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      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_EQ(algorithm_.is_muted(), 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
    644