src/media/filters/audio_renderer_algorithm_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 // The format of these tests are to enqueue a known amount of data and then
   6 // request the exact amount we expect in order to dequeue the known amount of
   7 // data.  This ensures that for any rate we are consuming input data at the
   8 // correct rate.  We always pass in a very large destination buffer with the
   9 // expectation that FillBuffer() will fill as much as it can but no more.
  10
  11 #include <algorithm>  // For std::min().
  12 #include <cmath>
  13 #include <vector>
  14
  15 #include "base/bind.h"
  16 #include "base/callback.h"
  17 #include "base/memory/scoped_ptr.h"
  18 #include "media/base/audio_buffer.h"
  19 #include "media/base/audio_bus.h"
  20 #include "media/base/buffers.h"
  21 #include "media/base/channel_layout.h"
  22 #include "media/base/test_helpers.h"
  23 #include "media/filters/audio_renderer_algorithm.h"
  24 #include "media/filters/wsola_internals.h"
  25 #include "testing/gtest/include/gtest/gtest.h"
  26
  27 namespace media {
  28
  29 const int kFrameSize = 250;
  30 const int kSamplesPerSecond = 3000;
  31 const int kOutputDurationInSec = 10;
  32
  33 static void FillWithSquarePulseTrain(
  34     int half_pulse_width, int offset, int num_samples, float* data) {
  35   ASSERT_GE(offset, 0);
  36   ASSERT_LE(offset, num_samples);
  37
  38   // Fill backward from |offset| - 1 toward zero, starting with -1, alternating
  39   // between -1 and 1 every |pulse_width| samples.
  40   float pulse = -1.0f;
  41   for (int n = offset - 1, k = 0; n >= 0; --n, ++k) {
  42     if (k >= half_pulse_width) {
  43       pulse = -pulse;
  44       k = 0;
  45     }
  46     data[n] = pulse;
  47   }
  48
  49   // Fill forward from |offset| towards the end, starting with 1, alternating
  50   // between 1 and -1 every |pulse_width| samples.
  51   pulse = 1.0f;
  52   for (int n = offset, k = 0; n < num_samples; ++n, ++k) {
  53     if (k >= half_pulse_width) {
  54       pulse = -pulse;
  55       k = 0;
  56     }
  57     data[n] = pulse;
  58   }
  59 }
  60
  61 static void FillWithSquarePulseTrain(
  62     int half_pulse_width, int offset, int channel, AudioBus* audio_bus) {
  63   FillWithSquarePulseTrain(half_pulse_width, offset, audio_bus->frames(),
  64                            audio_bus->channel(channel));
  65 }
  66
  67 class AudioRendererAlgorithmTest : public testing::Test {
  68  public:
  69   AudioRendererAlgorithmTest()
  70       : frames_enqueued_(0),
  71         channels_(0),
  72         channel_layout_(CHANNEL_LAYOUT_NONE),
  73         sample_format_(kUnknownSampleFormat),
  74         samples_per_second_(0),
  75         bytes_per_sample_(0) {
  76   }
  77
  78   virtual ~AudioRendererAlgorithmTest() {}
  79
  80   void Initialize() {
  81     Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatS16, 3000);
  82   }
  83
  84   void Initialize(ChannelLayout channel_layout,
  85                   SampleFormat sample_format,
  86                   int samples_per_second) {
  87     channels_ = ChannelLayoutToChannelCount(channel_layout);
  88     samples_per_second_ = samples_per_second;
  89     channel_layout_ = channel_layout;
  90     sample_format_ = sample_format;
  91     bytes_per_sample_ = SampleFormatToBytesPerChannel(sample_format);
  92     AudioParameters params(media::AudioParameters::AUDIO_PCM_LINEAR,
  93                            channel_layout,
  94                            samples_per_second,
  95                            bytes_per_sample_ * 8,
  96                            samples_per_second / 100);
  97     algorithm_.Initialize(1, params);
  98     FillAlgorithmQueue();
  99   }
 100
 101   void FillAlgorithmQueue() {
 102     // The value of the data is meaningless; we just want non-zero data to
 103     // differentiate it from muted data.
 104     scoped_refptr<AudioBuffer> buffer;
 105     while (!algorithm_.IsQueueFull()) {
 106       switch (sample_format_) {
 107         case kSampleFormatU8:
 108           buffer = MakeAudioBuffer<uint8>(
 109               sample_format_,
 110               channel_layout_,
 111               ChannelLayoutToChannelCount(channel_layout_),
 112               samples_per_second_,
 113               1,
 114               1,
 115               kFrameSize,
 116               kNoTimestamp());
 117           break;
 118         case kSampleFormatS16:
 119           buffer = MakeAudioBuffer<int16>(
 120               sample_format_,
 121               channel_layout_,
 122               ChannelLayoutToChannelCount(channel_layout_),
 123               samples_per_second_,
 124               1,
 125               1,
 126               kFrameSize,
 127               kNoTimestamp());
 128           break;
 129         case kSampleFormatS32:
 130           buffer = MakeAudioBuffer<int32>(
 131               sample_format_,
 132               channel_layout_,
 133               ChannelLayoutToChannelCount(channel_layout_),
 134               samples_per_second_,
 135               1,
 136               1,
 137               kFrameSize,
 138               kNoTimestamp());
 139           break;
 140         default:
 141           NOTREACHED() << "Unrecognized format " << sample_format_;
 142       }
 143       algorithm_.EnqueueBuffer(buffer);
 144       frames_enqueued_ += kFrameSize;
 145     }
 146   }
 147
 148   void CheckFakeData(AudioBus* audio_data, int frames_written) {
 149     // Check each channel individually.
 150     for (int ch = 0; ch < channels_; ++ch) {
 151       bool all_zero = true;
 152       for (int i = 0; i < frames_written && all_zero; ++i)
 153         all_zero = audio_data->channel(ch)[i] == 0.0f;
 154       ASSERT_FALSE(all_zero) << " for channel " << ch;
 155     }
 156   }
 157
 158   int ComputeConsumedFrames(int initial_frames_enqueued,
 159                             int initial_frames_buffered) {
 160     int frame_delta = frames_enqueued_ - initial_frames_enqueued;
 161     int buffered_delta = algorithm_.frames_buffered() - initial_frames_buffered;
 162     int consumed = frame_delta - buffered_delta;
 163     CHECK_GE(consumed, 0);
 164     return consumed;
 165   }
 166
 167   void TestPlaybackRate(double playback_rate) {
 168     const int kDefaultBufferSize = algorithm_.samples_per_second() / 100;
 169     const int kDefaultFramesRequested = kOutputDurationInSec *
 170         algorithm_.samples_per_second();
 171
 172     TestPlaybackRate(
 173         playback_rate, kDefaultBufferSize, kDefaultFramesRequested);
 174   }
 175
 176   void TestPlaybackRate(double playback_rate,
 177                         int buffer_size_in_frames,
 178                         int total_frames_requested) {
 179     int initial_frames_enqueued = frames_enqueued_;
 180     int initial_frames_buffered = algorithm_.frames_buffered();
 181     algorithm_.SetPlaybackRate(static_cast<float>(playback_rate));
 182
 183     scoped_ptr<AudioBus> bus =
 184         AudioBus::Create(channels_, buffer_size_in_frames);
 185     if (playback_rate == 0.0) {
 186       int frames_written =
 187           algorithm_.FillBuffer(bus.get(), buffer_size_in_frames);
 188       EXPECT_EQ(0, frames_written);
 189       return;
 190     }
 191
 192     int frames_remaining = total_frames_requested;
 193     bool first_fill_buffer = true;
 194     while (frames_remaining > 0) {
 195       int frames_requested = std::min(buffer_size_in_frames, frames_remaining);
 196       int frames_written = algorithm_.FillBuffer(bus.get(), frames_requested);
 197       ASSERT_GT(frames_written, 0) << "Requested: " << frames_requested
 198                                    << ", playing at " << playback_rate;
 199
 200       // Do not check data if it is first pull out and only one frame written.
 201       // The very first frame out of WSOLA is always zero because of
 202       // overlap-and-add window, which is zero for the first sample. Therefore,
 203       // if at very first buffer-fill only one frame is written, that is zero
 204       // which might cause exception in CheckFakeData().
 205       if (!first_fill_buffer || frames_written > 1)
 206         CheckFakeData(bus.get(), frames_written);
 207       first_fill_buffer = false;
 208       frames_remaining -= frames_written;
 209
 210       FillAlgorithmQueue();
 211     }
 212
 213     int frames_consumed =
 214         ComputeConsumedFrames(initial_frames_enqueued, initial_frames_buffered);
 215
 216     // If playing back at normal speed, we should always get back the same
 217     // number of bytes requested.
 218     if (playback_rate == 1.0) {
 219       EXPECT_EQ(total_frames_requested, frames_consumed);
 220       return;
 221     }
 222
 223     // Otherwise, allow |kMaxAcceptableDelta| difference between the target and
 224     // actual playback rate.
 225     // When |kSamplesPerSecond| and |total_frames_requested| are reasonably
 226     // large, one can expect less than a 1% difference in most cases. In our
 227     // current implementation, sped up playback is less accurate than slowed
 228     // down playback, and for playback_rate > 1, playback rate generally gets
 229     // less and less accurate the farther it drifts from 1 (though this is
 230     // nonlinear).
 231     double actual_playback_rate =
 232         1.0 * frames_consumed / total_frames_requested;
 233     EXPECT_NEAR(playback_rate, actual_playback_rate, playback_rate / 100.0);
 234   }
 235
 236   void WsolaTest(float playback_rate) {
 237     const int kSampleRateHz = 48000;
 238     const ChannelLayout kChannelLayout = CHANNEL_LAYOUT_STEREO;
 239     const int kBytesPerSample = 2;
 240     const int kNumFrames = kSampleRateHz / 100;  // 10 milliseconds.
 241
 242     channels_ = ChannelLayoutToChannelCount(kChannelLayout);
 243     AudioParameters params(AudioParameters::AUDIO_PCM_LINEAR, kChannelLayout,
 244                            kSampleRateHz, kBytesPerSample * 8, kNumFrames);
 245     algorithm_.Initialize(playback_rate, params);
 246
 247     // A pulse is 6 milliseconds (even number of samples).
 248     const int kPulseWidthSamples = 6 * kSampleRateHz / 1000;
 249     const int kHalfPulseWidthSamples = kPulseWidthSamples / 2;
 250
 251     // For the ease of implementation get 1 frame every call to FillBuffer().
 252     scoped_ptr<AudioBus> output = AudioBus::Create(channels_, 1);
 253
 254     // Input buffer to inject pulses.
 255     scoped_refptr<AudioBuffer> input =
 256         AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
 257                                   kChannelLayout,
 258                                   channels_,
 259                                   kSampleRateHz,
 260                                   kPulseWidthSamples);
 261
 262     const std::vector<uint8*>& channel_data = input->channel_data();
 263
 264     // Fill |input| channels.
 265     FillWithSquarePulseTrain(kHalfPulseWidthSamples, 0, kPulseWidthSamples,
 266                              reinterpret_cast<float*>(channel_data[0]));
 267     FillWithSquarePulseTrain(kHalfPulseWidthSamples, kHalfPulseWidthSamples,
 268                              kPulseWidthSamples,
 269                              reinterpret_cast<float*>(channel_data[1]));
 270
 271     // A buffer for the output until a complete pulse is created. Then
 272     // reference pulse is compared with this buffer.
 273     scoped_ptr<AudioBus> pulse_buffer = AudioBus::Create(
 274         channels_, kPulseWidthSamples);
 275
 276     const float kTolerance = 0.000001f;
 277     // Equivalent of 4 seconds.
 278     const int kNumRequestedPulses = kSampleRateHz * 4 / kPulseWidthSamples;
 279     for (int n = 0; n < kNumRequestedPulses; ++n) {
 280       int num_buffered_frames = 0;
 281       while (num_buffered_frames < kPulseWidthSamples) {
 282         int num_samples = algorithm_.FillBuffer(output.get(), 1);
 283         ASSERT_LE(num_samples, 1);
 284         if (num_samples > 0) {
 285           output->CopyPartialFramesTo(0, num_samples, num_buffered_frames,
 286                                       pulse_buffer.get());
 287           num_buffered_frames++;
 288         } else {
 289           algorithm_.EnqueueBuffer(input);
 290         }
 291       }
 292
 293       // Pulses in the first half of WSOLA AOL frame are not constructed
 294       // perfectly. Do not check them.
 295       if (n > 3) {
 296          for (int m = 0; m < channels_; ++m) {
 297           const float* pulse_ch = pulse_buffer->channel(m);
 298
 299           // Because of overlap-and-add we might have round off error.
 300           for (int k = 0; k < kPulseWidthSamples; ++k) {
 301             ASSERT_NEAR(reinterpret_cast<float*>(channel_data[m])[k],
 302                         pulse_ch[k], kTolerance) << " loop " << n
 303                                 << " channel/sample " << m << "/" << k;
 304           }
 305         }
 306       }
 307
 308       // Zero out the buffer to be sure the next comparison is relevant.
 309       pulse_buffer->Zero();
 310     }
 311   }
 312
 313  protected:
 314   AudioRendererAlgorithm algorithm_;
 315   int frames_enqueued_;
 316   int channels_;
 317   ChannelLayout channel_layout_;
 318   SampleFormat sample_format_;
 319   int samples_per_second_;
 320   int bytes_per_sample_;
 321 };
 322
 323 TEST_F(AudioRendererAlgorithmTest, FillBuffer_NormalRate) {
 324   Initialize();
 325   TestPlaybackRate(1.0);
 326 }
 327
 328 TEST_F(AudioRendererAlgorithmTest, FillBuffer_NearlyNormalFasterRate) {
 329   Initialize();
 330   TestPlaybackRate(1.0001);
 331 }
 332
 333 TEST_F(AudioRendererAlgorithmTest, FillBuffer_NearlyNormalSlowerRate) {
 334   Initialize();
 335   TestPlaybackRate(0.9999);
 336 }
 337
 338 TEST_F(AudioRendererAlgorithmTest, FillBuffer_OneAndAQuarterRate) {
 339   Initialize();
 340   TestPlaybackRate(1.25);
 341 }
 342
 343 TEST_F(AudioRendererAlgorithmTest, FillBuffer_OneAndAHalfRate) {
 344   Initialize();
 345   TestPlaybackRate(1.5);
 346 }
 347
 348 TEST_F(AudioRendererAlgorithmTest, FillBuffer_DoubleRate) {
 349   Initialize();
 350   TestPlaybackRate(2.0);
 351 }
 352
 353 TEST_F(AudioRendererAlgorithmTest, FillBuffer_EightTimesRate) {
 354   Initialize();
 355   TestPlaybackRate(8.0);
 356 }
 357
 358 TEST_F(AudioRendererAlgorithmTest, FillBuffer_ThreeQuartersRate) {
 359   Initialize();
 360   TestPlaybackRate(0.75);
 361 }
 362
 363 TEST_F(AudioRendererAlgorithmTest, FillBuffer_HalfRate) {
 364   Initialize();
 365   TestPlaybackRate(0.5);
 366 }
 367
 368 TEST_F(AudioRendererAlgorithmTest, FillBuffer_QuarterRate) {
 369   Initialize();
 370   TestPlaybackRate(0.25);
 371 }
 372
 373 TEST_F(AudioRendererAlgorithmTest, FillBuffer_Pause) {
 374   Initialize();
 375   TestPlaybackRate(0.0);
 376 }
 377
 378 TEST_F(AudioRendererAlgorithmTest, FillBuffer_SlowDown) {
 379   Initialize();
 380   TestPlaybackRate(4.5);
 381   TestPlaybackRate(3.0);
 382   TestPlaybackRate(2.0);
 383   TestPlaybackRate(1.0);
 384   TestPlaybackRate(0.5);
 385   TestPlaybackRate(0.25);
 386 }
 387
 388 TEST_F(AudioRendererAlgorithmTest, FillBuffer_SpeedUp) {
 389   Initialize();
 390   TestPlaybackRate(0.25);
 391   TestPlaybackRate(0.5);
 392   TestPlaybackRate(1.0);
 393   TestPlaybackRate(2.0);
 394   TestPlaybackRate(3.0);
 395   TestPlaybackRate(4.5);
 396 }
 397
 398 TEST_F(AudioRendererAlgorithmTest, FillBuffer_JumpAroundSpeeds) {
 399   Initialize();
 400   TestPlaybackRate(2.1);
 401   TestPlaybackRate(0.9);
 402   TestPlaybackRate(0.6);
 403   TestPlaybackRate(1.4);
 404   TestPlaybackRate(0.3);
 405 }
 406
 407 TEST_F(AudioRendererAlgorithmTest, FillBuffer_SmallBufferSize) {
 408   Initialize();
 409   static const int kBufferSizeInFrames = 1;
 410   static const int kFramesRequested = kOutputDurationInSec * kSamplesPerSecond;
 411   TestPlaybackRate(1.0, kBufferSizeInFrames, kFramesRequested);
 412   TestPlaybackRate(0.5, kBufferSizeInFrames, kFramesRequested);
 413   TestPlaybackRate(1.5, kBufferSizeInFrames, kFramesRequested);
 414 }
 415
 416 TEST_F(AudioRendererAlgorithmTest, FillBuffer_LargeBufferSize) {
 417   Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatS16, 44100);
 418   TestPlaybackRate(1.0);
 419   TestPlaybackRate(0.5);
 420   TestPlaybackRate(1.5);
 421 }
 422
 423 TEST_F(AudioRendererAlgorithmTest, FillBuffer_LowerQualityAudio) {
 424   Initialize(CHANNEL_LAYOUT_MONO, kSampleFormatU8, kSamplesPerSecond);
 425   TestPlaybackRate(1.0);
 426   TestPlaybackRate(0.5);
 427   TestPlaybackRate(1.5);
 428 }
 429
 430 TEST_F(AudioRendererAlgorithmTest, FillBuffer_HigherQualityAudio) {
 431   Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatS32, kSamplesPerSecond);
 432   TestPlaybackRate(1.0);
 433   TestPlaybackRate(0.5);
 434   TestPlaybackRate(1.5);
 435 }
 436
 437 TEST_F(AudioRendererAlgorithmTest, DotProduct) {
 438   const int kChannels = 3;
 439   const int kFrames = 20;
 440   const int kHalfPulseWidth = 2;
 441
 442   scoped_ptr<AudioBus> a = AudioBus::Create(kChannels, kFrames);
 443   scoped_ptr<AudioBus> b = AudioBus::Create(kChannels, kFrames);
 444
 445   scoped_ptr<float[]> dot_prod(new float[kChannels]);
 446
 447   FillWithSquarePulseTrain(kHalfPulseWidth, 0, 0, a.get());
 448   FillWithSquarePulseTrain(kHalfPulseWidth, 1, 1, a.get());
 449   FillWithSquarePulseTrain(kHalfPulseWidth, 2, 2, a.get());
 450
 451   FillWithSquarePulseTrain(kHalfPulseWidth, 0, 0, b.get());
 452   FillWithSquarePulseTrain(kHalfPulseWidth, 0, 1, b.get());
 453   FillWithSquarePulseTrain(kHalfPulseWidth, 0, 2, b.get());
 454
 455   internal::MultiChannelDotProduct(a.get(), 0, b.get(), 0, kFrames,
 456                                    dot_prod.get());
 457
 458   EXPECT_FLOAT_EQ(kFrames, dot_prod[0]);
 459   EXPECT_FLOAT_EQ(0, dot_prod[1]);
 460   EXPECT_FLOAT_EQ(-kFrames, dot_prod[2]);
 461
 462   internal::MultiChannelDotProduct(a.get(), 4, b.get(), 8, kFrames / 2,
 463                                    dot_prod.get());
 464
 465   EXPECT_FLOAT_EQ(kFrames / 2, dot_prod[0]);
 466   EXPECT_FLOAT_EQ(0, dot_prod[1]);
 467   EXPECT_FLOAT_EQ(-kFrames / 2, dot_prod[2]);
 468 }
 469
 470 TEST_F(AudioRendererAlgorithmTest, MovingBlockEnergy) {
 471   const int kChannels = 2;
 472   const int kFrames = 20;
 473   const int kFramesPerBlock = 3;
 474   const int kNumBlocks = kFrames - (kFramesPerBlock - 1);
 475   scoped_ptr<AudioBus> a = AudioBus::Create(kChannels, kFrames);
 476   scoped_ptr<float[]> energies(new float[kChannels * kNumBlocks]);
 477   float* ch_left = a->channel(0);
 478   float* ch_right = a->channel(1);
 479
 480   // Fill up both channels.
 481   for (int n = 0; n < kFrames; ++n) {
 482     ch_left[n] = n;
 483     ch_right[n] = kFrames - 1 - n;
 484   }
 485
 486   internal::MultiChannelMovingBlockEnergies(a.get(), kFramesPerBlock,
 487                                             energies.get());
 488
 489   // Check if the energy of candidate blocks of each channel computed correctly.
 490   for (int n = 0; n < kNumBlocks; ++n) {
 491     float expected_energy = 0;
 492     for (int k = 0; k < kFramesPerBlock; ++k)
 493       expected_energy += ch_left[n + k] * ch_left[n + k];
 494
 495     // Left (first) channel.
 496     EXPECT_FLOAT_EQ(expected_energy, energies[2 * n]);
 497
 498     expected_energy = 0;
 499     for (int k = 0; k < kFramesPerBlock; ++k)
 500       expected_energy += ch_right[n + k] * ch_right[n + k];
 501
 502     // Second (right) channel.
 503     EXPECT_FLOAT_EQ(expected_energy, energies[2 * n + 1]);
 504   }
 505 }
 506
 507 TEST_F(AudioRendererAlgorithmTest, FullAndDecimatedSearch) {
 508   const int kFramesInSearchRegion = 12;
 509   const int kChannels = 2;
 510   float ch_0[] = {
 511       0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 1.0f, 1.0f, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f };
 512   float ch_1[] = {
 513       0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.1f, 1.0f, 0.1f, 0.0f, 0.0f };
 514   ASSERT_EQ(sizeof(ch_0), sizeof(ch_1));
 515   ASSERT_EQ(static_cast<size_t>(kFramesInSearchRegion),
 516             sizeof(ch_0) / sizeof(*ch_0));
 517   scoped_ptr<AudioBus> search_region = AudioBus::Create(kChannels,
 518                                                         kFramesInSearchRegion);
 519   float* ch = search_region->channel(0);
 520   memcpy(ch, ch_0, sizeof(float) * kFramesInSearchRegion);
 521   ch = search_region->channel(1);
 522   memcpy(ch, ch_1, sizeof(float) * kFramesInSearchRegion);
 523
 524   const int kFramePerBlock = 4;
 525   float target_0[] = { 1.0f, 1.0f, 1.0f, 0.0f };
 526   float target_1[] = { 0.0f, 1.0f, 0.1f, 1.0f };
 527   ASSERT_EQ(sizeof(target_0), sizeof(target_1));
 528   ASSERT_EQ(static_cast<size_t>(kFramePerBlock),
 529             sizeof(target_0) / sizeof(*target_0));
 530
 531   scoped_ptr<AudioBus> target = AudioBus::Create(kChannels,
 532                                                  kFramePerBlock);
 533   ch = target->channel(0);
 534   memcpy(ch, target_0, sizeof(float) * kFramePerBlock);
 535   ch = target->channel(1);
 536   memcpy(ch, target_1, sizeof(float) * kFramePerBlock);
 537
 538   scoped_ptr<float[]> energy_target(new float[kChannels]);
 539
 540   internal::MultiChannelDotProduct(target.get(), 0, target.get(), 0,
 541                                    kFramePerBlock, energy_target.get());
 542
 543   ASSERT_EQ(3.f, energy_target[0]);
 544   ASSERT_EQ(2.01f, energy_target[1]);
 545
 546   const int kNumCandidBlocks = kFramesInSearchRegion - (kFramePerBlock - 1);
 547   scoped_ptr<float[]> energy_candid_blocks(new float[kNumCandidBlocks *
 548                                                      kChannels]);
 549
 550   internal::MultiChannelMovingBlockEnergies(
 551       search_region.get(), kFramePerBlock, energy_candid_blocks.get());
 552
 553   // Check the energy of the candidate blocks of the first channel.
 554   ASSERT_FLOAT_EQ(0, energy_candid_blocks[0]);
 555   ASSERT_FLOAT_EQ(0, energy_candid_blocks[2]);
 556   ASSERT_FLOAT_EQ(1, energy_candid_blocks[4]);
 557   ASSERT_FLOAT_EQ(2, energy_candid_blocks[6]);
 558   ASSERT_FLOAT_EQ(3, energy_candid_blocks[8]);
 559   ASSERT_FLOAT_EQ(3, energy_candid_blocks[10]);
 560   ASSERT_FLOAT_EQ(2, energy_candid_blocks[12]);
 561   ASSERT_FLOAT_EQ(1, energy_candid_blocks[14]);
 562   ASSERT_FLOAT_EQ(0, energy_candid_blocks[16]);
 563
 564   // Check the energy of the candidate blocks of the second channel.
 565   ASSERT_FLOAT_EQ(0, energy_candid_blocks[1]);
 566   ASSERT_FLOAT_EQ(0, energy_candid_blocks[3]);
 567   ASSERT_FLOAT_EQ(0, energy_candid_blocks[5]);
 568   ASSERT_FLOAT_EQ(0, energy_candid_blocks[7]);
 569   ASSERT_FLOAT_EQ(0.01f, energy_candid_blocks[9]);
 570   ASSERT_FLOAT_EQ(1.01f, energy_candid_blocks[11]);
 571   ASSERT_FLOAT_EQ(1.02f, energy_candid_blocks[13]);
 572   ASSERT_FLOAT_EQ(1.02f, energy_candid_blocks[15]);
 573   ASSERT_FLOAT_EQ(1.01f, energy_candid_blocks[17]);
 574
 575   // An interval which is of no effect.
 576   internal::Interval exclude_interval = std::make_pair(-100, -10);
 577   EXPECT_EQ(5, internal::FullSearch(
 578       0, kNumCandidBlocks - 1, exclude_interval, target.get(),
 579       search_region.get(), energy_target.get(), energy_candid_blocks.get()));
 580
 581   // Exclude the the best match.
 582   exclude_interval = std::make_pair(2, 5);
 583   EXPECT_EQ(7, internal::FullSearch(
 584       0, kNumCandidBlocks - 1, exclude_interval, target.get(),
 585       search_region.get(), energy_target.get(), energy_candid_blocks.get()));
 586
 587   // An interval which is of no effect.
 588   exclude_interval = std::make_pair(-100, -10);
 589   EXPECT_EQ(4, internal::DecimatedSearch(
 590       4, exclude_interval, target.get(), search_region.get(),
 591       energy_target.get(), energy_candid_blocks.get()));
 592
 593   EXPECT_EQ(5, internal::OptimalIndex(search_region.get(), target.get(),
 594                                       exclude_interval));
 595 }
 596
 597 TEST_F(AudioRendererAlgorithmTest, QuadraticInterpolation) {
 598   // Arbitrary coefficients.
 599   const float kA = 0.7f;
 600   const float kB = 1.2f;
 601   const float kC = 0.8f;
 602
 603   float y_values[3];
 604   y_values[0] = kA - kB + kC;
 605   y_values[1] = kC;
 606   y_values[2] = kA + kB + kC;
 607
 608   float extremum;
 609   float extremum_value;
 610
 611   internal::QuadraticInterpolation(y_values, &extremum, &extremum_value);
 612
 613   float x_star = -kB / (2.f * kA);
 614   float y_star = kA * x_star * x_star + kB * x_star + kC;
 615
 616   EXPECT_FLOAT_EQ(x_star, extremum);
 617   EXPECT_FLOAT_EQ(y_star, extremum_value);
 618 }
 619
 620 TEST_F(AudioRendererAlgorithmTest, QuadraticInterpolation_Colinear) {
 621   float y_values[3];
 622   y_values[0] = 1.0;
 623   y_values[1] = 1.0;
 624   y_values[2] = 1.0;
 625
 626   float extremum;
 627   float extremum_value;
 628
 629   internal::QuadraticInterpolation(y_values, &extremum, &extremum_value);
 630
 631   EXPECT_FLOAT_EQ(extremum, 0.0);
 632   EXPECT_FLOAT_EQ(extremum_value, 1.0);
 633 }
 634
 635 TEST_F(AudioRendererAlgorithmTest, WsolaSlowdown) {
 636   WsolaTest(0.6f);
 637 }
 638
 639 TEST_F(AudioRendererAlgorithmTest, WsolaSpeedup) {
 640   WsolaTest(1.6f);
 641 }
 642
 643 }  // namespace media