duplicate the webrtc content_browser tests and browser tests with --enable-audio-track-processing on

chrome/browser/media/chrome_webrtc_browsertest.cc

 // Copyright 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/memory/scoped_ptr.h"
 #include "base/path_service.h"
 #include "base/process/launch.h"
@@ skipping 20 matching lines @@
 #include "media/base/media_switches.h"
 #include "net/test/embedded_test_server/embedded_test_server.h"
 #include "testing/perf/perf_test.h"
 
 static const char kMainWebrtcTestHtmlPage[] =
     "/webrtc/webrtc_jsep01_test.html";
 
 // Top-level integration test for WebRTC. The test methods here must run
 // sequentially since they use a server binary on the system (hence they are
 // tagged as MANUAL).
-class WebRtcBrowserTest : public WebRtcTestBase {
+class WebRtcBrowserTest : public WebRtcTestBase,
+                          public testing::WithParamInterface<bool> {
  public:
+  WebRtcBrowserTest() {}
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
     PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
     DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
   }
 
   virtual void SetUpCommandLine(CommandLine* command_line) OVERRIDE {
     // Ensure the infobar is enabled, since we expect that in this test.
     EXPECT_FALSE(command_line->HasSwitch(switches::kUseFakeUIForMediaStream));
 
     // TODO(phoglund): allow this test to also run with real devices once we
     // get real webcam bots up.
     command_line->AppendSwitch(switches::kUseFakeDeviceForMediaStream);
 
     // Flag used by TestWebAudioMediaStream to force garbage collection.
     command_line->AppendSwitchASCII(switches::kJavaScriptFlags, "--expose-gc");
+
+    bool enable_audio_track_processing = GetParam();
+    if (enable_audio_track_processing)
+      command_line->AppendSwitch(switches::kEnableAudioTrackProcessing);
   }
 
   void PrintProcessMetrics(base::ProcessMetrics* process_metrics,
                            const std::string& suffix) {
     perf_test::PrintResult("cpu", "", "cpu" + suffix,
                            process_metrics->GetCPUUsage(),
                            "%", true);
     perf_test::PrintResult("memory", "", "ws_peak" + suffix,
                            process_metrics->GetPeakWorkingSetSize(),
                            "bytes", true);
@@ skipping 35 matching lines @@
     const base::DictionaryValue* result;
     if (!iterator.IsAtEnd() && iterator.value().GetAsDictionary(&result))
       return result;
 
     return NULL;
   }
 
   PeerConnectionServerRunner peerconnection_server_;
 };
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+static const bool kRunTestsWithFlag[] = { false, true };
+INSTANTIATE_TEST_CASE_P(WebRtcBrowserTests,
+                        WebRtcBrowserTest,
+                        testing::ValuesIn(kRunTestsWithFlag));
+
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        MANUAL_RunsAudioVideoWebRTCCallInTwoTabs) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   ASSERT_TRUE(peerconnection_server_.Start());
 
   content::WebContents* left_tab =
       OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
   content::WebContents* right_tab =
       OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
 
   EstablishCall(left_tab, right_tab);
 
   StartDetectingVideo(left_tab, "remote-view");
   StartDetectingVideo(right_tab, "remote-view");
 
   WaitForVideoToPlay(left_tab);
   WaitForVideoToPlay(right_tab);
 
   HangUp(left_tab);
   WaitUntilHangupVerified(left_tab);
   WaitUntilHangupVerified(right_tab);
 
   ASSERT_TRUE(peerconnection_server_.Stop());
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MANUAL_CpuUsage15Seconds) {
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CpuUsage15Seconds) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   ASSERT_TRUE(peerconnection_server_.Start());
 
   base::FilePath results_file;
   ASSERT_TRUE(base::CreateTemporaryFile(&results_file));
 
   content::WebContents* left_tab =
       OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
 
 #if defined(OS_MACOSX)
@@ skipping 30 matching lines @@
 
 #if !defined(OS_MACOSX)
   PrintProcessMetrics(renderer_process_metrics.get(), "_r");
 #endif
   PrintProcessMetrics(browser_process_metrics.get(), "_b");
 
   ASSERT_TRUE(peerconnection_server_.Stop());
 }
 
 // This is manual for its long execution time.
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        MANUAL_RunsAudioVideoCall60SecsAndLogsInternalMetrics) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   ASSERT_TRUE(peerconnection_server_.Start());
 
   ASSERT_GE(TestTimeouts::action_max_timeout().InSeconds(), 100) <<
       "This is a long-running test; you must specify "
       "--ui-test-action-max-timeout to have a value of at least 100000.";
 
   content::WebContents* left_tab =
       OpenTestPageAndGetUserMediaInNewTab(kMainWebrtcTestHtmlPage);
@@ skipping 29 matching lines @@
   PrintBweForVideoMetrics(*first_pc_dict);
   PrintMetricsForAllStreams(*first_pc_dict);
 
   HangUp(left_tab);
   WaitUntilHangupVerified(left_tab);
   WaitUntilHangupVerified(right_tab);
 
   ASSERT_TRUE(peerconnection_server_.Stop());
 }
 
-IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, TestWebAudioMediaStream) {
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, TestWebAudioMediaStream) {
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   GURL url(embedded_test_server()->GetURL("/webrtc/webaudio_crash.html"));
   ui_test_utils::NavigateToURL(browser(), url);
   content::WebContents* tab =
       browser()->tab_strip_model()->GetActiveWebContents();
 
   // A sleep is necessary to be able to detect the crash.
   SleepInJavascript(tab, 1000);
 
   ASSERT_FALSE(tab->IsCrashed());
 }