Relanding Switched main WebRTC browser tests to use a more realistic video

> Switched main WebRTC browser tests to use a more realistic video.

> After switching to fake device flags instead of software webcams, we
> noticed a ton of video-related metrics changed. This is because the
> spinning green ball is an unrealistic encode/decode example. This patch
> will use the new fake device file flag to play a more realistic video.

> BUG=352261, 343504
> Committed: https://src.chromium.org/viewvc/chrome?view=rev&revision=259887

TBR=tommi@chromium.org
BUG=352261, 343504

Review URL: https://codereview.chromium.org/216773002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@260213 0039d316-1c4b-4281-b951-d872f2087c98

chrome/browser/media/chrome_media_stream_infobar_browsertest.cc

@@ -119,9 +119,9 @@ IN_PROC_BROWSER_TEST_F(MediaStreamInfoBarTest,
 
   // Should fail with permission denied right away with no infobar popping up.
   GetUserMedia(tab_contents, kAudioVideoCallConstraints);
-  EXPECT_TRUE(PollingWaitUntil("obtainGetUserMediaResult()",
-                               kFailedWithPermissionDeniedError,
-                               tab_contents));
+  EXPECT_TRUE(test::PollingWaitUntil("obtainGetUserMediaResult()",
+                                     kFailedWithPermissionDeniedError,
+                                     tab_contents));
   InfoBarService* infobar_service =
       InfoBarService::FromWebContents(tab_contents);
   EXPECT_EQ(0u, infobar_service->infobar_count());

chrome/browser/media/chrome_webrtc_apprtc_browsertest.cc

@@ -119,7 +119,7 @@ class WebRtcApprtcBrowserTest : public WebRtcTestBase {
     // Apprtc will set remoteVideo.style.opacity to 1 when the call comes up.
     std::string javascript =
         "window.domAutomationController.send(remoteVideo.style.opacity)";
-    return PollingWaitUntil(javascript, "1", tab_contents);
+    return test::PollingWaitUntil(javascript, "1", tab_contents);
   }
 
   bool EvalInJavascriptFile(content::WebContents* tab_contents,

chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc

@@ -94,7 +94,7 @@ class WebRtcAudioQualityBrowserTest : public WebRtcTestBase,
  public:
   WebRtcAudioQualityBrowserTest() {}
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
-    PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
+    test::PeerConnectionServerRunner::KillAllPeerConnectionServers();
     DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
   }
 
@@ -137,10 +137,10 @@ class WebRtcAudioQualityBrowserTest : public WebRtcTestBase,
               ExecuteJavascript("negotiateCall()", from_tab));
 
     // Ensure the call gets up on both sides.
-    EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                                 "active", from_tab));
-    EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                                 "active", to_tab));
+    EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                       "active", from_tab));
+    EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                       "active", to_tab));
   }
 
   base::FilePath CreateTemporaryWaveFile() {
@@ -152,7 +152,7 @@ class WebRtcAudioQualityBrowserTest : public WebRtcTestBase,
     return wav_filename;
   }
 
-  PeerConnectionServerRunner peerconnection_server_;
+  test::PeerConnectionServerRunner peerconnection_server_;
 };
 
 class AudioRecorder {
@@ -419,7 +419,7 @@ IN_PROC_BROWSER_TEST_P(WebRtcAudioQualityBrowserTest,
   // because the ready state is ok on both sides. We sleep a bit between call
   // establishment and playing to avoid cutting of the beginning of the audio
   // file.
-  SleepInJavascript(left_tab, 2000);
+  test::SleepInJavascript(left_tab, 2000);
 
   base::FilePath recording = CreateTemporaryWaveFile();
 

chrome/browser/media/chrome_webrtc_browsertest.cc

@@ -32,18 +32,24 @@
 #include "net/test/embedded_test_server/embedded_test_server.h"
 #include "testing/perf/perf_test.h"
 
+// For fine-grained suppression.
+#if defined(OS_WIN)
+#include "base/win/windows_version.h"
+#endif
+
 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).
+// tagged as MANUAL). In addition, they need the reference videos which require
+// the webrtc.DEPS solution, which is not generally available on Chromium bots.
 class WebRtcBrowserTest : public WebRtcTestBase,
                           public testing::WithParamInterface<bool> {
  public:
   WebRtcBrowserTest() {}
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
-    PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
+    test::PeerConnectionServerRunner::KillAllPeerConnectionServers();
     DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
   }
 
@@ -51,8 +57,12 @@ class WebRtcBrowserTest : public WebRtcTestBase,
     // 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.
+    // Play a suitable, somewhat realistic video file.
+    base::FilePath input_video = test::GetReferenceVideosDir()
+        .Append(test::kReferenceFileName360p)
+        .AddExtension(test::kY4mFileExtension);
+    command_line->AppendSwitchPath(switches::kUseFileForFakeVideoCapture,
+                                   input_video);
     command_line->AppendSwitch(switches::kUseFakeDeviceForMediaStream);
 
     // Flag used by TestWebAudioMediaStream to force garbage collection.
@@ -113,7 +123,15 @@ class WebRtcBrowserTest : public WebRtcTestBase,
     return NULL;
   }
 
-  PeerConnectionServerRunner peerconnection_server_;
+  bool OnWinXp() {
+#if defined(OS_WIN)
+    return base::win::GetVersion() <= base::win::VERSION_XP;
+#else
+    return false;
+#endif
+  }
+
+  test::PeerConnectionServerRunner peerconnection_server_;
 };
 
 static const bool kRunTestsWithFlag[] = { false, true };
@@ -123,6 +141,9 @@ INSTANTIATE_TEST_CASE_P(WebRtcBrowserTests,
 
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        MANUAL_RunsAudioVideoWebRTCCallInTwoTabs) {
+  if (OnWinXp()) return;
+
+  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   ASSERT_TRUE(peerconnection_server_.Start());
 
@@ -147,6 +168,9 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
 }
 
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CpuUsage15Seconds) {
+  if (OnWinXp()) return;
+
+  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   ASSERT_TRUE(peerconnection_server_.Start());
 
@@ -182,7 +206,7 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CpuUsage15Seconds) {
 
   EstablishCall(left_tab, right_tab);
 
-  SleepInJavascript(left_tab, 15000);
+  test::SleepInJavascript(left_tab, 15000);
 
   HangUp(left_tab);
   WaitUntilHangupVerified(left_tab);
@@ -199,6 +223,9 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_CpuUsage15Seconds) {
 // This is manual for its long execution time.
 IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
                        MANUAL_RunsAudioVideoCall60SecsAndLogsInternalMetrics) {
+  if (OnWinXp()) return;
+
+  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   ASSERT_TRUE(peerconnection_server_.Start());
 
@@ -220,7 +247,7 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
   WaitForVideoToPlay(right_tab);
 
   // Let values stabilize, bandwidth ramp up, etc.
-  SleepInJavascript(left_tab, 60000);
+  test::SleepInJavascript(left_tab, 60000);
 
   // Start measurements.
   chrome::AddTabAt(browser(), GURL(), -1, true);
@@ -228,7 +255,7 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
   content::WebContents* webrtc_internals_tab =
       browser()->tab_strip_model()->GetActiveWebContents();
 
-  SleepInJavascript(left_tab, 10000);
+  test::SleepInJavascript(left_tab, 10000);
 
   scoped_ptr<base::DictionaryValue> all_data(
       GetWebrtcInternalsData(webrtc_internals_tab));
@@ -237,8 +264,8 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
   const base::DictionaryValue* first_pc_dict =
       GetDataOnFirstPeerConnection(all_data.get());
   ASSERT_TRUE(first_pc_dict != NULL);
-  PrintBweForVideoMetrics(*first_pc_dict);
-  PrintMetricsForAllStreams(*first_pc_dict);
+  test::PrintBweForVideoMetrics(*first_pc_dict);
+  test::PrintMetricsForAllStreams(*first_pc_dict);
 
   HangUp(left_tab);
   WaitUntilHangupVerified(left_tab);
@@ -247,7 +274,10 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest,
   ASSERT_TRUE(peerconnection_server_.Stop());
 }
 
-IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, TestWebAudioMediaStream) {
+IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, MANUAL_TestWebAudioMediaStream) {
+  if (OnWinXp()) return;
+
+  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
   ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
   GURL url(embedded_test_server()->GetURL("/webrtc/webaudio_crash.html"));
   ui_test_utils::NavigateToURL(browser(), url);
@@ -255,7 +285,7 @@ IN_PROC_BROWSER_TEST_P(WebRtcBrowserTest, TestWebAudioMediaStream) {
       browser()->tab_strip_model()->GetActiveWebContents();
 
   // A sleep is necessary to be able to detect the crash.
-  SleepInJavascript(tab, 1000);
+  test::SleepInJavascript(tab, 1000);
 
   ASSERT_FALSE(tab->IsCrashed());
 }

chrome/browser/media/chrome_webrtc_disable_encryption_flag_browsertest.cc

@@ -31,7 +31,7 @@ class WebRtcDisableEncryptionFlagBrowserTest : public WebRtcTestBase {
   virtual ~WebRtcDisableEncryptionFlagBrowserTest() {}
 
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
-    PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
+    test::PeerConnectionServerRunner::KillAllPeerConnectionServers();
     DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
   }
 
@@ -44,7 +44,7 @@ class WebRtcDisableEncryptionFlagBrowserTest : public WebRtcTestBase {
   }
 
  protected:
-  PeerConnectionServerRunner peerconnection_server_;
+  test::PeerConnectionServerRunner peerconnection_server_;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(WebRtcDisableEncryptionFlagBrowserTest);

chrome/browser/media/chrome_webrtc_typing_detection_browsertest.cc

@@ -52,7 +52,7 @@ class WebRtcTypingDetectionBrowserTest : public WebRtcTestBase {
   // TODO(phoglund): clean up duplication from audio quality browser test when
   // this test is complete and is proven to work.
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
-    PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
+    test::PeerConnectionServerRunner::KillAllPeerConnectionServers();
   }
 
   bool HasAllRequiredResources() {
@@ -88,13 +88,13 @@ class WebRtcTypingDetectionBrowserTest : public WebRtcTestBase {
               ExecuteJavascript("negotiateCall()", from_tab));
 
     // Ensure the call gets up on both sides.
-    EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                                 "active", from_tab));
-    EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                                 "active", to_tab));
+    EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                       "active", from_tab));
+    EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                       "active", to_tab));
   }
 
-  PeerConnectionServerRunner peerconnection_server_;
+  test::PeerConnectionServerRunner peerconnection_server_;
 };
 
 // TODO(phoglund): enable when fully implemented.
@@ -133,13 +133,13 @@ IN_PROC_BROWSER_TEST_F(WebRtcTypingDetectionBrowserTest,
   // because the ready state is ok on both sides. We sleep a bit between call
   // establishment and playing to avoid cutting of the beginning of the audio
   // file.
-  SleepInJavascript(left_tab, 2000);
+  test::SleepInJavascript(left_tab, 2000);
 
   PlayAudioFile(left_tab);
 
   // TODO(phoglund): simulate key presses, look for changes in typing detection
   // state.
-  SleepInJavascript(left_tab, 10000);
+  test::SleepInJavascript(left_tab, 10000);
 
   HangUp(left_tab);
   WaitUntilHangupVerified(left_tab);

chrome/browser/media/chrome_webrtc_video_quality_browsertest.cc

@@ -61,14 +61,6 @@ static const char kHomeEnvName[] =
 // The working dir should be in the user's home folder.
 static const base::FilePath::CharType kWorkingDirName[] =
     FILE_PATH_LITERAL("webrtc_video_quality");
-static const base::FilePath::CharType kReferenceVideosDirName[] =
-    FILE_PATH_LITERAL("webrtc.DEPS/webrtc_videos");
-static const base::FilePath::CharType kReferenceFileName360p[] =
-    FILE_PATH_LITERAL("reference_video_640x360_30fps");
-static const base::FilePath::CharType kYuvFileExtension[] =
-    FILE_PATH_LITERAL("yuv");
-static const base::FilePath::CharType kY4mFileExtension[] =
-    FILE_PATH_LITERAL("y4m");
 static const base::FilePath::CharType kCapturedYuvFileName[] =
     FILE_PATH_LITERAL("captured_video.yuv");
 static const base::FilePath::CharType kStatsFileName[] =
@@ -84,14 +76,6 @@ static const int k360pHeight = 360;
 // too!
 static const char kPyWebSocketPortNumber[] = "12221";
 
-const char kAdviseOnGclientSolution[] =
-    "You need to add this solution to your .gclient to run this test:\n"
-    "{\n"
-    "  \"name\"        : \"webrtc.DEPS\",\n"
-    "  \"url\"         : \"svn://svn.chromium.org/chrome/trunk/deps/"
-    "third_party/webrtc/webrtc.DEPS\",\n"
-    "}";
-
 // Test the video quality of the WebRTC output.
 //
 // Prerequisites: This test case must run on a machine with a chrome playing
@@ -122,15 +106,16 @@ class WebRtcVideoQualityBrowserTest : public WebRtcTestBase {
         environment_(base::Environment::Create()) {}
 
   virtual void SetUpInProcessBrowserTestFixture() OVERRIDE {
-    PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem();
+    test::PeerConnectionServerRunner::KillAllPeerConnectionServers();
     DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
   }
 
   virtual void SetUpCommandLine(CommandLine* command_line) OVERRIDE {
     // Set up the command line option with the expected file name. We will check
     // its existence in HasAllRequiredResources().
-    webrtc_reference_video_y4m_ = GetReferenceVideosDir()
-        .Append(kReferenceFileName360p).AddExtension(kY4mFileExtension);
+    webrtc_reference_video_y4m_ = test::GetReferenceVideosDir()
+        .Append(test::kReferenceFileName360p)
+        .AddExtension(test::kY4mFileExtension);
     command_line->AppendSwitchPath(switches::kUseFileForFakeVideoCapture,
                                    webrtc_reference_video_y4m_);
     command_line->AppendSwitch(switches::kUseFakeDeviceForMediaStream);
@@ -145,27 +130,9 @@ class WebRtcVideoQualityBrowserTest : public WebRtcTestBase {
           "files:" << GetWorkingDir().value();
       return false;
     }
-    if (!base::PathExists(GetReferenceVideosDir())) {
-      LOG(ERROR) << "Cannot find the working directory for the reference video "
-          "files, expected at " << GetReferenceVideosDir().value() << ". " <<
-          kAdviseOnGclientSolution;
-      return false;
-    }
-    base::FilePath webrtc_reference_video_yuv = GetReferenceVideosDir()
-        .Append(kReferenceFileName360p).AddExtension(kYuvFileExtension);
-    if (!base::PathExists(webrtc_reference_video_yuv)) {
-      LOG(ERROR) << "Missing YUV reference video to be used for quality"
-          << " comparison, expected at " << webrtc_reference_video_yuv.value()
-          << ". " << kAdviseOnGclientSolution;
-      return false;
-    }
-    if (!base::PathExists(webrtc_reference_video_y4m_)) {
-      LOG(ERROR) << "Missing Y4M reference video to be used for quality"
-          << " comparison, expected at "<< webrtc_reference_video_y4m_.value()
-          << ". " << kAdviseOnGclientSolution;
-      return false;
-    }
-    return true;
+
+    // Ensure we have the required input files.
+    return test::HasReferenceFilesInCheckout();
   }
 
   bool StartPyWebSocketServer() {
@@ -311,23 +278,7 @@ class WebRtcVideoQualityBrowserTest : public WebRtcTestBase {
     return base::FilePath(native_home_dir).Append(kWorkingDirName);
   }
 
-  base::FilePath GetReferenceVideosDir() {
-    // FilePath does not tolerate relative paths, and we want to hang the
-    // kReferenceVideosDirName at the same level as Chromium codebase, so we
-    // need to subtract the trailing .../src manually from the path.
-    const size_t src_token_length = 3u;
-    const base::FilePath::StringType src_token(FILE_PATH_LITERAL("src"));
-    base::FilePath::StringType path = GetSourceDir().value();
-    DCHECK_GT(path.size(), src_token_length);
-    std::size_t found = path.rfind(src_token);
-    if (found != std::string::npos)
-      path.replace(found,
-                   src_token_length,
-                   base::FilePath::StringType(FILE_PATH_LITERAL("")));
-    return base::FilePath(path).Append(kReferenceVideosDirName);
-  }
-
-  PeerConnectionServerRunner peerconnection_server_;
+  test::PeerConnectionServerRunner peerconnection_server_;
 
  private:
   base::FilePath GetSourceDir() {
@@ -379,7 +330,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcVideoQualityBrowserTest,
   // sending frames take quite a bit of time.
   int polling_interval_msec = 1000;
 
-  EXPECT_TRUE(PollingWaitUntil(
+  EXPECT_TRUE(test::PollingWaitUntil(
       "doneFrameCapturing()", "done-capturing", right_tab,
       polling_interval_msec));
 
@@ -387,7 +338,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcVideoQualityBrowserTest,
   WaitUntilHangupVerified(left_tab);
   WaitUntilHangupVerified(right_tab);
 
-  EXPECT_TRUE(PollingWaitUntil(
+  EXPECT_TRUE(test::PollingWaitUntil(
       "haveMoreFramesToSend()", "no-more-frames", right_tab,
       polling_interval_msec));
 
@@ -406,7 +357,8 @@ IN_PROC_BROWSER_TEST_F(WebRtcVideoQualityBrowserTest,
       k360pWidth,
       k360pHeight,
       GetWorkingDir().Append(kCapturedYuvFileName),
-      GetReferenceVideosDir().Append(kReferenceFileName360p).AddExtension(
-          kYuvFileExtension),
+      test::GetReferenceVideosDir()
+          .Append(test::kReferenceFileName360p)
+          .AddExtension(test::kYuvFileExtension),
       GetWorkingDir().Append(kStatsFileName)));
 }

chrome/browser/media/webrtc_browsertest_base.cc

@@ -91,8 +91,8 @@ void WebRtcTestBase::GetUserMediaWithSpecificConstraintsAndAccept(
 
   // Wait for WebRTC to call the success callback.
   const char kOkGotStream[] = "ok-got-stream";
-  EXPECT_TRUE(PollingWaitUntil("obtainGetUserMediaResult()", kOkGotStream,
-                               tab_contents));
+  EXPECT_TRUE(test::PollingWaitUntil("obtainGetUserMediaResult()", kOkGotStream,
+                                     tab_contents));
 }
 
 void WebRtcTestBase::GetUserMediaAndDeny(content::WebContents* tab_contents) {
@@ -108,8 +108,9 @@ void WebRtcTestBase::GetUserMediaWithSpecificConstraintsAndDeny(
   CloseInfoBarInTab(tab_contents, infobar);
 
   // Wait for WebRTC to call the fail callback.
-  EXPECT_TRUE(PollingWaitUntil("obtainGetUserMediaResult()",
-                               kFailedWithPermissionDeniedError, tab_contents));
+  EXPECT_TRUE(test::PollingWaitUntil("obtainGetUserMediaResult()",
+                                     kFailedWithPermissionDeniedError,
+                                     tab_contents));
 }
 
 void WebRtcTestBase::GetUserMediaAndDismiss(
@@ -120,9 +121,9 @@ void WebRtcTestBase::GetUserMediaAndDismiss(
   CloseInfoBarInTab(tab_contents, infobar);
 
   // A dismiss should be treated like a deny.
-  EXPECT_TRUE(PollingWaitUntil("obtainGetUserMediaResult()",
-                               kFailedWithPermissionDismissedError,
-                               tab_contents));
+  EXPECT_TRUE(test::PollingWaitUntil("obtainGetUserMediaResult()",
+                                     kFailedWithPermissionDismissedError,
+                                     tab_contents));
 }
 
 void WebRtcTestBase::GetUserMedia(content::WebContents* tab_contents,
@@ -232,7 +233,7 @@ void WebRtcTestBase::ConnectToPeerConnectionServer(
     content::WebContents* tab_contents) const {
   std::string javascript = base::StringPrintf(
       "connect('http://localhost:%s', '%s');",
-      PeerConnectionServerRunner::kDefaultPort, peer_name.c_str());
+      test::PeerConnectionServerRunner::kDefaultPort, peer_name.c_str());
   EXPECT_EQ("ok-connected", ExecuteJavascript(javascript, tab_contents));
 }
 
@@ -247,10 +248,10 @@ void WebRtcTestBase::EstablishCall(content::WebContents* from_tab,
   EXPECT_EQ("ok-negotiating", ExecuteJavascript("negotiateCall()", from_tab));
 
   // Ensure the call gets up on both sides.
-  EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                               "active", from_tab));
-  EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                               "active", to_tab));
+  EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                     "active", from_tab));
+  EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                     "active", to_tab));
 }
 
 void WebRtcTestBase::HangUp(content::WebContents* from_tab) const {
@@ -259,8 +260,8 @@ void WebRtcTestBase::HangUp(content::WebContents* from_tab) const {
 
 void WebRtcTestBase::WaitUntilHangupVerified(
     content::WebContents* tab_contents) const {
-  EXPECT_TRUE(PollingWaitUntil("getPeerConnectionReadyState()",
-                               "no-peer-connection", tab_contents));
+  EXPECT_TRUE(test::PollingWaitUntil("getPeerConnectionReadyState()",
+                                     "no-peer-connection", tab_contents));
 }
 
 void WebRtcTestBase::DetectErrorsInJavaScript() {
@@ -277,6 +278,6 @@ void WebRtcTestBase::StartDetectingVideo(
 
 void WebRtcTestBase::WaitForVideoToPlay(
     content::WebContents* tab_contents) const {
-  EXPECT_TRUE(PollingWaitUntil("isVideoPlaying()", "video-playing",
-                               tab_contents));
+  EXPECT_TRUE(test::PollingWaitUntil("isVideoPlaying()", "video-playing",
+                                     tab_contents));
 }

chrome/browser/media/webrtc_browsertest_common.cc

@@ -15,8 +15,75 @@
 #include "chrome/browser/ui/browser_tabstrip.h"
 #include "content/public/test/browser_test_utils.h"
 
+namespace test {
+
+const base::FilePath::CharType kReferenceVideosDirName[] =
+    FILE_PATH_LITERAL("webrtc.DEPS/webrtc_videos");
+const base::FilePath::CharType kReferenceFileName360p[] =
+    FILE_PATH_LITERAL("reference_video_640x360_30fps");
+const base::FilePath::CharType kYuvFileExtension[] = FILE_PATH_LITERAL("yuv");
+const base::FilePath::CharType kY4mFileExtension[] = FILE_PATH_LITERAL("y4m");
+
+// This message describes how to modify your .gclient to get the reference
+// video files downloaded for you.
+static const char kAdviseOnGclientSolution[] =
+    "You need to add this solution to your .gclient to run this test:\n"
+    "{\n"
+    "  \"name\"        : \"webrtc.DEPS\",\n"
+    "  \"url\"         : \"svn://svn.chromium.org/chrome/trunk/deps/"
+    "third_party/webrtc/webrtc.DEPS\",\n"
+    "}";
+
 const int kDefaultPollIntervalMsec = 250;
 
+base::FilePath GetReferenceVideosDir() {
+  // FilePath does not tolerate relative paths, and we want to hang the
+  // kReferenceVideosDirName at the same level as Chromium codebase, so we
+  // need to subtract the trailing .../src manually from the path.
+  const size_t src_token_length = 3u;
+  const base::FilePath::StringType src_token(FILE_PATH_LITERAL("src"));
+
+  base::FilePath source_dir;
+  PathService::Get(base::DIR_SOURCE_ROOT, &source_dir);
+
+  base::FilePath::StringType path = source_dir.value();
+  DCHECK_GT(path.size(), src_token_length);
+  std::size_t found = path.rfind(src_token);
+  if (found != std::string::npos)
+    path.erase(found, src_token_length);
+  return base::FilePath(path).Append(kReferenceVideosDirName);
+}
+
+bool HasReferenceFilesInCheckout() {
+  if (!base::PathExists(GetReferenceVideosDir())) {
+    LOG(ERROR)
+        << "Cannot find the working directory for the reference video "
+        << "files, expected at " << GetReferenceVideosDir().value() << ". " <<
+        kAdviseOnGclientSolution;
+    return false;
+  }
+  base::FilePath webrtc_reference_video_yuv = GetReferenceVideosDir()
+      .Append(kReferenceFileName360p).AddExtension(kYuvFileExtension);
+  if (!base::PathExists(webrtc_reference_video_yuv)) {
+    LOG(ERROR)
+        << "Missing YUV reference video to be used for quality"
+        << " comparison, expected at " << webrtc_reference_video_yuv.value()
+        << ". " << kAdviseOnGclientSolution;
+    return false;
+  }
+
+  base::FilePath webrtc_reference_video_y4m = GetReferenceVideosDir()
+      .Append(kReferenceFileName360p).AddExtension(kY4mFileExtension);
+  if (!base::PathExists(webrtc_reference_video_y4m)) {
+    LOG(ERROR)
+        << "Missing Y4M reference video to be used for quality"
+        << " comparison, expected at "<< webrtc_reference_video_y4m.value()
+        << ". " << kAdviseOnGclientSolution;
+    return false;
+  }
+  return true;
+}
+
 bool SleepInJavascript(content::WebContents* tab_contents, int timeout_msec) {
   const std::string javascript = base::StringPrintf(
       "setTimeout(function() {"
@@ -62,9 +129,10 @@ bool PollingWaitUntil(const std::string& javascript,
       LOG(ERROR) << "Failed to sleep.";
     }
   }
-  LOG(ERROR) << "Timed out while waiting for " << javascript <<
-      " to evaluate to " << evaluates_to << "; last result was '" << result <<
-      "'";
+  LOG(ERROR)
+      << "Timed out while waiting for " << javascript
+      << " to evaluate to " << evaluates_to << "; last result was '" << result
+      << "'";
   return false;
 }
 
@@ -86,8 +154,9 @@ bool PeerConnectionServerRunner::Start() {
   peerconnection_server = peerconnection_server.Append(kServerExecutable);
 
   if (!base::PathExists(peerconnection_server)) {
-    LOG(ERROR) << "Missing " << kServerExecutable << ". You must build "
-        "it so it ends up next to the browser test binary.";
+    LOG(ERROR)
+        << "Missing " << kServerExecutable << ". You must build "
+        << "it so it ends up next to the browser test binary.";
     return false;
   }
 
@@ -103,7 +172,7 @@ bool PeerConnectionServerRunner::Stop() {
   return base::KillProcess(server_pid_, 0, false);
 }
 
-void PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem() {
+void PeerConnectionServerRunner::KillAllPeerConnectionServers() {
   if (!base::KillProcesses(kServerExecutable, -1, NULL)) {
     LOG(ERROR) << "Failed to kill instances of " << kServerExecutable << ".";
     return;
@@ -111,3 +180,5 @@ void PeerConnectionServerRunner::KillAllPeerConnectionServersOnCurrentSystem() {
   base::WaitForProcessesToExit(kServerExecutable,
                                base::TimeDelta::FromSeconds(5), NULL);
 }
+
+}  // namespace test

chrome/browser/media/webrtc_browsertest_common.h

@@ -7,12 +7,31 @@
 
 #include <string>
 
+#include "base/files/file_path.h"
 #include "base/process/process_handle.h"
 
 namespace content {
 class WebContents;
 }
 
+namespace test {
+
+// Reference video locations.
+
+// Checks if the user has the reference video files, returns true if so.
+// If the user's checkout don't have these dirs, they need to configure their
+// .gclient as described in kAdviseOnGclientSolution. The reason for this is
+// that we don't want to burden regular chrome devs with downloading these
+// sizable reference files by default.
+bool HasReferenceFilesInCheckout();
+
+// Retrieves the reference video dir, to which file names can be appended.
+base::FilePath GetReferenceVideosDir();
+
+extern const base::FilePath::CharType kReferenceFileName360p[];
+extern const base::FilePath::CharType kYuvFileExtension[];
+extern const base::FilePath::CharType kY4mFileExtension[];
+
 // Executes javascript code which will sleep for |timeout_msec| milliseconds.
 // Returns true on success.
 bool SleepInJavascript(content::WebContents* tab_contents, int timeout_msec);
@@ -44,10 +63,12 @@ class PeerConnectionServerRunner {
   // Stops the peerconnection server.
   bool Stop();
 
-  static void KillAllPeerConnectionServersOnCurrentSystem();
+  static void KillAllPeerConnectionServers();
 
  private:
   base::ProcessHandle server_pid_;
 };
 
+}  // namespace test
+
 #endif  // CHROME_BROWSER_MEDIA_WEBRTC_BROWSERTEST_COMMON_H_

chrome/browser/media/webrtc_browsertest_perf.cc

@@ -172,6 +172,8 @@ static std::set<std::string> FindAllSsrcIdentifiers(
   return result;
 }
 
+namespace test {
+
 void PrintBweForVideoMetrics(const base::DictionaryValue& pc_dict) {
   const std::string kBweStatsKey = "bweforvideo";
   std::string value;
@@ -218,3 +220,5 @@ void PrintMetricsForAllStreams(const base::DictionaryValue& pc_dict) {
                                            << " is. ";
   }
 }
+
+}  // namespace test

chrome/browser/media/webrtc_browsertest_perf.h

@@ -9,6 +9,8 @@ namespace base {
 class DictionaryValue;
 }
 
+namespace test {
+
 // These functions takes parsed data (on one peer connection) from the
 // peerConnectionDataStore object that is backing webrtc-internals and prints
 // metrics they consider interesting using testing/perf/perf_test.h primitives.
@@ -20,4 +22,6 @@ class DictionaryValue;
 void PrintBweForVideoMetrics(const base::DictionaryValue& pc_dict);
 void PrintMetricsForAllStreams(const base::DictionaryValue& pc_dict);
 
+}  // namespace test
+
 #endif  // CHROME_BROWSER_MEDIA_WEBRTC_BROWSERTEST_PERF_H_

chrome/test/data/webrtc/webrtc_jsep01_test.html

@@ -16,10 +16,8 @@
       <td>Remote Video</td>
     </tr>
     <tr>
-      <td><video width="320" height="240" id="local-view"
-          autoplay="autoplay"></video></td>
-      <td><video width="320" height="240" id="remote-view"
-          autoplay="autoplay"></video></td>
+      <td><video id="local-view" autoplay></video></td>
+      <td><video id="remote-view" autoplay></video></td>
     </tr>
   </table>
 </body>