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..2995fa3dafa4a584c62030f35421e5277de6c054 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. |
@@ -80,19 +84,8 @@ const int kMaxAgcSegmentDiffMs = |
// their own input. |
// |
// On Linux: |
-// 1. # sudo apt-get install pavucontrol sox |
+// 1. # sudo apt-get install 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. |
@@ -100,35 +93,11 @@ 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 |
+// 5. (For Chrome bots) Ensure sox is 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 |
kjellander_chromium
2017/03/21 14:07:26
Remove this line too :)
|
// 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 +155,22 @@ 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, |
- 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 +231,34 @@ 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"); |
+ 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 +485,25 @@ 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 +518,9 @@ 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 +536,16 @@ 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; |
} |
void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia( |
@@ -671,9 +561,8 @@ 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 +583,6 @@ 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 +594,17 @@ 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 +655,8 @@ 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()); |