Stuff audio buffer directly rather than signal/slot

audiohandler.cpp

@@ -81,7 +81,7 @@ bool audioHandler::init(audioSetup setupIn)
 		setup.bits = 16;
 	}
 
-	ringBuf = new wilt::Ring<audioPacket>(100); // Should be customizable.
+	ringBuf = new wilt::Ring<audioPacket>(8); // Should be customizable.
 
 	tempBuf.sent = 0;
 

udphandler.cpp

@@ -945,8 +945,8 @@ void udpAudio::dataReceived()
                     tempAudio.data = r.mid(0x18);
                     // Prefer signal/slot to forward audio as it is thread/safe
                     // Need to do more testing but latency appears fine.
-                    //audioLatency = rxaudio->incomingAudio(tempAudio);
-                    emit haveAudioData(tempAudio);
+                    rxaudio->incomingAudio(tempAudio);
+                    //emit haveAudioData(tempAudio);
                     audioLatency = rxaudio->getLatency();
                 }
                 break;

udpserver.cpp

@@ -653,7 +653,9 @@ void udpServer::audioReceived()
                     tempAudio.sent = 0;
                     tempAudio.data = r.mid(0x18);
                     //qInfo(logUdpServer()) << "sending tx audio " << in->seq;
-                    emit haveAudioData(tempAudio);
+                    //emit haveAudioData(tempAudio);
+                    txaudio->incomingAudio(tempAudio);
+
                 }
             }
             break;