.new_with_change_func(self.create_answer_done)


@@ -205,9 +224,9 @@ class WebRTCClient:
         answer = reply.get_value('answer')
         sdp_message = answer.sdp
         mids = [sdp_message.get_media(i).get_attribute_val('mid')
         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())]
+                for i in range(sdp_message.medias_len())]

         user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')
         user_fragments = [sdp_message.get_media(i).get_attribute_val('ice-ufrag')

-                               for i in range(sdp_message.medias_len())]
+                          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')
         self.events.sdp_info.put_nowait((mids, user_fragments))
         sdp_answer = sdp_message.as_text()
         log.info(f'Send SDP answer')


@@ -258,38 +277,16 @@ class WebRTCClient:
             self.pipe.set_state(Gst.State.NULL)
 
 
             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()
     try:
         events = Events()

+        webrtc = WebRTCClient(events, rtmp_uri)
+        signaling = SignalingClient(events, laplace_uri)

 
 

-        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))
+        webrtc_task = asyncio.Task(webrtc.run())
+        signaling_task = asyncio.Task(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()
 
         for task in done:
             task.result()


@@ -299,15 +296,43 @@ async def run(uri):
         print(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():
 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 = 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)
     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__':
 
 
 if __name__ == '__main__':