media/audio/win/audio_low_latency_output_win_unittest.cc

   1 // Copyright (c) 2012 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/files/file_util.h"
  11 #include "base/memory/scoped_ptr.h"
  12 #include "base/message_loop/message_loop.h"
  13 #include "base/path_service.h"
  14 #include "base/test/test_timeouts.h"
  15 #include "base/time/time.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/audio_unittest_util.h"
  20 #include "media/audio/mock_audio_source_callback.h"
  21 #include "media/audio/win/audio_low_latency_output_win.h"
  22 #include "media/audio/win/core_audio_util_win.h"
  23 #include "media/base/decoder_buffer.h"
  24 #include "media/base/seekable_buffer.h"
  25 #include "media/base/test_data_util.h"
  26 #include "testing/gmock/include/gmock/gmock.h"
  27 #include "testing/gmock_mutant.h"
  28 #include "testing/gtest/include/gtest/gtest.h"
  29
  30 using ::testing::_;
  31 using ::testing::AnyNumber;
  32 using ::testing::AtLeast;
  33 using ::testing::Between;
  34 using ::testing::CreateFunctor;
  35 using ::testing::DoAll;
  36 using ::testing::Gt;
  37 using ::testing::InvokeWithoutArgs;
  38 using ::testing::NotNull;
  39 using ::testing::Return;
  40 using base::win::ScopedCOMInitializer;
  41
  42 namespace media {
  43
  44 static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw";
  45 static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw";
  46 static const size_t kFileDurationMs = 20000;
  47 static const size_t kNumFileSegments = 2;
  48 static const int kBitsPerSample = 16;
  49 static const size_t kMaxDeltaSamples = 1000;
  50 static const char kDeltaTimeMsFileName[] = "delta_times_ms.txt";
  51
  52 MATCHER_P(HasValidDelay, value, "") {
  53   // It is difficult to come up with a perfect test condition for the delay
  54   // estimation. For now, verify that the produced output delay is always
  55   // larger than the selected buffer size.
  56   return arg >= value;
  57 }
  58
  59 // Used to terminate a loop from a different thread than the loop belongs to.
  60 // |loop| should be a MessageLoopProxy.
  61 ACTION_P(QuitLoop, loop) {
  62   loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
  63 }
  64
  65 // This audio source implementation should be used for manual tests only since
  66 // it takes about 20 seconds to play out a file.
  67 class ReadFromFileAudioSource : public AudioOutputStream::AudioSourceCallback {
  68  public:
  69   explicit ReadFromFileAudioSource(const std::string& name)
  70     : pos_(0),
  71       previous_call_time_(base::TimeTicks::Now()),
  72       text_file_(NULL),
  73       elements_to_write_(0) {
  74     // Reads a test file from media/test/data directory.
  75     file_ = ReadTestDataFile(name);
  76
  77     // Creates an array that will store delta times between callbacks.
  78     // The content of this array will be written to a text file at
  79     // destruction and can then be used for off-line analysis of the exact
  80     // timing of callbacks. The text file will be stored in media/test/data.
  81     delta_times_.reset(new int[kMaxDeltaSamples]);
  82   }
  83
  84   ~ReadFromFileAudioSource() override {
  85     // Get complete file path to output file in directory containing
  86     // media_unittests.exe.
  87     base::FilePath file_name;
  88     EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_name));
  89     file_name = file_name.AppendASCII(kDeltaTimeMsFileName);
  90
  91     EXPECT_TRUE(!text_file_);
  92     text_file_ = base::OpenFile(file_name, "wt");
  93     DLOG_IF(ERROR, !text_file_) << "Failed to open log file.";
  94
  95     // Write the array which contains delta times to a text file.
  96     size_t elements_written = 0;
  97     while (elements_written < elements_to_write_) {
  98       fprintf(text_file_, "%d\n", delta_times_[elements_written]);
  99       ++elements_written;
 100     }
 101
 102     base::CloseFile(text_file_);
 103   }
 104
 105   // AudioOutputStream::AudioSourceCallback implementation.
 106   int OnMoreData(AudioBus* audio_bus, uint32 total_bytes_delay) override {
 107     // Store time difference between two successive callbacks in an array.
 108     // These values will be written to a file in the destructor.
 109     const base::TimeTicks now_time = base::TimeTicks::Now();
 110     const int diff = (now_time - previous_call_time_).InMilliseconds();
 111     previous_call_time_ = now_time;
 112     if (elements_to_write_ < kMaxDeltaSamples) {
 113       delta_times_[elements_to_write_] = diff;
 114       ++elements_to_write_;
 115     }
 116
 117     int max_size =
 118         audio_bus->frames() * audio_bus->channels() * kBitsPerSample / 8;
 119
 120     // Use samples read from a data file and fill up the audio buffer
 121     // provided to us in the callback.
 122     if (pos_ + static_cast<int>(max_size) > file_size())
 123       max_size = file_size() - pos_;
 124     int frames = max_size / (audio_bus->channels() * kBitsPerSample / 8);
 125     if (max_size) {
 126       audio_bus->FromInterleaved(
 127           file_->data() + pos_, frames, kBitsPerSample / 8);
 128       pos_ += max_size;
 129     }
 130     return frames;
 131   }
 132
 133   void OnError(AudioOutputStream* stream) override {}
 134
 135   int file_size() { return file_->data_size(); }
 136
 137  private:
 138   scoped_refptr<DecoderBuffer> file_;
 139   scoped_ptr<int[]> delta_times_;
 140   int pos_;
 141   base::TimeTicks previous_call_time_;
 142   FILE* text_file_;
 143   size_t elements_to_write_;
 144 };
 145
 146 static bool ExclusiveModeIsEnabled() {
 147   return (WASAPIAudioOutputStream::GetShareMode() ==
 148           AUDCLNT_SHAREMODE_EXCLUSIVE);
 149 }
 150
 151 static bool HasCoreAudioAndOutputDevices(AudioManager* audio_man) {
 152   // The low-latency (WASAPI-based) version requires Windows Vista or higher.
 153   // TODO(henrika): note that we use Wave today to query the number of
 154   // existing output devices.
 155   return CoreAudioUtil::IsSupported() && audio_man->HasAudioOutputDevices();
 156 }
 157
 158 // Convenience method which creates a default AudioOutputStream object but
 159 // also allows the user to modify the default settings.
 160 class AudioOutputStreamWrapper {
 161  public:
 162   explicit AudioOutputStreamWrapper(AudioManager* audio_manager)
 163       : audio_man_(audio_manager),
 164         format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
 165         bits_per_sample_(kBitsPerSample) {
 166     AudioParameters preferred_params;
 167     EXPECT_TRUE(SUCCEEDED(CoreAudioUtil::GetPreferredAudioParameters(
 168         eRender, eConsole, &preferred_params)));
 169     channel_layout_ = preferred_params.channel_layout();
 170     sample_rate_ = preferred_params.sample_rate();
 171     samples_per_packet_ = preferred_params.frames_per_buffer();
 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   // sample rate and frame size are modified.
 190   AudioOutputStream* Create(int sample_rate, int samples_per_packet) {
 191     sample_rate_ = sample_rate;
 192     samples_per_packet_ = samples_per_packet;
 193     return CreateOutputStream();
 194   }
 195
 196   AudioParameters::Format format() const { return format_; }
 197   int channels() const { return ChannelLayoutToChannelCount(channel_layout_); }
 198   int bits_per_sample() const { return bits_per_sample_; }
 199   int sample_rate() const { return sample_rate_; }
 200   int samples_per_packet() const { return samples_per_packet_; }
 201
 202  private:
 203   AudioOutputStream* CreateOutputStream() {
 204     AudioOutputStream* aos = audio_man_->MakeAudioOutputStream(
 205         AudioParameters(format_, channel_layout_, sample_rate_,
 206                         bits_per_sample_, samples_per_packet_),
 207         std::string());
 208     EXPECT_TRUE(aos);
 209     return aos;
 210   }
 211
 212   AudioManager* audio_man_;
 213   AudioParameters::Format format_;
 214   ChannelLayout channel_layout_;
 215   int bits_per_sample_;
 216   int sample_rate_;
 217   int samples_per_packet_;
 218 };
 219
 220 // Convenience method which creates a default AudioOutputStream object.
 221 static AudioOutputStream* CreateDefaultAudioOutputStream(
 222     AudioManager* audio_manager) {
 223   AudioOutputStreamWrapper aosw(audio_manager);
 224   AudioOutputStream* aos = aosw.Create();
 225   return aos;
 226 }
 227
 228 // Verify that we can retrieve the current hardware/mixing sample rate
 229 // for the default audio device.
 230 // TODO(henrika): modify this test when we support full device enumeration.
 231 TEST(WASAPIAudioOutputStreamTest, HardwareSampleRate) {
 232   // Skip this test in exclusive mode since the resulting rate is only utilized
 233   // for shared mode streams.
 234   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 235   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()) &&
 236                           ExclusiveModeIsEnabled());
 237
 238   // Default device intended for games, system notification sounds,
 239   // and voice commands.
 240   int fs = static_cast<int>(
 241       WASAPIAudioOutputStream::HardwareSampleRate(std::string()));
 242   EXPECT_GE(fs, 0);
 243 }
 244
 245 // Test Create(), Close() calling sequence.
 246 TEST(WASAPIAudioOutputStreamTest, CreateAndClose) {
 247   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 248   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 249   AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 250   aos->Close();
 251 }
 252
 253 // Test Open(), Close() calling sequence.
 254 TEST(WASAPIAudioOutputStreamTest, OpenAndClose) {
 255   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 256   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 257   AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 258   EXPECT_TRUE(aos->Open());
 259   aos->Close();
 260 }
 261
 262 // Test Open(), Start(), Close() calling sequence.
 263 TEST(WASAPIAudioOutputStreamTest, OpenStartAndClose) {
 264   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 265   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 266   AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 267   EXPECT_TRUE(aos->Open());
 268   MockAudioSourceCallback source;
 269   EXPECT_CALL(source, OnError(aos))
 270       .Times(0);
 271   aos->Start(&source);
 272   aos->Close();
 273 }
 274
 275 // Test Open(), Start(), Stop(), Close() calling sequence.
 276 TEST(WASAPIAudioOutputStreamTest, OpenStartStopAndClose) {
 277   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 278   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 279   AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 280   EXPECT_TRUE(aos->Open());
 281   MockAudioSourceCallback source;
 282   EXPECT_CALL(source, OnError(aos))
 283       .Times(0);
 284   aos->Start(&source);
 285   aos->Stop();
 286   aos->Close();
 287 }
 288
 289 // Test SetVolume(), GetVolume()
 290 TEST(WASAPIAudioOutputStreamTest, Volume) {
 291   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 292   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 293   AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 294
 295   // Initial volume should be full volume (1.0).
 296   double volume = 0.0;
 297   aos->GetVolume(&volume);
 298   EXPECT_EQ(1.0, volume);
 299
 300   // Verify some valid volume settings.
 301   aos->SetVolume(0.0);
 302   aos->GetVolume(&volume);
 303   EXPECT_EQ(0.0, volume);
 304
 305   aos->SetVolume(0.5);
 306   aos->GetVolume(&volume);
 307   EXPECT_EQ(0.5, volume);
 308
 309   aos->SetVolume(1.0);
 310   aos->GetVolume(&volume);
 311   EXPECT_EQ(1.0, volume);
 312
 313   // Ensure that invalid volume setting have no effect.
 314   aos->SetVolume(1.5);
 315   aos->GetVolume(&volume);
 316   EXPECT_EQ(1.0, volume);
 317
 318   aos->SetVolume(-0.5);
 319   aos->GetVolume(&volume);
 320   EXPECT_EQ(1.0, volume);
 321
 322   aos->Close();
 323 }
 324
 325 // Test some additional calling sequences.
 326 TEST(WASAPIAudioOutputStreamTest, MiscCallingSequences) {
 327   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 328   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 329
 330   AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 331   WASAPIAudioOutputStream* waos = static_cast<WASAPIAudioOutputStream*>(aos);
 332
 333   // Open(), Open() is a valid calling sequence (second call does nothing).
 334   EXPECT_TRUE(aos->Open());
 335   EXPECT_TRUE(aos->Open());
 336
 337   MockAudioSourceCallback source;
 338
 339   // Start(), Start() is a valid calling sequence (second call does nothing).
 340   aos->Start(&source);
 341   EXPECT_TRUE(waos->started());
 342   aos->Start(&source);
 343   EXPECT_TRUE(waos->started());
 344
 345   // Stop(), Stop() is a valid calling sequence (second call does nothing).
 346   aos->Stop();
 347   EXPECT_FALSE(waos->started());
 348   aos->Stop();
 349   EXPECT_FALSE(waos->started());
 350
 351   // Start(), Stop(), Start(), Stop().
 352   aos->Start(&source);
 353   EXPECT_TRUE(waos->started());
 354   aos->Stop();
 355   EXPECT_FALSE(waos->started());
 356   aos->Start(&source);
 357   EXPECT_TRUE(waos->started());
 358   aos->Stop();
 359   EXPECT_FALSE(waos->started());
 360
 361   aos->Close();
 362 }
 363
 364 // Use preferred packet size and verify that rendering starts.
 365 TEST(WASAPIAudioOutputStreamTest, ValidPacketSize) {
 366   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 367   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 368
 369   base::MessageLoopForUI loop;
 370   MockAudioSourceCallback source;
 371
 372   // Create default WASAPI output stream which plays out in stereo using
 373   // the shared mixing rate. The default buffer size is 10ms.
 374   AudioOutputStreamWrapper aosw(audio_manager.get());
 375   AudioOutputStream* aos = aosw.Create();
 376   EXPECT_TRUE(aos->Open());
 377
 378   // Derive the expected size in bytes of each packet.
 379   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
 380       (aosw.bits_per_sample() / 8);
 381
 382   // Wait for the first callback and verify its parameters.
 383   EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
 384       .WillOnce(DoAll(
 385           QuitLoop(loop.message_loop_proxy()),
 386           Return(aosw.samples_per_packet())));
 387
 388   aos->Start(&source);
 389   loop.PostDelayedTask(FROM_HERE, base::MessageLoop::QuitClosure(),
 390                        TestTimeouts::action_timeout());
 391   loop.Run();
 392   aos->Stop();
 393   aos->Close();
 394 }
 395
 396 // This test is intended for manual tests and should only be enabled
 397 // when it is required to play out data from a local PCM file.
 398 // By default, GTest will print out YOU HAVE 1 DISABLED TEST.
 399 // To include disabled tests in test execution, just invoke the test program
 400 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS
 401 // environment variable to a value greater than 0.
 402 // The test files are approximately 20 seconds long.
 403 TEST(WASAPIAudioOutputStreamTest, DISABLED_ReadFromStereoFile) {
 404   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 405   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()));
 406
 407   AudioOutputStreamWrapper aosw(audio_manager.get());
 408   AudioOutputStream* aos = aosw.Create();
 409   EXPECT_TRUE(aos->Open());
 410
 411   std::string file_name;
 412   if (aosw.sample_rate() == 48000) {
 413     file_name = kSpeechFile_16b_s_48k;
 414   } else if (aosw.sample_rate() == 44100) {
 415     file_name = kSpeechFile_16b_s_44k;
 416   } else if (aosw.sample_rate() == 96000) {
 417     // Use 48kHz file at 96kHz as well. Will sound like Donald Duck.
 418     file_name = kSpeechFile_16b_s_48k;
 419   } else {
 420     FAIL() << "This test supports 44.1, 48kHz and 96kHz only.";
 421     return;
 422   }
 423   ReadFromFileAudioSource file_source(file_name);
 424
 425   DVLOG(0) << "File name      : " << file_name.c_str();
 426   DVLOG(0) << "Sample rate    : " << aosw.sample_rate();
 427   DVLOG(0) << "Bits per sample: " << aosw.bits_per_sample();
 428   DVLOG(0) << "#channels      : " << aosw.channels();
 429   DVLOG(0) << "File size      : " << file_source.file_size();
 430   DVLOG(0) << "#file segments : " << kNumFileSegments;
 431   DVLOG(0) << ">> Listen to the stereo file while playing...";
 432
 433   for (int i = 0; i < kNumFileSegments; i++) {
 434     // Each segment will start with a short (~20ms) block of zeros, hence
 435     // some short glitches might be heard in this test if kNumFileSegments
 436     // is larger than one. The exact length of the silence period depends on
 437     // the selected sample rate.
 438     aos->Start(&file_source);
 439     base::PlatformThread::Sleep(
 440         base::TimeDelta::FromMilliseconds(kFileDurationMs / kNumFileSegments));
 441     aos->Stop();
 442   }
 443
 444   DVLOG(0) << ">> Stereo file playout has stopped.";
 445   aos->Close();
 446 }
 447
 448 // Verify that we can open the output stream in exclusive mode using a
 449 // certain set of audio parameters and a sample rate of 48kHz.
 450 // The expected outcomes of each setting in this test has been derived
 451 // manually using log outputs (--v=1).
 452 TEST(WASAPIAudioOutputStreamTest, ExclusiveModeBufferSizesAt48kHz) {
 453   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 454   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()) &&
 455                           ExclusiveModeIsEnabled());
 456
 457   AudioOutputStreamWrapper aosw(audio_manager.get());
 458
 459   // 10ms @ 48kHz shall work.
 460   // Note that, this is the same size as we can use for shared-mode streaming
 461   // but here the endpoint buffer delay is only 10ms instead of 20ms.
 462   AudioOutputStream* aos = aosw.Create(48000, 480);
 463   EXPECT_TRUE(aos->Open());
 464   aos->Close();
 465
 466   // 5ms @ 48kHz does not work due to misalignment.
 467   // This test will propose an aligned buffer size of 5.3333ms.
 468   // Note that we must call Close() even is Open() fails since Close() also
 469   // deletes the object and we want to create a new object in the next test.
 470   aos = aosw.Create(48000, 240);
 471   EXPECT_FALSE(aos->Open());
 472   aos->Close();
 473
 474   // 5.3333ms @ 48kHz should work (see test above).
 475   aos = aosw.Create(48000, 256);
 476   EXPECT_TRUE(aos->Open());
 477   aos->Close();
 478
 479   // 2.6667ms is smaller than the minimum supported size (=3ms).
 480   aos = aosw.Create(48000, 128);
 481   EXPECT_FALSE(aos->Open());
 482   aos->Close();
 483
 484   // 3ms does not correspond to an aligned buffer size.
 485   // This test will propose an aligned buffer size of 3.3333ms.
 486   aos = aosw.Create(48000, 144);
 487   EXPECT_FALSE(aos->Open());
 488   aos->Close();
 489
 490   // 3.3333ms @ 48kHz <=> smallest possible buffer size we can use.
 491   aos = aosw.Create(48000, 160);
 492   EXPECT_TRUE(aos->Open());
 493   aos->Close();
 494 }
 495
 496 // Verify that we can open the output stream in exclusive mode using a
 497 // certain set of audio parameters and a sample rate of 44.1kHz.
 498 // The expected outcomes of each setting in this test has been derived
 499 // manually using log outputs (--v=1).
 500 TEST(WASAPIAudioOutputStreamTest, ExclusiveModeBufferSizesAt44kHz) {
 501   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 502   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()) &&
 503                           ExclusiveModeIsEnabled());
 504
 505   AudioOutputStreamWrapper aosw(audio_manager.get());
 506
 507   // 10ms @ 44.1kHz does not work due to misalignment.
 508   // This test will propose an aligned buffer size of 10.1587ms.
 509   AudioOutputStream* aos = aosw.Create(44100, 441);
 510   EXPECT_FALSE(aos->Open());
 511   aos->Close();
 512
 513   // 10.1587ms @ 44.1kHz shall work (see test above).
 514   aos = aosw.Create(44100, 448);
 515   EXPECT_TRUE(aos->Open());
 516   aos->Close();
 517
 518   // 5.8050ms @ 44.1 should work.
 519   aos = aosw.Create(44100, 256);
 520   EXPECT_TRUE(aos->Open());
 521   aos->Close();
 522
 523   // 4.9887ms @ 44.1kHz does not work to misalignment.
 524   // This test will propose an aligned buffer size of 5.0794ms.
 525   // Note that we must call Close() even is Open() fails since Close() also
 526   // deletes the object and we want to create a new object in the next test.
 527   aos = aosw.Create(44100, 220);
 528   EXPECT_FALSE(aos->Open());
 529   aos->Close();
 530
 531   // 5.0794ms @ 44.1kHz shall work (see test above).
 532   aos = aosw.Create(44100, 224);
 533   EXPECT_TRUE(aos->Open());
 534   aos->Close();
 535
 536   // 2.9025ms is smaller than the minimum supported size (=3ms).
 537   aos = aosw.Create(44100, 132);
 538   EXPECT_FALSE(aos->Open());
 539   aos->Close();
 540
 541   // 3.01587ms is larger than the minimum size but is not aligned.
 542   // This test will propose an aligned buffer size of 3.6281ms.
 543   aos = aosw.Create(44100, 133);
 544   EXPECT_FALSE(aos->Open());
 545   aos->Close();
 546
 547   // 3.6281ms @ 44.1kHz <=> smallest possible buffer size we can use.
 548   aos = aosw.Create(44100, 160);
 549   EXPECT_TRUE(aos->Open());
 550   aos->Close();
 551 }
 552
 553 // Verify that we can open and start the output stream in exclusive mode at
 554 // the lowest possible delay at 48kHz.
 555 TEST(WASAPIAudioOutputStreamTest, ExclusiveModeMinBufferSizeAt48kHz) {
 556   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 557   ABORT_AUDIO_TEST_IF_NOT(HasCoreAudioAndOutputDevices(audio_manager.get()) &&
 558                           ExclusiveModeIsEnabled());
 559
 560   base::MessageLoopForUI loop;
 561   MockAudioSourceCallback source;
 562
 563   // Create exclusive-mode WASAPI output stream which plays out in stereo
 564   // using the minimum buffer size at 48kHz sample rate.
 565   AudioOutputStreamWrapper aosw(audio_manager.get());
 566   AudioOutputStream* aos = aosw.Create(48000, 160);
 567   EXPECT_TRUE(aos->Open());
 568
 569   // Derive the expected size in bytes of each packet.
 570   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
 571       (aosw.bits_per_sample() / 8);
 572
 573  // Wait for the first callback and verify its parameters.
 574   EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
 575       .WillOnce(DoAll(
 576           QuitLoop(loop.message_loop_proxy()),
 577           Return(aosw.samples_per_packet())))
 578       .WillRepeatedly(Return(aosw.samples_per_packet()));
 579
 580   aos->Start(&source);
 581   loop.PostDelayedTask(FROM_HERE, base::MessageLoop::QuitClosure(),
 582                        TestTimeouts::action_timeout());
 583   loop.Run();
 584   aos->Stop();
 585   aos->Close();
 586 }
 587
 588 // Verify that we can open and start the output stream in exclusive mode at
 589 // the lowest possible delay at 44.1kHz.
 590 TEST(WASAPIAudioOutputStreamTest, ExclusiveModeMinBufferSizeAt44kHz) {
 591   ABORT_AUDIO_TEST_IF_NOT(ExclusiveModeIsEnabled());
 592   scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting());
 593
 594   base::MessageLoopForUI loop;
 595   MockAudioSourceCallback source;
 596
 597   // Create exclusive-mode WASAPI output stream which plays out in stereo
 598   // using the minimum buffer size at 44.1kHz sample rate.
 599   AudioOutputStreamWrapper aosw(audio_manager.get());
 600   AudioOutputStream* aos = aosw.Create(44100, 160);
 601   EXPECT_TRUE(aos->Open());
 602
 603   // Derive the expected size in bytes of each packet.
 604   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
 605       (aosw.bits_per_sample() / 8);
 606
 607   // Wait for the first callback and verify its parameters.
 608   EXPECT_CALL(source, OnMoreData(NotNull(), HasValidDelay(bytes_per_packet)))
 609     .WillOnce(DoAll(
 610         QuitLoop(loop.message_loop_proxy()),
 611         Return(aosw.samples_per_packet())))
 612     .WillRepeatedly(Return(aosw.samples_per_packet()));
 613
 614   aos->Start(&source);
 615   loop.PostDelayedTask(FROM_HERE, base::MessageLoop::QuitClosure(),
 616                         TestTimeouts::action_timeout());
 617   loop.Run();
 618   aos->Stop();
 619   aos->Close();
 620 }
 621
 622 }  // namespace media