chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc - Issue 2753543010: WebRTC: Use the MediaStream Recording API for the audio_quality_browsertest.

Unified Diff: chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc

Issue 2753543010: WebRTC: Use the MediaStream Recording API for the audio_quality_browsertest. (Closed)

Patch Set: Removed obsolete comments. Created 3 years, 9 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Index: chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc

diff --git a/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc b/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc

index 8107494b852a4fe7df29dbdfdbec0877ab1eee6a..f34221b275dcd36704bb0eba3d5ce512831093e4 100644

--- a/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc

+++ b/chrome/browser/media/webrtc/webrtc_audio_quality_browsertest.cc

@@ -6,6 +6,7 @@

#include <ctime>

+#include "base/base64.h"

#include "base/command_line.h"

#include "base/files/file_enumerator.h"

#include "base/files/file_util.h"

@@ -50,6 +51,9 @@ static const char kReferenceFileRelativeUrl[] =

static const char kWebRtcAudioTestHtmlPage[] =

"/webrtc/webrtc_audio_quality_test.html";

+// How often to ask the test page whether the audio recording is completed.

+const int kPollingIntervalInMs = 1000;

// 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.

@@ -76,23 +80,8 @@ const int kMaxAgcSegmentDiffMs =

// 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.

+// 1. # sudo apt-get install sox

// On Mac:

// TODO(phoglund): download sox from gs instead.

@@ -100,35 +89,10 @@ const int kMaxAgcSegmentDiffMs =

// 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

+// 5. (For Chrome bots) Ensure sox is reachable from the env the test

+// executes in (sox tends to install in /opt/, which generally isn't in the

+// Chrome bots' env). For instance, run

// 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 'stereo mix' device - this is what your speakers output.

-// If you don't have one, your driver doesn't support stereo mix devices.

-// Some drivers use different names for the mix device though (like "Wave").

-// Right click > Properties.

-// 4. Ensure "listen to this device" is unchecked, otherwise you get echo.

-// 5. Ensure the mix device is the default recording device.

-// 6. 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() {}

@@ -186,147 +150,22 @@ class MAYBE_WebRtcAudioQualityBrowserTest : public WebRtcTestBase {

"setMediaElementMuted('" + element_id + "', true)", tab_contents));

}

+ void WriteCapturedAudio(content::WebContents* capturing_tab,

+ const base::FilePath& audio_filename);

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,

mcasas 2017/03/21 16:54:22 |recording| is confusing, what about s/recording/r

- const std::string& constraints,

- const base::TimeDelta recording_time);

+ const std::string& constraints);

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() {

@@ -387,6 +226,34 @@ bool RemoveSilence(const base::FilePath& input_file,

return ok;

}

+// Runs ffmpeg on the captured webm video and writes it to a .wav file.

+bool RunWebmToWavConverter(const base::FilePath& webm_audio_filename,

+ const base::FilePath& wav_audio_filename) {

+ base::FilePath path_to_ffmpeg = test::GetToolForPlatform("ffmpeg");

mcasas 2017/03/21 16:54:22 nit: const here and in l.252, l.486, l.493, 495 et

+ if (!base::PathExists(path_to_ffmpeg)) {

+ LOG(ERROR) << "Missing ffmpeg: should be in " << path_to_ffmpeg.value();

+ return false;

+ }

+ // Set up ffmpeg to output at a certain bitrate (-ab). This is hopefully set

+ // high enough to avoid degrading audio quality too much.

+ base::CommandLine ffmpeg_command(path_to_ffmpeg);

+ ffmpeg_command.AppendArg("-i");

+ ffmpeg_command.AppendArgPath(webm_audio_filename);

+ ffmpeg_command.AppendArg("-ab");

+ ffmpeg_command.AppendArg("300k");

+ ffmpeg_command.AppendArg("-y");

+ ffmpeg_command.AppendArgPath(wav_audio_filename);

+ // We produce an output file that will later be used as an input to the

+ // barcode decoder and frame analyzer tools.

+ DVLOG(0) << "Running " << ffmpeg_command.GetCommandLineString();

+ std::string result;

+ bool ok = base::GetAppOutputAndError(ffmpeg_command, &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

@@ -613,6 +480,25 @@ void ComputeAndPrintPesqResults(const base::FilePath& reference_file,

} // namespace

+void MAYBE_WebRtcAudioQualityBrowserTest::WriteCapturedAudio(

+ content::WebContents* capturing_tab,

+ const base::FilePath& audio_filename) {

+ base::FilePath audio_filename_webm =

+ audio_filename.AddExtension(FILE_PATH_LITERAL(".webm"));

+ std::string base64_encoded_audio =

+ ExecuteJavascript("getRecordedAudioAsBase64()", capturing_tab);

+ std::string recorded_audio;

+ ASSERT_TRUE(base::Base64Decode(base64_encoded_audio, &recorded_audio));

+ base::File audio_file(audio_filename_webm,

+ base::File::FLAG_CREATE | base::File::FLAG_WRITE);

+ size_t written =

+ audio_file.Write(0, recorded_audio.c_str(), recorded_audio.length());

+ ASSERT_EQ(recorded_audio.length(), written);

+ RunWebmToWavConverter(audio_filename_webm, audio_filename);

// Sets up a two-way WebRTC call and records its output to |recording|, using

// getUserMedia.

@@ -627,11 +513,9 @@ void ComputeAndPrintPesqResults(const base::FilePath& reference_file,

void MAYBE_WebRtcAudioQualityBrowserTest::SetupAndRecordAudioCall(

const base::FilePath& reference_file,

const base::FilePath& recording,

- const std::string& constraints,

- const base::TimeDelta recording_time) {

+ const std::string& constraints) {

ASSERT_TRUE(embedded_test_server()->Start());

ASSERT_TRUE(test::HasReferenceFilesInCheckout());

- ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());

ConfigureFakeDeviceToPlayFile(reference_file);

@@ -647,15 +531,16 @@ void MAYBE_WebRtcAudioQualityBrowserTest::SetupAndRecordAudioCall(

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;

+ EXPECT_TRUE(test::PollingWaitUntil("doneCapturingAudio()", "done-capturing",

+ right_tab, kPollingIntervalInMs));

HangUp(left_tab);

+ WriteCapturedAudio(right_tab, recording);

+ DVLOG(0) << "Done recording to " << recording.value() << std::endl;

mcasas 2017/03/21 16:54:22 s/recording.value()/recording.MaybeAsASCII()/ and

}

void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia(

@@ -671,9 +556,8 @@ void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia(

test::GetReferenceFilesDir().Append(kReferenceFile);

base::FilePath recording = CreateTemporaryWaveFile();

- ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall(

- reference_file, recording, constraints,

- base::TimeDelta::FromSeconds(30)));

+ ASSERT_NO_FATAL_FAILURE(

+ SetupAndRecordAudioCall(reference_file, recording, constraints));

ComputeAndPrintPesqResults(reference_file, recording, perf_modifier);

DeleteFileUnlessTestFailed(recording, false);

@@ -694,8 +578,6 @@ IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,

ASSERT_TRUE(test::HasReferenceFilesInCheckout());

ASSERT_TRUE(embedded_test_server()->Start());

- ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent());

content::WebContents* left_tab =

OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage);

content::WebContents* right_tab =

@@ -707,19 +589,17 @@ IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest,

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;

+ EXPECT_TRUE(test::PollingWaitUntil("doneCapturingAudio()", "done-capturing",

+ right_tab, kPollingIntervalInMs));

HangUp(left_tab);

+ WriteCapturedAudio(right_tab, recording);

+ DVLOG(0) << "Done recording to " << recording.value() << std::endl;

// Compare with the reference file on disk (this is the same file we played

// through WebAudio earlier).

base::FilePath reference_file =

@@ -770,9 +650,8 @@ void MAYBE_WebRtcAudioQualityBrowserTest::TestAutoGainControl(

test::GetReferenceFilesDir().Append(reference_filename);

base::FilePath recording = CreateTemporaryWaveFile();

- ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall(

- reference_file, recording, constraints,

- base::TimeDelta::FromSeconds(30)));

+ ASSERT_NO_FATAL_FAILURE(

+ SetupAndRecordAudioCall(reference_file, recording, constraints));

base::ScopedTempDir split_ref_files;

ASSERT_TRUE(split_ref_files.CreateUniqueTempDir());

« no previous file with comments | « no previous file | chrome/test/data/webrtc/audio_extraction.js » ('j') | chrome/test/data/webrtc/audio_extraction.js » ('J')