Cleaned up WebRTC browser test js, removed unneeded stuff.

chrome/test/data/webrtc/webrtc_audio_quality_test.html

Index: chrome/test/data/webrtc/webrtc_audio_quality_test.html
diff --git a/chrome/test/data/webrtc/webrtc_audio_quality_test.html b/chrome/test/data/webrtc/webrtc_audio_quality_test.html
index db849fc9403f849613085cf349a0c4fbb1ecee87..f281afc2416386913272a1c6a52dbd68b3c0de78 100644
--- a/chrome/test/data/webrtc/webrtc_audio_quality_test.html
+++ b/chrome/test/data/webrtc/webrtc_audio_quality_test.html
@@ -4,9 +4,8 @@
   <title>WebRTC Automated Test</title>
   <script type="text/javascript" src="adapter.js"></script>
   <script type="text/javascript" src="test_functions.js"></script>
-  <script type="text/javascript" src="message_handling.js"></script>
+  <script type="text/javascript" src="peerconnection.js"></script>
   <script type="text/javascript" src="getusermedia.js"></script>
-  <script type="text/javascript" src="jsep01_call.js"></script>
   <script type="text/javascript" src="video_detector.js"></script>
   <script type="text/javascript" src="webaudio.js"></script>
 </head>