Cleanup: remove prefix chrome_ of chrome/browser/media/chrome* files

chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc

Index: chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
diff --git a/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc b/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
deleted file mode 100644
index 2c2a8687eb91cbfa0dfa53232879a8fd5d89ddf9..0000000000000000000000000000000000000000
--- a/chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
+++ /dev/null
@@ -1,812 +0,0 @@
-// 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 <stddef.h>
-
-#include <ctime>
-
-#include "base/command_line.h"
-#include "base/files/file_enumerator.h"
-#include "base/files/file_util.h"
-#include "base/files/scoped_temp_dir.h"
-#include "base/macros.h"
-#include "base/process/launch.h"
-#include "base/process/process.h"
-#include "base/scoped_native_library.h"
-#include "base/strings/string_number_conversions.h"
-#include "base/strings/string_util.h"
-#include "base/strings/stringprintf.h"
-#include "base/strings/utf_string_conversions.h"
-#include "build/build_config.h"
-#include "chrome/browser/media/webrtc_browsertest_audio.h"
-#include "chrome/browser/media/webrtc_browsertest_base.h"
-#include "chrome/browser/media/webrtc_browsertest_common.h"
-#include "chrome/browser/profiles/profile.h"
-#include "chrome/browser/ui/browser.h"
-#include "chrome/browser/ui/browser_tabstrip.h"
-#include "chrome/browser/ui/tabs/tab_strip_model.h"
-#include "chrome/common/chrome_paths.h"
-#include "chrome/common/chrome_switches.h"
-#include "chrome/test/base/ui_test_utils.h"
-#include "content/public/common/content_switches.h"
-#include "content/public/test/browser_test_utils.h"
-#include "media/audio/audio_parameters.h"
-#include "media/base/media_switches.h"
-#include "net/test/embedded_test_server/embedded_test_server.h"
-#include "testing/perf/perf_test.h"
-
-namespace {
-
-static const base::FilePath::CharType kReferenceFile[] =
-    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.
-static const char kReferenceFileRelativeUrl[] =
-    "resources/speech_44kHz_16bit_stereo.wav";
-
-static const char kWebRtcAudioTestHtmlPage[] =
-    "/webrtc/webrtc_audio_quality_test.html";
-
-// For the AGC test, there are 6 speech segments split on silence. If one
-// segment is significantly different in length compared to the same segment in
-// the reference file, there's something fishy going on.
-const int kMaxAgcSegmentDiffMs =
-#if defined(OS_MACOSX)
-  // Something is different on Mac; http://crbug.com/477653.
-  600;
-#else
-  200;
-#endif
-
-#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
-
-}  // namespace
-
-// 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
-// pesq binary for your platform (and sox.exe on windows). Read more on how
-// resources are managed in chrome/test/data/webrtc/resources/README.
-//
-// This test will only work on machines that have been configured to record
-// their own input.
-//
-// On Linux:
-// 1. # sudo apt-get install pavucontrol sox
-// 2. For the user who will run the test: # pavucontrol
-// 3. In a separate terminal, # arecord dummy
-// 4. In pavucontrol, go to the recording tab.
-// 5. For the ALSA plugin [aplay]: ALSA Capture from, change from <x> to
-//    <Monitor of x>, where x is whatever your primary sound device is called.
-// 6. Try launching chrome as the target user on the target machine, try
-//    playing, say, a YouTube video, and record with # arecord -f dat tmp.dat.
-//    Verify the recording with aplay (should have recorded what you played
-//    from chrome).
-//
-// Note: the volume for ALL your input devices will be forced to 100% by
-//       running this test on Linux.
-//
-// On Mac:
-// TODO(phoglund): download sox from gs instead.
-// 1. Get SoundFlower: http://rogueamoeba.com/freebies/soundflower/download.php
-// 2. Install it + reboot.
-// 3. Install MacPorts (http://www.macports.org/).
-// 4. Install sox: sudo port install sox.
-// 5. (For Chrome bots) Ensure sox and rec are reachable from the env the test
-//    executes in (sox and rec tends to install in /opt/, which generally isn't
-//    in the Chrome bots' env). For instance, run
-//    sudo ln -s /opt/local/bin/rec /usr/local/bin/rec
-//    sudo ln -s /opt/local/bin/sox /usr/local/bin/sox
-// 6. In Sound Preferences, set both input and output to Soundflower (2ch).
-//    Note: You will no longer hear audio on this machine, and it will no
-//    longer use any built-in mics.
-// 7. Try launching chrome as the target user on the target machine, try
-//    playing, say, a YouTube video, and record with 'rec test.wav trim 0 5'.
-//    Stop the video in chrome and try playing back the file; you should hear
-//    a recording of the video (note; if you play back on the target machine
-//    you must revert the changes in step 3 first).
-//
-// On Windows 7:
-// 1. Control panel > Sound > Manage audio devices.
-// 2. In the recording tab, right-click in an empty space in the pane with the
-//    devices. Tick 'show disabled devices'.
-// 3. You should see a 'stero mix' device - this is what your speakers output.
-//    Right click > Properties.
-// 4. In the Listen tab for the mix device, check the 'listen to this device'
-//    checkbox. Ensure the mix device is the default recording device.
-// 5. Launch chrome and try playing a video with sound. You should see
-//    in the volume meter for the mix device. Configure the mix device to have
-//    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 MAYBE_WebRtcAudioQualityBrowserTest : public WebRtcTestBase {
- public:
-  MAYBE_WebRtcAudioQualityBrowserTest() {}
-  void SetUpInProcessBrowserTestFixture() override {
-    DetectErrorsInJavaScript();  // Look for errors in our rather complex js.
-  }
-
-  void SetUpCommandLine(base::CommandLine* command_line) override {
-    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);
-
-    // Add loopback interface such that there is always connectivity.
-    command_line->AppendSwitch(switches::kAllowLoopbackInPeerConnection);
-  }
-
-  void ConfigureFakeDeviceToPlayFile(const base::FilePath& wav_file_path) {
-    base::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 PlayAudioFileThroughWebAudio(content::WebContents* tab_contents) {
-    EXPECT_EQ("ok-playing", ExecuteJavascript("playAudioFile()", tab_contents));
-  }
-
-  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;
-  }
-
-  void MuteMediaElement(const std::string& element_id,
-                        content::WebContents* tab_contents) {
-    EXPECT_EQ("ok-muted", ExecuteJavascript(
-        "setMediaElementMuted('" + element_id + "', true)", tab_contents));
-  }
-
- protected:
-  void TestAutoGainControl(const base::FilePath::StringType& reference_filename,
-                           const std::string& constraints,
-                           const std::string& perf_modifier);
-  void SetupAndRecordAudioCall(const base::FilePath& reference_file,
-                               const base::FilePath& recording,
-                               const std::string& constraints,
-                               const base::TimeDelta recording_time);
-  void TestWithFakeDeviceGetUserMedia(const std::string& constraints,
-                                      const std::string& perf_modifier);
-};
-
-namespace {
-
-class AudioRecorder {
- public:
-  AudioRecorder() {}
-  ~AudioRecorder() {}
-
-  // Starts the recording program for the specified duration. Returns true
-  // on success. We record in 16-bit 44.1 kHz Stereo (mostly because that's
-  // what SoundRecorder.exe will give us and we can't change that).
-  bool StartRecording(base::TimeDelta recording_time,
-                      const base::FilePath& output_file) {
-    EXPECT_FALSE(recording_application_.IsValid())
-        << "Tried to record, but is already recording.";
-
-    int duration_sec = static_cast<int>(recording_time.InSeconds());
-    base::CommandLine command_line(base::CommandLine::NO_PROGRAM);
-
-#if defined(OS_WIN)
-    // This disable is required to run SoundRecorder.exe on 64-bit Windows
-    // from a 32-bit binary. We need to load the wow64 disable function from
-    // the DLL since it doesn't exist on Windows XP.
-    base::ScopedNativeLibrary kernel32_lib(base::FilePath(L"kernel32"));
-    if (kernel32_lib.is_valid()) {
-      typedef BOOL (WINAPI* Wow64DisableWow64FSRedirection)(PVOID*);
-      Wow64DisableWow64FSRedirection wow_64_disable_wow_64_fs_redirection;
-      wow_64_disable_wow_64_fs_redirection =
-          reinterpret_cast<Wow64DisableWow64FSRedirection>(
-              kernel32_lib.GetFunctionPointer(
-                  "Wow64DisableWow64FsRedirection"));
-      if (wow_64_disable_wow_64_fs_redirection != NULL) {
-        PVOID* ignored = NULL;
-        wow_64_disable_wow_64_fs_redirection(ignored);
-      }
-    }
-
-    char duration_in_hms[128] = {0};
-    struct tm duration_tm = {0};
-    duration_tm.tm_sec = duration_sec;
-    EXPECT_NE(0u, strftime(duration_in_hms, arraysize(duration_in_hms),
-                           "%H:%M:%S", &duration_tm));
-
-    command_line.SetProgram(
-        base::FilePath(FILE_PATH_LITERAL("SoundRecorder.exe")));
-    command_line.AppendArg("/FILE");
-    command_line.AppendArgPath(output_file);
-    command_line.AppendArg("/DURATION");
-    command_line.AppendArg(duration_in_hms);
-#elif defined(OS_MACOSX)
-    command_line.SetProgram(base::FilePath("rec"));
-    command_line.AppendArg("-b");
-    command_line.AppendArg("16");
-    command_line.AppendArg("-q");
-    command_line.AppendArgPath(output_file);
-    command_line.AppendArg("trim");
-    command_line.AppendArg("0");
-    command_line.AppendArg(base::IntToString(duration_sec));
-#else
-    command_line.SetProgram(base::FilePath("arecord"));
-    command_line.AppendArg("-d");
-    command_line.AppendArg(base::IntToString(duration_sec));
-    command_line.AppendArg("-f");
-    command_line.AppendArg("cd");
-    command_line.AppendArg("-c");
-    command_line.AppendArg("2");
-    command_line.AppendArgPath(output_file);
-#endif
-
-    DVLOG(0) << "Running " << command_line.GetCommandLineString();
-    recording_application_ =
-        base::LaunchProcess(command_line, base::LaunchOptions());
-    return recording_application_.IsValid();
-  }
-
-  // Joins the recording program. Returns true on success.
-  bool WaitForRecordingToEnd() {
-    int exit_code = -1;
-    recording_application_.WaitForExit(&exit_code);
-    return exit_code == 0;
-  }
- private:
-  base::Process recording_application_;
-};
-
-bool ForceMicrophoneVolumeTo100Percent() {
-#if defined(OS_WIN)
-  // Note: the force binary isn't in tools since it's one of our own.
-  base::CommandLine command_line(test::GetReferenceFilesDir().Append(
-      FILE_PATH_LITERAL("force_mic_volume_max.exe")));
-  DVLOG(0) << "Running " << command_line.GetCommandLineString();
-  std::string result;
-  if (!base::GetAppOutput(command_line, &result)) {
-    LOG(ERROR) << "Failed to set source volume: output was " << result;
-    return false;
-  }
-#elif defined(OS_MACOSX)
-  base::CommandLine command_line(
-      base::FilePath(FILE_PATH_LITERAL("osascript")));
-  command_line.AppendArg("-e");
-  command_line.AppendArg("set volume input volume 100");
-  command_line.AppendArg("-e");
-  command_line.AppendArg("set volume output volume 85");
-
-  std::string result;
-  if (!base::GetAppOutput(command_line, &result)) {
-    LOG(ERROR) << "Failed to set source volume: output was " << result;
-    return false;
-  }
-#else
-  // Just force the volume of, say the first 5 devices. A machine will rarely
-  // have more input sources than that. This is way easier than finding the
-  // input device we happen to be using.
-  for (int device_index = 0; device_index < 5; ++device_index) {
-    std::string result;
-    const std::string kHundredPercentVolume = "65536";
-    base::CommandLine command_line(base::FilePath(FILE_PATH_LITERAL("pacmd")));
-    command_line.AppendArg("set-source-volume");
-    command_line.AppendArg(base::IntToString(device_index));
-    command_line.AppendArg(kHundredPercentVolume);
-    DVLOG(0) << "Running " << command_line.GetCommandLineString();
-    if (!base::GetAppOutput(command_line, &result)) {
-      LOG(ERROR) << "Failed to set source volume: output was " << result;
-      return false;
-    }
-  }
-#endif
-  return true;
-}
-
-// Sox is the "Swiss army knife" of audio processing. We mainly use it for
-// silence trimming. See http://sox.sourceforge.net.
-base::CommandLine MakeSoxCommandLine() {
-#if defined(OS_WIN)
-  base::FilePath sox_path = test::GetToolForPlatform("sox");
-  if (!base::PathExists(sox_path)) {
-    LOG(ERROR) << "Missing sox.exe binary in " << sox_path.value()
-               << "; you may have to provide this binary yourself.";
-    return base::CommandLine(base::CommandLine::NO_PROGRAM);
-  }
-  base::CommandLine command_line(sox_path);
-#else
-  // TODO(phoglund): call checked-in sox rather than system sox on mac/linux.
-  // Same for rec invocations on Mac, above.
-  base::CommandLine command_line(base::FilePath(FILE_PATH_LITERAL("sox")));
-#endif
-  return command_line;
-}
-
-// Removes silence from beginning and end of the |input_audio_file| and writes
-// the result to the |output_audio_file|. Returns true on success.
-bool RemoveSilence(const base::FilePath& input_file,
-                   const base::FilePath& output_file) {
-  // SOX documentation for silence command: http://sox.sourceforge.net/sox.html
-  // To remove the silence from both beginning and end of the audio file, we
-  // call sox silence command twice: once on normal file and again on its
-  // reverse, then we reverse the final output.
-  // Silence parameters are (in sequence):
-  // ABOVE_PERIODS: The period for which silence occurs. Value 1 is used for
-  //                 silence at beginning of audio.
-  // DURATION: the amount of time in seconds that non-silence must be detected
-  //           before sox stops trimming audio.
-  // THRESHOLD: value used to indicate what sample value is treats as silence.
-  const char* kAbovePeriods = "1";
-  const char* kDuration = "2";
-  const char* kTreshold = "1.5%";
-
-  base::CommandLine command_line = MakeSoxCommandLine();
-  if (command_line.GetProgram().empty())
-    return false;
-  command_line.AppendArgPath(input_file);
-  command_line.AppendArgPath(output_file);
-  command_line.AppendArg("silence");
-  command_line.AppendArg(kAbovePeriods);
-  command_line.AppendArg(kDuration);
-  command_line.AppendArg(kTreshold);
-  command_line.AppendArg("reverse");
-  command_line.AppendArg("silence");
-  command_line.AppendArg(kAbovePeriods);
-  command_line.AppendArg(kDuration);
-  command_line.AppendArg(kTreshold);
-  command_line.AppendArg("reverse");
-
-  DVLOG(0) << "Running " << command_line.GetCommandLineString();
-  std::string result;
-  bool ok = base::GetAppOutput(command_line, &result);
-  DVLOG(0) << "Output was:\n\n" << result;
-  return ok;
-}
-
-// Looks for 0.2 second audio segments surrounded by silences under 0.3% audio
-// power and splits the input file on those silences. Output files are written
-// according to the output file template (e.g. /tmp/out.wav writes
-// /tmp/out001.wav, /tmp/out002.wav, etc if there are two silence-padded
-// regions in the file). The silences between speech segments must be at
-// least 500 ms for this to be reliable.
-bool SplitFileOnSilence(const base::FilePath& input_file,
-                        const base::FilePath& output_file_template) {
-  base::CommandLine command_line = MakeSoxCommandLine();
-  if (command_line.GetProgram().empty())
-    return false;
-
-  // These are experimentally determined and work on the files we use.
-  const char* kAbovePeriods = "1";
-  const char* kUnderPeriods = "1";
-  const char* kDuration = "0.2";
-  const char* kTreshold = "0.5%";
-  command_line.AppendArgPath(input_file);
-  command_line.AppendArgPath(output_file_template);
-  command_line.AppendArg("silence");
-  command_line.AppendArg(kAbovePeriods);
-  command_line.AppendArg(kDuration);
-  command_line.AppendArg(kTreshold);
-  command_line.AppendArg(kUnderPeriods);
-  command_line.AppendArg(kDuration);
-  command_line.AppendArg(kTreshold);
-  command_line.AppendArg(":");
-  command_line.AppendArg("newfile");
-  command_line.AppendArg(":");
-  command_line.AppendArg("restart");
-
-  DVLOG(0) << "Running " << command_line.GetCommandLineString();
-  std::string result;
-  bool ok = base::GetAppOutput(command_line, &result);
-  DVLOG(0) << "Output was:\n\n" << result;
-  return ok;
-}
-
-bool CanParseAsFloat(const std::string& value) {
-  return atof(value.c_str()) != 0 || value == "0";
-}
-
-// Runs PESQ to compare |reference_file| to a |actual_file|. The |sample_rate|
-// can be either 16000 or 8000.
-//
-// PESQ is only mono-aware, so the files should preferably be recorded in mono.
-// Furthermore it expects the file to be 16 rather than 32 bits, even though
-// 32 bits might work. The audio bandwidth of the two files should be the same
-// e.g. don't compare a 32 kHz file to a 8 kHz file.
-//
-// The raw score in MOS is written to |raw_mos|, whereas the MOS-LQO score is
-// written to mos_lqo. The scores are returned as floats in string form (e.g.
-// "3.145", etc). Returns true on success.
-bool RunPesq(const base::FilePath& reference_file,
-             const base::FilePath& actual_file,
-             int sample_rate, std::string* raw_mos, std::string* mos_lqo) {
-  // PESQ will break if the paths are too long (!).
-  EXPECT_LT(reference_file.value().length(), 128u);
-  EXPECT_LT(actual_file.value().length(), 128u);
-
-  base::FilePath pesq_path = test::GetToolForPlatform("pesq");
-  if (!base::PathExists(pesq_path)) {
-    LOG(ERROR) << "Missing PESQ binary in " << pesq_path.value()
-               << "; you may have to provide this binary yourself.";
-    return false;
-  }
-
-  base::CommandLine command_line(pesq_path);
-  command_line.AppendArg(base::StringPrintf("+%d", sample_rate));
-  command_line.AppendArgPath(reference_file);
-  command_line.AppendArgPath(actual_file);
-
-  DVLOG(0) << "Running " << command_line.GetCommandLineString();
-  std::string result;
-  if (!base::GetAppOutput(command_line, &result)) {
-    LOG(ERROR) << "Failed to run PESQ.";
-    return false;
-  }
-  DVLOG(0) << "Output was:\n\n" << result;
-
-  const std::string result_anchor = "Prediction (Raw MOS, MOS-LQO):  = ";
-  std::size_t anchor_pos = result.find(result_anchor);
-  if (anchor_pos == std::string::npos) {
-    LOG(ERROR) << "PESQ was not able to compute a score; we probably recorded "
-        << "only silence. Please check the output/input volume levels.";
-    return false;
-  }
-
-  // There are two tab-separated numbers on the format x.xxx, e.g. 5 chars each.
-  std::size_t first_number_pos = anchor_pos + result_anchor.length();
-  *raw_mos = result.substr(first_number_pos, 5);
-  EXPECT_TRUE(CanParseAsFloat(*raw_mos)) << "Failed to parse raw MOS number.";
-  *mos_lqo = result.substr(first_number_pos + 5 + 1, 5);
-  EXPECT_TRUE(CanParseAsFloat(*mos_lqo)) << "Failed to parse MOS LQO number.";
-
-  return true;
-}
-
-base::FilePath CreateTemporaryWaveFile() {
-  base::FilePath filename;
-  EXPECT_TRUE(base::CreateTemporaryFile(&filename));
-  base::FilePath wav_filename =
-      filename.AddExtension(FILE_PATH_LITERAL(".wav"));
-  EXPECT_TRUE(base::Move(filename, wav_filename));
-  return wav_filename;
-}
-
-void DeleteFileUnlessTestFailed(const base::FilePath& path, bool recursive) {
-  if (::testing::Test::HasFailure())
-    printf("Test failed; keeping recording(s) at\n\t%" PRFilePath ".\n",
-           path.value().c_str());
-  else
-    EXPECT_TRUE(base::DeleteFile(path, recursive));
-}
-
-std::vector<base::FilePath> ListWavFilesInDir(const base::FilePath& dir) {
-  base::FileEnumerator files(dir, false, base::FileEnumerator::FILES,
-                             FILE_PATH_LITERAL("*.wav"));
-
-  std::vector<base::FilePath> result;
-  for (base::FilePath name = files.Next(); !name.empty(); name = files.Next())
-    result.push_back(name);
-  return result;
-}
-
-// Splits |to_split| into sub-files based on silence. The file you use must have
-// at least 500 ms periods of silence between speech segments for this to be
-// reliable.
-void SplitFileOnSilenceIntoDir(const base::FilePath& to_split,
-                               const base::FilePath& workdir) {
-  // First trim beginning and end since they are tricky for the splitter.
-  base::FilePath trimmed_audio = CreateTemporaryWaveFile();
-
-  ASSERT_TRUE(RemoveSilence(to_split, trimmed_audio));
-  DVLOG(0) << "Trimmed silence: " << trimmed_audio.value() << std::endl;
-
-  ASSERT_TRUE(SplitFileOnSilence(
-      trimmed_audio, workdir.Append(FILE_PATH_LITERAL("output.wav"))));
-  DeleteFileUnlessTestFailed(trimmed_audio, false);
-}
-
-// Computes the difference between the actual and reference segment. A positive
-// number x means the actual file is x dB stronger than the reference.
-float AnalyzeOneSegment(const base::FilePath& ref_segment,
-                        const base::FilePath& actual_segment,
-                        int segment_number) {
-  media::AudioParameters ref_parameters;
-  media::AudioParameters actual_parameters;
-  float ref_energy =
-      test::ComputeAudioEnergyForWavFile(ref_segment, &ref_parameters);
-  float actual_energy =
-      test::ComputeAudioEnergyForWavFile(actual_segment, &actual_parameters);
-
-  base::TimeDelta difference_in_length = ref_parameters.GetBufferDuration() -
-                                         actual_parameters.GetBufferDuration();
-
-  EXPECT_LE(difference_in_length,
-            base::TimeDelta::FromMilliseconds(kMaxAgcSegmentDiffMs))
-      << "Segments differ " << difference_in_length.InMilliseconds() << " ms "
-      << "in length for segment " << segment_number << "; we're likely "
-      << "comparing unrelated segments or silence splitting is busted.";
-
-  return actual_energy - ref_energy;
-}
-
-std::string MakeTraceName(const base::FilePath& ref_filename,
-                          size_t segment_number) {
-  std::string ascii_filename;
-#if defined(OS_WIN)
-  ascii_filename = base::WideToUTF8(ref_filename.BaseName().value());
-#else
-  ascii_filename = ref_filename.BaseName().value();
-#endif
-  return base::StringPrintf(
-      "%s_segment_%d", ascii_filename.c_str(), (int)segment_number);
-}
-
-void AnalyzeSegmentsAndPrintResult(
-    const std::vector<base::FilePath>& ref_segments,
-    const std::vector<base::FilePath>& actual_segments,
-    const base::FilePath& reference_file,
-    const std::string& perf_modifier) {
-  ASSERT_GT(ref_segments.size(), 0u)
-      << "Failed to split reference file on silence; sox is likely broken.";
-  ASSERT_EQ(ref_segments.size(), actual_segments.size())
-      << "The recording did not result in the same number of audio segments "
-      << "after on splitting on silence; WebRTC must have deformed the audio "
-      << "too much.";
-
-  for (size_t i = 0; i < ref_segments.size(); i++) {
-    float difference_in_decibel = AnalyzeOneSegment(ref_segments[i],
-                                                    actual_segments[i],
-                                                    i);
-    std::string trace_name = MakeTraceName(reference_file, i);
-    perf_test::PrintResult("agc_energy_diff", perf_modifier, trace_name,
-                           difference_in_decibel, "dB", false);
-  }
-}
-
-void ComputeAndPrintPesqResults(const base::FilePath& reference_file,
-                                const base::FilePath& recording,
-                                const std::string& perf_modifier) {
-  base::FilePath trimmed_reference = CreateTemporaryWaveFile();
-  base::FilePath trimmed_recording = CreateTemporaryWaveFile();
-
-  ASSERT_TRUE(RemoveSilence(reference_file, trimmed_reference));
-  ASSERT_TRUE(RemoveSilence(recording, trimmed_recording));
-
-  std::string raw_mos;
-  std::string mos_lqo;
-  bool succeeded = RunPesq(trimmed_reference, trimmed_recording, 16000,
-                           &raw_mos, &mos_lqo);
-  EXPECT_TRUE(succeeded) << "Failed to run PESQ.";
-  if (succeeded) {
-    perf_test::PrintResult(
-        "audio_pesq", perf_modifier, "raw_mos", raw_mos, "score", true);
-    perf_test::PrintResult(
-        "audio_pesq", perf_modifier, "mos_lqo", mos_lqo, "score", true);
-  }
-
-  DeleteFileUnlessTestFailed(trimmed_reference, false);
-  DeleteFileUnlessTestFailed(trimmed_recording, false);
-}
-
-}  // namespace
-
-// Sets up a two-way WebRTC call and records its output to |recording|, using
-// getUserMedia.
-//
-// |reference_file| should have at least five seconds of silence in the
-// beginning: otherwise all the reference audio will not be picked up by the
-// recording. Note that the reference file will start playing as soon as the
-// audio device is up following the getUserMedia call in the left tab. The time
-// it takes to negotiate a call isn't deterministic, but five seconds should be
-// plenty of time. Similarly, the recording time should be enough to catch the
-// whole reference file. If you then silence-trim the reference file and actual
-// file, you should end up with two time-synchronized files.
-void MAYBE_WebRtcAudioQualityBrowserTest::SetupAndRecordAudioCall(
-    const base::FilePath& reference_file,
-    const base::FilePath& recording,
-    const std::string& constraints,
-    const base::TimeDelta recording_time) {
-  ASSERT_TRUE(embedded_test_server()->Start());
-  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
-  ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());
-
-  ConfigureFakeDeviceToPlayFile(reference_file);
-
-  // Create a two-way call. Mute one of the receivers though; that way it will
-  // be receiving audio bytes, but we will not be playing out of both elements.
-  GURL test_page = embedded_test_server()->GetURL(kWebRtcAudioTestHtmlPage);
-  content::WebContents* left_tab =
-      OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints);
-  SetupPeerconnectionWithLocalStream(left_tab);
-  MuteMediaElement("remote-view", left_tab);
-
-  content::WebContents* right_tab =
-      OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints);
-  SetupPeerconnectionWithLocalStream(right_tab);
-
-  AudioRecorder recorder;
-  ASSERT_TRUE(recorder.StartRecording(recording_time, recording));
-
-  NegotiateCall(left_tab, right_tab);
-
-  ASSERT_TRUE(recorder.WaitForRecordingToEnd());
-  DVLOG(0) << "Done recording to " << recording.value() << std::endl;
-
-  HangUp(left_tab);
-}
-
-void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia(
-    const std::string& constraints,
-    const std::string& perf_modifier) {
-  if (OnWin8()) {
-    // http://crbug.com/379798.
-    LOG(ERROR) << "This test is not implemented for Windows XP/Win8.";
-    return;
-  }
-
-  base::FilePath reference_file =
-      test::GetReferenceFilesDir().Append(kReferenceFile);
-  base::FilePath recording = CreateTemporaryWaveFile();
-
-  ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall(
-      reference_file, recording, constraints,
-      base::TimeDelta::FromSeconds(30)));
-
-  ComputeAndPrintPesqResults(reference_file, recording, perf_modifier);
-  DeleteFileUnlessTestFailed(recording, false);
-}
-
-IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
-                       MANUAL_TestCallQualityWithAudioFromFakeDevice) {
-  TestWithFakeDeviceGetUserMedia(kAudioOnlyCallConstraints, "_getusermedia");
-}
-
-IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
-                       MANUAL_TestCallQualityWithAudioFromWebAudio) {
-  if (OnWin8()) {
-    // http://crbug.com/379798.
-    LOG(ERROR) << "This test is not implemented for Windows XP/Win8.";
-    return;
-  }
-  ASSERT_TRUE(test::HasReferenceFilesInCheckout());
-  ASSERT_TRUE(embedded_test_server()->Start());
-
-  ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());
-
-  content::WebContents* left_tab =
-      OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);
-  content::WebContents* right_tab =
-      OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);
-
-  AddAudioFileToWebAudio(kReferenceFileRelativeUrl, left_tab);
-
-  NegotiateCall(left_tab, right_tab);
-
-  base::FilePath recording = CreateTemporaryWaveFile();
-
-  // Note: the sound clip is 21.6 seconds: record for 25 seconds to get some
-  // safety margins on each side.
-  AudioRecorder recorder;
-  ASSERT_TRUE(recorder.StartRecording(base::TimeDelta::FromSeconds(25),
-                                      recording));
-
-  PlayAudioFileThroughWebAudio(left_tab);
-
-  ASSERT_TRUE(recorder.WaitForRecordingToEnd());
-  DVLOG(0) << "Done recording to " << recording.value() << std::endl;
-
-  HangUp(left_tab);
-
-  // Compare with the reference file on disk (this is the same file we played
-  // through WebAudio earlier).
-  base::FilePath reference_file =
-      test::GetReferenceFilesDir().Append(kReferenceFile);
-  ComputeAndPrintPesqResults(reference_file, recording, "_webaudio");
-}
-
-/**
- * The auto gain control test plays a file into the fake microphone. Then it
- * sets up a one-way WebRTC call with audio only and records Chrome's output on
- * the receiving side using the audio loopback provided by the quality test
- * (see the class comments for more details).
- *
- * Then both the recording and reference file are split on silence. This creates
- * a number of segments with speech in them. The reason for this is to provide
- * a kind of synchronization mechanism so the start of each speech segment is
- * compared to the start of the corresponding speech segment. This is because we
- * will experience inevitable clock drift between the system clock (which runs
- * the fake microphone) and the sound card (which runs play-out). Effectively
- * re-synchronizing on each segment mitigates this.
- *
- * The silence splitting is inherently sensitive to the sound file we run on.
- * Therefore the reference file must have at least 500 ms of pure silence
- * between speech segments; the test will fail if the output produces more
- * segments than the reference.
- *
- * The test reports the difference in decibel between the reference and output
- * file per 10 ms interval in each speech segment. A value of 6 means the
- * output was 6 dB louder than the reference, presumably because the AGC applied
- * gain to the signal.
- *
- * The test only exercises digital AGC for now.
- *
- * We record in CD format here (44.1 kHz) because that's what the fake input
- * device currently supports, and we want to be able to compare directly. See
- * http://crbug.com/421054.
- */
-void MAYBE_WebRtcAudioQualityBrowserTest::TestAutoGainControl(
-    const base::FilePath::StringType& reference_filename,
-    const std::string& constraints,
-    const std::string& perf_modifier) {
-  if (OnWin8()) {
-    // http://crbug.com/379798.
-    LOG(ERROR) << "This test is not implemented for Windows XP/Win8.";
-    return;
-  }
-  base::FilePath reference_file =
-      test::GetReferenceFilesDir().Append(reference_filename);
-  base::FilePath recording = CreateTemporaryWaveFile();
-
-  ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall(
-      reference_file, recording, constraints,
-      base::TimeDelta::FromSeconds(30)));
-
-  base::ScopedTempDir split_ref_files;
-  ASSERT_TRUE(split_ref_files.CreateUniqueTempDir());
-  ASSERT_NO_FATAL_FAILURE(
-      SplitFileOnSilenceIntoDir(reference_file, split_ref_files.path()));
-  std::vector<base::FilePath> ref_segments =
-      ListWavFilesInDir(split_ref_files.path());
-
-  base::ScopedTempDir split_actual_files;
-  ASSERT_TRUE(split_actual_files.CreateUniqueTempDir());
-  ASSERT_NO_FATAL_FAILURE(
-      SplitFileOnSilenceIntoDir(recording, split_actual_files.path()));
-
-  // Keep the recording and split files if the analysis fails.
-  base::FilePath actual_files_dir = split_actual_files.Take();
-  std::vector<base::FilePath> actual_segments =
-      ListWavFilesInDir(actual_files_dir);
-
-  AnalyzeSegmentsAndPrintResult(
-      ref_segments, actual_segments, reference_file, perf_modifier);
-
-  DeleteFileUnlessTestFailed(recording, false);
-  DeleteFileUnlessTestFailed(actual_files_dir, true);
-}
-
-// The AGC should apply non-zero gain here.
-IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
-                       MANUAL_TestAutoGainControlOnLowAudio) {
-  ASSERT_NO_FATAL_FAILURE(TestAutoGainControl(
-      kReferenceFile, kAudioOnlyCallConstraints, "_with_agc"));
-}
-
-// Since the AGC is off here there should be no gain at all.
-IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,
-                       MANUAL_TestAutoGainIsOffWithAudioProcessingOff) {
-  const char* kAudioCallWithoutAudioProcessing =
-      "{audio: { mandatory: { echoCancellation: false } } }";
-  ASSERT_NO_FATAL_FAILURE(TestAutoGainControl(
-      kReferenceFile, kAudioCallWithoutAudioProcessing, "_no_agc"));
-}