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/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/fake_audio_log_factory.h"
  17 #include "media/base/seekable_buffer.h"
  18 #include "testing/gmock/include/gmock/gmock.h"
  19 #include "testing/gtest/include/gtest/gtest.h"
  20
  21 #if defined(USE_ALSA)
  22 #include "media/audio/alsa/audio_manager_alsa.h"
  23 #elif defined(OS_MACOSX)
  24 #include "media/audio/mac/audio_manager_mac.h"
  25 #elif defined(OS_WIN)
  26 #include "media/audio/win/audio_manager_win.h"
  27 #include "media/audio/win/core_audio_util_win.h"
  28 #elif defined(OS_ANDROID)
  29 #include "media/audio/android/audio_manager_android.h"
  30 #else
  31 #include "media/audio/fake_audio_manager.h"
  32 #endif
  33
  34 namespace media {
  35
  36 #if defined(USE_ALSA)
  37 typedef AudioManagerAlsa AudioManagerAnyPlatform;
  38 #elif defined(OS_MACOSX)
  39 typedef AudioManagerMac AudioManagerAnyPlatform;
  40 #elif defined(OS_WIN)
  41 typedef AudioManagerWin AudioManagerAnyPlatform;
  42 #elif defined(OS_ANDROID)
  43 typedef AudioManagerAndroid AudioManagerAnyPlatform;
  44 #else
  45 typedef FakeAudioManager AudioManagerAnyPlatform;
  46 #endif
  47
  48 // Limits the number of delay measurements we can store in an array and
  49 // then write to file at end of the WASAPIAudioInputOutputFullDuplex test.
  50 static const size_t kMaxDelayMeasurements = 1000;
  51
  52 // Name of the output text file. The output file will be stored in the
  53 // directory containing media_unittests.exe.
  54 // Example: \src\build\Debug\audio_delay_values_ms.txt.
  55 // See comments for the WASAPIAudioInputOutputFullDuplex test for more details
  56 // about the file format.
  57 static const char kDelayValuesFileName[] = "audio_delay_values_ms.txt";
  58
  59 // Contains delay values which are reported during the full-duplex test.
  60 // Total delay = |buffer_delay_ms| + |input_delay_ms| + |output_delay_ms|.
  61 struct AudioDelayState {
  62   AudioDelayState()
  63       : delta_time_ms(0),
  64         buffer_delay_ms(0),
  65         input_delay_ms(0),
  66         output_delay_ms(0) {
  67   }
  68
  69   // Time in milliseconds since last delay report. Typical value is ~10 [ms].
  70   int delta_time_ms;
  71
  72   // Size of internal sync buffer. Typical value is ~0 [ms].
  73   int buffer_delay_ms;
  74
  75   // Reported capture/input delay. Typical value is ~10 [ms].
  76   int input_delay_ms;
  77
  78   // Reported render/output delay. Typical value is ~40 [ms].
  79   int output_delay_ms;
  80 };
  81
  82 // This class mocks the platform specific audio manager and overrides
  83 // the GetMessageLoop() method to ensure that we can run our tests on
  84 // the main thread instead of the audio thread.
  85 class MockAudioManager : public AudioManagerAnyPlatform {
  86  public:
  87   MockAudioManager() : AudioManagerAnyPlatform(&fake_audio_log_factory_) {}
  88   virtual ~MockAudioManager() {}
  89
  90   virtual scoped_refptr<base::SingleThreadTaskRunner> GetTaskRunner() OVERRIDE {
  91     return base::MessageLoop::current()->message_loop_proxy();
  92   }
  93
  94  private:
  95   FakeAudioLogFactory fake_audio_log_factory_;
  96   DISALLOW_COPY_AND_ASSIGN(MockAudioManager);
  97 };
  98
  99 // Test fixture class.
 100 class AudioLowLatencyInputOutputTest : public testing::Test {
 101  protected:
 102   AudioLowLatencyInputOutputTest() {}
 103
 104   virtual ~AudioLowLatencyInputOutputTest() {}
 105
 106   AudioManager* audio_manager() { return &mock_audio_manager_; }
 107   base::MessageLoopForUI* message_loop() { return &message_loop_; }
 108
 109   // Convenience method which ensures that we are not running on the build
 110   // bots and that at least one valid input and output device can be found.
 111   bool CanRunAudioTests() {
 112     bool input = audio_manager()->HasAudioInputDevices();
 113     bool output = audio_manager()->HasAudioOutputDevices();
 114     LOG_IF(WARNING, !input) << "No input device detected.";
 115     LOG_IF(WARNING, !output) << "No output device detected.";
 116     return input && output;
 117   }
 118
 119  private:
 120   base::MessageLoopForUI message_loop_;
 121   MockAudioManager mock_audio_manager_;
 122
 123   DISALLOW_COPY_AND_ASSIGN(AudioLowLatencyInputOutputTest);
 124 };
 125
 126 // This audio source/sink implementation should be used for manual tests
 127 // only since delay measurements are stored on an output text file.
 128 // All incoming/recorded audio packets are stored in an intermediate media
 129 // buffer which the renderer reads from when it needs audio for playout.
 130 // The total effect is that recorded audio is played out in loop back using
 131 // a sync buffer as temporary storage.
 132 class FullDuplexAudioSinkSource
 133     : public AudioInputStream::AudioInputCallback,
 134       public AudioOutputStream::AudioSourceCallback {
 135  public:
 136   FullDuplexAudioSinkSource(int sample_rate,
 137                             int samples_per_packet,
 138                             int channels)
 139     : sample_rate_(sample_rate),
 140       samples_per_packet_(samples_per_packet),
 141       channels_(channels),
 142       input_elements_to_write_(0),
 143       output_elements_to_write_(0),
 144       previous_write_time_(base::TimeTicks::Now()) {
 145     // Size in bytes of each audio frame (4 bytes for 16-bit stereo PCM).
 146     frame_size_ = (16 / 8) * channels_;
 147
 148     // Start with the smallest possible buffer size. It will be increased
 149     // dynamically during the test if required.
 150     buffer_.reset(
 151         new media::SeekableBuffer(0, samples_per_packet_ * frame_size_));
 152
 153     frames_to_ms_ = static_cast<double>(1000.0 / sample_rate_);
 154     delay_states_.reset(new AudioDelayState[kMaxDelayMeasurements]);
 155   }
 156
 157   virtual ~FullDuplexAudioSinkSource() {
 158     // Get complete file path to output file in the directory containing
 159     // media_unittests.exe. Example: src/build/Debug/audio_delay_values_ms.txt.
 160     base::FilePath file_name;
 161     EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_name));
 162     file_name = file_name.AppendASCII(kDelayValuesFileName);
 163
 164     FILE* text_file = base::OpenFile(file_name, "wt");
 165     DLOG_IF(ERROR, !text_file) << "Failed to open log file.";
 166     VLOG(0) << ">> Output file " << file_name.value() << " has been created.";
 167
 168     // Write the array which contains time-stamps, buffer size and
 169     // audio delays values to a text file.
 170     size_t elements_written = 0;
 171     while (elements_written <
 172         std::min(input_elements_to_write_, output_elements_to_write_)) {
 173       const AudioDelayState state = delay_states_[elements_written];
 174       fprintf(text_file, "%d %d %d %d\n",
 175               state.delta_time_ms,
 176               state.buffer_delay_ms,
 177               state.input_delay_ms,
 178               state.output_delay_ms);
 179       ++elements_written;
 180     }
 181
 182     base::CloseFile(text_file);
 183   }
 184
 185   // AudioInputStream::AudioInputCallback.
 186   virtual void OnData(AudioInputStream* stream,
 187                       const uint8* src, uint32 size,
 188                       uint32 hardware_delay_bytes,
 189                       double volume) OVERRIDE {
 190     base::AutoLock lock(lock_);
 191
 192     // Update three components in the AudioDelayState for this recorded
 193     // audio packet.
 194     const base::TimeTicks now_time = base::TimeTicks::Now();
 195     const int diff = (now_time - previous_write_time_).InMilliseconds();
 196     previous_write_time_ = now_time;
 197     if (input_elements_to_write_ < kMaxDelayMeasurements) {
 198       delay_states_[input_elements_to_write_].delta_time_ms = diff;
 199       delay_states_[input_elements_to_write_].buffer_delay_ms =
 200           BytesToMilliseconds(buffer_->forward_bytes());
 201       delay_states_[input_elements_to_write_].input_delay_ms =
 202           BytesToMilliseconds(hardware_delay_bytes);
 203       ++input_elements_to_write_;
 204     }
 205
 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   virtual void OnError(AudioInputStream* stream) OVERRIDE {}
 217
 218   // AudioOutputStream::AudioSourceCallback.
 219   virtual int OnMoreData(AudioBus* audio_bus,
 220                          AudioBuffersState buffers_state) OVERRIDE {
 221     base::AutoLock lock(lock_);
 222
 223     // Update one component in the AudioDelayState for the packet
 224     // which is about to be played out.
 225     if (output_elements_to_write_ < kMaxDelayMeasurements) {
 226       int output_delay_bytes = buffers_state.hardware_delay_bytes;
 227 #if defined(OS_WIN)
 228       // Special fix for Windows in combination with Wave where the
 229       // pending bytes field of the audio buffer state is used to
 230       // report the delay.
 231       if (!CoreAudioUtil::IsSupported()) {
 232         output_delay_bytes = buffers_state.pending_bytes;
 233       }
 234 #endif
 235       delay_states_[output_elements_to_write_].output_delay_ms =
 236           BytesToMilliseconds(output_delay_bytes);
 237       ++output_elements_to_write_;
 238     }
 239
 240     int size;
 241     const uint8* source;
 242     // Read the data from the seekable media buffer which contains
 243     // captured data at the same size and sample rate as the output side.
 244     if (buffer_->GetCurrentChunk(&source, &size) && size > 0) {
 245       EXPECT_EQ(channels_, audio_bus->channels());
 246       size = std::min(audio_bus->frames() * frame_size_, size);
 247       EXPECT_EQ(static_cast<size_t>(size) % sizeof(*audio_bus->channel(0)), 0U);
 248       audio_bus->FromInterleaved(
 249           source, size / frame_size_, frame_size_ / channels_);
 250       buffer_->Seek(size);
 251       return size / frame_size_;
 252     }
 253
 254     return 0;
 255   }
 256
 257   virtual void OnError(AudioOutputStream* stream) OVERRIDE {}
 258
 259  protected:
 260   // Converts from bytes to milliseconds taking the sample rate and size
 261   // of an audio frame into account.
 262   int BytesToMilliseconds(uint32 delay_bytes) const {
 263     return static_cast<int>((delay_bytes / frame_size_) * frames_to_ms_ + 0.5);
 264   }
 265
 266  private:
 267   base::Lock lock_;
 268   scoped_ptr<media::SeekableBuffer> buffer_;
 269   int sample_rate_;
 270   int samples_per_packet_;
 271   int channels_;
 272   int frame_size_;
 273   double frames_to_ms_;
 274   scoped_ptr<AudioDelayState[]> delay_states_;
 275   size_t input_elements_to_write_;
 276   size_t output_elements_to_write_;
 277   base::TimeTicks previous_write_time_;
 278 };
 279
 280 class AudioInputStreamTraits {
 281  public:
 282   typedef AudioInputStream StreamType;
 283
 284   static AudioParameters GetDefaultAudioStreamParameters(
 285       AudioManager* audio_manager) {
 286     return audio_manager->GetInputStreamParameters(
 287         AudioManagerBase::kDefaultDeviceId);
 288   }
 289
 290   static StreamType* CreateStream(AudioManager* audio_manager,
 291       const AudioParameters& params) {
 292     return audio_manager->MakeAudioInputStream(params,
 293       AudioManagerBase::kDefaultDeviceId);
 294   }
 295 };
 296
 297 class AudioOutputStreamTraits {
 298  public:
 299   typedef AudioOutputStream StreamType;
 300
 301   static AudioParameters GetDefaultAudioStreamParameters(
 302       AudioManager* audio_manager) {
 303     return audio_manager->GetDefaultOutputStreamParameters();
 304   }
 305
 306   static StreamType* CreateStream(AudioManager* audio_manager,
 307       const AudioParameters& params) {
 308     return audio_manager->MakeAudioOutputStream(params, std::string());
 309   }
 310 };
 311
 312 // Traits template holding a trait of StreamType. It encapsulates
 313 // AudioInputStream and AudioOutputStream stream types.
 314 template <typename StreamTraits>
 315 class StreamWrapper {
 316  public:
 317   typedef typename StreamTraits::StreamType StreamType;
 318
 319   explicit StreamWrapper(AudioManager* audio_manager)
 320       :
 321         audio_manager_(audio_manager),
 322         format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
 323 #if defined(OS_ANDROID)
 324         channel_layout_(CHANNEL_LAYOUT_MONO),
 325 #else
 326         channel_layout_(CHANNEL_LAYOUT_STEREO),
 327 #endif
 328         bits_per_sample_(16) {
 329     // Use the preferred sample rate.
 330     const AudioParameters& params =
 331         StreamTraits::GetDefaultAudioStreamParameters(audio_manager_);
 332     sample_rate_ = params.sample_rate();
 333
 334     // Use the preferred buffer size. Note that the input side uses the same
 335     // size as the output side in this implementation.
 336     samples_per_packet_ = params.frames_per_buffer();
 337   }
 338
 339   virtual ~StreamWrapper() {}
 340
 341   // Creates an Audio[Input|Output]Stream stream object using default
 342   // parameters.
 343   StreamType* Create() {
 344     return CreateStream();
 345   }
 346
 347   int channels() const {
 348     return ChannelLayoutToChannelCount(channel_layout_);
 349   }
 350   int bits_per_sample() const { return bits_per_sample_; }
 351   int sample_rate() const { return sample_rate_; }
 352   int samples_per_packet() const { return samples_per_packet_; }
 353
 354  private:
 355   StreamType* CreateStream() {
 356     StreamType* stream = StreamTraits::CreateStream(audio_manager_,
 357         AudioParameters(format_, channel_layout_, sample_rate_,
 358             bits_per_sample_, samples_per_packet_));
 359     EXPECT_TRUE(stream);
 360     return stream;
 361   }
 362
 363   AudioManager* audio_manager_;
 364   AudioParameters::Format format_;
 365   ChannelLayout channel_layout_;
 366   int bits_per_sample_;
 367   int sample_rate_;
 368   int samples_per_packet_;
 369 };
 370
 371 typedef StreamWrapper<AudioInputStreamTraits> AudioInputStreamWrapper;
 372 typedef StreamWrapper<AudioOutputStreamTraits> AudioOutputStreamWrapper;
 373
 374 // This test is intended for manual tests and should only be enabled
 375 // when it is required to make a real-time test of audio in full duplex and
 376 // at the same time create a text file which contains measured delay values.
 377 // The file can later be analyzed off line using e.g. MATLAB.
 378 // MATLAB example:
 379 //   D=load('audio_delay_values_ms.txt');
 380 //   x=cumsum(D(:,1));
 381 //   plot(x, D(:,2), x, D(:,3), x, D(:,4), x, D(:,2)+D(:,3)+D(:,4));
 382 //   axis([0, max(x), 0, max(D(:,2)+D(:,3)+D(:,4))+10]);
 383 //   legend('buffer delay','input delay','output delay','total delay');
 384 //   xlabel('time [msec]')
 385 //   ylabel('delay [msec]')
 386 //   title('Full-duplex audio delay measurement');
 387 TEST_F(AudioLowLatencyInputOutputTest, DISABLED_FullDuplexDelayMeasurement) {
 388   if (!CanRunAudioTests())
 389     return;
 390
 391   AudioInputStreamWrapper aisw(audio_manager());
 392   AudioInputStream* ais = aisw.Create();
 393   EXPECT_TRUE(ais);
 394
 395   AudioOutputStreamWrapper aosw(audio_manager());
 396   AudioOutputStream* aos = aosw.Create();
 397   EXPECT_TRUE(aos);
 398
 399   // This test only supports identical parameters in both directions.
 400   // TODO(henrika): it is possible to cut delay here by using different
 401   // buffer sizes for input and output.
 402   if (aisw.sample_rate() != aosw.sample_rate() ||
 403       aisw.samples_per_packet() != aosw.samples_per_packet() ||
 404       aisw.channels()!= aosw.channels() ||
 405       aisw.bits_per_sample() != aosw.bits_per_sample()) {
 406     LOG(ERROR) << "This test requires symmetric input and output parameters. "
 407         "Ensure that sample rate and number of channels are identical in "
 408         "both directions";
 409     aos->Close();
 410     ais->Close();
 411     return;
 412   }
 413
 414   EXPECT_TRUE(ais->Open());
 415   EXPECT_TRUE(aos->Open());
 416
 417   FullDuplexAudioSinkSource full_duplex(
 418       aisw.sample_rate(), aisw.samples_per_packet(), aisw.channels());
 419
 420   VLOG(0) << ">> You should now be able to hear yourself in loopback...";
 421   DVLOG(0) << "   sample_rate       : " << aisw.sample_rate();
 422   DVLOG(0) << "   samples_per_packet: " << aisw.samples_per_packet();
 423   DVLOG(0) << "   channels          : " << aisw.channels();
 424
 425   ais->Start(&full_duplex);
 426   aos->Start(&full_duplex);
 427
 428   // Wait for approximately 10 seconds. The user shall hear his own voice
 429   // in loop back during this time. At the same time, delay recordings are
 430   // performed and stored in the output text file.
 431   message_loop()->PostDelayedTask(FROM_HERE,
 432       base::MessageLoop::QuitClosure(), TestTimeouts::action_timeout());
 433   message_loop()->Run();
 434
 435   aos->Stop();
 436   ais->Stop();
 437
 438   // All Close() operations that run on the mocked audio thread,
 439   // should be synchronous and not post additional close tasks to
 440   // mocked the audio thread. Hence, there is no need to call
 441   // message_loop()->RunUntilIdle() after the Close() methods.
 442   aos->Close();
 443   ais->Close();
 444 }
 445
 446 }  // namespace media