Renaming remaining WebRTC browser tests.

chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc

Index: chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
diff --git a/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc b/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
index fe70fab81c718f9410e37328fafd9a5796f74080..a3ddf07c639b6fe27bf7889058ba6963040b2fd5 100644
--- a/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
+++ b/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
@@ -89,7 +89,7 @@ static base::FilePath GetTestDataDir() {
 //    50 / 100 in level. Also go into the playback tab, right-click Speakers,
 //    and set that level to 50 / 100. Otherwise you will get distortion in
 //    the recording.
-class WebrtcAudioQualityBrowserTest : public WebRtcTestBase {
+class WebRtcAudioQualityBrowserTest : public WebRtcTestBase {
  public:
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
     PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
@@ -377,7 +377,7 @@ bool RunPesq(const base::FilePath& reference_file,
 #define MAYBE_MANUAL_TestAudioQuality DISABLED_MANUAL_TestAudioQuality
 #endif
 
-IN_PROC_BROWSER_TEST_F(WebrtcAudioQualityBrowserTest,
+IN_PROC_BROWSER_TEST_F(WebRtcAudioQualityBrowserTest,
                        MAYBE_MANUAL_TestAudioQuality) {
 #if defined(OS_WIN)
   if (base::win::GetVersion() < base::win::VERSION_VISTA) {