Cleanup.

diff --git a/lagarde.py b/lagarde.py
index c33b912339ac13742de2383f6476b6127969c5c1..63782061dee0ed7288a2e482a6de7f6573ee07b2 100755 (executable)
--- a/lagarde.py
+++ b/lagarde.py
@@ -5,7 +5,6 @@ import json
 import logging
 import ssl
 import queue
 import logging
 import ssl
 import queue

-from typing import Optional, List

 
 import gi
 import websockets
 
 import gi
 import websockets


@@ -19,31 +18,9 @@ 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()


@@ -180,12 +157,9 @@ class WebRTCClient:
         caps = pad.get_current_caps()
         padsize = caps.get_size()
 
         caps = pad.get_current_caps()
         padsize = caps.get_size()
 

-        log.info(f'>>>> {padsize} {caps}')
-

         for i in range(padsize):
             s = caps.get_structure(i)  # Gst.Structure
             name = s.get_name()
         for i in range(padsize):
             s = caps.get_structure(i)  # Gst.Structure
             name = s.get_name()

-            log.info(f'###### {name}')

             if name.startswith('video'):
                 q = Gst.ElementFactory.make('queue')
                 conv = Gst.ElementFactory.make('videoconvert')
             if name.startswith('video'):
                 q = Gst.ElementFactory.make('queue')
                 conv = Gst.ElementFactory.make('videoconvert')


@@ -197,8 +171,6 @@ class WebRTCClient:
                 flmux = Gst.ElementFactory.make('flvmux')
                 sink = Gst.ElementFactory.make('rtmpsink')
                 sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa')
                 flmux = Gst.ElementFactory.make('flvmux')
                 sink = Gst.ElementFactory.make('rtmpsink')
                 sink.set_property('location', 'rtmp://192.168.1.46:1935/gregoa')

-                # sink.set_property('location', 'rtmp://bla:1936/gregoa')
-                print(sink.props.location, dir(sink.props))

                 assert q and conv and enc and capsfilter and flmux and sink
 
                 self.pipe.add(q)
                 assert q and conv and enc and capsfilter and flmux and sink
 
                 self.pipe.add(q)


@@ -213,9 +185,6 @@ class WebRTCClient:
                 pad_link_return = pad.link(q_pad_sink)
                 assert pad_link_return == Gst.PadLinkReturn.OK
 
                 pad_link_return = pad.link(q_pad_sink)
                 assert pad_link_return == Gst.PadLinkReturn.OK
 

-                # ok = element.link(q)
-                # assert ok
-

                 ok = q.link(conv)
                 assert ok
                 ok = conv.link(enc)
                 ok = q.link(conv)
                 assert ok
                 ok = conv.link(enc)


@@ -228,8 +197,6 @@ class WebRTCClient:
                 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()

-                #print(dir(Gst.DebugGraphDetails))
-                #Gst.debug_bin_to_dot_data(element, Gst.DebugGraphDetails.ALL)

 
             elif name.startswith('audio'):
                 q = Gst.ElementFactory.make('queue')
 
             elif name.startswith('audio'):
                 q = Gst.ElementFactory.make('queue')


@@ -317,15 +284,13 @@ async def run_repeated(task):
 async def run(uri):
     try:
         events = Events()
 async def run(uri):
     try:
         events = Events()

-        # rtsp = RtspServer()

         webrtc = WebRTCClient(events)
         signaling = SignalingClient(events, uri)
 
         webrtc_task = asyncio.Task(webrtc.run())
         signaling_task = asyncio.Task(signaling.run())
 
         webrtc = WebRTCClient(events)
         signaling = SignalingClient(events, uri)
 
         webrtc_task = asyncio.Task(webrtc.run())
         signaling_task = asyncio.Task(signaling.run())
 

-        done, pending = await asyncio.wait([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()