| Index: media/filters/audio_renderer_algorithm_unittest.cc
|
| diff --git a/media/filters/audio_renderer_algorithm_unittest.cc b/media/filters/audio_renderer_algorithm_unittest.cc
|
| deleted file mode 100644
|
| index 47ce30e84a66c8703df63cb2138f5b98bb4a3dfc..0000000000000000000000000000000000000000
|
| --- a/media/filters/audio_renderer_algorithm_unittest.cc
|
| +++ /dev/null
|
| @@ -1,685 +0,0 @@
|
| -// Copyright (c) 2012 The Chromium Authors. All rights reserved.
|
| -// Use of this source code is governed by a BSD-style license that can be
|
| -// found in the LICENSE file.
|
| -//
|
| -// The format of these tests are to enqueue a known amount of data and then
|
| -// request the exact amount we expect in order to dequeue the known amount of
|
| -// data. This ensures that for any rate we are consuming input data at the
|
| -// correct rate. We always pass in a very large destination buffer with the
|
| -// expectation that FillBuffer() will fill as much as it can but no more.
|
| -
|
| -#include <algorithm> // For std::min().
|
| -#include <cmath>
|
| -#include <vector>
|
| -
|
| -#include "base/bind.h"
|
| -#include "base/callback.h"
|
| -#include "base/memory/scoped_ptr.h"
|
| -#include "media/base/audio_buffer.h"
|
| -#include "media/base/audio_bus.h"
|
| -#include "media/base/buffers.h"
|
| -#include "media/base/channel_layout.h"
|
| -#include "media/base/test_helpers.h"
|
| -#include "media/filters/audio_renderer_algorithm.h"
|
| -#include "media/filters/wsola_internals.h"
|
| -#include "testing/gtest/include/gtest/gtest.h"
|
| -
|
| -namespace media {
|
| -
|
| -const int kFrameSize = 250;
|
| -const int kSamplesPerSecond = 3000;
|
| -const int kOutputDurationInSec = 10;
|
| -
|
| -static void FillWithSquarePulseTrain(
|
| - int half_pulse_width, int offset, int num_samples, float* data) {
|
| - ASSERT_GE(offset, 0);
|
| - ASSERT_LE(offset, num_samples);
|
| -
|
| - // Fill backward from |offset| - 1 toward zero, starting with -1, alternating
|
| - // between -1 and 1 every |pulse_width| samples.
|
| - float pulse = -1.0f;
|
| - for (int n = offset - 1, k = 0; n >= 0; --n, ++k) {
|
| - if (k >= half_pulse_width) {
|
| - pulse = -pulse;
|
| - k = 0;
|
| - }
|
| - data[n] = pulse;
|
| - }
|
| -
|
| - // Fill forward from |offset| towards the end, starting with 1, alternating
|
| - // between 1 and -1 every |pulse_width| samples.
|
| - pulse = 1.0f;
|
| - for (int n = offset, k = 0; n < num_samples; ++n, ++k) {
|
| - if (k >= half_pulse_width) {
|
| - pulse = -pulse;
|
| - k = 0;
|
| - }
|
| - data[n] = pulse;
|
| - }
|
| -}
|
| -
|
| -static void FillWithSquarePulseTrain(
|
| - int half_pulse_width, int offset, int channel, AudioBus* audio_bus) {
|
| - FillWithSquarePulseTrain(half_pulse_width, offset, audio_bus->frames(),
|
| - audio_bus->channel(channel));
|
| -}
|
| -
|
| -class AudioRendererAlgorithmTest : public testing::Test {
|
| - public:
|
| - AudioRendererAlgorithmTest()
|
| - : frames_enqueued_(0),
|
| - channels_(0),
|
| - channel_layout_(CHANNEL_LAYOUT_NONE),
|
| - sample_format_(kUnknownSampleFormat),
|
| - samples_per_second_(0),
|
| - bytes_per_sample_(0) {
|
| - }
|
| -
|
| - ~AudioRendererAlgorithmTest() override {}
|
| -
|
| - void Initialize() {
|
| - Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatS16, 3000);
|
| - }
|
| -
|
| - void Initialize(ChannelLayout channel_layout,
|
| - SampleFormat sample_format,
|
| - int samples_per_second) {
|
| - channels_ = ChannelLayoutToChannelCount(channel_layout);
|
| - samples_per_second_ = samples_per_second;
|
| - channel_layout_ = channel_layout;
|
| - sample_format_ = sample_format;
|
| - bytes_per_sample_ = SampleFormatToBytesPerChannel(sample_format);
|
| - AudioParameters params(media::AudioParameters::AUDIO_PCM_LINEAR,
|
| - channel_layout,
|
| - samples_per_second,
|
| - bytes_per_sample_ * 8,
|
| - samples_per_second / 100);
|
| - algorithm_.Initialize(params);
|
| - FillAlgorithmQueue();
|
| - }
|
| -
|
| - void FillAlgorithmQueue() {
|
| - // The value of the data is meaningless; we just want non-zero data to
|
| - // differentiate it from muted data.
|
| - scoped_refptr<AudioBuffer> buffer;
|
| - while (!algorithm_.IsQueueFull()) {
|
| - switch (sample_format_) {
|
| - case kSampleFormatU8:
|
| - buffer = MakeAudioBuffer<uint8>(
|
| - sample_format_,
|
| - channel_layout_,
|
| - ChannelLayoutToChannelCount(channel_layout_),
|
| - samples_per_second_,
|
| - 1,
|
| - 1,
|
| - kFrameSize,
|
| - kNoTimestamp());
|
| - break;
|
| - case kSampleFormatS16:
|
| - buffer = MakeAudioBuffer<int16>(
|
| - sample_format_,
|
| - channel_layout_,
|
| - ChannelLayoutToChannelCount(channel_layout_),
|
| - samples_per_second_,
|
| - 1,
|
| - 1,
|
| - kFrameSize,
|
| - kNoTimestamp());
|
| - break;
|
| - case kSampleFormatS32:
|
| - buffer = MakeAudioBuffer<int32>(
|
| - sample_format_,
|
| - channel_layout_,
|
| - ChannelLayoutToChannelCount(channel_layout_),
|
| - samples_per_second_,
|
| - 1,
|
| - 1,
|
| - kFrameSize,
|
| - kNoTimestamp());
|
| - break;
|
| - default:
|
| - NOTREACHED() << "Unrecognized format " << sample_format_;
|
| - }
|
| - algorithm_.EnqueueBuffer(buffer);
|
| - frames_enqueued_ += kFrameSize;
|
| - }
|
| - }
|
| -
|
| - bool VerifyAudioData(AudioBus* bus, int offset, int frames, float value) {
|
| - for (int ch = 0; ch < bus->channels(); ++ch) {
|
| - for (int i = offset; i < offset + frames; ++i) {
|
| - if (bus->channel(ch)[i] != value)
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| - }
|
| -
|
| - bool AudioDataIsMuted(AudioBus* audio_data, int frames_written) {
|
| - return VerifyAudioData(audio_data, 0, frames_written, 0);
|
| - }
|
| -
|
| - int ComputeConsumedFrames(int initial_frames_enqueued,
|
| - int initial_frames_buffered) {
|
| - int frame_delta = frames_enqueued_ - initial_frames_enqueued;
|
| - int buffered_delta = algorithm_.frames_buffered() - initial_frames_buffered;
|
| - int consumed = frame_delta - buffered_delta;
|
| - CHECK_GE(consumed, 0);
|
| - return consumed;
|
| - }
|
| -
|
| - void TestPlaybackRate(double playback_rate) {
|
| - const int kDefaultBufferSize = algorithm_.samples_per_second() / 100;
|
| - const int kDefaultFramesRequested = kOutputDurationInSec *
|
| - algorithm_.samples_per_second();
|
| -
|
| - TestPlaybackRate(
|
| - playback_rate, kDefaultBufferSize, kDefaultFramesRequested);
|
| - }
|
| -
|
| - void TestPlaybackRate(double playback_rate,
|
| - int buffer_size_in_frames,
|
| - int total_frames_requested) {
|
| - int initial_frames_enqueued = frames_enqueued_;
|
| - int initial_frames_buffered = algorithm_.frames_buffered();
|
| -
|
| - scoped_ptr<AudioBus> bus =
|
| - AudioBus::Create(channels_, buffer_size_in_frames);
|
| - if (playback_rate == 0.0) {
|
| - int frames_written = algorithm_.FillBuffer(
|
| - bus.get(), 0, buffer_size_in_frames, playback_rate);
|
| - EXPECT_EQ(0, frames_written);
|
| - return;
|
| - }
|
| -
|
| - bool expect_muted = (playback_rate < 0.5 || playback_rate > 4);
|
| -
|
| - int frames_remaining = total_frames_requested;
|
| - bool first_fill_buffer = true;
|
| - while (frames_remaining > 0) {
|
| - int frames_requested = std::min(buffer_size_in_frames, frames_remaining);
|
| - int frames_written =
|
| - algorithm_.FillBuffer(bus.get(), 0, frames_requested, playback_rate);
|
| - ASSERT_GT(frames_written, 0) << "Requested: " << frames_requested
|
| - << ", playing at " << playback_rate;
|
| -
|
| - // Do not check data if it is first pull out and only one frame written.
|
| - // The very first frame out of WSOLA is always zero because of
|
| - // overlap-and-add window, which is zero for the first sample. Therefore,
|
| - // if at very first buffer-fill only one frame is written, that is zero
|
| - // which might cause exception in CheckFakeData().
|
| - if (!first_fill_buffer || frames_written > 1)
|
| - ASSERT_EQ(expect_muted, AudioDataIsMuted(bus.get(), frames_written));
|
| - first_fill_buffer = false;
|
| - frames_remaining -= frames_written;
|
| -
|
| - FillAlgorithmQueue();
|
| - }
|
| -
|
| - int frames_consumed =
|
| - ComputeConsumedFrames(initial_frames_enqueued, initial_frames_buffered);
|
| -
|
| - // If playing back at normal speed, we should always get back the same
|
| - // number of bytes requested.
|
| - if (playback_rate == 1.0) {
|
| - EXPECT_EQ(total_frames_requested, frames_consumed);
|
| - return;
|
| - }
|
| -
|
| - // Otherwise, allow |kMaxAcceptableDelta| difference between the target and
|
| - // actual playback rate.
|
| - // When |kSamplesPerSecond| and |total_frames_requested| are reasonably
|
| - // large, one can expect less than a 1% difference in most cases. In our
|
| - // current implementation, sped up playback is less accurate than slowed
|
| - // down playback, and for playback_rate > 1, playback rate generally gets
|
| - // less and less accurate the farther it drifts from 1 (though this is
|
| - // nonlinear).
|
| - double actual_playback_rate =
|
| - 1.0 * frames_consumed / total_frames_requested;
|
| - EXPECT_NEAR(playback_rate, actual_playback_rate, playback_rate / 100.0);
|
| - }
|
| -
|
| - void WsolaTest(float playback_rate) {
|
| - const int kSampleRateHz = 48000;
|
| - const ChannelLayout kChannelLayout = CHANNEL_LAYOUT_STEREO;
|
| - const int kBytesPerSample = 2;
|
| - const int kNumFrames = kSampleRateHz / 100; // 10 milliseconds.
|
| -
|
| - channels_ = ChannelLayoutToChannelCount(kChannelLayout);
|
| - AudioParameters params(AudioParameters::AUDIO_PCM_LINEAR, kChannelLayout,
|
| - kSampleRateHz, kBytesPerSample * 8, kNumFrames);
|
| - algorithm_.Initialize(params);
|
| -
|
| - // A pulse is 6 milliseconds (even number of samples).
|
| - const int kPulseWidthSamples = 6 * kSampleRateHz / 1000;
|
| - const int kHalfPulseWidthSamples = kPulseWidthSamples / 2;
|
| -
|
| - // For the ease of implementation get 1 frame every call to FillBuffer().
|
| - scoped_ptr<AudioBus> output = AudioBus::Create(channels_, 1);
|
| -
|
| - // Input buffer to inject pulses.
|
| - scoped_refptr<AudioBuffer> input =
|
| - AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
|
| - kChannelLayout,
|
| - channels_,
|
| - kSampleRateHz,
|
| - kPulseWidthSamples);
|
| -
|
| - const std::vector<uint8*>& channel_data = input->channel_data();
|
| -
|
| - // Fill |input| channels.
|
| - FillWithSquarePulseTrain(kHalfPulseWidthSamples, 0, kPulseWidthSamples,
|
| - reinterpret_cast<float*>(channel_data[0]));
|
| - FillWithSquarePulseTrain(kHalfPulseWidthSamples, kHalfPulseWidthSamples,
|
| - kPulseWidthSamples,
|
| - reinterpret_cast<float*>(channel_data[1]));
|
| -
|
| - // A buffer for the output until a complete pulse is created. Then
|
| - // reference pulse is compared with this buffer.
|
| - scoped_ptr<AudioBus> pulse_buffer = AudioBus::Create(
|
| - channels_, kPulseWidthSamples);
|
| -
|
| - const float kTolerance = 0.000001f;
|
| - // Equivalent of 4 seconds.
|
| - const int kNumRequestedPulses = kSampleRateHz * 4 / kPulseWidthSamples;
|
| - for (int n = 0; n < kNumRequestedPulses; ++n) {
|
| - int num_buffered_frames = 0;
|
| - while (num_buffered_frames < kPulseWidthSamples) {
|
| - int num_samples =
|
| - algorithm_.FillBuffer(output.get(), 0, 1, playback_rate);
|
| - ASSERT_LE(num_samples, 1);
|
| - if (num_samples > 0) {
|
| - output->CopyPartialFramesTo(0, num_samples, num_buffered_frames,
|
| - pulse_buffer.get());
|
| - num_buffered_frames++;
|
| - } else {
|
| - algorithm_.EnqueueBuffer(input);
|
| - }
|
| - }
|
| -
|
| - // Pulses in the first half of WSOLA AOL frame are not constructed
|
| - // perfectly. Do not check them.
|
| - if (n > 3) {
|
| - for (int m = 0; m < channels_; ++m) {
|
| - const float* pulse_ch = pulse_buffer->channel(m);
|
| -
|
| - // Because of overlap-and-add we might have round off error.
|
| - for (int k = 0; k < kPulseWidthSamples; ++k) {
|
| - ASSERT_NEAR(reinterpret_cast<float*>(channel_data[m])[k],
|
| - pulse_ch[k], kTolerance) << " loop " << n
|
| - << " channel/sample " << m << "/" << k;
|
| - }
|
| - }
|
| - }
|
| -
|
| - // Zero out the buffer to be sure the next comparison is relevant.
|
| - pulse_buffer->Zero();
|
| - }
|
| - }
|
| -
|
| - protected:
|
| - AudioRendererAlgorithm algorithm_;
|
| - int frames_enqueued_;
|
| - int channels_;
|
| - ChannelLayout channel_layout_;
|
| - SampleFormat sample_format_;
|
| - int samples_per_second_;
|
| - int bytes_per_sample_;
|
| -};
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_NormalRate) {
|
| - Initialize();
|
| - TestPlaybackRate(1.0);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_NearlyNormalFasterRate) {
|
| - Initialize();
|
| - TestPlaybackRate(1.0001);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_NearlyNormalSlowerRate) {
|
| - Initialize();
|
| - TestPlaybackRate(0.9999);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_OneAndAQuarterRate) {
|
| - Initialize();
|
| - TestPlaybackRate(1.25);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_OneAndAHalfRate) {
|
| - Initialize();
|
| - TestPlaybackRate(1.5);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_DoubleRate) {
|
| - Initialize();
|
| - TestPlaybackRate(2.0);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_EightTimesRate) {
|
| - Initialize();
|
| - TestPlaybackRate(8.0);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_ThreeQuartersRate) {
|
| - Initialize();
|
| - TestPlaybackRate(0.75);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_HalfRate) {
|
| - Initialize();
|
| - TestPlaybackRate(0.5);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_QuarterRate) {
|
| - Initialize();
|
| - TestPlaybackRate(0.25);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_Pause) {
|
| - Initialize();
|
| - TestPlaybackRate(0.0);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_SlowDown) {
|
| - Initialize();
|
| - TestPlaybackRate(4.5);
|
| - TestPlaybackRate(3.0);
|
| - TestPlaybackRate(2.0);
|
| - TestPlaybackRate(1.0);
|
| - TestPlaybackRate(0.5);
|
| - TestPlaybackRate(0.25);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_SpeedUp) {
|
| - Initialize();
|
| - TestPlaybackRate(0.25);
|
| - TestPlaybackRate(0.5);
|
| - TestPlaybackRate(1.0);
|
| - TestPlaybackRate(2.0);
|
| - TestPlaybackRate(3.0);
|
| - TestPlaybackRate(4.5);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_JumpAroundSpeeds) {
|
| - Initialize();
|
| - TestPlaybackRate(2.1);
|
| - TestPlaybackRate(0.9);
|
| - TestPlaybackRate(0.6);
|
| - TestPlaybackRate(1.4);
|
| - TestPlaybackRate(0.3);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_SmallBufferSize) {
|
| - Initialize();
|
| - static const int kBufferSizeInFrames = 1;
|
| - static const int kFramesRequested = kOutputDurationInSec * kSamplesPerSecond;
|
| - TestPlaybackRate(1.0, kBufferSizeInFrames, kFramesRequested);
|
| - TestPlaybackRate(0.5, kBufferSizeInFrames, kFramesRequested);
|
| - TestPlaybackRate(1.5, kBufferSizeInFrames, kFramesRequested);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_LargeBufferSize) {
|
| - Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatS16, 44100);
|
| - TestPlaybackRate(1.0);
|
| - TestPlaybackRate(0.5);
|
| - TestPlaybackRate(1.5);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_LowerQualityAudio) {
|
| - Initialize(CHANNEL_LAYOUT_MONO, kSampleFormatU8, kSamplesPerSecond);
|
| - TestPlaybackRate(1.0);
|
| - TestPlaybackRate(0.5);
|
| - TestPlaybackRate(1.5);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBuffer_HigherQualityAudio) {
|
| - Initialize(CHANNEL_LAYOUT_STEREO, kSampleFormatS32, kSamplesPerSecond);
|
| - TestPlaybackRate(1.0);
|
| - TestPlaybackRate(0.5);
|
| - TestPlaybackRate(1.5);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, DotProduct) {
|
| - const int kChannels = 3;
|
| - const int kFrames = 20;
|
| - const int kHalfPulseWidth = 2;
|
| -
|
| - scoped_ptr<AudioBus> a = AudioBus::Create(kChannels, kFrames);
|
| - scoped_ptr<AudioBus> b = AudioBus::Create(kChannels, kFrames);
|
| -
|
| - scoped_ptr<float[]> dot_prod(new float[kChannels]);
|
| -
|
| - FillWithSquarePulseTrain(kHalfPulseWidth, 0, 0, a.get());
|
| - FillWithSquarePulseTrain(kHalfPulseWidth, 1, 1, a.get());
|
| - FillWithSquarePulseTrain(kHalfPulseWidth, 2, 2, a.get());
|
| -
|
| - FillWithSquarePulseTrain(kHalfPulseWidth, 0, 0, b.get());
|
| - FillWithSquarePulseTrain(kHalfPulseWidth, 0, 1, b.get());
|
| - FillWithSquarePulseTrain(kHalfPulseWidth, 0, 2, b.get());
|
| -
|
| - internal::MultiChannelDotProduct(a.get(), 0, b.get(), 0, kFrames,
|
| - dot_prod.get());
|
| -
|
| - EXPECT_FLOAT_EQ(kFrames, dot_prod[0]);
|
| - EXPECT_FLOAT_EQ(0, dot_prod[1]);
|
| - EXPECT_FLOAT_EQ(-kFrames, dot_prod[2]);
|
| -
|
| - internal::MultiChannelDotProduct(a.get(), 4, b.get(), 8, kFrames / 2,
|
| - dot_prod.get());
|
| -
|
| - EXPECT_FLOAT_EQ(kFrames / 2, dot_prod[0]);
|
| - EXPECT_FLOAT_EQ(0, dot_prod[1]);
|
| - EXPECT_FLOAT_EQ(-kFrames / 2, dot_prod[2]);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, MovingBlockEnergy) {
|
| - const int kChannels = 2;
|
| - const int kFrames = 20;
|
| - const int kFramesPerBlock = 3;
|
| - const int kNumBlocks = kFrames - (kFramesPerBlock - 1);
|
| - scoped_ptr<AudioBus> a = AudioBus::Create(kChannels, kFrames);
|
| - scoped_ptr<float[]> energies(new float[kChannels * kNumBlocks]);
|
| - float* ch_left = a->channel(0);
|
| - float* ch_right = a->channel(1);
|
| -
|
| - // Fill up both channels.
|
| - for (int n = 0; n < kFrames; ++n) {
|
| - ch_left[n] = n;
|
| - ch_right[n] = kFrames - 1 - n;
|
| - }
|
| -
|
| - internal::MultiChannelMovingBlockEnergies(a.get(), kFramesPerBlock,
|
| - energies.get());
|
| -
|
| - // Check if the energy of candidate blocks of each channel computed correctly.
|
| - for (int n = 0; n < kNumBlocks; ++n) {
|
| - float expected_energy = 0;
|
| - for (int k = 0; k < kFramesPerBlock; ++k)
|
| - expected_energy += ch_left[n + k] * ch_left[n + k];
|
| -
|
| - // Left (first) channel.
|
| - EXPECT_FLOAT_EQ(expected_energy, energies[2 * n]);
|
| -
|
| - expected_energy = 0;
|
| - for (int k = 0; k < kFramesPerBlock; ++k)
|
| - expected_energy += ch_right[n + k] * ch_right[n + k];
|
| -
|
| - // Second (right) channel.
|
| - EXPECT_FLOAT_EQ(expected_energy, energies[2 * n + 1]);
|
| - }
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FullAndDecimatedSearch) {
|
| - const int kFramesInSearchRegion = 12;
|
| - const int kChannels = 2;
|
| - float ch_0[] = {
|
| - 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 1.0f, 1.0f, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f };
|
| - float ch_1[] = {
|
| - 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.1f, 1.0f, 0.1f, 0.0f, 0.0f };
|
| - ASSERT_EQ(sizeof(ch_0), sizeof(ch_1));
|
| - ASSERT_EQ(static_cast<size_t>(kFramesInSearchRegion),
|
| - sizeof(ch_0) / sizeof(*ch_0));
|
| - scoped_ptr<AudioBus> search_region = AudioBus::Create(kChannels,
|
| - kFramesInSearchRegion);
|
| - float* ch = search_region->channel(0);
|
| - memcpy(ch, ch_0, sizeof(float) * kFramesInSearchRegion);
|
| - ch = search_region->channel(1);
|
| - memcpy(ch, ch_1, sizeof(float) * kFramesInSearchRegion);
|
| -
|
| - const int kFramePerBlock = 4;
|
| - float target_0[] = { 1.0f, 1.0f, 1.0f, 0.0f };
|
| - float target_1[] = { 0.0f, 1.0f, 0.1f, 1.0f };
|
| - ASSERT_EQ(sizeof(target_0), sizeof(target_1));
|
| - ASSERT_EQ(static_cast<size_t>(kFramePerBlock),
|
| - sizeof(target_0) / sizeof(*target_0));
|
| -
|
| - scoped_ptr<AudioBus> target = AudioBus::Create(kChannels,
|
| - kFramePerBlock);
|
| - ch = target->channel(0);
|
| - memcpy(ch, target_0, sizeof(float) * kFramePerBlock);
|
| - ch = target->channel(1);
|
| - memcpy(ch, target_1, sizeof(float) * kFramePerBlock);
|
| -
|
| - scoped_ptr<float[]> energy_target(new float[kChannels]);
|
| -
|
| - internal::MultiChannelDotProduct(target.get(), 0, target.get(), 0,
|
| - kFramePerBlock, energy_target.get());
|
| -
|
| - ASSERT_EQ(3.f, energy_target[0]);
|
| - ASSERT_EQ(2.01f, energy_target[1]);
|
| -
|
| - const int kNumCandidBlocks = kFramesInSearchRegion - (kFramePerBlock - 1);
|
| - scoped_ptr<float[]> energy_candid_blocks(new float[kNumCandidBlocks *
|
| - kChannels]);
|
| -
|
| - internal::MultiChannelMovingBlockEnergies(
|
| - search_region.get(), kFramePerBlock, energy_candid_blocks.get());
|
| -
|
| - // Check the energy of the candidate blocks of the first channel.
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[0]);
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[2]);
|
| - ASSERT_FLOAT_EQ(1, energy_candid_blocks[4]);
|
| - ASSERT_FLOAT_EQ(2, energy_candid_blocks[6]);
|
| - ASSERT_FLOAT_EQ(3, energy_candid_blocks[8]);
|
| - ASSERT_FLOAT_EQ(3, energy_candid_blocks[10]);
|
| - ASSERT_FLOAT_EQ(2, energy_candid_blocks[12]);
|
| - ASSERT_FLOAT_EQ(1, energy_candid_blocks[14]);
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[16]);
|
| -
|
| - // Check the energy of the candidate blocks of the second channel.
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[1]);
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[3]);
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[5]);
|
| - ASSERT_FLOAT_EQ(0, energy_candid_blocks[7]);
|
| - ASSERT_FLOAT_EQ(0.01f, energy_candid_blocks[9]);
|
| - ASSERT_FLOAT_EQ(1.01f, energy_candid_blocks[11]);
|
| - ASSERT_FLOAT_EQ(1.02f, energy_candid_blocks[13]);
|
| - ASSERT_FLOAT_EQ(1.02f, energy_candid_blocks[15]);
|
| - ASSERT_FLOAT_EQ(1.01f, energy_candid_blocks[17]);
|
| -
|
| - // An interval which is of no effect.
|
| - internal::Interval exclude_interval = std::make_pair(-100, -10);
|
| - EXPECT_EQ(5, internal::FullSearch(
|
| - 0, kNumCandidBlocks - 1, exclude_interval, target.get(),
|
| - search_region.get(), energy_target.get(), energy_candid_blocks.get()));
|
| -
|
| - // Exclude the the best match.
|
| - exclude_interval = std::make_pair(2, 5);
|
| - EXPECT_EQ(7, internal::FullSearch(
|
| - 0, kNumCandidBlocks - 1, exclude_interval, target.get(),
|
| - search_region.get(), energy_target.get(), energy_candid_blocks.get()));
|
| -
|
| - // An interval which is of no effect.
|
| - exclude_interval = std::make_pair(-100, -10);
|
| - EXPECT_EQ(4, internal::DecimatedSearch(
|
| - 4, exclude_interval, target.get(), search_region.get(),
|
| - energy_target.get(), energy_candid_blocks.get()));
|
| -
|
| - EXPECT_EQ(5, internal::OptimalIndex(search_region.get(), target.get(),
|
| - exclude_interval));
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, QuadraticInterpolation) {
|
| - // Arbitrary coefficients.
|
| - const float kA = 0.7f;
|
| - const float kB = 1.2f;
|
| - const float kC = 0.8f;
|
| -
|
| - float y_values[3];
|
| - y_values[0] = kA - kB + kC;
|
| - y_values[1] = kC;
|
| - y_values[2] = kA + kB + kC;
|
| -
|
| - float extremum;
|
| - float extremum_value;
|
| -
|
| - internal::QuadraticInterpolation(y_values, &extremum, &extremum_value);
|
| -
|
| - float x_star = -kB / (2.f * kA);
|
| - float y_star = kA * x_star * x_star + kB * x_star + kC;
|
| -
|
| - EXPECT_FLOAT_EQ(x_star, extremum);
|
| - EXPECT_FLOAT_EQ(y_star, extremum_value);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, QuadraticInterpolation_Colinear) {
|
| - float y_values[3];
|
| - y_values[0] = 1.0;
|
| - y_values[1] = 1.0;
|
| - y_values[2] = 1.0;
|
| -
|
| - float extremum;
|
| - float extremum_value;
|
| -
|
| - internal::QuadraticInterpolation(y_values, &extremum, &extremum_value);
|
| -
|
| - EXPECT_FLOAT_EQ(extremum, 0.0);
|
| - EXPECT_FLOAT_EQ(extremum_value, 1.0);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, WsolaSlowdown) {
|
| - WsolaTest(0.6f);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, WsolaSpeedup) {
|
| - WsolaTest(1.6f);
|
| -}
|
| -
|
| -TEST_F(AudioRendererAlgorithmTest, FillBufferOffset) {
|
| - Initialize();
|
| -
|
| - scoped_ptr<AudioBus> bus = AudioBus::Create(channels_, kFrameSize);
|
| -
|
| - // Verify that the first half of |bus| remains zero and the last half is
|
| - // filled appropriately at normal, above normal, below normal, and muted
|
| - // rates.
|
| - const int kHalfSize = kFrameSize / 2;
|
| - const float kAudibleRates[] = {1.0f, 2.0f, 0.5f};
|
| - for (size_t i = 0; i < arraysize(kAudibleRates); ++i) {
|
| - SCOPED_TRACE(kAudibleRates[i]);
|
| - bus->Zero();
|
| -
|
| - const int frames_filled = algorithm_.FillBuffer(
|
| - bus.get(), kHalfSize, kHalfSize, kAudibleRates[i]);
|
| - ASSERT_EQ(kHalfSize, frames_filled);
|
| - ASSERT_TRUE(VerifyAudioData(bus.get(), 0, kHalfSize, 0));
|
| - ASSERT_FALSE(VerifyAudioData(bus.get(), kHalfSize, kHalfSize, 0));
|
| - }
|
| -
|
| - const float kMutedRates[] = {5.0f, 0.25f};
|
| - for (size_t i = 0; i < arraysize(kMutedRates); ++i) {
|
| - SCOPED_TRACE(kMutedRates[i]);
|
| - for (int ch = 0; ch < bus->channels(); ++ch)
|
| - std::fill(bus->channel(ch), bus->channel(ch) + bus->frames(), 1.0f);
|
| -
|
| - const int frames_filled =
|
| - algorithm_.FillBuffer(bus.get(), kHalfSize, kHalfSize, kMutedRates[i]);
|
| - ASSERT_EQ(kHalfSize, frames_filled);
|
| - ASSERT_FALSE(VerifyAudioData(bus.get(), 0, kHalfSize, 0));
|
| - ASSERT_TRUE(VerifyAudioData(bus.get(), kHalfSize, kHalfSize, 0));
|
| - }
|
| -}
|
| -
|
| -} // namespace media
|
|
|