Refactored how WebRTC content browser tests talk to their javascript.

content/browser/media/webrtc_browsertest.cc

 // Copyright (c) 2012 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/strings/stringprintf.h"
 #include "base/values.h"
 #include "content/browser/media/webrtc_internals.h"
 #include "content/browser/web_contents/web_contents_impl.h"
 #include "content/public/common/content_switches.h"
@@ skipping 16 matching lines @@
   WebRtcBrowserTest() {}
   virtual ~WebRtcBrowserTest() {}
 
   // Convenience function since most peerconnection-call.html tests just load
   // the page, kick off some javascript and wait for the title to change to OK.
   void MakeTypicalPeerConnectionCall(const std::string& javascript) {
     ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
 
     GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
     NavigateToURL(shell(), url);
-    ExecuteTestAndWaitForOk(javascript);
+
+    DisableOpusIfOnAndroid();
+    ExecuteJavascriptAndWaitForOk(javascript);
   }
 
-  void ExecuteTestAndWaitForOk(const std::string& javascript) {
+  void DisableOpusIfOnAndroid() {
 #if defined (OS_ANDROID)
     // Always force iSAC 16K on Android for now (Opus is broken).
-    ASSERT_TRUE(ExecuteJavascript("forceIsac16KInSdp();"));
+    EXPECT_EQ("isac-forced",
+              ExecuteJavascriptAndReturnResult("forceIsac16KInSdp();"));
 #endif
-
-    ASSERT_TRUE(ExecuteJavascript(javascript));
-    ExpectTitle("OK");
   }
 };
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
 // Timing out on ARM linux bot: http://crbug.com/238490
 #define MAYBE_CanSetupVideoCall DISABLED_CanSetupVideoCall
 #else
 #define MAYBE_CanSetupVideoCall CanSetupVideoCall
 #endif
 
@@ skipping 266 matching lines @@
 
   base::FilePath dump_file;
   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 
   // This fakes the behavior of another open tab with webrtc-internals, and
   // enabling AEC dump in that tab.
   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
-  ExecuteTestAndWaitForOk("call({video: true, audio: true});");
+  DisableOpusIfOnAndroid();
+  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);
 
   base::DeleteFile(dump_file, false);
 }
 
 #if defined(OS_LINUX) && !defined(OS_CHROMEOS) && defined(ARCH_CPU_ARM_FAMILY)
@@ skipping 15 matching lines @@
   base::FilePath dump_file;
   ASSERT_TRUE(CreateTemporaryFile(&dump_file));
 
   // This fakes the behavior of another open tab with webrtc-internals, and
   // enabling AEC dump in that tab, then disabling it.
   WebRTCInternals::GetInstance()->FileSelected(dump_file, -1, NULL);
   WebRTCInternals::GetInstance()->DisableAecDump();
 
   GURL url(embedded_test_server()->GetURL("/media/peerconnection-call.html"));
   NavigateToURL(shell(), url);
-  ExecuteTestAndWaitForOk("call({video: true, audio: true});");
+  DisableOpusIfOnAndroid();
+  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_EQ(0, file_size);
 
   base::DeleteFile(dump_file, false);
 }
 
 }  // namespace content