Implementing WebRTC Auto Gain Control (AGC) test (minus verification).

This implements the test scaffolding. We still need a tool to analyze
the audio energy of the output file and ensure it matches the energy
level of the input file.

The test will be disabled for now until the tool is in place.

The next patches will be to wire up the tool and, after that, write
more exciting tests which verify the AGC actually kicks in and adjusts
the level (this test just ensures it doesn't zero the volume level).

BUG=303259

Review URL: https://codereview.chromium.org/753993002

Cr-Commit-Position: refs/heads/master@{#307448}

2 files changed, 102 insertions, 42 deletions

diff --git a/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc b/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
index 15ec593..f0d5516 100644
--- a/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
+++ b/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
@@ -34,6 +34,9 @@ static const base::FilePath::CharType kReferenceFile[] =
     FILE_PATH_LITERAL("human-voice-linux.wav");
 #endif
 
+static const base::FilePath::CharType kAgcTestReferenceFile[] =
+    FILE_PATH_LITERAL("speech_44kHz_16bit_stereo.wav");
+
 // The javascript will load the reference file relative to its location,
 // which is in /webrtc on the web server. The files we are looking for are in
 // webrtc/resources in the chrome/test/data folder.
@@ -46,9 +49,16 @@ static const char kReferenceFileRelativeUrl[] =
     "resources/human-voice-linux.wav";
 #endif
 
-static const char kMainWebrtcTestHtmlPage[] =
+static const char kWebRtcAudioTestHtmlPage[] =
     "/webrtc/webrtc_audio_quality_test.html";
 
+#if defined(OS_LINUX) || defined(OS_WIN) || defined(OS_MACOSX)
+#define MAYBE_WebRtcAudioQualityBrowserTest WebRtcAudioQualityBrowserTest
+#else
+// Not implemented on Android, ChromeOS etc.
+#define MAYBE_WebRtcAudioQualityBrowserTest DISABLED_WebRtcAudioQualityBrowserTest
+#endif
+
 // Test we can set up a WebRTC call and play audio through it.
 //
 // If you're not a googler and want to run this test, you need to provide a
@@ -105,29 +115,36 @@ static const char kMainWebrtcTestHtmlPage[] =
 //    50 / 100 in level. Also go into the playback tab, right-click Speakers,
 //    and set that level to 50 / 100. Otherwise you will get distortion in
 //    the recording.
-class WebRtcAudioQualityBrowserTest : public WebRtcTestBase {
+class MAYBE_WebRtcAudioQualityBrowserTest : public WebRtcTestBase {
  public:
-  WebRtcAudioQualityBrowserTest() {}
+  MAYBE_WebRtcAudioQualityBrowserTest() {}
   void SetUpInProcessBrowserTestFixture() override {
     DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
   }
 
   void SetUpCommandLine(CommandLine* command_line) override {
-    // This test expects real device handling and requires a real webcam / audio
-    // device; it will not work with fake devices.
-    EXPECT_FALSE(command_line->HasSwitch(
-        switches::kUseFakeDeviceForMediaStream));
     EXPECT_FALSE(command_line->HasSwitch(
         switches::kUseFakeUIForMediaStream));
+
+    // The WebAudio-based tests don't care what devices are available to
+    // getUserMedia, and the getUserMedia-based tests will play back a file
+    // through the fake device using using --use-file-for-fake-audio-capture.
+    command_line->AppendSwitch(switches::kUseFakeDeviceForMediaStream);
   }
 
-  void AddAudioFile(const std::string& input_file_relative_url,
-                    content::WebContents* tab_contents) {
+  void ConfigureFakeDeviceToPlayFile(const base::FilePath& wav_file_path) {
+    CommandLine::ForCurrentProcess()->AppendSwitchPath(
+        switches::kUseFileForFakeAudioCapture, wav_file_path);
+  }
+
+  void AddAudioFileToWebAudio(const std::string& input_file_relative_url,
+                              content::WebContents* tab_contents) {
+    // This calls into webaudio.js.
     EXPECT_EQ("ok-added", ExecuteJavascript(
         "addAudioFile('" + input_file_relative_url + "')", tab_contents));
   }
 
-  void PlayAudioFile(content::WebContents* tab_contents) {
+  void PlayAudioFileThroughWebAudio(content::WebContents* tab_contents) {
     EXPECT_EQ("ok-playing", ExecuteJavascript("playAudioFile()", tab_contents));
   }
 
@@ -139,6 +156,20 @@ class WebRtcAudioQualityBrowserTest : public WebRtcTestBase {
     EXPECT_TRUE(base::Move(filename, wav_filename));
     return wav_filename;
   }
+
+  content::WebContents* OpenPageWithoutGetUserMedia(const char* url) {
+    chrome::AddTabAt(browser(), GURL(), -1, true);
+    ui_test_utils::NavigateToURL(
+        browser(), embedded_test_server()->GetURL(url));
+    content::WebContents* tab =
+        browser()->tab_strip_model()->GetActiveWebContents();
+
+    // Prepare the peer connections manually in this test since we don't add
+    // getUserMedia-derived media streams in this test like the other tests.
+    EXPECT_EQ("ok-peerconnection-created",
+              ExecuteJavascript("preparePeerConnection()", tab));
+    return tab;
+  }
 };
 
 class AudioRecorder {
@@ -391,15 +422,8 @@ bool RunPesq(const base::FilePath& reference_file,
   return true;
 }
 
-#if defined(OS_LINUX) || defined(OS_WIN) || defined(OS_MACOSX)
-#define MAYBE_MANUAL_TestAudioQuality MANUAL_TestAudioQuality
-#else
-// Not implemented on Android, ChromeOS etc.
-#define MAYBE_MANUAL_TestAudioQuality DISABLED_MANUAL_TestAudioQuality
-#endif
-
-IN_PROC_BROWSER_TEST_F(WebRtcAudioQualityBrowserTest,
-                       MAYBE_MANUAL_TestAudioQuality) {
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
+                       MANUAL_TestAudioQuality) {
   if (OnWinXp()) {
     LOG(ERROR) << "This test is not implemented for Windows XP.";
     return;
@@ -414,32 +438,18 @@ IN_PROC_BROWSER_TEST_F(WebRtcAudioQualityBrowserTest,
 
   ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());
 
-  ui_test_utils::NavigateToURL(
-      browser(), embedded_test_server()->GetURL(kMainWebrtcTestHtmlPage));
   content::WebContents* left_tab =
-      browser()->tab_strip_model()->GetActiveWebContents();
-
-  chrome::AddTabAt(browser(), GURL(), -1, true);
+      OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);
   content::WebContents* right_tab =
-      browser()->tab_strip_model()->GetActiveWebContents();
-  ui_test_utils::NavigateToURL(
-      browser(), embedded_test_server()->GetURL(kMainWebrtcTestHtmlPage));
-
-  // Prepare the peer connections manually in this test since we don't add
-  // getUserMedia-derived media streams in this test like the other tests.
-  EXPECT_EQ("ok-peerconnection-created",
-            ExecuteJavascript("preparePeerConnection()", left_tab));
-  EXPECT_EQ("ok-peerconnection-created",
-            ExecuteJavascript("preparePeerConnection()", right_tab));
+      OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);
 
-  AddAudioFile(kReferenceFileRelativeUrl, left_tab);
+  AddAudioFileToWebAudio(kReferenceFileRelativeUrl, left_tab);
 
   NegotiateCall(left_tab, right_tab);
 
   // Note: the media flow isn't necessarily established on the connection just
   // 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.
+  // establishment and playing to avoid cutting off the beginning of the stream.
   test::SleepInJavascript(left_tab, 2000);
 
   base::FilePath recording = CreateTemporaryWaveFile();
@@ -450,7 +460,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcAudioQualityBrowserTest,
   static int kRecordingTimeSeconds = 15;
   ASSERT_TRUE(recorder.StartRecording(kRecordingTimeSeconds, recording, true));
 
-  PlayAudioFile(left_tab);
+  PlayAudioFileThroughWebAudio(left_tab);
 
   ASSERT_TRUE(recorder.WaitForRecordingToEnd());
   DVLOG(0) << "Done recording to " << recording.value() << std::endl;
@@ -475,3 +485,57 @@ IN_PROC_BROWSER_TEST_F(WebRtcAudioQualityBrowserTest,
   EXPECT_TRUE(base::DeleteFile(recording, false));
   EXPECT_TRUE(base::DeleteFile(trimmed_recording, false));
 }
+
+IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
+                       MANUAL_TestAutoGainControlIncreasesEnergyForLowAudio) {
+  if (OnWinXp()) {
+    LOG(ERROR) << "This test is not implemented for Windows XP.";
+    return;
+  }
+  if (OnWin8()) {
+    // http://crbug.com/379798.
+    LOG(ERROR) << "Temporarily disabled for Win 8.";
+    return;
+  }
+  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
+  ASSERT_TRUE(embedded_test_server()->InitializeAndWaitUntilReady());
+
+  ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());
+
+  ConfigureFakeDeviceToPlayFile(
+      test::GetReferenceFilesDir().Append(kAgcTestReferenceFile));
+
+  // Create a one-way call.
+  GURL test_page = embedded_test_server()->GetURL(kWebRtcAudioTestHtmlPage);
+  content::WebContents* left_tab =
+      OpenPageAndGetUserMediaInNewTabWithConstraints(test_page,
+                                                     kAudioOnlyCallConstraints);
+  SetupPeerconnectionWithLocalStream(left_tab);
+
+  content::WebContents* right_tab =
+      OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);
+
+  NegotiateCall(left_tab, right_tab);
+
+  base::FilePath recording = CreateTemporaryWaveFile();
+
+  AudioRecorder recorder;
+  static int kRecordingTimeSeconds = 10;
+  ASSERT_TRUE(recorder.StartRecording(kRecordingTimeSeconds, recording, true));
+
+  ASSERT_TRUE(recorder.WaitForRecordingToEnd());
+  DVLOG(0) << "Done recording to " << recording.value() << std::endl;
+
+  HangUp(left_tab);
+
+  base::FilePath trimmed_recording = CreateTemporaryWaveFile();
+
+  ASSERT_TRUE(RemoveSilence(recording, trimmed_recording));
+  DVLOG(0) << "Trimmed silence: " << trimmed_recording.value() << std::endl;
+
+  // TODO(phoglund): invoke bjornv's audio energy analysis tool on the trimmed
+  // recording and log the result.
+
+  EXPECT_TRUE(base::DeleteFile(recording, false));
+  EXPECT_TRUE(base::DeleteFile(trimmed_recording, false));
+}
diff --git a/chrome/browser/media/webrtc_browsertest_base.cc b/chrome/browser/media/webrtc_browsertest_base.cc
index ea00f7a..93404d8 100644
--- a/chrome/browser/media/webrtc_browsertest_base.cc
+++ b/chrome/browser/media/webrtc_browsertest_base.cc
@@ -185,10 +185,6 @@ WebRtcTestBase::OpenPageAndGetUserMediaInNewTabWithConstraints(
     const std::string& constraints) const {
   chrome::AddTabAt(browser(), GURL(), -1, true);
   ui_test_utils::NavigateToURL(browser(), url);
-#if defined (OS_LINUX)
-  // Load the page again on Linux to work around crbug.com/281268.
-  ui_test_utils::NavigateToURL(browser(), url);
-#endif
   content::WebContents* new_tab =
       browser()->tab_strip_model()->GetActiveWebContents();
   GetUserMediaWithSpecificConstraintsAndAccept(new_tab, constraints);