split pipe

raspberry/roberto/camera/camera_gstreamer_webrtc.py

@@ -39,18 +39,31 @@ import json
 #    webrtcbin name=sendrecv
 #'''
 
+#PIPELINE_DESC = '''
+#webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
+# v4l2src device=/dev/video0 ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
+# queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
+# audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc !rtpopuspay !
+# queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
+#'''
+
 PIPELINE_DESC = '''
-webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
+tee name=audiotee ! queue ! fakesink
- v4l2src device=/dev/video0 ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
+ audiotestsrc is-live=true wave=red-noise ! queue ! opusenc ! rtpopuspay !
- queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
+ queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! audiotee.
- audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc !rtpopuspay !
- queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
 '''
 
+#PIPELINE_DESC = '''
+#tee name=audiotee ! queue ! fakesink
+# v4l2src device=/dev/video0 ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
+# queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! audiotee.
+#'''
+
 # rpicamsrc name=src preview=0 fullscreen=0 ! h264parse ! omxh264dec ! glimagesink sync=0
 
 class WebRTCCamera(Thread):
     def __init__(self):
+        self.pipe = None
         self.sid = None
         self._num_clients = 0
         self.connected = False
@@ -71,11 +84,16 @@ class WebRTCCamera(Thread):
         while True:
             item = self._queue.get()
             if item['job'] == "connect_client":
-                self.start_pipeline(item['sid'])
+                if not self.pipe:
+                    self.start_global_pipeline()
+                self.start_client_pipeline(item['sid'])
                 self._num_clients += 1
             elif item['job'] == "disconnect_client":
-                self.close_pipeline(item['sid'])
+                self.close_client_pipeline(item['sid'])
                 self._num_clients -= 1
+                if self._num_clients == 0:
+                    print("last client left. stopping global pipeline")
+                    self.stop_global_pipeline()
             else:
                 print("unknown job: %s" % item['job'])
             self._queue.task_done()
@@ -103,18 +121,42 @@ class WebRTCCamera(Thread):
     def disconnect_client(self, sid, room):
         self._queue.put({'job':'disconnect_client', 'sid':sid})
 
-    def start_pipeline(self, client_sid):
+    def start_global_pipeline(self):
         print("STARTING PIPELINE")
+        self.pipe = Gst.parse_launch(PIPELINE_DESC)
+        self.pipe.set_state(Gst.State.PLAYING)
+
+    def stop_global_pipeline(self):
+        self.pipe.set_state(Gst.State.NULL)
+        self.pipe = None
+
+    def start_client_pipeline(self, client_sid):
+        print("starting client pipeline")
         self._peers[client_sid] = {}
-        pipe = Gst.parse_launch(PIPELINE_DESC)
+        q = Gst.ElementFactory.make('queue')
-        self._peers[client_sid]['pipe'] = pipe
+        webrtc = Gst.ElementFactory.make('webrtcbin', client_sid)
-        webrtc = pipe.get_by_name('sendrecv')
         self._peers[client_sid]['webrtc'] = webrtc
+        self.pipe.add(q, webrtc)
+#        q.link(webrtc.get_request_pad('sink_%u'))
+        q.link(webrtc)
+        tee = self.pipe.get_by_name('audiotee')
+#        tee.link(q.get_static_pad('sink'))
+        tee.link(q)
+
+        self._peers[client_sid]['pipe'] = q
+        
         webrtc.connect('on-negotiation-needed', self.on_negotiation_needed, client_sid)
         webrtc.connect('on-ice-candidate', self.on_ice_candidate, client_sid)
-        pipe.set_state(Gst.State.PLAYING)
+        self.pipe.sync_children_states()
+        webrtc.sync_children_states()
+
+    def close_client_pipeline(self, client_sid):
+        webrtc = self._peers[client_sid]['webrtc']
+        self.pipe.remove(webrtc)
+        self.pipe.remove(self._peers[client_sid]['pipe'])
+        tee = self.pipe.get_by_name('audiotee')
+        #tee.release_request_pad(webrtc.get_request_pad('sink_%u'))
 
-    def close_pipeline(self, client_sid):
         self._peers[client_sid]['pipe'].set_state(Gst.State.NULL)
         self._peers[client_sid]['pipe'] = None
         self._peers[client_sid]['webrtc'] = None