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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include <stddef.h> | 5 #include <stddef.h> |
6 | 6 |
7 #include <ctime> | 7 #include <ctime> |
8 | 8 |
9 #include "base/base64.h" | |
9 #include "base/command_line.h" | 10 #include "base/command_line.h" |
10 #include "base/files/file_enumerator.h" | 11 #include "base/files/file_enumerator.h" |
11 #include "base/files/file_util.h" | 12 #include "base/files/file_util.h" |
12 #include "base/files/scoped_temp_dir.h" | 13 #include "base/files/scoped_temp_dir.h" |
13 #include "base/macros.h" | 14 #include "base/macros.h" |
14 #include "base/process/launch.h" | 15 #include "base/process/launch.h" |
15 #include "base/process/process.h" | 16 #include "base/process/process.h" |
16 #include "base/scoped_native_library.h" | 17 #include "base/scoped_native_library.h" |
17 #include "base/strings/string_number_conversions.h" | 18 #include "base/strings/string_number_conversions.h" |
18 #include "base/strings/string_util.h" | 19 #include "base/strings/string_util.h" |
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43 | 44 |
44 // The javascript will load the reference file relative to its location, | 45 // The javascript will load the reference file relative to its location, |
45 // which is in /webrtc on the web server. The files we are looking for are in | 46 // which is in /webrtc on the web server. The files we are looking for are in |
46 // webrtc/resources in the chrome/test/data folder. | 47 // webrtc/resources in the chrome/test/data folder. |
47 static const char kReferenceFileRelativeUrl[] = | 48 static const char kReferenceFileRelativeUrl[] = |
48 "resources/speech_44kHz_16bit_stereo.wav"; | 49 "resources/speech_44kHz_16bit_stereo.wav"; |
49 | 50 |
50 static const char kWebRtcAudioTestHtmlPage[] = | 51 static const char kWebRtcAudioTestHtmlPage[] = |
51 "/webrtc/webrtc_audio_quality_test.html"; | 52 "/webrtc/webrtc_audio_quality_test.html"; |
52 | 53 |
54 // How often to ask the test page whether the audio recording is completed. | |
55 const int kPollingIntervalInMs = 1000; | |
56 | |
53 // For the AGC test, there are 6 speech segments split on silence. If one | 57 // For the AGC test, there are 6 speech segments split on silence. If one |
54 // segment is significantly different in length compared to the same segment in | 58 // segment is significantly different in length compared to the same segment in |
55 // the reference file, there's something fishy going on. | 59 // the reference file, there's something fishy going on. |
56 const int kMaxAgcSegmentDiffMs = | 60 const int kMaxAgcSegmentDiffMs = |
57 #if defined(OS_MACOSX) | 61 #if defined(OS_MACOSX) |
58 // Something is different on Mac; http://crbug.com/477653. | 62 // Something is different on Mac; http://crbug.com/477653. |
59 600; | 63 600; |
60 #else | 64 #else |
61 200; | 65 200; |
62 #endif | 66 #endif |
63 | 67 |
64 #if defined(OS_LINUX) || defined(OS_WIN) || defined(OS_MACOSX) | 68 #if defined(OS_LINUX) || defined(OS_WIN) || defined(OS_MACOSX) |
65 #define MAYBE_WebRtcAudioQualityBrowserTest WebRtcAudioQualityBrowserTest | 69 #define MAYBE_WebRtcAudioQualityBrowserTest WebRtcAudioQualityBrowserTest |
66 #else | 70 #else |
67 // Not implemented on Android, ChromeOS etc. | 71 // Not implemented on Android, ChromeOS etc. |
68 #define MAYBE_WebRtcAudioQualityBrowserTest DISABLED_WebRtcAudioQualityBrowserTe st | 72 #define MAYBE_WebRtcAudioQualityBrowserTest DISABLED_WebRtcAudioQualityBrowserTe st |
69 #endif | 73 #endif |
70 | 74 |
71 } // namespace | 75 } // namespace |
72 | 76 |
73 // Test we can set up a WebRTC call and play audio through it. | 77 // Test we can set up a WebRTC call and play audio through it. |
74 // | 78 // |
75 // If you're not a googler and want to run this test, you need to provide a | 79 // If you're not a googler and want to run this test, you need to provide a |
76 // pesq binary for your platform (and sox.exe on windows). Read more on how | 80 // pesq binary for your platform (and sox.exe on windows). Read more on how |
77 // resources are managed in chrome/test/data/webrtc/resources/README. | 81 // resources are managed in chrome/test/data/webrtc/resources/README. |
78 // | 82 // |
79 // This test will only work on machines that have been configured to record | |
80 // their own input. | |
81 // | |
82 // On Linux: | 83 // On Linux: |
83 // 1. # sudo apt-get install pavucontrol sox | 84 // 1. # sudo apt-get install sox |
84 // 2. For the user who will run the test: # pavucontrol | |
85 // 3. In a separate terminal, # arecord dummy | |
86 // 4. In pavucontrol, go to the recording tab. | |
87 // 5. For the ALSA plugin [aplay]: ALSA Capture from, change from <x> to | |
88 // <Monitor of x>, where x is whatever your primary sound device is called. | |
89 // 6. Try launching chrome as the target user on the target machine, try | |
90 // playing, say, a YouTube video, and record with # arecord -f dat tmp.dat. | |
91 // Verify the recording with aplay (should have recorded what you played | |
92 // from chrome). | |
93 // | |
94 // Note: the volume for ALL your input devices will be forced to 100% by | |
95 // running this test on Linux. | |
96 // | 85 // |
97 // On Mac: | 86 // On Mac: |
98 // TODO(phoglund): download sox from gs instead. | 87 // TODO(phoglund): download sox from gs instead. |
99 // 1. Get SoundFlower: http://rogueamoeba.com/freebies/soundflower/download.php | 88 // 1. Get SoundFlower: http://rogueamoeba.com/freebies/soundflower/download.php |
100 // 2. Install it + reboot. | 89 // 2. Install it + reboot. |
101 // 3. Install MacPorts (http://www.macports.org/). | 90 // 3. Install MacPorts (http://www.macports.org/). |
102 // 4. Install sox: sudo port install sox. | 91 // 4. Install sox: sudo port install sox. |
103 // 5. (For Chrome bots) Ensure sox and rec are reachable from the env the test | 92 // 5. (For Chrome bots) Ensure sox is reachable from the env the test |
104 // executes in (sox and rec tends to install in /opt/, which generally isn't | 93 // executes in (sox tends to install in /opt/, which generally isn't in the |
105 // in the Chrome bots' env). For instance, run | 94 // Chrome bots' env). For instance, run |
106 // sudo ln -s /opt/local/bin/rec /usr/local/bin/rec | |
107 // sudo ln -s /opt/local/bin/sox /usr/local/bin/sox | 95 // sudo ln -s /opt/local/bin/sox /usr/local/bin/sox |
108 // 6. In Sound Preferences, set both input and output to Soundflower (2ch). | |
109 // Note: You will no longer hear audio on this machine, and it will no | |
110 // longer use any built-in mics. | |
111 // 7. Try launching chrome as the target user on the target machine, try | |
112 // playing, say, a YouTube video, and record with 'rec test.wav trim 0 5'. | |
113 // Stop the video in chrome and try playing back the file; you should hear | |
114 // a recording of the video (note; if you play back on the target machine | |
115 // you must revert the changes in step 3 first). | |
116 // | |
117 // On Windows 7: | |
118 // 1. Control panel > Sound > Manage audio devices. | |
119 // 2. In the recording tab, right-click in an empty space in the pane with the | |
120 // devices. Tick 'show disabled devices'. | |
121 // 3. You should see a 'stereo mix' device - this is what your speakers output. | |
122 // If you don't have one, your driver doesn't support stereo mix devices. | |
123 // Some drivers use different names for the mix device though (like "Wave"). | |
124 // Right click > Properties. | |
125 // 4. Ensure "listen to this device" is unchecked, otherwise you get echo. | |
126 // 5. Ensure the mix device is the default recording device. | |
127 // 6. Launch chrome and try playing a video with sound. You should see | |
128 // in the volume meter for the mix device. Configure the mix device to have | |
129 // 50 / 100 in level. Also go into the playback tab, right-click Speakers, | |
130 // and set that level to 50 / 100. Otherwise you will get distortion in | |
131 // the recording. | |
132 class MAYBE_WebRtcAudioQualityBrowserTest : public WebRtcTestBase { | 96 class MAYBE_WebRtcAudioQualityBrowserTest : public WebRtcTestBase { |
133 public: | 97 public: |
134 MAYBE_WebRtcAudioQualityBrowserTest() {} | 98 MAYBE_WebRtcAudioQualityBrowserTest() {} |
135 void SetUpInProcessBrowserTestFixture() override { | 99 void SetUpInProcessBrowserTestFixture() override { |
136 DetectErrorsInJavaScript(); // Look for errors in our rather complex js. | 100 DetectErrorsInJavaScript(); // Look for errors in our rather complex js. |
137 } | 101 } |
138 | 102 |
139 void SetUpCommandLine(base::CommandLine* command_line) override { | 103 void SetUpCommandLine(base::CommandLine* command_line) override { |
140 EXPECT_FALSE(command_line->HasSwitch( | 104 EXPECT_FALSE(command_line->HasSwitch( |
141 switches::kUseFakeUIForMediaStream)); | 105 switches::kUseFakeUIForMediaStream)); |
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179 ExecuteJavascript("preparePeerConnection()", tab)); | 143 ExecuteJavascript("preparePeerConnection()", tab)); |
180 return tab; | 144 return tab; |
181 } | 145 } |
182 | 146 |
183 void MuteMediaElement(const std::string& element_id, | 147 void MuteMediaElement(const std::string& element_id, |
184 content::WebContents* tab_contents) { | 148 content::WebContents* tab_contents) { |
185 EXPECT_EQ("ok-muted", ExecuteJavascript( | 149 EXPECT_EQ("ok-muted", ExecuteJavascript( |
186 "setMediaElementMuted('" + element_id + "', true)", tab_contents)); | 150 "setMediaElementMuted('" + element_id + "', true)", tab_contents)); |
187 } | 151 } |
188 | 152 |
153 void WriteCapturedAudio(content::WebContents* capturing_tab, | |
154 const base::FilePath& audio_filename); | |
155 | |
189 protected: | 156 protected: |
190 void TestAutoGainControl(const base::FilePath::StringType& reference_filename, | 157 void TestAutoGainControl(const base::FilePath::StringType& reference_filename, |
191 const std::string& constraints, | 158 const std::string& constraints, |
192 const std::string& perf_modifier); | 159 const std::string& perf_modifier); |
193 void SetupAndRecordAudioCall(const base::FilePath& reference_file, | 160 void SetupAndRecordAudioCall(const base::FilePath& reference_file, |
194 const base::FilePath& recording, | 161 const base::FilePath& recording, |
mcasas
2017/03/21 16:54:22
|recording| is confusing, what about
s/recording/r
| |
195 const std::string& constraints, | 162 const std::string& constraints); |
196 const base::TimeDelta recording_time); | |
197 void TestWithFakeDeviceGetUserMedia(const std::string& constraints, | 163 void TestWithFakeDeviceGetUserMedia(const std::string& constraints, |
198 const std::string& perf_modifier); | 164 const std::string& perf_modifier); |
199 }; | 165 }; |
200 | 166 |
201 namespace { | 167 namespace { |
202 | 168 |
203 class AudioRecorder { | |
204 public: | |
205 AudioRecorder() {} | |
206 ~AudioRecorder() {} | |
207 | |
208 // Starts the recording program for the specified duration. Returns true | |
209 // on success. We record in 16-bit 44.1 kHz Stereo (mostly because that's | |
210 // what SoundRecorder.exe will give us and we can't change that). | |
211 bool StartRecording(base::TimeDelta recording_time, | |
212 const base::FilePath& output_file) { | |
213 EXPECT_FALSE(recording_application_.IsValid()) | |
214 << "Tried to record, but is already recording."; | |
215 | |
216 int duration_sec = static_cast<int>(recording_time.InSeconds()); | |
217 base::CommandLine command_line(base::CommandLine::NO_PROGRAM); | |
218 | |
219 #if defined(OS_WIN) | |
220 // This disable is required to run SoundRecorder.exe on 64-bit Windows | |
221 // from a 32-bit binary. We need to load the wow64 disable function from | |
222 // the DLL since it doesn't exist on Windows XP. | |
223 base::ScopedNativeLibrary kernel32_lib(base::FilePath(L"kernel32")); | |
224 if (kernel32_lib.is_valid()) { | |
225 typedef BOOL (WINAPI* Wow64DisableWow64FSRedirection)(PVOID*); | |
226 Wow64DisableWow64FSRedirection wow_64_disable_wow_64_fs_redirection; | |
227 wow_64_disable_wow_64_fs_redirection = | |
228 reinterpret_cast<Wow64DisableWow64FSRedirection>( | |
229 kernel32_lib.GetFunctionPointer( | |
230 "Wow64DisableWow64FsRedirection")); | |
231 if (wow_64_disable_wow_64_fs_redirection != NULL) { | |
232 PVOID* ignored = NULL; | |
233 wow_64_disable_wow_64_fs_redirection(ignored); | |
234 } | |
235 } | |
236 | |
237 char duration_in_hms[128] = {0}; | |
238 struct tm duration_tm = {0}; | |
239 duration_tm.tm_sec = duration_sec; | |
240 EXPECT_NE(0u, strftime(duration_in_hms, arraysize(duration_in_hms), | |
241 "%H:%M:%S", &duration_tm)); | |
242 | |
243 command_line.SetProgram( | |
244 base::FilePath(FILE_PATH_LITERAL("SoundRecorder.exe"))); | |
245 command_line.AppendArg("/FILE"); | |
246 command_line.AppendArgPath(output_file); | |
247 command_line.AppendArg("/DURATION"); | |
248 command_line.AppendArg(duration_in_hms); | |
249 #elif defined(OS_MACOSX) | |
250 command_line.SetProgram(base::FilePath("rec")); | |
251 command_line.AppendArg("-b"); | |
252 command_line.AppendArg("16"); | |
253 command_line.AppendArg("-q"); | |
254 command_line.AppendArgPath(output_file); | |
255 command_line.AppendArg("trim"); | |
256 command_line.AppendArg("0"); | |
257 command_line.AppendArg(base::IntToString(duration_sec)); | |
258 #else | |
259 command_line.SetProgram(base::FilePath("arecord")); | |
260 command_line.AppendArg("-d"); | |
261 command_line.AppendArg(base::IntToString(duration_sec)); | |
262 command_line.AppendArg("-f"); | |
263 command_line.AppendArg("cd"); | |
264 command_line.AppendArg("-c"); | |
265 command_line.AppendArg("2"); | |
266 command_line.AppendArgPath(output_file); | |
267 #endif | |
268 | |
269 DVLOG(0) << "Running " << command_line.GetCommandLineString(); | |
270 recording_application_ = | |
271 base::LaunchProcess(command_line, base::LaunchOptions()); | |
272 return recording_application_.IsValid(); | |
273 } | |
274 | |
275 // Joins the recording program. Returns true on success. | |
276 bool WaitForRecordingToEnd() { | |
277 int exit_code = -1; | |
278 recording_application_.WaitForExit(&exit_code); | |
279 return exit_code == 0; | |
280 } | |
281 private: | |
282 base::Process recording_application_; | |
283 }; | |
284 | |
285 bool ForceMicrophoneVolumeTo100Percent() { | |
286 #if defined(OS_WIN) | |
287 // Note: the force binary isn't in tools since it's one of our own. | |
288 base::CommandLine command_line(test::GetReferenceFilesDir().Append( | |
289 FILE_PATH_LITERAL("force_mic_volume_max.exe"))); | |
290 DVLOG(0) << "Running " << command_line.GetCommandLineString(); | |
291 std::string result; | |
292 if (!base::GetAppOutput(command_line, &result)) { | |
293 LOG(ERROR) << "Failed to set source volume: output was " << result; | |
294 return false; | |
295 } | |
296 #elif defined(OS_MACOSX) | |
297 base::CommandLine command_line( | |
298 base::FilePath(FILE_PATH_LITERAL("osascript"))); | |
299 command_line.AppendArg("-e"); | |
300 command_line.AppendArg("set volume input volume 100"); | |
301 command_line.AppendArg("-e"); | |
302 command_line.AppendArg("set volume output volume 85"); | |
303 | |
304 std::string result; | |
305 if (!base::GetAppOutput(command_line, &result)) { | |
306 LOG(ERROR) << "Failed to set source volume: output was " << result; | |
307 return false; | |
308 } | |
309 #else | |
310 // Just force the volume of, say the first 5 devices. A machine will rarely | |
311 // have more input sources than that. This is way easier than finding the | |
312 // input device we happen to be using. | |
313 for (int device_index = 0; device_index < 5; ++device_index) { | |
314 std::string result; | |
315 const std::string kHundredPercentVolume = "65536"; | |
316 base::CommandLine command_line(base::FilePath(FILE_PATH_LITERAL("pacmd"))); | |
317 command_line.AppendArg("set-source-volume"); | |
318 command_line.AppendArg(base::IntToString(device_index)); | |
319 command_line.AppendArg(kHundredPercentVolume); | |
320 DVLOG(0) << "Running " << command_line.GetCommandLineString(); | |
321 if (!base::GetAppOutput(command_line, &result)) { | |
322 LOG(ERROR) << "Failed to set source volume: output was " << result; | |
323 return false; | |
324 } | |
325 } | |
326 #endif | |
327 return true; | |
328 } | |
329 | |
330 // Sox is the "Swiss army knife" of audio processing. We mainly use it for | 169 // Sox is the "Swiss army knife" of audio processing. We mainly use it for |
331 // silence trimming. See http://sox.sourceforge.net. | 170 // silence trimming. See http://sox.sourceforge.net. |
332 base::CommandLine MakeSoxCommandLine() { | 171 base::CommandLine MakeSoxCommandLine() { |
333 #if defined(OS_WIN) | 172 #if defined(OS_WIN) |
334 base::FilePath sox_path = test::GetToolForPlatform("sox"); | 173 base::FilePath sox_path = test::GetToolForPlatform("sox"); |
335 if (!base::PathExists(sox_path)) { | 174 if (!base::PathExists(sox_path)) { |
336 LOG(ERROR) << "Missing sox.exe binary in " << sox_path.value() | 175 LOG(ERROR) << "Missing sox.exe binary in " << sox_path.value() |
337 << "; you may have to provide this binary yourself."; | 176 << "; you may have to provide this binary yourself."; |
338 return base::CommandLine(base::CommandLine::NO_PROGRAM); | 177 return base::CommandLine(base::CommandLine::NO_PROGRAM); |
339 } | 178 } |
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380 command_line.AppendArg(kTreshold); | 219 command_line.AppendArg(kTreshold); |
381 command_line.AppendArg("reverse"); | 220 command_line.AppendArg("reverse"); |
382 | 221 |
383 DVLOG(0) << "Running " << command_line.GetCommandLineString(); | 222 DVLOG(0) << "Running " << command_line.GetCommandLineString(); |
384 std::string result; | 223 std::string result; |
385 bool ok = base::GetAppOutput(command_line, &result); | 224 bool ok = base::GetAppOutput(command_line, &result); |
386 DVLOG(0) << "Output was:\n\n" << result; | 225 DVLOG(0) << "Output was:\n\n" << result; |
387 return ok; | 226 return ok; |
388 } | 227 } |
389 | 228 |
229 // Runs ffmpeg on the captured webm video and writes it to a .wav file. | |
230 bool RunWebmToWavConverter(const base::FilePath& webm_audio_filename, | |
231 const base::FilePath& wav_audio_filename) { | |
232 base::FilePath path_to_ffmpeg = test::GetToolForPlatform("ffmpeg"); | |
mcasas
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nit: const here and in l.252, l.486, l.493, 495 et
| |
233 if (!base::PathExists(path_to_ffmpeg)) { | |
234 LOG(ERROR) << "Missing ffmpeg: should be in " << path_to_ffmpeg.value(); | |
235 return false; | |
236 } | |
237 | |
238 // Set up ffmpeg to output at a certain bitrate (-ab). This is hopefully set | |
239 // high enough to avoid degrading audio quality too much. | |
240 base::CommandLine ffmpeg_command(path_to_ffmpeg); | |
241 ffmpeg_command.AppendArg("-i"); | |
242 ffmpeg_command.AppendArgPath(webm_audio_filename); | |
243 ffmpeg_command.AppendArg("-ab"); | |
244 ffmpeg_command.AppendArg("300k"); | |
245 ffmpeg_command.AppendArg("-y"); | |
246 ffmpeg_command.AppendArgPath(wav_audio_filename); | |
247 | |
248 // We produce an output file that will later be used as an input to the | |
249 // barcode decoder and frame analyzer tools. | |
250 DVLOG(0) << "Running " << ffmpeg_command.GetCommandLineString(); | |
251 std::string result; | |
252 bool ok = base::GetAppOutputAndError(ffmpeg_command, &result); | |
253 DVLOG(0) << "Output was:\n\n" << result; | |
254 return ok; | |
255 } | |
256 | |
390 // Looks for 0.2 second audio segments surrounded by silences under 0.3% audio | 257 // Looks for 0.2 second audio segments surrounded by silences under 0.3% audio |
391 // power and splits the input file on those silences. Output files are written | 258 // power and splits the input file on those silences. Output files are written |
392 // according to the output file template (e.g. /tmp/out.wav writes | 259 // according to the output file template (e.g. /tmp/out.wav writes |
393 // /tmp/out001.wav, /tmp/out002.wav, etc if there are two silence-padded | 260 // /tmp/out001.wav, /tmp/out002.wav, etc if there are two silence-padded |
394 // regions in the file). The silences between speech segments must be at | 261 // regions in the file). The silences between speech segments must be at |
395 // least 500 ms for this to be reliable. | 262 // least 500 ms for this to be reliable. |
396 bool SplitFileOnSilence(const base::FilePath& input_file, | 263 bool SplitFileOnSilence(const base::FilePath& input_file, |
397 const base::FilePath& output_file_template) { | 264 const base::FilePath& output_file_template) { |
398 base::CommandLine command_line = MakeSoxCommandLine(); | 265 base::CommandLine command_line = MakeSoxCommandLine(); |
399 if (command_line.GetProgram().empty()) | 266 if (command_line.GetProgram().empty()) |
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606 perf_test::PrintResult( | 473 perf_test::PrintResult( |
607 "audio_pesq", perf_modifier, "mos_lqo", mos_lqo, "score", true); | 474 "audio_pesq", perf_modifier, "mos_lqo", mos_lqo, "score", true); |
608 } | 475 } |
609 | 476 |
610 DeleteFileUnlessTestFailed(trimmed_reference, false); | 477 DeleteFileUnlessTestFailed(trimmed_reference, false); |
611 DeleteFileUnlessTestFailed(trimmed_recording, false); | 478 DeleteFileUnlessTestFailed(trimmed_recording, false); |
612 } | 479 } |
613 | 480 |
614 } // namespace | 481 } // namespace |
615 | 482 |
483 void MAYBE_WebRtcAudioQualityBrowserTest::WriteCapturedAudio( | |
484 content::WebContents* capturing_tab, | |
485 const base::FilePath& audio_filename) { | |
486 base::FilePath audio_filename_webm = | |
487 audio_filename.AddExtension(FILE_PATH_LITERAL(".webm")); | |
488 | |
489 std::string base64_encoded_audio = | |
490 ExecuteJavascript("getRecordedAudioAsBase64()", capturing_tab); | |
491 std::string recorded_audio; | |
492 ASSERT_TRUE(base::Base64Decode(base64_encoded_audio, &recorded_audio)); | |
493 base::File audio_file(audio_filename_webm, | |
494 base::File::FLAG_CREATE | base::File::FLAG_WRITE); | |
495 size_t written = | |
496 audio_file.Write(0, recorded_audio.c_str(), recorded_audio.length()); | |
497 ASSERT_EQ(recorded_audio.length(), written); | |
498 | |
499 RunWebmToWavConverter(audio_filename_webm, audio_filename); | |
500 } | |
501 | |
616 // Sets up a two-way WebRTC call and records its output to |recording|, using | 502 // Sets up a two-way WebRTC call and records its output to |recording|, using |
617 // getUserMedia. | 503 // getUserMedia. |
618 // | 504 // |
619 // |reference_file| should have at least five seconds of silence in the | 505 // |reference_file| should have at least five seconds of silence in the |
620 // beginning: otherwise all the reference audio will not be picked up by the | 506 // beginning: otherwise all the reference audio will not be picked up by the |
621 // recording. Note that the reference file will start playing as soon as the | 507 // recording. Note that the reference file will start playing as soon as the |
622 // audio device is up following the getUserMedia call in the left tab. The time | 508 // audio device is up following the getUserMedia call in the left tab. The time |
623 // it takes to negotiate a call isn't deterministic, but five seconds should be | 509 // it takes to negotiate a call isn't deterministic, but five seconds should be |
624 // plenty of time. Similarly, the recording time should be enough to catch the | 510 // plenty of time. Similarly, the recording time should be enough to catch the |
625 // whole reference file. If you then silence-trim the reference file and actual | 511 // whole reference file. If you then silence-trim the reference file and actual |
626 // file, you should end up with two time-synchronized files. | 512 // file, you should end up with two time-synchronized files. |
627 void MAYBE_WebRtcAudioQualityBrowserTest::SetupAndRecordAudioCall( | 513 void MAYBE_WebRtcAudioQualityBrowserTest::SetupAndRecordAudioCall( |
628 const base::FilePath& reference_file, | 514 const base::FilePath& reference_file, |
629 const base::FilePath& recording, | 515 const base::FilePath& recording, |
630 const std::string& constraints, | 516 const std::string& constraints) { |
631 const base::TimeDelta recording_time) { | |
632 ASSERT_TRUE(embedded_test_server()->Start()); | 517 ASSERT_TRUE(embedded_test_server()->Start()); |
633 ASSERT_TRUE(test::HasReferenceFilesInCheckout()); | 518 ASSERT_TRUE(test::HasReferenceFilesInCheckout()); |
634 ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent()); | |
635 | 519 |
636 ConfigureFakeDeviceToPlayFile(reference_file); | 520 ConfigureFakeDeviceToPlayFile(reference_file); |
637 | 521 |
638 // Create a two-way call. Mute one of the receivers though; that way it will | 522 // Create a two-way call. Mute one of the receivers though; that way it will |
639 // be receiving audio bytes, but we will not be playing out of both elements. | 523 // be receiving audio bytes, but we will not be playing out of both elements. |
640 GURL test_page = embedded_test_server()->GetURL(kWebRtcAudioTestHtmlPage); | 524 GURL test_page = embedded_test_server()->GetURL(kWebRtcAudioTestHtmlPage); |
641 content::WebContents* left_tab = | 525 content::WebContents* left_tab = |
642 OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints); | 526 OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints); |
643 SetupPeerconnectionWithLocalStream(left_tab); | 527 SetupPeerconnectionWithLocalStream(left_tab); |
644 MuteMediaElement("remote-view", left_tab); | 528 MuteMediaElement("remote-view", left_tab); |
645 | 529 |
646 content::WebContents* right_tab = | 530 content::WebContents* right_tab = |
647 OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints); | 531 OpenPageAndGetUserMediaInNewTabWithConstraints(test_page, constraints); |
648 SetupPeerconnectionWithLocalStream(right_tab); | 532 SetupPeerconnectionWithLocalStream(right_tab); |
649 | 533 |
650 AudioRecorder recorder; | |
651 ASSERT_TRUE(recorder.StartRecording(recording_time, recording)); | |
652 | |
653 NegotiateCall(left_tab, right_tab); | 534 NegotiateCall(left_tab, right_tab); |
654 | 535 |
655 ASSERT_TRUE(recorder.WaitForRecordingToEnd()); | 536 EXPECT_TRUE(test::PollingWaitUntil("doneCapturingAudio()", "done-capturing", |
656 DVLOG(0) << "Done recording to " << recording.value() << std::endl; | 537 right_tab, kPollingIntervalInMs)); |
657 | 538 |
658 HangUp(left_tab); | 539 HangUp(left_tab); |
540 | |
541 WriteCapturedAudio(right_tab, recording); | |
542 | |
543 DVLOG(0) << "Done recording to " << recording.value() << std::endl; | |
mcasas
2017/03/21 16:54:22
s/recording.value()/recording.MaybeAsASCII()/ and
| |
659 } | 544 } |
660 | 545 |
661 void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia( | 546 void MAYBE_WebRtcAudioQualityBrowserTest::TestWithFakeDeviceGetUserMedia( |
662 const std::string& constraints, | 547 const std::string& constraints, |
663 const std::string& perf_modifier) { | 548 const std::string& perf_modifier) { |
664 if (OnWin8()) { | 549 if (OnWin8()) { |
665 // http://crbug.com/379798. | 550 // http://crbug.com/379798. |
666 LOG(ERROR) << "This test is not implemented for Windows XP/Win8."; | 551 LOG(ERROR) << "This test is not implemented for Windows XP/Win8."; |
667 return; | 552 return; |
668 } | 553 } |
669 | 554 |
670 base::FilePath reference_file = | 555 base::FilePath reference_file = |
671 test::GetReferenceFilesDir().Append(kReferenceFile); | 556 test::GetReferenceFilesDir().Append(kReferenceFile); |
672 base::FilePath recording = CreateTemporaryWaveFile(); | 557 base::FilePath recording = CreateTemporaryWaveFile(); |
673 | 558 |
674 ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall( | 559 ASSERT_NO_FATAL_FAILURE( |
675 reference_file, recording, constraints, | 560 SetupAndRecordAudioCall(reference_file, recording, constraints)); |
676 base::TimeDelta::FromSeconds(30))); | |
677 | 561 |
678 ComputeAndPrintPesqResults(reference_file, recording, perf_modifier); | 562 ComputeAndPrintPesqResults(reference_file, recording, perf_modifier); |
679 DeleteFileUnlessTestFailed(recording, false); | 563 DeleteFileUnlessTestFailed(recording, false); |
680 } | 564 } |
681 | 565 |
682 IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest, | 566 IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest, |
683 MANUAL_TestCallQualityWithAudioFromFakeDevice) { | 567 MANUAL_TestCallQualityWithAudioFromFakeDevice) { |
684 TestWithFakeDeviceGetUserMedia(kAudioOnlyCallConstraints, "_getusermedia"); | 568 TestWithFakeDeviceGetUserMedia(kAudioOnlyCallConstraints, "_getusermedia"); |
685 } | 569 } |
686 | 570 |
687 IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest, | 571 IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest, |
688 MANUAL_TestCallQualityWithAudioFromWebAudio) { | 572 MANUAL_TestCallQualityWithAudioFromWebAudio) { |
689 if (OnWin8()) { | 573 if (OnWin8()) { |
690 // http://crbug.com/379798. | 574 // http://crbug.com/379798. |
691 LOG(ERROR) << "This test is not implemented for Windows XP/Win8."; | 575 LOG(ERROR) << "This test is not implemented for Windows XP/Win8."; |
692 return; | 576 return; |
693 } | 577 } |
694 ASSERT_TRUE(test::HasReferenceFilesInCheckout()); | 578 ASSERT_TRUE(test::HasReferenceFilesInCheckout()); |
695 ASSERT_TRUE(embedded_test_server()->Start()); | 579 ASSERT_TRUE(embedded_test_server()->Start()); |
696 | 580 |
697 ASSERT_TRUE(ForceMicrophoneVolumeTo100Percent()); | |
698 | |
699 content::WebContents* left_tab = | 581 content::WebContents* left_tab = |
700 OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage); | 582 OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage); |
701 content::WebContents* right_tab = | 583 content::WebContents* right_tab = |
702 OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage); | 584 OpenPageWithoutGetUserMedia(kWebRtcAudioTestHtmlPage); |
703 | 585 |
704 AddAudioFileToWebAudio(kReferenceFileRelativeUrl, left_tab); | 586 AddAudioFileToWebAudio(kReferenceFileRelativeUrl, left_tab); |
705 | 587 |
706 NegotiateCall(left_tab, right_tab); | 588 NegotiateCall(left_tab, right_tab); |
707 | 589 |
708 base::FilePath recording = CreateTemporaryWaveFile(); | 590 base::FilePath recording = CreateTemporaryWaveFile(); |
709 | 591 |
710 // Note: the sound clip is 21.6 seconds: record for 25 seconds to get some | |
711 // safety margins on each side. | |
712 AudioRecorder recorder; | |
713 ASSERT_TRUE(recorder.StartRecording(base::TimeDelta::FromSeconds(25), | |
714 recording)); | |
715 | |
716 PlayAudioFileThroughWebAudio(left_tab); | 592 PlayAudioFileThroughWebAudio(left_tab); |
717 | 593 |
718 ASSERT_TRUE(recorder.WaitForRecordingToEnd()); | 594 EXPECT_TRUE(test::PollingWaitUntil("doneCapturingAudio()", "done-capturing", |
719 DVLOG(0) << "Done recording to " << recording.value() << std::endl; | 595 right_tab, kPollingIntervalInMs)); |
720 | 596 |
721 HangUp(left_tab); | 597 HangUp(left_tab); |
722 | 598 |
599 WriteCapturedAudio(right_tab, recording); | |
600 | |
601 DVLOG(0) << "Done recording to " << recording.value() << std::endl; | |
602 | |
723 // Compare with the reference file on disk (this is the same file we played | 603 // Compare with the reference file on disk (this is the same file we played |
724 // through WebAudio earlier). | 604 // through WebAudio earlier). |
725 base::FilePath reference_file = | 605 base::FilePath reference_file = |
726 test::GetReferenceFilesDir().Append(kReferenceFile); | 606 test::GetReferenceFilesDir().Append(kReferenceFile); |
727 ComputeAndPrintPesqResults(reference_file, recording, "_webaudio"); | 607 ComputeAndPrintPesqResults(reference_file, recording, "_webaudio"); |
728 } | 608 } |
729 | 609 |
730 /** | 610 /** |
731 * The auto gain control test plays a file into the fake microphone. Then it | 611 * The auto gain control test plays a file into the fake microphone. Then it |
732 * sets up a one-way WebRTC call with audio only and records Chrome's output on | 612 * sets up a one-way WebRTC call with audio only and records Chrome's output on |
(...skipping 30 matching lines...) Expand all Loading... | |
763 const std::string& perf_modifier) { | 643 const std::string& perf_modifier) { |
764 if (OnWin8()) { | 644 if (OnWin8()) { |
765 // http://crbug.com/379798. | 645 // http://crbug.com/379798. |
766 LOG(ERROR) << "This test is not implemented for Windows XP/Win8."; | 646 LOG(ERROR) << "This test is not implemented for Windows XP/Win8."; |
767 return; | 647 return; |
768 } | 648 } |
769 base::FilePath reference_file = | 649 base::FilePath reference_file = |
770 test::GetReferenceFilesDir().Append(reference_filename); | 650 test::GetReferenceFilesDir().Append(reference_filename); |
771 base::FilePath recording = CreateTemporaryWaveFile(); | 651 base::FilePath recording = CreateTemporaryWaveFile(); |
772 | 652 |
773 ASSERT_NO_FATAL_FAILURE(SetupAndRecordAudioCall( | 653 ASSERT_NO_FATAL_FAILURE( |
774 reference_file, recording, constraints, | 654 SetupAndRecordAudioCall(reference_file, recording, constraints)); |
775 base::TimeDelta::FromSeconds(30))); | |
776 | 655 |
777 base::ScopedTempDir split_ref_files; | 656 base::ScopedTempDir split_ref_files; |
778 ASSERT_TRUE(split_ref_files.CreateUniqueTempDir()); | 657 ASSERT_TRUE(split_ref_files.CreateUniqueTempDir()); |
779 ASSERT_NO_FATAL_FAILURE( | 658 ASSERT_NO_FATAL_FAILURE( |
780 SplitFileOnSilenceIntoDir(reference_file, split_ref_files.GetPath())); | 659 SplitFileOnSilenceIntoDir(reference_file, split_ref_files.GetPath())); |
781 std::vector<base::FilePath> ref_segments = | 660 std::vector<base::FilePath> ref_segments = |
782 ListWavFilesInDir(split_ref_files.GetPath()); | 661 ListWavFilesInDir(split_ref_files.GetPath()); |
783 | 662 |
784 base::ScopedTempDir split_actual_files; | 663 base::ScopedTempDir split_actual_files; |
785 ASSERT_TRUE(split_actual_files.CreateUniqueTempDir()); | 664 ASSERT_TRUE(split_actual_files.CreateUniqueTempDir()); |
(...skipping 20 matching lines...) Expand all Loading... | |
806 } | 685 } |
807 | 686 |
808 // Since the AGC is off here there should be no gain at all. | 687 // Since the AGC is off here there should be no gain at all. |
809 IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest, | 688 IN_PROC_BROWSER_TEST_F(MAYBE_WebRtcAudioQualityBrowserTest, |
810 MANUAL_TestAutoGainIsOffWithAudioProcessingOff) { | 689 MANUAL_TestAutoGainIsOffWithAudioProcessingOff) { |
811 const char* kAudioCallWithoutAudioProcessing = | 690 const char* kAudioCallWithoutAudioProcessing = |
812 "{audio: { mandatory: { echoCancellation: false } } }"; | 691 "{audio: { mandatory: { echoCancellation: false } } }"; |
813 ASSERT_NO_FATAL_FAILURE(TestAutoGainControl( | 692 ASSERT_NO_FATAL_FAILURE(TestAutoGainControl( |
814 kReferenceFile, kAudioCallWithoutAudioProcessing, "_no_agc")); | 693 kReferenceFile, kAudioCallWithoutAudioProcessing, "_no_agc")); |
815 } | 694 } |
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