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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/memory/scoped_ptr.h" |
| 11 #include "base/test/test_timeouts.h" |
| 12 #include "base/win/scoped_com_initializer.h" |
| 13 #include "media/audio/audio_io.h" |
| 14 #include "media/audio/audio_manager.h" |
| 15 #include "media/audio/win/audio_low_latency_output_win.h" |
| 16 #include "media/base/seekable_buffer.h" |
| 17 #include "media/base/test_data_util.h" |
| 18 #include "testing/gmock/include/gmock/gmock.h" |
| 19 #include "testing/gtest/include/gtest/gtest.h" |
| 20 |
| 21 using ::testing::_; |
| 22 using ::testing::AnyNumber; |
| 23 using ::testing::Between; |
| 24 using ::testing::Gt; |
| 25 using ::testing::NotNull; |
| 26 using ::testing::Return; |
| 27 using base::win::ScopedCOMInitializer; |
| 28 |
| 29 namespace media { |
| 30 |
| 31 static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw"; |
| 32 static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw"; |
| 33 |
| 34 MATCHER_P(HasValidDelay, value, "") { |
| 35 // It is difficult to come up with a perfect test condition for the delay |
| 36 // estimation. For now, verify that the produced output delay is always |
| 37 // larger than the selected buffer size. |
| 38 return arg.hardware_delay_bytes > value.hardware_delay_bytes; |
| 39 } |
| 40 |
| 41 class MockAudioSourceCallback : public AudioOutputStream::AudioSourceCallback { |
| 42 public: |
| 43 MOCK_METHOD4(OnMoreData, uint32(AudioOutputStream* stream, |
| 44 uint8* dest, |
| 45 uint32 max_size, |
| 46 AudioBuffersState buffers_state)); |
| 47 MOCK_METHOD2(OnError, void(AudioOutputStream* stream, int code)); |
| 48 }; |
| 49 |
| 50 // This audio source implementation should be used for manual tests only since |
| 51 // it takes about 20 seconds to play out a file. |
| 52 class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback { |
| 53 public: |
| 54 explicit ReadFromFileAudioSource(const std::string& name) : pos_(0) { |
| 55 // Reads a test file from media/test/data directory and stores it in |
| 56 // a scoped_array. |
| 57 ReadTestDataFile(name, &file_, &file_size_); |
| 58 file_size_ = file_size_; |
| 59 } |
| 60 |
| 61 virtual ~ReadFromFileAudioSource() {} |
| 62 |
| 63 // AudioOutputStream::AudioSourceCallback implementation. |
| 64 virtual uint32 OnMoreData(AudioOutputStream* stream, |
| 65 uint8* dest, |
| 66 uint32 max_size, |
| 67 AudioBuffersState buffers_state) { |
| 68 if (pos_ + static_cast<int>(max_size) > file_size_) |
| 69 max_size = file_size_ - pos_; |
| 70 if (max_size) { |
| 71 memcpy(dest, &file_[pos_], max_size); |
| 72 pos_ += max_size; |
| 73 } |
| 74 return max_size; |
| 75 } |
| 76 |
| 77 virtual void OnError(AudioOutputStream* stream, int code) {} |
| 78 |
| 79 int file_size() { return file_size_; } |
| 80 |
| 81 private: |
| 82 scoped_array<uint8> file_; |
| 83 int file_size_; |
| 84 int pos_; |
| 85 }; |
| 86 |
| 87 // Convenience method which ensures that we are not running on the build |
| 88 // bots and that at least one valid output device can be found. |
| 89 static bool CanRunAudioTests() { |
| 90 scoped_ptr<base::Environment> env(base::Environment::Create()); |
| 91 if (env->HasVar("CHROME_HEADLESS")) |
| 92 return false; |
| 93 AudioManager* audio_man = AudioManager::GetAudioManager(); |
| 94 if (NULL == audio_man) |
| 95 return false; |
| 96 // TODO(henrika): note that we use Wave today to query the number of |
| 97 // existing output devices. |
| 98 return audio_man->HasAudioOutputDevices(); |
| 99 } |
| 100 |
| 101 // Convenience method which creates a default AudioOutputStream object but |
| 102 // also allows the user to modify the default settings. |
| 103 class AudioOutputStreamWrapper { |
| 104 public: |
| 105 AudioOutputStreamWrapper() |
| 106 : com_init_(ScopedCOMInitializer::kMTA), |
| 107 audio_man_(AudioManager::GetAudioManager()), |
| 108 format_(AudioParameters::AUDIO_PCM_LOW_LATENCY), |
| 109 channel_layout_(CHANNEL_LAYOUT_STEREO), |
| 110 bits_per_sample_(16) { |
| 111 // Use native/mixing sample rate and 10ms frame size as default. |
| 112 sample_rate_ = static_cast<int>( |
| 113 WASAPIAudioOutputStream::HardwareSampleRate(eConsole)); |
| 114 samples_per_packet_ = sample_rate_ / 100; |
| 115 DCHECK(sample_rate_); |
| 116 } |
| 117 |
| 118 ~AudioOutputStreamWrapper() {} |
| 119 |
| 120 // Creates AudioOutputStream object using default parameters. |
| 121 AudioOutputStream* Create() { |
| 122 return CreateOutputStream(); |
| 123 } |
| 124 |
| 125 // Creates AudioOutputStream object using non-default parameters where the |
| 126 // frame size is modified. |
| 127 AudioOutputStream* Create(int samples_per_packet) { |
| 128 samples_per_packet_ = samples_per_packet; |
| 129 return CreateOutputStream(); |
| 130 } |
| 131 |
| 132 // Creates AudioOutputStream object using non-default parameters where the |
| 133 // channel layout is modified. |
| 134 AudioOutputStream* Create(ChannelLayout channel_layout) { |
| 135 channel_layout_ = channel_layout; |
| 136 return CreateOutputStream(); |
| 137 } |
| 138 |
| 139 AudioParameters::Format format() const { return format_; } |
| 140 int channels() const { return ChannelLayoutToChannelCount(channel_layout_); } |
| 141 int bits_per_sample() const { return bits_per_sample_; } |
| 142 int sample_rate() const { return sample_rate_; } |
| 143 int samples_per_packet() const { return samples_per_packet_; } |
| 144 |
| 145 private: |
| 146 AudioOutputStream* CreateOutputStream() { |
| 147 AudioOutputStream* aos = audio_man_->MakeAudioOutputStream( |
| 148 AudioParameters(format_, channel_layout_, sample_rate_, |
| 149 bits_per_sample_, samples_per_packet_)); |
| 150 EXPECT_TRUE(aos); |
| 151 return aos; |
| 152 } |
| 153 |
| 154 ScopedCOMInitializer com_init_; |
| 155 AudioManager* audio_man_; |
| 156 AudioParameters::Format format_; |
| 157 ChannelLayout channel_layout_; |
| 158 int bits_per_sample_; |
| 159 int sample_rate_; |
| 160 int samples_per_packet_; |
| 161 }; |
| 162 |
| 163 // Convenience method which creates a default AudioOutputStream object. |
| 164 static AudioOutputStream* CreateDefaultAudioOutputStream() { |
| 165 AudioOutputStreamWrapper aosw; |
| 166 AudioOutputStream* aos = aosw.Create(); |
| 167 return aos; |
| 168 } |
| 169 |
| 170 // Verify that we can retrieve the current hardware/mixing sample rate |
| 171 // for all supported device roles. The ERole enumeration defines constants |
| 172 // that indicate the role that the system/user has assigned to an audio |
| 173 // endpoint device. |
| 174 // TODO(henrika): modify this test when we support full device enumeration. |
| 175 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestHardwareSampleRate) { |
| 176 if (!CanRunAudioTests()) |
| 177 return; |
| 178 |
| 179 ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA); |
| 180 |
| 181 // Default device intended for games, system notification sounds, |
| 182 // and voice commands. |
| 183 int fs = static_cast<int>( |
| 184 WASAPIAudioOutputStream::HardwareSampleRate(eConsole)); |
| 185 EXPECT_GE(fs, 0); |
| 186 |
| 187 // Default communication device intended for e.g. VoIP communication. |
| 188 fs = static_cast<int>( |
| 189 WASAPIAudioOutputStream::HardwareSampleRate(eCommunications)); |
| 190 EXPECT_GE(fs, 0); |
| 191 |
| 192 // Multimedia device for music, movies and live music recording. |
| 193 fs = static_cast<int>( |
| 194 WASAPIAudioOutputStream::HardwareSampleRate(eMultimedia)); |
| 195 EXPECT_GE(fs, 0); |
| 196 } |
| 197 |
| 198 // Test Create(), Close() calling sequence. |
| 199 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestCreateAndClose) { |
| 200 if (!CanRunAudioTests()) |
| 201 return; |
| 202 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 203 aos->Close(); |
| 204 } |
| 205 |
| 206 // Test Open(), Close() calling sequence. |
| 207 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenAndClose) { |
| 208 if (!CanRunAudioTests()) |
| 209 return; |
| 210 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 211 EXPECT_TRUE(aos->Open()); |
| 212 aos->Close(); |
| 213 } |
| 214 |
| 215 // Test Open(), Start(), Close() calling sequence. |
| 216 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartAndClose) { |
| 217 if (!CanRunAudioTests()) |
| 218 return; |
| 219 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 220 EXPECT_TRUE(aos->Open()); |
| 221 MockAudioSourceCallback source; |
| 222 EXPECT_CALL(source, OnError(aos, _)) |
| 223 .Times(0); |
| 224 aos->Start(&source); |
| 225 aos->Close(); |
| 226 } |
| 227 |
| 228 // Test Open(), Start(), Stop(), Close() calling sequence. |
| 229 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartStopAndClose) { |
| 230 if (!CanRunAudioTests()) |
| 231 return; |
| 232 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 233 EXPECT_TRUE(aos->Open()); |
| 234 MockAudioSourceCallback source; |
| 235 EXPECT_CALL(source, OnError(aos, _)) |
| 236 .Times(0); |
| 237 aos->Start(&source); |
| 238 aos->Stop(); |
| 239 aos->Close(); |
| 240 } |
| 241 |
| 242 // Test SetVolume(), GetVolume() |
| 243 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestVolume) { |
| 244 if (!CanRunAudioTests()) |
| 245 return; |
| 246 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 247 |
| 248 // Initial volume should be full volume (1.0). |
| 249 double volume = 0.0; |
| 250 aos->GetVolume(&volume); |
| 251 EXPECT_TRUE(volume == 1.0); |
| 252 |
| 253 // Verify some valid volume settings. |
| 254 aos->SetVolume(0.0); |
| 255 aos->GetVolume(&volume); |
| 256 EXPECT_TRUE(volume == 0.0); |
| 257 |
| 258 aos->SetVolume(0.5); |
| 259 aos->GetVolume(&volume); |
| 260 EXPECT_TRUE(volume == 0.5); |
| 261 |
| 262 aos->SetVolume(1.0); |
| 263 aos->GetVolume(&volume); |
| 264 EXPECT_TRUE(volume == 1.0); |
| 265 |
| 266 // Ensure that invalid volume setting have no effect. |
| 267 aos->SetVolume(1.5); |
| 268 aos->GetVolume(&volume); |
| 269 EXPECT_TRUE(volume == 1.0); |
| 270 |
| 271 aos->SetVolume(-0.5); |
| 272 aos->GetVolume(&volume); |
| 273 EXPECT_TRUE(volume == 1.0); |
| 274 |
| 275 aos->Close(); |
| 276 } |
| 277 |
| 278 // Test some additional calling sequences. |
| 279 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMiscCallingSequences) { |
| 280 if (!CanRunAudioTests()) |
| 281 return; |
| 282 AudioOutputStream* aos = CreateDefaultAudioOutputStream(); |
| 283 WASAPIAudioOutputStream* waos = static_cast<WASAPIAudioOutputStream*>(aos); |
| 284 |
| 285 // Open(), Open() should fail the second time. |
| 286 EXPECT_TRUE(aos->Open()); |
| 287 EXPECT_FALSE(aos->Open()); |
| 288 |
| 289 MockAudioSourceCallback source; |
| 290 |
| 291 // Start(), Start() is a valid calling sequence (second call does nothing). |
| 292 aos->Start(&source); |
| 293 EXPECT_TRUE(waos->started()); |
| 294 aos->Start(&source); |
| 295 EXPECT_TRUE(waos->started()); |
| 296 |
| 297 // Stop(), Stop() is a valid calling sequence (second call does nothing). |
| 298 aos->Stop(); |
| 299 EXPECT_FALSE(waos->started()); |
| 300 aos->Stop(); |
| 301 EXPECT_FALSE(waos->started()); |
| 302 |
| 303 aos->Close(); |
| 304 } |
| 305 |
| 306 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizes) { |
| 307 if (!CanRunAudioTests()) |
| 308 return; |
| 309 |
| 310 // 10 ms packet size. |
| 311 |
| 312 // Create default WASAPI output stream which plays out in stereo using |
| 313 // the shared mixing rate. The default buffer size is 10ms. |
| 314 AudioOutputStreamWrapper aosw; |
| 315 AudioOutputStream* aos = aosw.Create(); |
| 316 EXPECT_TRUE(aos->Open()); |
| 317 |
| 318 // Derive the expected size in bytes of each packet. |
| 319 uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 320 (aosw.bits_per_sample() / 8); |
| 321 |
| 322 // Set up expected minimum delay estimation. |
| 323 AudioBuffersState state(0, bytes_per_packet); |
| 324 |
| 325 MockAudioSourceCallback source; |
| 326 |
| 327 // We use 10ms packets and will run the test for ~100ms. Given that the |
| 328 // startup sequence takes some time, it is reasonable to expect 5-12 |
| 329 // callbacks in this time period. All should ask for the same size and |
| 330 // contain a valid delay estimate. |
| 331 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 332 HasValidDelay(state))) |
| 333 .Times(Between(5, 10)) |
| 334 .WillRepeatedly(Return(bytes_per_packet)); |
| 335 |
| 336 aos->Start(&source); |
| 337 base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms()); |
| 338 aos->Stop(); |
| 339 |
| 340 // Store current packet size (to be used in the subsequent tests). |
| 341 int samples_per_packet_10ms = aosw.samples_per_packet(); |
| 342 |
| 343 aos->Close(); |
| 344 |
| 345 // 20 ms packet size. |
| 346 |
| 347 aos = aosw.Create(2 * samples_per_packet_10ms); |
| 348 EXPECT_TRUE(aos->Open()); |
| 349 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 350 (aosw.bits_per_sample() / 8); |
| 351 |
| 352 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 353 HasValidDelay(state))) |
| 354 .Times(Between(5, 10)) |
| 355 .WillRepeatedly(Return(bytes_per_packet)); |
| 356 |
| 357 aos->Start(&source); |
| 358 base::PlatformThread::Sleep(2 * TestTimeouts::tiny_timeout_ms()); |
| 359 aos->Stop(); |
| 360 |
| 361 aos->Close(); |
| 362 |
| 363 // 40 ms packet size. |
| 364 |
| 365 aos = aosw.Create(4 * samples_per_packet_10ms); |
| 366 EXPECT_TRUE(aos->Open()); |
| 367 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 368 (aosw.bits_per_sample() / 8); |
| 369 |
| 370 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 371 HasValidDelay(state))) |
| 372 .Times(Between(5, 10)) |
| 373 .WillRepeatedly(Return(bytes_per_packet)); |
| 374 |
| 375 aos->Start(&source); |
| 376 base::PlatformThread::Sleep(4 * TestTimeouts::tiny_timeout_ms()); |
| 377 aos->Stop(); |
| 378 |
| 379 aos->Close(); |
| 380 |
| 381 // 50 ms packet size. |
| 382 |
| 383 aos = aosw.Create(samples_per_packet_10ms * 5); |
| 384 EXPECT_TRUE(aos->Open()); |
| 385 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 386 (aosw.bits_per_sample() / 8); |
| 387 |
| 388 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 389 HasValidDelay(state))) |
| 390 .Times(Between(5, 10)) |
| 391 .WillRepeatedly(Return(bytes_per_packet)); |
| 392 |
| 393 aos->Start(&source); |
| 394 base::PlatformThread::Sleep(5 * TestTimeouts::tiny_timeout_ms()); |
| 395 aos->Stop(); |
| 396 |
| 397 aos->Close(); |
| 398 |
| 399 // 5 ms packet size. |
| 400 |
| 401 aos = aosw.Create(samples_per_packet_10ms / 2); |
| 402 EXPECT_TRUE(aos->Open()); |
| 403 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 404 (aosw.bits_per_sample() / 8); |
| 405 |
| 406 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 407 HasValidDelay(state))) |
| 408 .Times(Between(2 * 5, 2 * 10)) |
| 409 .WillRepeatedly(Return(bytes_per_packet)); |
| 410 |
| 411 aos->Start(&source); |
| 412 base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms()); |
| 413 aos->Stop(); |
| 414 |
| 415 aos->Close(); |
| 416 |
| 417 // 512 samples (independent of sample rate) |
| 418 |
| 419 aos = aosw.Create(512); |
| 420 EXPECT_TRUE(aos->Open()); |
| 421 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 422 (aosw.bits_per_sample() / 8); |
| 423 |
| 424 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 425 HasValidDelay(state))) |
| 426 .WillRepeatedly(Return(bytes_per_packet)); |
| 427 |
| 428 aos->Start(&source); |
| 429 base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms()); |
| 430 aos->Stop(); |
| 431 |
| 432 aos->Close(); |
| 433 } |
| 434 |
| 435 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMonoStereo) { |
| 436 if (!CanRunAudioTests()) |
| 437 return; |
| 438 |
| 439 // CHANNEL_LAYOUT_MONO |
| 440 |
| 441 // Create default WASAPI output stream which plays out in *mono* using |
| 442 // the shared mixing rate. The default buffer size is 10ms. |
| 443 AudioOutputStreamWrapper aosw; |
| 444 AudioOutputStream* aos = aosw.Create(CHANNEL_LAYOUT_MONO); |
| 445 EXPECT_TRUE(aos->Open()); |
| 446 |
| 447 // Derive the expected size in bytes of each packet. |
| 448 uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 449 (aosw.bits_per_sample() / 8); |
| 450 |
| 451 // Set up expected minimum delay estimation. |
| 452 AudioBuffersState state(0, bytes_per_packet); |
| 453 |
| 454 MockAudioSourceCallback source; |
| 455 |
| 456 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 457 HasValidDelay(state))) |
| 458 .Times(Between(5, 10)) |
| 459 .WillRepeatedly(Return(bytes_per_packet)); |
| 460 |
| 461 aos->Start(&source); |
| 462 base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms()); |
| 463 aos->Stop(); |
| 464 aos->Close(); |
| 465 |
| 466 // CHANNEL_LAYOUT_STEREO |
| 467 |
| 468 // Create default WASAPI output stream which plays out in *stereo* using |
| 469 // the shared mixing rate. The default buffer size is 10ms. |
| 470 aos = aosw.Create(CHANNEL_LAYOUT_STEREO); |
| 471 EXPECT_TRUE(aos->Open()); |
| 472 |
| 473 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() * |
| 474 (aosw.bits_per_sample() / 8); |
| 475 |
| 476 state.pending_bytes = 0; |
| 477 state.hardware_delay_bytes = bytes_per_packet; |
| 478 |
| 479 EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet, |
| 480 HasValidDelay(state))) |
| 481 .Times(Between(5, 10)) |
| 482 .WillRepeatedly(Return(bytes_per_packet)); |
| 483 |
| 484 aos->Start(&source); |
| 485 base::PlatformThread::Sleep(TestTimeouts::tiny_timeout_ms()); |
| 486 aos->Stop(); |
| 487 aos->Close(); |
| 488 } |
| 489 |
| 490 // This test is intended for manual tests and should only be enabled |
| 491 // when it is required to store the captured data on a local file. |
| 492 // By default, GTest will print out YOU HAVE 1 DISABLED TEST. |
| 493 // To include disabled tests in test execution, just invoke the test program |
| 494 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS |
| 495 // environment variable to a value greater than 0. |
| 496 // The test files are approximately 20 seconds long. |
| 497 TEST(WinAudioOutputTest, DISABLED_WASAPIAudioOutputStreamReadFromFile) { |
| 498 if (!CanRunAudioTests()) |
| 499 return; |
| 500 |
| 501 AudioOutputStreamWrapper aosw; |
| 502 AudioOutputStream* aos = aosw.Create(); |
| 503 EXPECT_TRUE(aos->Open()); |
| 504 |
| 505 std::string file_name; |
| 506 if (aosw.sample_rate() == 48000) { |
| 507 file_name = kSpeechFile_16b_s_48k; |
| 508 } else if (aosw.sample_rate() == 44100) { |
| 509 file_name = kSpeechFile_16b_s_44k; |
| 510 } else { |
| 511 fprintf(stderr, "This test supports 44.1 and 48kHz only.\n"); |
| 512 return; |
| 513 } |
| 514 |
| 515 ReadFromFileAudioSource file_source(file_name); |
| 516 fprintf(stderr, " File name : %s\n", file_name.c_str()); |
| 517 fprintf(stderr, " Sample rate: %d\n", aosw.sample_rate()); |
| 518 fprintf(stderr, " File size : %d\n", file_source.file_size()); |
| 519 fprintf(stderr, " >> Listen to the file while playing...\n"); |
| 520 aos->Start(&file_source); |
| 521 base::PlatformThread::Sleep(2 * TestTimeouts::action_timeout_ms()); |
| 522 aos->Stop(); |
| 523 fprintf(stderr, " >> File playout has stopped.\n"); |
| 524 aos->Close(); |
| 525 } |
| 526 |
| 527 } // namespace media |
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