media/audio/audio_low_latency_input_output_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 "base/basictypes.h"
   6 #include "base/environment.h"
   7 #include "base/files/file_util.h"
   8 #include "base/memory/scoped_ptr.h"
   9 #include "base/path_service.h"
  10 #include "base/synchronization/lock.h"
  11 #include "base/test/test_timeouts.h"
  12 #include "base/time/time.h"
  13 #include "build/build_config.h"
  14 #include "media/audio/audio_io.h"
  15 #include "media/audio/audio_manager_base.h"
  16 #include "media/audio/audio_unittest_util.h"
  17 #include "media/audio/fake_audio_log_factory.h"
  18 #include "media/base/seekable_buffer.h"
  19 #include "testing/gmock/include/gmock/gmock.h"
  20 #include "testing/gtest/include/gtest/gtest.h"
  21
  22 #if defined(USE_PULSEAUDIO)
  23 #include "media/audio/pulse/audio_manager_pulse.h"
  24 #elif defined(USE_ALSA)
  25 #include "media/audio/alsa/audio_manager_alsa.h"
  26 #elif defined(USE_CRAS)
  27 #include "media/audio/cras/audio_manager_cras.h"
  28 #elif defined(OS_MACOSX)
  29 #include "media/audio/mac/audio_manager_mac.h"
  30 #elif defined(OS_WIN)
  31 #include "media/audio/win/audio_manager_win.h"
  32 #include "media/audio/win/core_audio_util_win.h"
  33 #elif defined(OS_ANDROID)
  34 #include "media/audio/android/audio_manager_android.h"
  35 #else
  36 #include "media/audio/fake_audio_manager.h"
  37 #endif
  38
  39 namespace media {
  40
  41 #if defined(USE_PULSEAUDIO)
  42 typedef AudioManagerPulse AudioManagerAnyPlatform;
  43 #elif defined(USE_ALSA)
  44 typedef AudioManagerAlsa AudioManagerAnyPlatform;
  45 #elif defined(USE_CRAS)
  46 typedef AudioManagerCras AudioManagerAnyPlatform;
  47 #elif defined(OS_MACOSX)
  48 typedef AudioManagerMac AudioManagerAnyPlatform;
  49 #elif defined(OS_WIN)
  50 typedef AudioManagerWin AudioManagerAnyPlatform;
  51 #elif defined(OS_ANDROID)
  52 typedef AudioManagerAndroid AudioManagerAnyPlatform;
  53 #else
  54 typedef FakeAudioManager AudioManagerAnyPlatform;
  55 #endif
  56
  57 // Limits the number of delay measurements we can store in an array and
  58 // then write to file at end of the WASAPIAudioInputOutputFullDuplex test.
  59 static const size_t kMaxDelayMeasurements = 1000;
  60
  61 // Name of the output text file. The output file will be stored in the
  62 // directory containing media_unittests.exe.
  63 // Example: \src\build\Debug\audio_delay_values_ms.txt.
  64 // See comments for the WASAPIAudioInputOutputFullDuplex test for more details
  65 // about the file format.
  66 static const char kDelayValuesFileName[] = "audio_delay_values_ms.txt";
  67
  68 // Contains delay values which are reported during the full-duplex test.
  69 // Total delay = |buffer_delay_ms| + |input_delay_ms| + |output_delay_ms|.
  70 struct AudioDelayState {
  71   AudioDelayState()
  72       : delta_time_ms(0),
  73         buffer_delay_ms(0),
  74         input_delay_ms(0),
  75         output_delay_ms(0) {
  76   }
  77
  78   // Time in milliseconds since last delay report. Typical value is ~10 [ms].
  79   int delta_time_ms;
  80
  81   // Size of internal sync buffer. Typical value is ~0 [ms].
  82   int buffer_delay_ms;
  83
  84   // Reported capture/input delay. Typical value is ~10 [ms].
  85   int input_delay_ms;
  86
  87   // Reported render/output delay. Typical value is ~40 [ms].
  88   int output_delay_ms;
  89 };
  90
  91 // This class mocks the platform specific audio manager and overrides
  92 // the GetMessageLoop() method to ensure that we can run our tests on
  93 // the main thread instead of the audio thread.
  94 class MockAudioManager : public AudioManagerAnyPlatform {
  95  public:
  96   MockAudioManager() : AudioManagerAnyPlatform(&fake_audio_log_factory_) {}
  97   ~MockAudioManager() override {}
  98
  99   scoped_refptr<base::SingleThreadTaskRunner> GetTaskRunner() override {
 100     return base::MessageLoop::current()->task_runner();
 101   }
 102
 103  private:
 104   FakeAudioLogFactory fake_audio_log_factory_;
 105   DISALLOW_COPY_AND_ASSIGN(MockAudioManager);
 106 };
 107
 108 // Test fixture class.
 109 class AudioLowLatencyInputOutputTest : public testing::Test {
 110  protected:
 111   AudioLowLatencyInputOutputTest() {}
 112
 113   ~AudioLowLatencyInputOutputTest() override {}
 114
 115   AudioManager* audio_manager() { return &mock_audio_manager_; }
 116   base::MessageLoopForUI* message_loop() { return &message_loop_; }
 117
 118  private:
 119   base::MessageLoopForUI message_loop_;
 120   MockAudioManager mock_audio_manager_;
 121
 122   DISALLOW_COPY_AND_ASSIGN(AudioLowLatencyInputOutputTest);
 123 };
 124
 125 // This audio source/sink implementation should be used for manual tests
 126 // only since delay measurements are stored on an output text file.
 127 // All incoming/recorded audio packets are stored in an intermediate media
 128 // buffer which the renderer reads from when it needs audio for playout.
 129 // The total effect is that recorded audio is played out in loop back using
 130 // a sync buffer as temporary storage.
 131 class FullDuplexAudioSinkSource
 132     : public AudioInputStream::AudioInputCallback,
 133       public AudioOutputStream::AudioSourceCallback {
 134  public:
 135   FullDuplexAudioSinkSource(int sample_rate,
 136                             int samples_per_packet,
 137                             int channels)
 138     : sample_rate_(sample_rate),
 139       samples_per_packet_(samples_per_packet),
 140       channels_(channels),
 141       input_elements_to_write_(0),
 142       output_elements_to_write_(0),
 143       previous_write_time_(base::TimeTicks::Now()) {
 144     // Size in bytes of each audio frame (4 bytes for 16-bit stereo PCM).
 145     frame_size_ = (16 / 8) * channels_;
 146
 147     // Start with the smallest possible buffer size. It will be increased
 148     // dynamically during the test if required.
 149     buffer_.reset(
 150         new media::SeekableBuffer(0, samples_per_packet_ * frame_size_));
 151
 152     frames_to_ms_ = static_cast<double>(1000.0 / sample_rate_);
 153     delay_states_.reset(new AudioDelayState[kMaxDelayMeasurements]);
 154   }
 155
 156   ~FullDuplexAudioSinkSource() override {
 157     // Get complete file path to output file in the directory containing
 158     // media_unittests.exe. Example: src/build/Debug/audio_delay_values_ms.txt.
 159     base::FilePath file_name;
 160     EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_name));
 161     file_name = file_name.AppendASCII(kDelayValuesFileName);
 162
 163     FILE* text_file = base::OpenFile(file_name, "wt");
 164     DLOG_IF(ERROR, !text_file) << "Failed to open log file.";
 165     VLOG(0) << ">> Output file " << file_name.value() << " has been created.";
 166
 167     // Write the array which contains time-stamps, buffer size and
 168     // audio delays values to a text file.
 169     size_t elements_written = 0;
 170     while (elements_written <
 171         std::min(input_elements_to_write_, output_elements_to_write_)) {
 172       const AudioDelayState state = delay_states_[elements_written];
 173       fprintf(text_file, "%d %d %d %d\n",
 174               state.delta_time_ms,
 175               state.buffer_delay_ms,
 176               state.input_delay_ms,
 177               state.output_delay_ms);
 178       ++elements_written;
 179     }
 180
 181     base::CloseFile(text_file);
 182   }
 183
 184   // AudioInputStream::AudioInputCallback.
 185   void OnData(AudioInputStream* stream,
 186               const AudioBus* src,
 187               uint32 hardware_delay_bytes,
 188               double volume) override {
 189     base::AutoLock lock(lock_);
 190
 191     // Update three components in the AudioDelayState for this recorded
 192     // audio packet.
 193     const base::TimeTicks now_time = base::TimeTicks::Now();
 194     const int diff = (now_time - previous_write_time_).InMilliseconds();
 195     previous_write_time_ = now_time;
 196     if (input_elements_to_write_ < kMaxDelayMeasurements) {
 197       delay_states_[input_elements_to_write_].delta_time_ms = diff;
 198       delay_states_[input_elements_to_write_].buffer_delay_ms =
 199           BytesToMilliseconds(buffer_->forward_bytes());
 200       delay_states_[input_elements_to_write_].input_delay_ms =
 201           BytesToMilliseconds(hardware_delay_bytes);
 202       ++input_elements_to_write_;
 203     }
 204
 205     // TODO(henrika): fix this and use AudioFifo instead.
 206     // Store the captured audio packet in a seekable media buffer.
 207     // if (!buffer_->Append(src, size)) {
 208     // An attempt to write outside the buffer limits has been made.
 209     // Double the buffer capacity to ensure that we have a buffer large
 210     // enough to handle the current sample test scenario.
 211     //   buffer_->set_forward_capacity(2 * buffer_->forward_capacity());
 212     //   buffer_->Clear();
 213     // }
 214   }
 215
 216   void OnError(AudioInputStream* stream) override {}
 217
 218   // AudioOutputStream::AudioSourceCallback.
 219   int OnMoreData(AudioBus* audio_bus, uint32 total_bytes_delay) override {
 220     base::AutoLock lock(lock_);
 221
 222     // Update one component in the AudioDelayState for the packet
 223     // which is about to be played out.
 224     if (output_elements_to_write_ < kMaxDelayMeasurements) {
 225       delay_states_[output_elements_to_write_].output_delay_ms =
 226           BytesToMilliseconds(total_bytes_delay);
 227       ++output_elements_to_write_;
 228     }
 229
 230     int size;
 231     const uint8* source;
 232     // Read the data from the seekable media buffer which contains
 233     // captured data at the same size and sample rate as the output side.
 234     if (buffer_->GetCurrentChunk(&source, &size) && size > 0) {
 235       EXPECT_EQ(channels_, audio_bus->channels());
 236       size = std::min(audio_bus->frames() * frame_size_, size);
 237       EXPECT_EQ(static_cast<size_t>(size) % sizeof(*audio_bus->channel(0)), 0U);
 238       audio_bus->FromInterleaved(
 239           source, size / frame_size_, frame_size_ / channels_);
 240       buffer_->Seek(size);
 241       return size / frame_size_;
 242     }
 243
 244     return 0;
 245   }
 246
 247   void OnError(AudioOutputStream* stream) override {}
 248
 249  protected:
 250   // Converts from bytes to milliseconds taking the sample rate and size
 251   // of an audio frame into account.
 252   int BytesToMilliseconds(uint32 delay_bytes) const {
 253     return static_cast<int>((delay_bytes / frame_size_) * frames_to_ms_ + 0.5);
 254   }
 255
 256  private:
 257   base::Lock lock_;
 258   scoped_ptr<media::SeekableBuffer> buffer_;
 259   int sample_rate_;
 260   int samples_per_packet_;
 261   int channels_;
 262   int frame_size_;
 263   double frames_to_ms_;
 264   scoped_ptr<AudioDelayState[]> delay_states_;
 265   size_t input_elements_to_write_;
 266   size_t output_elements_to_write_;
 267   base::TimeTicks previous_write_time_;
 268 };
 269
 270 class AudioInputStreamTraits {
 271  public:
 272   typedef AudioInputStream StreamType;
 273
 274   static AudioParameters GetDefaultAudioStreamParameters(
 275       AudioManager* audio_manager) {
 276     return audio_manager->GetInputStreamParameters(
 277         AudioManagerBase::kDefaultDeviceId);
 278   }
 279
 280   static StreamType* CreateStream(AudioManager* audio_manager,
 281       const AudioParameters& params) {
 282     return audio_manager->MakeAudioInputStream(params,
 283       AudioManagerBase::kDefaultDeviceId);
 284   }
 285 };
 286
 287 class AudioOutputStreamTraits {
 288  public:
 289   typedef AudioOutputStream StreamType;
 290
 291   static AudioParameters GetDefaultAudioStreamParameters(
 292       AudioManager* audio_manager) {
 293     return audio_manager->GetDefaultOutputStreamParameters();
 294   }
 295
 296   static StreamType* CreateStream(AudioManager* audio_manager,
 297       const AudioParameters& params) {
 298     return audio_manager->MakeAudioOutputStream(params, std::string());
 299   }
 300 };
 301
 302 // Traits template holding a trait of StreamType. It encapsulates
 303 // AudioInputStream and AudioOutputStream stream types.
 304 template <typename StreamTraits>
 305 class StreamWrapper {
 306  public:
 307   typedef typename StreamTraits::StreamType StreamType;
 308
 309   explicit StreamWrapper(AudioManager* audio_manager)
 310       :
 311         audio_manager_(audio_manager),
 312         format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
 313 #if defined(OS_ANDROID)
 314         channel_layout_(CHANNEL_LAYOUT_MONO),
 315 #else
 316         channel_layout_(CHANNEL_LAYOUT_STEREO),
 317 #endif
 318         bits_per_sample_(16) {
 319     // Use the preferred sample rate.
 320     const AudioParameters& params =
 321         StreamTraits::GetDefaultAudioStreamParameters(audio_manager_);
 322     sample_rate_ = params.sample_rate();
 323
 324     // Use the preferred buffer size. Note that the input side uses the same
 325     // size as the output side in this implementation.
 326     samples_per_packet_ = params.frames_per_buffer();
 327   }
 328
 329   virtual ~StreamWrapper() {}
 330
 331   // Creates an Audio[Input|Output]Stream stream object using default
 332   // parameters.
 333   StreamType* Create() {
 334     return CreateStream();
 335   }
 336
 337   int channels() const {
 338     return ChannelLayoutToChannelCount(channel_layout_);
 339   }
 340   int bits_per_sample() const { return bits_per_sample_; }
 341   int sample_rate() const { return sample_rate_; }
 342   int samples_per_packet() const { return samples_per_packet_; }
 343
 344  private:
 345   StreamType* CreateStream() {
 346     StreamType* stream = StreamTraits::CreateStream(audio_manager_,
 347         AudioParameters(format_, channel_layout_, sample_rate_,
 348             bits_per_sample_, samples_per_packet_));
 349     EXPECT_TRUE(stream);
 350     return stream;
 351   }
 352
 353   AudioManager* audio_manager_;
 354   AudioParameters::Format format_;
 355   ChannelLayout channel_layout_;
 356   int bits_per_sample_;
 357   int sample_rate_;
 358   int samples_per_packet_;
 359 };
 360
 361 typedef StreamWrapper<AudioInputStreamTraits> AudioInputStreamWrapper;
 362 typedef StreamWrapper<AudioOutputStreamTraits> AudioOutputStreamWrapper;
 363
 364 // This test is intended for manual tests and should only be enabled
 365 // when it is required to make a real-time test of audio in full duplex and
 366 // at the same time create a text file which contains measured delay values.
 367 // The file can later be analyzed off line using e.g. MATLAB.
 368 // MATLAB example:
 369 //   D=load('audio_delay_values_ms.txt');
 370 //   x=cumsum(D(:,1));
 371 //   plot(x, D(:,2), x, D(:,3), x, D(:,4), x, D(:,2)+D(:,3)+D(:,4));
 372 //   axis([0, max(x), 0, max(D(:,2)+D(:,3)+D(:,4))+10]);
 373 //   legend('buffer delay','input delay','output delay','total delay');
 374 //   xlabel('time [msec]')
 375 //   ylabel('delay [msec]')
 376 //   title('Full-duplex audio delay measurement');
 377 TEST_F(AudioLowLatencyInputOutputTest, DISABLED_FullDuplexDelayMeasurement) {
 378   ABORT_AUDIO_TEST_IF_NOT(audio_manager()->HasAudioInputDevices() &&
 379                           audio_manager()->HasAudioOutputDevices());
 380
 381   AudioInputStreamWrapper aisw(audio_manager());
 382   AudioInputStream* ais = aisw.Create();
 383   EXPECT_TRUE(ais);
 384
 385   AudioOutputStreamWrapper aosw(audio_manager());
 386   AudioOutputStream* aos = aosw.Create();
 387   EXPECT_TRUE(aos);
 388
 389   // This test only supports identical parameters in both directions.
 390   // TODO(henrika): it is possible to cut delay here by using different
 391   // buffer sizes for input and output.
 392   if (aisw.sample_rate() != aosw.sample_rate() ||
 393       aisw.samples_per_packet() != aosw.samples_per_packet() ||
 394       aisw.channels()!= aosw.channels() ||
 395       aisw.bits_per_sample() != aosw.bits_per_sample()) {
 396     LOG(ERROR) << "This test requires symmetric input and output parameters. "
 397         "Ensure that sample rate and number of channels are identical in "
 398         "both directions";
 399     aos->Close();
 400     ais->Close();
 401     return;
 402   }
 403
 404   EXPECT_TRUE(ais->Open());
 405   EXPECT_TRUE(aos->Open());
 406
 407   FullDuplexAudioSinkSource full_duplex(
 408       aisw.sample_rate(), aisw.samples_per_packet(), aisw.channels());
 409
 410   VLOG(0) << ">> You should now be able to hear yourself in loopback...";
 411   DVLOG(0) << "   sample_rate       : " << aisw.sample_rate();
 412   DVLOG(0) << "   samples_per_packet: " << aisw.samples_per_packet();
 413   DVLOG(0) << "   channels          : " << aisw.channels();
 414
 415   ais->Start(&full_duplex);
 416   aos->Start(&full_duplex);
 417
 418   // Wait for approximately 10 seconds. The user shall hear his own voice
 419   // in loop back during this time. At the same time, delay recordings are
 420   // performed and stored in the output text file.
 421   message_loop()->PostDelayedTask(FROM_HERE,
 422       base::MessageLoop::QuitClosure(), TestTimeouts::action_timeout());
 423   message_loop()->Run();
 424
 425   aos->Stop();
 426   ais->Stop();
 427
 428   // All Close() operations that run on the mocked audio thread,
 429   // should be synchronous and not post additional close tasks to
 430   // mocked the audio thread. Hence, there is no need to call
 431   // message_loop()->RunUntilIdle() after the Close() methods.
 432   aos->Close();
 433   ais->Close();
 434 }
 435
 436 }  // namespace media