Refactored how WebRTC content browser tests talk to their javascript.

content/browser/media/webrtc_aecdump_browsertest.cc

 // Copyright (c) 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/command_line.h"
 #include "base/file_util.h"
 #include "base/json/json_reader.h"
 #include "base/platform_file.h"
 #include "base/strings/stringprintf.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 49 matching lines @@
 // This tests will make a complete PeerConnection-based call, verify that
 // video is playing for the call, and verify that a non-empty AEC dump file
 // exists.
 IN_PROC_BROWSER_TEST_F(WebRtcAecDumpBrowserTest, MAYBE_CallWithAecDump) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
 
 #if defined (OS_ANDROID)
-    // Always force iSAC 16K on Android for now (Opus is broken).
-    ASSERT_TRUE(ExecuteJavascript("forceIsac16KInSdp();"));
+  // Always force iSAC 16K on Android for now (Opus is broken).
+  EXPECT_EQ("isac-forced",
+            ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();"));
 #endif
 
-  EXPECT_TRUE(ExecuteJavascript("call({video: true, audio: true});"));
-  ExpectTitle("OK");
+  ExecuteJavascriptAndWaitForOk("call({video: true, audio: true});");
 
   EXPECT_TRUE(base::PathExists(dump_file_));
   int64 file_size = 0;
   EXPECT_TRUE(base::GetFileSize(dump_file_, &file_size));
   EXPECT_GT(file_size, 0);
 }
 
 }  // namespace content