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| 1 // Copyright (c) 2011 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #include <windows.h> |
| 6 #include <mmsystem.h> |
| 7 |
| 8 #include "base/basictypes.h" |
| 9 #include "base/environment.h" |
| 10 #include "base/file_util.h" |
| 11 #include "base/memory/scoped_ptr.h" |
| 12 #include "base/message_loop.h" |
| 13 #include "base/test/test_timeouts.h" |
| 14 #include "base/time.h" |
| 15 #include "base/path_service.h" |
| 16 #include "base/win/scoped_com_initializer.h" |
| 17 #include "media/audio/audio_io.h" |
| 18 #include "media/audio/audio_manager.h" |
| 19 #include "media/audio/win/audio_low_latency_output_win.h" |
| 20 #include "media/base/seekable_buffer.h" |
| 21 #include "media/base/test_data_util.h" |
| 22 #include "testing/gmock_mutant.h" |
| 23 #include "testing/gmock/include/gmock/gmock.h" |
| 24 #include "testing/gtest/include/gtest/gtest.h" |
| 25 |
| 26 using ::testing::_; |
| 27 using ::testing::AnyNumber; |
| 28 using ::testing::Between; |
| 29 using ::testing::CreateFunctor; |
| 30 using ::testing::DoAll; |
| 31 using ::testing::Gt; |
| 32 using ::testing::InvokeWithoutArgs; |
| 33 using ::testing::NotNull; |
| 34 using ::testing::Return; |
| 35 using base::win::ScopedCOMInitializer; |
| 36 |
| 37 namespace media { |
| 38 |
| 39 static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw"; |
| 40 static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw"; |
| 41 static const size_t kFileDurationMs = 20000; |
| 42 |
| 43 static const size_t kMaxDeltaSamples = 1000; |
| 44 static const char* kDeltaTimeMsFileName = "delta_times_ms.txt"; |
| 45 |
| 46 MATCHER_P(HasValidDelay, value, "") { |
| 47 // It is difficult to come up with a perfect test condition for the delay |
| 48 // estimation. For now, verify that the produced output delay is always |
| 49 // larger than the selected buffer size. |
| 50 return arg.hardware_delay_bytes > value.hardware_delay_bytes; |
| 51 } |
| 52 |
| 53 class MockAudioSourceCallback : public AudioOutputStream::AudioSourceCallback { |
| 54 public: |
| 55 MOCK_METHOD4(OnMoreData, uint32(AudioOutputStream* stream, |
| 56 uint8* dest, |
| 57 uint32 max_size, |
| 58 AudioBuffersState buffers_state)); |
| 59 MOCK_METHOD2(OnError, void(AudioOutputStream* stream, int code)); |
| 60 }; |
| 61 |
| 62 // This audio source implementation should be used for manual tests only since |
| 63 // it takes about 20 seconds to play out a file. |
| 64 class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback { |
| 65 public: |
| 66 explicit ReadFromFileAudioSource(const std::string& name) |
| 67 : pos_(0), |
| 68 previous_call_time_(base::Time::Now()), |
| 69 text_file_(NULL), |
| 70 elements_to_write_(0) { |
| 71 // Reads a test file from media/test/data directory and stores it in |
| 72 // a scoped_array. |
| 73 ReadTestDataFile(name, &file_, &file_size_); |
| 74 file_size_ = file_size_; |
| 75 |
| 76 // Creates an array that will store delta times between callbacks. |
| 77 // The content of this array will be written to a text file at |
| 78 // destruction and can then be used for off-line analysis of the exact |
| 79 // timing of callbacks. The text file will be stored in media/test/data. |
| 80 delta_times_.reset(new int[kMaxDeltaSamples]); |
| 81 } |
| 82 |
| 83 virtual ~ReadFromFileAudioSource() { |
| 84 // Get complete file path to output file in directory containing |
| 85 // media_unittests.exe. |
| 86 FilePath file_name; |
| 87 EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_name)); |
| 88 file_name = file_name.AppendASCII(kDeltaTimeMsFileName); |
| 89 |
| 90 EXPECT_TRUE(!text_file_); |
| 91 text_file_ = file_util::OpenFile(file_name, "wt"); |
| 92 DLOG_IF(ERROR, !text_file_) << "Failed to open log file."; |
| 93 |
| 94 // Write the array which contains delta times to a text file. |
| 95 size_t elements_written = 0; |
| 96 while (elements_written < elements_to_write_) { |
| 97 fprintf(text_file_, "%d\n", delta_times_[elements_written]); |
| 98 ++elements_written; |
| 99 } |
| 100 |
| 101 file_util::CloseFile(text_file_); |
| 102 } |
| 103 |
| 104 // AudioOutputStream::AudioSourceCallback implementation. |
| 105 virtual uint32 OnMoreData(AudioOutputStream* stream, |
| 106 uint8* dest, |
| 107 uint32 max_size, |
| 108 AudioBuffersState buffers_state) { |
| 109 // Store time difference between two successive callbacks in an array. |
| 110 // These values will be written to a file in the destructor. |
| 111 int diff = (base::Time::Now() - previous_call_time_).InMilliseconds(); |
| 112 previous_call_time_ = base::Time::Now(); |
| 113 if (elements_to_write_ < kMaxDeltaSamples) { |
| 114 delta_times_[elements_to_write_] = diff; |
| 115 ++elements_to_write_; |
| 116 } |
| 117 |
| 118 // Use samples read from a data file and fill up the audio buffer |
| 119 // provided to us in the callback. |
| 120 if (pos_ + static_cast<int>(max_size) > file_size_) |
| 121 max_size = file_size_ - pos_; |
| 122 if (max_size) { |
| 123 memcpy(dest, &file_[pos_], max_size); |
| 124 pos_ += max_size; |
| 125 } |
| 126 return max_size; |
| 127 } |
| 128 |
| 129 virtual void OnError(AudioOutputStream* stream, int code) {} |
| 130 |
| 131 int file_size() { return file_size_; } |
| 132 |
| 133 private: |
| 134 scoped_array<uint8> file_; |
| 135 scoped_array<int> delta_times_; |
| 136 int file_size_; |
| 137 int pos_; |
| 138 base::Time previous_call_time_; |
| 139 FILE* text_file_; |
| 140 size_t elements_to_write_; |
| 141 }; |
| 142 |
| 143 // Convenience method which ensures that we are not running on the build |
| 144 // bots and that at least one valid output device can be found. |
| 145 static bool CanRunAudioTests() { |
| 146 scoped_ptr<base::Environment> env(base::Environment::Create()); |
| 147 if (env->HasVar("CHROME_HEADLESS")) |
| 148 return false; |
| 149 AudioManager* audio_man = AudioManager::GetAudioManager(); |
| 150 if (NULL == audio_man) |
| 151 return false; |
| 152 // TODO(henrika): note that we use Wave today to query the number of |
| 153 // existing output devices. |
| 154 return audio_man->HasAudioOutputDevices(); |
| 155 } |
| 156 |
| 157 // Convenience method which creates a default AudioOutputStream object but |
| 158 // also allows the user to modify the default settings. |
| 159 class AudioOutputStreamWrapper { |
| 160 public: |
| 161 AudioOutputStreamWrapper() |
| 162 : com_init_(ScopedCOMInitializer::kMTA), |
| 163 audio_man_(AudioManager::GetAudioManager()), |
| 164 format_(AudioParameters::AUDIO_PCM_LOW_LATENCY), |
| 165 channel_layout_(CHANNEL_LAYOUT_STEREO), |
| 166 bits_per_sample_(16) { |
| 167 // Use native/mixing sample rate and 10ms frame size as default. |
| 168 sample_rate_ = static_cast<int>( |
| 169 WASAPIAudioOutputStream::HardwareSampleRate(eConsole)); |
| 170 samples_per_packet_ = sample_rate_ / 100; |
| 171 DCHECK(sample_rate_); |
| 172 } |
| 173 |
| 174 ~AudioOutputStreamWrapper() {} |
| 175 |
| 176 // Creates AudioOutputStream object using default parameters. |
| 177 AudioOutputStream* Create() { |
| 178 return CreateOutputStream(); |
| 179 } |
| 180 |
| 181 // Creates AudioOutputStream object using non-default parameters where the |
| 182 // frame size is modified. |
| 183 AudioOutputStream* Create(int samples_per_packet) { |
| 184 samples_per_packet_ = samples_per_packet; |
| 185 return CreateOutputStream(); |
| 186 } |
| 187 |
| 188 // Creates AudioOutputStream object using non-default parameters where the |
| 189 // channel layout is modified. |
| 190 AudioOutputStream* Create(ChannelLayout channel_layout) { |
| 191 channel_layout_ = channel_layout; |
| 192 return CreateOutputStream(); |
| 193 } |
| 194 |
| 195 AudioParameters::Format format() const { return format_; } |
| 196 int channels() const { return ChannelLayoutToChannelCount(channel_layout_); } |
| 197 int bits_per_sample() const { return bits_per_sample_; } |
| 198 int sample_rate() const { return sample_rate_; } |
| 199 int samples_per_packet() const { return samples_per_packet_; } |
| 200 |
| 201 private: |
| 202 AudioOutputStream* CreateOutputStream() { |
| 203 AudioOutputStream* aos = audio_man_->MakeAudioOutputStream( |
| 204 AudioParameters(format_, channel_layout_, sample_rate_, |
| 205 bits_per_sample_, samples_per_packet_)); |
| 206 EXPECT_TRUE(aos); |
| 207 return aos; |
| 208 } |
| 209 |
| 210 ScopedCOMInitializer com_init_; |
| 211 AudioManager* audio_man_; |
| 212 AudioParameters::Format format_; |
| 213 ChannelLayout channel_layout_; |
| 214 int bits_per_sample_; |
| 215 int sample_rate_; |
| 216 int samples_per_packet_; |
| 217 }; |
| 218 |
| 219 // Convenience method which creates a default AudioOutputStream object. |
| 220 static AudioOutputStream* CreateDefaultAudioOutputStream() { |
| 221 AudioOutputStreamWrapper aosw; |
| 222 AudioOutputStream* aos = aosw.Create(); |
| 223 return aos; |
| 224 } |
| 225 |
| 226 static void QuitMessageLoop(base::MessageLoopProxy* proxy) { |
| 227 proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask()); |
| 228 } |
| 229 |
| 230 // Verify that we can retrieve the current hardware/mixing sample rate |
| 231 // for all supported device roles. The ERole enumeration defines constants |
| 232 // that indicate the role that the system/user has assigned to an audio |
| 233 // endpoint device. |
| 234 // TODO(henrika): modify this test when we support full device enumeration. |
| 235 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestHardwareSampleRate) { |
| 236 if (!CanRunAudioTests()) |
| 237 return; |
| 238 |
| 239 ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA); |
| 240 |
| 241 // Default device intended for games, system notification sounds, |
| 242 // and voice commands. |
| 243 int fs = static_cast<int>( |
| 244 WASAPIAudioOutputStream::HardwareSampleRate(eConsole)); |
| 245 EXPECT_GE(fs, 0); |
| 246 |
| 247 // Default communication device intended for e.g. VoIP communication. |
| 248 fs = static_cast<int>( |
| 249 WASAPIAudioOutputStream::HardwareSampleRate(eCommunications)); |
| 250 EXPECT_GE(fs, 0); |
| 251 |
| 252 // Multimedia device for music, movies and live music recording. |
| 253 fs = static_cast<int>( |
| 254 WASAPIAudioOutputStream::HardwareSampleRate(eMultimedia)); |
| 255 EXPECT_GE(fs, 0); |
| 256 } |
| 257 |
| 258 // Test Create(), Close() calling sequence. |
| 259 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestCreateAndClose) { |
| 260 if (!CanRunAudioTests()) |
| 261 return; |
| 262 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 263 aos->Close(); |
| 264 } |
| 265 |
| 266 // Test Open(), Close() calling sequence. |
| 267 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenAndClose) { |
| 268 if (!CanRunAudioTests()) |
| 269 return; |
| 270 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 271 EXPECT_TRUE(aos->Open()); |
| 272 aos->Close(); |
| 273 } |
| 274 |
| 275 // Test Open(), Start(), Close() calling sequence. |
| 276 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartAndClose) { |
| 277 if (!CanRunAudioTests()) |
| 278 return; |
| 279 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 280 EXPECT_TRUE(aos->Open()); |
| 281 MockAudioSourceCallback source; |
| 282 EXPECT_CALL(source, OnError(aos, _)) |
| 283 .Times(0); |
| 284 aos->Start(&source); |
| 285 aos->Close(); |
| 286 } |
| 287 |
| 288 // Test Open(), Start(), Stop(), Close() calling sequence. |
| 289 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartStopAndClose) { |
| 290 if (!CanRunAudioTests()) |
| 291 return; |
| 292 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 293 EXPECT_TRUE(aos->Open()); |
| 294 MockAudioSourceCallback source; |
| 295 EXPECT_CALL(source, OnError(aos, _)) |
| 296 .Times(0); |
| 297 aos->Start(&source); |
| 298 aos->Stop(); |
| 299 aos->Close(); |
| 300 } |
| 301 |
| 302 // Test SetVolume(), GetVolume() |
| 303 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestVolume) { |
| 304 if (!CanRunAudioTests()) |
| 305 return; |
| 306 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 307 |
| 308 // Initial volume should be full volume (1.0). |
| 309 double volume = 0.0; |
| 310 aos->GetVolume(&volume); |
| 311 EXPECT_EQ(1.0, volume); |
| 312 |
| 313 // Verify some valid volume settings. |
| 314 aos->SetVolume(0.0); |
| 315 aos->GetVolume(&volume); |
| 316 EXPECT_EQ(0.0, volume); |
| 317 |
| 318 aos->SetVolume(0.5); |
| 319 aos->GetVolume(&volume); |
| 320 EXPECT_EQ(0.5, volume); |
| 321 |
| 322 aos->SetVolume(1.0); |
| 323 aos->GetVolume(&volume); |
| 324 EXPECT_EQ(1.0, volume); |
| 325 |
| 326 // Ensure that invalid volume setting have no effect. |
| 327 aos->SetVolume(1.5); |
| 328 aos->GetVolume(&volume); |
| 329 EXPECT_EQ(1.0, volume); |
| 330 |
| 331 aos->SetVolume(-0.5); |
| 332 aos->GetVolume(&volume); |
| 333 EXPECT_EQ(1.0, volume); |
| 334 |
| 335 aos->Close(); |
| 336 } |
| 337 |
| 338 // Test some additional calling sequences. |
| 339 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMiscCallingSequences) { |
| 340 if (!CanRunAudioTests()) |
| 341 return; |
| 342 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 343 WASAPIAudioOutputStream* waos = static_cast<WASAPIAudioOutputStream*>(aos); |
| 344 |
| 345 // Open(), Open() is a valid calling sequence (second call does nothing). |
| 346 EXPECT_TRUE(aos->Open()); |
| 347 EXPECT_TRUE(aos->Open()); |
| 348 |
| 349 MockAudioSourceCallback source; |
| 350 |
| 351 // Start(), Start() is a valid calling sequence (second call does nothing). |
| 352 aos->Start(&source); |
| 353 EXPECT_TRUE(waos->started()); |
| 354 aos->Start(&source); |
| 355 EXPECT_TRUE(waos->started()); |
| 356 |
| 357 // Stop(), Stop() is a valid calling sequence (second call does nothing). |
| 358 aos->Stop(); |
| 359 EXPECT_FALSE(waos->started()); |
| 360 aos->Stop(); |
| 361 EXPECT_FALSE(waos->started()); |
| 362 |
| 363 aos->Close(); |
| 364 } |
| 365 |
| 366 // Use default packet size (10ms) and verify that rendering starts. |
| 367 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizeInMilliseconds) { |
| 368 if (!CanRunAudioTests()) |
| 369 return; |
| 370 |
| 371 MessageLoopForUI loop; |
| 372 scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy()); |
| 373 |
| 374 MockAudioSourceCallback source; |
| 375 |
| 376 // Create default WASAPI output stream which plays out in stereo using |
| 377 // the shared mixing rate. The default buffer size is 10ms. |
| 378 AudioOutputStreamWrapper aosw; |
| 379 AudioOutputStream* aos = aosw.Create(); |
| 380 EXPECT_TRUE(aos->Open()); |
| 381 |
| 382 // Derive the expected size in bytes of each packet. |
| 383 uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 384 (aosw.bits_per_sample() / 8); |
| 385 |
| 386 // Set up expected minimum delay estimation. |
| 387 AudioBuffersState state(0, bytes_per_packet); |
| 388 |
| 389 // Wait for the first callback and verify its parameters. |
| 390 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 391 HasValidDelay(state))) |
| 392 .WillOnce( |
| 393 DoAll( |
| 394 InvokeWithoutArgs( |
| 395 CreateFunctor(&QuitMessageLoop, proxy.get())), |
| 396 Return(bytes_per_packet))); |
| 397 |
| 398 aos->Start(&source); |
| 399 loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(), |
| 400 TestTimeouts::action_timeout_ms()); |
| 401 loop.Run(); |
| 402 aos->Stop(); |
| 403 aos->Close(); |
| 404 } |
| 405 |
| 406 // Use a fixed packets size (independent of sample rate) and verify |
| 407 // that rendering starts. |
| 408 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizeInSamples) { |
| 409 if (!CanRunAudioTests()) |
| 410 return; |
| 411 |
| 412 MessageLoopForUI loop; |
| 413 scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy()); |
| 414 |
| 415 MockAudioSourceCallback source; |
| 416 |
| 417 // Create default WASAPI output stream which plays out in stereo using |
| 418 // the shared mixing rate. The buffer size is set to 1024 samples. |
| 419 AudioOutputStreamWrapper aosw; |
| 420 AudioOutputStream* aos = aosw.Create(1024); |
| 421 EXPECT_TRUE(aos->Open()); |
| 422 |
| 423 // Derive the expected size in bytes of each packet. |
| 424 uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 425 (aosw.bits_per_sample() / 8); |
| 426 |
| 427 // Set up expected minimum delay estimation. |
| 428 AudioBuffersState state(0, bytes_per_packet); |
| 429 |
| 430 // Wait for the first callback and verify its parameters. |
| 431 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 432 HasValidDelay(state))) |
| 433 .WillOnce( |
| 434 DoAll( |
| 435 InvokeWithoutArgs( |
| 436 CreateFunctor(&QuitMessageLoop, proxy.get())), |
| 437 Return(bytes_per_packet))); |
| 438 |
| 439 aos->Start(&source); |
| 440 loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(), |
| 441 TestTimeouts::action_timeout_ms()); |
| 442 loop.Run(); |
| 443 aos->Stop(); |
| 444 aos->Close(); |
| 445 } |
| 446 |
| 447 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMono) { |
| 448 if (!CanRunAudioTests()) |
| 449 return; |
| 450 |
| 451 MessageLoopForUI loop; |
| 452 scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy()); |
| 453 |
| 454 MockAudioSourceCallback source; |
| 455 |
| 456 // Create default WASAPI output stream which plays out in *mono* using |
| 457 // the shared mixing rate. The default buffer size is 10ms. |
| 458 AudioOutputStreamWrapper aosw; |
| 459 AudioOutputStream* aos = aosw.Create(CHANNEL_LAYOUT_MONO); |
| 460 EXPECT_TRUE(aos->Open()); |
| 461 |
| 462 // Derive the expected size in bytes of each packet. |
| 463 uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 464 (aosw.bits_per_sample() / 8); |
| 465 |
| 466 // Set up expected minimum delay estimation. |
| 467 AudioBuffersState state(0, bytes_per_packet); |
| 468 |
| 469 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 470 HasValidDelay(state))) |
| 471 .WillOnce( |
| 472 DoAll( |
| 473 InvokeWithoutArgs( |
| 474 CreateFunctor(&QuitMessageLoop, proxy.get())), |
| 475 Return(bytes_per_packet))); |
| 476 |
| 477 aos->Start(&source); |
| 478 loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(), |
| 479 TestTimeouts::action_timeout_ms()); |
| 480 loop.Run(); |
| 481 aos->Stop(); |
| 482 aos->Close(); |
| 483 } |
| 484 |
| 485 // This test is intended for manual tests and should only be enabled |
| 486 // when it is required to store the captured data on a local file. |
| 487 // By default, GTest will print out YOU HAVE 1 DISABLED TEST. |
| 488 // To include disabled tests in test execution, just invoke the test program |
| 489 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS |
| 490 // environment variable to a value greater than 0. |
| 491 // The test files are approximately 20 seconds long. |
| 492 TEST(WinAudioOutputTest, DISABLE_WASAPIAudioOutputStreamReadFromFile) { |
| 493 if (!CanRunAudioTests()) |
| 494 return; |
| 495 |
| 496 AudioOutputStreamWrapper aosw; |
| 497 AudioOutputStream* aos = aosw.Create(); |
| 498 EXPECT_TRUE(aos->Open()); |
| 499 |
| 500 std::string file_name; |
| 501 if (aosw.sample_rate() == 48000) { |
| 502 file_name = kSpeechFile_16b_s_48k; |
| 503 } else if (aosw.sample_rate() == 44100) { |
| 504 file_name = kSpeechFile_16b_s_44k; |
| 505 } else if (aosw.sample_rate() == 96000) { |
| 506 // Use 48kHz file at 96kHz as well. Will sound like Donald Duck. |
| 507 file_name = kSpeechFile_16b_s_48k; |
| 508 } else { |
| 509 FAIL() << "This test supports 44.1, 48kHz and 96kHz only."; |
| 510 return; |
| 511 } |
| 512 ReadFromFileAudioSource file_source(file_name); |
| 513 int file_duration_ms = kFileDurationMs; |
| 514 |
| 515 LOG(INFO) << "File name : " << file_name.c_str(); |
| 516 LOG(INFO) << "Sample rate: " << aosw.sample_rate(); |
| 517 LOG(INFO) << "File size : " << file_source.file_size(); |
| 518 LOG(INFO) << ">> Listen to the file while playing..."; |
| 519 |
| 520 aos->Start(&file_source); |
| 521 base::PlatformThread::Sleep(file_duration_ms); |
| 522 aos->Stop(); |
| 523 |
| 524 LOG(INFO) << ">> File playout has stopped."; |
| 525 aos->Close(); |
| 526 } |
| 527 |
| 528 } // namespace media |
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