Refactored how WebRTC content browser tests talk to their javascript.

content/browser/media/webrtc_browsertest.cc

Index: content/browser/media/webrtc_browsertest.cc
diff --git a/content/browser/media/webrtc_browsertest.cc b/content/browser/media/webrtc_browsertest.cc
index 8ecc8b2fd22787620912a2895a93233bedd16253..4a8a4c0bcb7ef7072d800ca837c9220c43a20aab 100644
--- a/content/browser/media/webrtc_browsertest.cc
+++ b/content/browser/media/webrtc_browsertest.cc
@@ -34,17 +34,7 @@ class WebRtcBrowserTest : public WebRtcContentBrowserTest {
 
     GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
     NavigateToURL(shell(), url);
-    ExecuteTestAndWaitForOk(javascript);
-  }
-
-  void ExecuteTestAndWaitForOk(const std::string& javascript) {
-#if defined (OS_ANDROID)
-    // Always force iSAC 16K on Android for now (Opus is broken).
-    ASSERT_TRUE(ExecuteJavascript("forceIsac16KInSdp();"));
-#endif
-
-    ASSERT_TRUE(ExecuteJavascript(javascript));
-    ExpectTitle("OK");
+    ExecuteJavascriptAndWaitForOk(javascript);
   }
 };
 
@@ -331,7 +321,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CallWithAecDump) {
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
-  ExecuteTestAndWaitForOk("call({video: true, audio: true});");
+  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 
   EXPECT_TRUE(base::PathExists(dump_file));
   int64 file_size = 0;
@@ -367,7 +357,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
-  ExecuteTestAndWaitForOk("call({video: true, audio: true});");
+  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 
   EXPECT_TRUE(base::PathExists(dump_file));
   int64 file_size = 0;