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Diffstat (limited to 'src/3rdparty/resonance-audio/resonance_audio/dsp')
76 files changed, 17149 insertions, 0 deletions
diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter.cc new file mode 100644 index 000000000..ca489cce3 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter.cc @@ -0,0 +1,149 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/biquad_filter.h" + +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +// This value has been chosen empirically after a series of experiments +// performed by kellyi@ and alperg@. It was found that crossfading over 256 +// samples yielded no audible glitching artefacts and an acceptable amount of +// delay. + +const size_t kIdealSamplesToIterate = 256U; + +} // namespace + +BiquadFilter::BiquadFilter(const BiquadCoefficients& coefficients, + size_t frames_per_buffer) + : biquad_delay_line_({{0.0f, 0.0f}}), + interpolate_flag_(false), + interpolate_counter_(0), + old_delay_line_({{0.0f, 0.0f}}), + samples_to_iterate_over_(frames_per_buffer > kIdealSamplesToIterate + ? kIdealSamplesToIterate + : frames_per_buffer), + fade_scale_(1.0f / static_cast<float>(samples_to_iterate_over_)), + old_coefficients_() { + DCHECK_GT(frames_per_buffer, 0U); + CHECK_GT(coefficients_.a[0], kEpsilonFloat); + SetCoefficients(coefficients); +} + +void BiquadFilter::SetCoefficients(const BiquadCoefficients& coefficients) { + coefficients_ = coefficients; + // Normalize the coefficients for use in the |FilterSample()| function. + coefficients_.a[1] /= coefficients_.a[0]; + coefficients_.a[2] /= coefficients_.a[0]; + coefficients_.b[0] /= coefficients_.a[0]; + coefficients_.b[1] /= coefficients_.a[0]; + coefficients_.b[2] /= coefficients_.a[0]; +} + +void BiquadFilter::Filter(const AudioBuffer::Channel& input_channel, + AudioBuffer::Channel* output_channel) { + DCHECK(output_channel); + DCHECK_EQ(input_channel.size(), output_channel->size()); + + if (interpolate_flag_) { + for (size_t frame = 0; frame < input_channel.size(); ++frame) { + // Biquad coefficients are updated here. + UpdateInterpolate(); + (*output_channel)[frame] = InterpolateFilterSample(input_channel[frame]); + } + } else { + for (size_t frame = 0; frame < input_channel.size(); ++frame) { + (*output_channel)[frame] = FilterSample( + input_channel[frame], &biquad_delay_line_, coefficients_); + } + } +} + +void BiquadFilter::InterpolateToCoefficients( + const BiquadCoefficients& coefficients) { + interpolate_flag_ = true; + // Reset the counter so we perform the update over samples_to_iterate_over_ + // samples. + interpolate_counter_ = 0; + // Store the old coefficients, update the new ones and transfer data between + // delay lines. + old_coefficients_ = coefficients_; + coefficients_ = coefficients; + old_delay_line_ = biquad_delay_line_; +} + +void BiquadFilter::Clear() { + biquad_delay_line_ = {{0.0f, 0.0f}}; + interpolate_flag_ = false; + interpolate_counter_ = 0; + old_delay_line_ = {{0.0f, 0.0f}}; +} + +float BiquadFilter::FilterSample(float input, std::array<float, 2>* delay_line, + const BiquadCoefficients& coefficients) { + // Using A Direct Form II Implementation Difference equation: + // Source: Digital Signal Processing Principles Algorithms and Applications + // Fourth Edition. John G. Prolakis and Dimitris G. Manolakis - Chapter 9 + // w[n] = x[n] - (a1/a0)*w[n-1] - (a2/a0)*w[n-2] + // y(n) = (b0/a0)*w[n] + (b1/a0)*w[n-1] + (b2/a0)*w[n-2] + // where x[n] is input, w[n] is storage and y[n] is output. + // The division by a0 has been performed in Biquad::SetCoefficients. + + // define a temporary storage value w. + const float w = input - (*delay_line)[0] * coefficients.a[1] - + (*delay_line)[1] * coefficients.a[2]; + + // Do second half of the calculation to generate output. + const float output = w * coefficients.b[0] + + (*delay_line)[0] * coefficients.b[1] + + (*delay_line)[1] * coefficients.b[2]; + + // Update delay line. + (*delay_line)[1] = (*delay_line)[0]; + (*delay_line)[0] = w; + + return output; +} + +void BiquadFilter::UpdateInterpolate() { + if (++interpolate_counter_ > samples_to_iterate_over_) { + interpolate_flag_ = false; + } +} + +float BiquadFilter::InterpolateFilterSample(float input) { + const float new_filter_output = + FilterSample(input, &biquad_delay_line_, coefficients_); + + if (!interpolate_flag_) { + return new_filter_output; + } else { + // Process the "old" filter values. + const float old_filter_output = + FilterSample(input, &old_delay_line_, old_coefficients_); + // A linear crossfade between old_filter_output and new_filter_output, + // stepsize is fade_scale_. + const float weight = fade_scale_ * static_cast<float>(interpolate_counter_); + const float sample_diff = new_filter_output - old_filter_output; + return weight * sample_diff + old_filter_output; + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter.h new file mode 100644 index 000000000..2e05c3276 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter.h @@ -0,0 +1,137 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_BIQUAD_FILTER_H_ +#define RESONANCE_AUDIO_DSP_BIQUAD_FILTER_H_ + +#include <array> + +#include "base/audio_buffer.h" +#include "base/logging.h" + +namespace vraudio { + +// Set of transfer function coefficients. +struct BiquadCoefficients { + // Constructor takes as its arguments 6 floats representing the transfer + // function coefficients of a biquad filter. + BiquadCoefficients(float a0, float a1, float a2, float b0, float b1, float b2) + : a({{a0, a1, a2}}), b({{b0, b1, b2}}) {} + + // Default constructor that sets a0 and b0 == 1 so that if the coefficients + // are used, the filter is stable and has no effect on input. i.e. input + // appears to just pass through. + BiquadCoefficients() : BiquadCoefficients(1.0, 0.0, 0.0, 1.0, 0.0, 0.0) {} + + // The denominator quadratic coefficients. + std::array<float, 3> a; + // The numerator quadratic coefficients. + std::array<float, 3> b; +}; + +// Class representing a biquad filter as its transfer function coefficients. +// This class also performs filtering. +class BiquadFilter { + public: + // Constructs a BiquadFilter using a BiquadCoefficients struct. + // + // @param coefficients A BiquadCoefficients to set the internal + // |coefficients_|. + // @param frames_per_buffer The number of frames in each data buffer to be + // processed. This is used to set the number of samples to iterate over + // during a change of filter state. + BiquadFilter(const BiquadCoefficients& coefficients, + size_t frames_per_buffer); + + // Filter method for use with AudioBuffer::Channel. + // + // @param input_channel An AudioBuffer::Channel of input. + // @param output_channel A pointer to an AudioBuffer::Channel for output. + void Filter(const AudioBuffer::Channel& input_channel, + AudioBuffer::Channel* output_channel); + + // Sets the fiter's coefficents to those passed, with all + // coefficients bar a0 scaled by 1/a0. + // + // @param coefficients A set of BiquadCoefficients. + void SetCoefficients(const BiquadCoefficients& coefficients); + + // Sets the target coefficients to be interpolated. + // + // @param coefficents The BiquadCoefficients we wish to interpolate to over + // samples_to_iterate_over_ samples of the next input buffer. + void InterpolateToCoefficients(const BiquadCoefficients& coefficients); + + // Clears the internal state of the filter, except for coefficients. + void Clear(); + + private: + friend class BiquadFilterInterpolateTest; + + // Filters a single sample of input. + // + // @param input A single input sample from a Planar or Interleaved buffer. + // @param delay_line The delay_line to use (important for interpolation). + // @param coefficients The biquad coeffients for use in filtering. + // @return An output value to be placed in a Planar or Interleaved buffer. + float FilterSample(float input, std::array<float, 2>* delay_line, + const BiquadCoefficients& coefficients); + + // If InterpolateToState() has been called to assign new filter coefficients, + // this function will be called samples_to_iterate_over_ times within the next + // call of the Filter() function to slowly transition to the new coefficients + // which were passed to InterpolateToState() previously. + void UpdateInterpolate(); + + // Filters a single sample of input while transitioning between coefficients. + // Performs a linear crossfade over samples_to_iterate_over_ samples. + // + // @param input A single input sample from a Planar or Interleaved buffer. + // @return Output value to be placed in an audio buffer. + float InterpolateFilterSample(float input); + + // Stores the memory state of the biquad + std::array<float, 2> biquad_delay_line_; + + // Flag that denotes whether or not we are transitioning to another set of + // coefficients via interpolation. + bool interpolate_flag_; + + // Counter that denotes how far we are into transitioning to another set of + // coefficients via interpolation. + size_t interpolate_counter_; + + // Stores the memory state of the biquad. + std::array<float, 2> old_delay_line_; + + // Number of samples over which to apply the filter coefficients. + size_t samples_to_iterate_over_; + + // Value used to crossfade between filter outputs. + float fade_scale_; + + // A set of coefficient which are updated as we interpolate between filter + // coefficients. + BiquadCoefficients old_coefficients_; + + // Represents and maintains the state of the biquad filter in terms of its + // transfer function coefficients. + BiquadCoefficients coefficients_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_BIQUAD_FILTER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter_test.cc new file mode 100644 index 000000000..c8ac6de78 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/biquad_filter_test.cc @@ -0,0 +1,303 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/biquad_filter.h" + +#include <algorithm> +#include <cmath> +#include <numeric> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +const size_t kTestInputDataSize = 16; + +const float kTestFilterCoefficients[] = {1.0f, 0.0f, 0.0, 1.0f, 0.1f, 0.1f}; + +const size_t kIdealSamplesToIterate = 256; + +// For use with std::transform in the StabilityTest. +float operator_abs(float f) { return std::abs(f); } + +} // namespace + +// Tests that the filtering of a mono signal produces the correct output. +TEST(BiquadFilterTest, FilterTest) { + const size_t kNumIterations = 1; + + const BiquadCoefficients kCoefficients( + kTestFilterCoefficients[0], kTestFilterCoefficients[1], + kTestFilterCoefficients[2], kTestFilterCoefficients[3], + kTestFilterCoefficients[4], kTestFilterCoefficients[5]); + + AudioBuffer input(kNumMonoChannels, kTestInputDataSize); + AudioBuffer output(kNumMonoChannels, kTestInputDataSize); + // Set the input AudioBuffer to a Kronecker delta. + input.Clear(); + input[0][0] = 1.0f; + + // This vector will accumulate the output from the filter over time (Only + // works with mono channel). + std::vector<float> output_accumulator; + output_accumulator.reserve(kNumIterations * kTestInputDataSize); + + // Create a biquad filter initialized with transfer function coefficients. + BiquadFilter biquad(kCoefficients, kTestInputDataSize); + + // Perform filtering. + for (size_t i = 0; i < kNumIterations; ++i) { + biquad.Filter(input[0], &output[0]); + output_accumulator.insert(output_accumulator.end(), output[0].begin(), + output[0].end()); + } + + // Since the denominator of the biquad coefficients is [1 0 0] we can expect + // the impulse response to be equal to the numerator. + for (size_t i = 0; i < output_accumulator.size(); ++i) { + if (i < 3U) { + EXPECT_NEAR(kTestFilterCoefficients[3 + i], output_accumulator[i], + kEpsilonFloat); + } else { + EXPECT_EQ(0.0f, output_accumulator[i]); + } + } +} + +// Tests that the filtering of a mono signal produces the correct output. +TEST(BiquadFilterTest, InplaceFilterTest) { + const size_t kNumIterations = 1; + + const BiquadCoefficients kCoefficients( + kTestFilterCoefficients[0], kTestFilterCoefficients[1], + kTestFilterCoefficients[2], kTestFilterCoefficients[3], + kTestFilterCoefficients[4], kTestFilterCoefficients[5]); + + AudioBuffer input(kNumMonoChannels, kTestInputDataSize); + // Set the input AudioBuffer to a Kronecker delta. + input.Clear(); + input[0][0] = 1.0f; + + // This vector will accumulate the output from the filter over time (Only + // works with mono channel). + std::vector<float> output_accumulator; + output_accumulator.reserve(kTestInputDataSize); + + // Create a biquad filter initialized with transfer function coefficients. + BiquadFilter biquad(kCoefficients, kTestInputDataSize); + + // Perform inplace filtering of |input| vector. + for (size_t i = 0; i < kNumIterations; ++i) { + biquad.Filter(input[0], &input[0]); + output_accumulator.insert(output_accumulator.end(), input[0].begin(), + input[0].end()); + } + + // Since the denominator of the biquad coefficients is [1 0 0] we can expect + // the impulse response to be equal to the numerator. + for (size_t i = 0; i < output_accumulator.size(); ++i) { + if (i < 3) { + EXPECT_NEAR(kTestFilterCoefficients[3 + i], output_accumulator[i], + kEpsilonFloat); + } else { + EXPECT_EQ(0.0f, output_accumulator[i]); + } + } +} + +// Tests whether the interpolation stops after kIdealSamplesToIterate samples, +// given a long enough buffer. +TEST(BiquadFilterTest, InterpolationStopsTest) { + const size_t kFramesPerBuffer = 512; + + std::vector<std::vector<float>> planar_input_data( + kNumMonoChannels, std::vector<float>(kFramesPerBuffer)); + planar_input_data[0][0] = 1.0f; + planar_input_data[0][256] = 1.0f; + + AudioBuffer input_planar(kNumMonoChannels, kFramesPerBuffer); + AudioBuffer output_planar(kNumMonoChannels, kFramesPerBuffer); + input_planar = planar_input_data; + + // Instantiate Biquad for planar data. The impulse response of the default is: + // 1, 0, 0, ...... + BiquadFilter biquad(BiquadCoefficients(), kFramesPerBuffer); + + // Coefficients we wish to interpolate to. The impulse response of this filter + // is: 1, 0, 1, ...... + const BiquadCoefficients kNextCoefficients = {1.0f, 0.0f, 0.0f, + 1.0f, 0.0f, 1.0f}; + + biquad.InterpolateToCoefficients(kNextCoefficients); + biquad.Filter(input_planar[0], &output_planar[0]); + + // Based on the known impulse responses we can see that if elements 256, 257 + // and 258 have the values 1, 0, 1, then only the new filter coefficients are + // contributing and we have stopped crossfading. Note: The value of 256 here + // comes from kIdealSamplesToIterate defined in biquad_filter.cc + EXPECT_EQ(1.0f, output_planar[0][kIdealSamplesToIterate]); + EXPECT_EQ(0.0f, output_planar[0][kIdealSamplesToIterate + 1]); + EXPECT_EQ(1.0f, output_planar[0][kIdealSamplesToIterate + 2]); +} + +// Tests whether the |BiquadFilter| remains stable when its z-domain poles lie +// very close to the unit circle. +TEST(BiquadFilterTest, StabilityTest) { + static const size_t kBufferSize = 1024; + const size_t kIterations = 200; + // The following filter was designed in MATLAB with a very slow decay rate. + // (There are both poles and zeros with magnitude 0.999). Even 200'000 samples + // after the initial impulse the response will have only died away, in an + // oscillating manner, to ~1/700 of its peak value. + static const BiquadCoefficients kHighQCoefficients( + 1.0f, -0.907804951302441f, 0.999869108872718f, 15279.8745150008f, 0.0f, + -15279.8745150008f); + BiquadFilter biquad(kHighQCoefficients, kBufferSize); + + AudioBuffer input(kNumMonoChannels, kBufferSize); + AudioBuffer output(kNumMonoChannels, kBufferSize); + // Set the input AudioBuffer to a Kronecker delta. + input.Clear(); + input[0][0] = 1.0f; + + // This vector will accumulate the output from the filter over time (Only + // works with mono channel). + std::vector<float> output_accumulator; + output_accumulator.reserve(kBufferSize * kIterations); + + // Filter once with the Kronecker delta. + biquad.Filter(input[0], &output[0]); + output_accumulator.insert(output_accumulator.end(), output[0].begin(), + output[0].end()); + + // Perform filtering with all zero input. + input[0][0] = 0.0f; + for (size_t i = 1; i < kIterations; ++i) { + biquad.Filter(input[0], &output[0]); + output_accumulator.insert(output_accumulator.end(), output[0].begin(), + output[0].end()); + } + + // Test that the signal is decaying over time (i.e. filter is stable). + for (size_t i = 0; i < output_accumulator.size() - kBufferSize; + i += kBufferSize) { + std::vector<float> absolute_first_half(kBufferSize / 2); + std::transform(output_accumulator.begin() + i, + output_accumulator.begin() + i + kBufferSize / 2, + absolute_first_half.begin(), operator_abs); + std::vector<float> absolute_second_half(kBufferSize / 2); + std::transform(output_accumulator.begin() + 1 + i + kBufferSize / 2, + output_accumulator.begin() + i + kBufferSize, + absolute_second_half.begin(), operator_abs); + const float sum_first_half = std::accumulate( + absolute_first_half.begin(), absolute_first_half.end(), 0.0f); + const float sum_second_half = std::accumulate( + absolute_second_half.begin(), absolute_second_half.end(), 0.0f); + EXPECT_LT(sum_second_half, sum_first_half); + } +} + +class BiquadFilterInterpolateTest : public ::testing::Test { + protected: + BiquadFilterInterpolateTest() {} + // Virtual methods from ::testing::Test + ~BiquadFilterInterpolateTest() override {} + void SetUp() override {} + void TearDown() override {} + + // Returns true if |filter|'s internal coefficients are equal to those of + // |expected_coefficients| after scaling by a0. + bool TestInternalCoefficients( + BiquadFilter* filter, const BiquadCoefficients& expected_coefficients) { + bool return_value = true; + return_value &= + expected_coefficients.a[0] - filter->coefficients_.a[0] < kEpsilonFloat; + + // From this point we must take account of the scaling by 1/a0. + const float a0 = filter->coefficients_.a[0]; + return_value &= + expected_coefficients.b[0] - filter->coefficients_.b[0] * a0 < + kEpsilonFloat; + for (int i = 1; i < 3; ++i) { + return_value &= + expected_coefficients.b[i] - filter->coefficients_.b[i] * a0 < + kEpsilonFloat; + return_value &= + expected_coefficients.a[i] - filter->coefficients_.a[i] * a0 < + kEpsilonFloat; + } + return return_value; + } +}; + +// Tests whether updating the filter's coefficients with the +// InterpolateToCoefficients reaches the correct coefficients after filtering a +// block of data and whether the samples_to_iterate_over_ value is set +// correctly. +TEST_F(BiquadFilterInterpolateTest, InterpolatedUpdateTest) { + const size_t kFramesPerBuffer = 8; + + const std::vector<std::vector<float>> kPlanarInputData = { + {1.0f, 2.0f, 3.0f, 4.0f, 5.0f, 6.0f, 7.0f, 8.0f}}; + + const std::vector<float> kExpectedOutputAfterChange = { + 8.0f, 10.0f, 4.0f, 6.0f, 8.0f, 10.0f, 12.0f, 14.0f}; + + // Default constructor sets a = {1.0, 0.0, 0.0} and b = {1.0, 0.0, 0.0}. + const BiquadCoefficients kCoefficients; + + // Create input and output AudioBuffers. + AudioBuffer input_planar(kNumMonoChannels, kFramesPerBuffer); + AudioBuffer output_planar(kNumMonoChannels, kFramesPerBuffer); + input_planar = kPlanarInputData; + + // Instantiate Biquad for planar data. + BiquadFilter biquad(kCoefficients, kFramesPerBuffer); + + // Perform filtering on kNumMonoChannels interleaved. + biquad.Filter(input_planar[0], &output_planar[0]); + + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR(output_planar[0][i], kPlanarInputData[0][i], kEpsilonFloat); + } + + // Coefficients we wish to interpolate to. + static const BiquadCoefficients kNextCoefficients = {1.0f, 0.0f, 0.0f, + 1.0f, 0.0f, 1.0f}; + + biquad.InterpolateToCoefficients(kNextCoefficients); + + // Filter once to transition and once to flush out the state. + biquad.Filter(input_planar[0], &output_planar[0]); + biquad.Filter(input_planar[0], &output_planar[0]); + + // Now the output should be just from the new state. + biquad.Filter(input_planar[0], &output_planar[0]); + + // Now check that we transitioned properly, i.e. the output is as expected. + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR(output_planar[0][i], kExpectedOutputAfterChange[i], + kEpsilonFloat); + } + + // Now check that we have the new coefficients. + EXPECT_TRUE(TestInternalCoefficients(&biquad, kNextCoefficients)); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter.cc new file mode 100644 index 000000000..375001bac --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter.cc @@ -0,0 +1,45 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/channel_converter.h" + +#include <cmath> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/simd_utils.h" + +namespace vraudio { + +void ConvertStereoFromMono(const AudioBuffer& input, AudioBuffer* output) { + DCHECK(output); + DCHECK_EQ(input.num_channels(), kNumMonoChannels); + DCHECK_EQ(output->num_channels(), kNumStereoChannels); + DCHECK_EQ(input.num_frames(), output->num_frames()); + StereoFromMonoSimd(input.num_frames(), &input[0][0], &(*output)[0][0], + &(*output)[1][0]); +} + +void ConvertMonoFromStereo(const AudioBuffer& input, AudioBuffer* output) { + DCHECK(output); + DCHECK_EQ(input.num_channels(), kNumStereoChannels); + DCHECK_EQ(output->num_channels(), kNumMonoChannels); + DCHECK_EQ(input.num_frames(), output->num_frames()); + MonoFromStereoSimd(input.num_frames(), &input[0][0], &input[1][0], + &(*output)[0][0]); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter.h new file mode 100644 index 000000000..26d825ae4 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter.h @@ -0,0 +1,40 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_CHANNEL_CONVERTER_H_ +#define RESONANCE_AUDIO_DSP_CHANNEL_CONVERTER_H_ + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Converts a mono |input| buffer to a stereo |output| buffer by preserving the +// signal energy. +// +// @param input Mono input buffer. +// @param output Pointer to a stereo output buffer. +void ConvertStereoFromMono(const AudioBuffer& input, AudioBuffer* output); + +// Converts a stereo |input| buffer to a mono |output| buffer by preserving the +// signal energy. +// +// @param input Stereo input buffer. +// @param output Pointer to a mono output buffer. +void ConvertMonoFromStereo(const AudioBuffer& input, AudioBuffer* output); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_CHANNEL_CONVERTER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter_test.cc new file mode 100644 index 000000000..3d6928ad5 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/channel_converter_test.cc @@ -0,0 +1,80 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/channel_converter.h" + +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +// Number of frames for test buffers. +const size_t kNumFrames = 5; + +// Tests that the correct stereo output buffer is obtained from the converter +// given an arbitrary mono input buffer. +TEST(ChannelConverterTest, ConvertStereoFromMonoTest) { + const std::vector<std::vector<float>> kMonoInput = { + {1.0f, 2.0f, 3.0f, 4.0f, 5.0f}}; + const std::vector<std::vector<float>> kExpectedStereoOutput = { + {0.707107f, 1.414214f, 2.121320f, 2.828427f, 3.535534f}, + {0.707107f, 1.414214f, 2.121320f, 2.828427f, 3.535534f}}; + + // Initialize the test buffers. + AudioBuffer mono_input(kNumMonoChannels, kNumFrames); + mono_input = kMonoInput; + AudioBuffer stereo_output(kNumStereoChannels, kNumFrames); + // Process the input buffer. + ConvertStereoFromMono(mono_input, &stereo_output); + // Compare the output buffer against the expected output. + for (size_t channel = 0; channel < kNumStereoChannels; ++channel) { + for (size_t frame = 0; frame < kNumFrames; ++frame) { + EXPECT_NEAR(stereo_output[channel][frame], + kExpectedStereoOutput[channel][frame], kEpsilonFloat); + } + } +} + +// Tests that the correct mono output buffer is obtained from the converter +// given an arbitrary stereo input buffer. +TEST(ChannelConverterTest, ConvertMonoFromStereoTest) { + const std::vector<std::vector<float>> kStereoInput = { + {0.707107f, 1.414214f, 2.121320f, 2.828427f, 3.535534f}, + {0.707107f, 1.414214f, 2.121320f, 2.828427f, 3.535534f}}; + + const std::vector<std::vector<float>> kExpectedMonoOutput = { + {1.0f, 2.0f, 3.0f, 4.0f, 5.0f}}; + + // Initialize the test buffers. + AudioBuffer stereo_input(kNumStereoChannels, kNumFrames); + stereo_input = kStereoInput; + AudioBuffer mono_output(kNumMonoChannels, kNumFrames); + // Process the input buffer. + ConvertMonoFromStereo(stereo_input, &mono_output); + // Compare the output buffer against the expected output. + for (size_t frame = 0; frame < kNumFrames; ++frame) { + EXPECT_NEAR(mono_output[0][frame], kExpectedMonoOutput[0][frame], + kEpsilonFloat); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer.cc new file mode 100644 index 000000000..5ef2be2c9 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer.cc @@ -0,0 +1,120 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/circular_buffer.h" + +#include <algorithm> + +#include "base/constants_and_types.h" + +namespace vraudio { + +CircularBuffer::CircularBuffer(size_t buffer_length, size_t num_input_frames, + size_t num_output_frames) + : num_input_frames_(num_input_frames), + num_output_frames_(num_output_frames), + buffer_(kNumMonoChannels, buffer_length), + write_cursor_(0), + read_cursor_(0), + num_valid_frames_(0) { + CHECK_GE(buffer_length, num_input_frames + num_output_frames); +} + +bool CircularBuffer::InsertBuffer(const AudioBuffer::Channel& input) { + DCHECK_EQ(input.size(), num_input_frames_); + + if (num_valid_frames_ + num_input_frames_ > buffer_.num_frames()) { + return false; + } + // Remaining space after the write cursor. + const size_t forward_write_space = read_cursor_ <= write_cursor_ + ? buffer_.num_frames() - write_cursor_ + : read_cursor_ - write_cursor_; + + // Copy the input into the buffer. + AudioBuffer::Channel* buffer_channel = &buffer_[0]; + if (forward_write_space >= num_input_frames_) { + DCHECK_LE(buffer_channel->begin() + write_cursor_ + num_input_frames_, + buffer_channel->end()); + std::copy(input.begin(), input.end(), + buffer_channel->begin() + write_cursor_); + } else { + DCHECK_LE(buffer_channel->begin() + write_cursor_ + forward_write_space, + buffer_channel->end()); + DCHECK_LT(input.begin() + forward_write_space, input.end()); + std::copy(input.begin(), input.begin() + forward_write_space, + buffer_channel->begin() + write_cursor_); + DCHECK_LE(buffer_channel->begin() + forward_write_space, + buffer_channel->end()); + std::copy(input.begin() + forward_write_space, input.end(), + buffer_channel->begin()); + } + + write_cursor_ = (write_cursor_ + num_input_frames_) % buffer_.num_frames(); + num_valid_frames_ += num_input_frames_; + return true; +} + +bool CircularBuffer::RetrieveBuffer(AudioBuffer::Channel* output) { + return RetrieveBufferWithOffset(/*offset=*/0, output); +} + +bool CircularBuffer::RetrieveBufferWithOffset(size_t offset, + AudioBuffer::Channel* output) { + DCHECK_LE(output->begin() + num_output_frames_ + offset, output->end()); + + if (num_valid_frames_ < num_output_frames_) { + return false; + } + // Remaining space after the read cursor. + const size_t forward_read_space = read_cursor_ < write_cursor_ + ? write_cursor_ - read_cursor_ + : buffer_.num_frames() - read_cursor_; + + // Copy the buffer values into the output. + AudioBuffer::Channel* buffer_channel = &buffer_[0]; + if (forward_read_space >= num_output_frames_) { + DCHECK_LE(buffer_channel->begin() + read_cursor_ + num_output_frames_, + buffer_channel->end()); + std::copy(buffer_channel->begin() + read_cursor_, + buffer_channel->begin() + read_cursor_ + num_output_frames_, + output->begin() + offset); + } else { + DCHECK_LE(buffer_channel->begin() + read_cursor_ + forward_read_space, + buffer_channel->end()); + DCHECK_LE(output->begin() + forward_read_space + offset, output->end()); + std::copy(buffer_channel->begin() + read_cursor_, + buffer_channel->begin() + read_cursor_ + forward_read_space, + output->begin() + offset); + DCHECK_GE(buffer_channel->begin() + num_output_frames_ - forward_read_space, + buffer_channel->begin()); + DCHECK_LE(output->begin() + offset + num_output_frames_, output->end()); + std::copy(buffer_channel->begin(), + buffer_channel->begin() + num_output_frames_ - forward_read_space, + output->begin() + offset + forward_read_space); + } + read_cursor_ = (read_cursor_ + num_output_frames_) % buffer_.num_frames(); + num_valid_frames_ -= num_output_frames_; + return true; +} + +void CircularBuffer::Clear() { + read_cursor_ = 0; + write_cursor_ = 0; + num_valid_frames_ = 0; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer.h new file mode 100644 index 000000000..e21f5817d --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer.h @@ -0,0 +1,96 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_CIRCULAR_BUFFER_H_ +#define RESONANCE_AUDIO_DSP_CIRCULAR_BUFFER_H_ + +#include "base/audio_buffer.h" + +namespace vraudio { +// Class that implements a simple mono circular buffer, accepting input and +// output of different length. +class CircularBuffer { + public: + // Constructs a circular buffer. + // + // @param buffer_length The length of the Circular buffer. This value must be + // at least |num_input_frames| + |num_output_frames| so that we can always + // either add or remove data. + // @param num_input_frames Length of the input buffers in frames. + // @param num_output_frames Length of the output buffers in frames. + CircularBuffer(size_t buffer_length, size_t num_input_frames, + size_t num_output_frames); + + // Inserts a buffer of mono input into the |CircularBuffer| if space permits. + // + // @param input A channel of input data |num_input_frames| in length. + // @return True if there was space in the buffer and the input was + // successfully inserted, false otherwise. + bool InsertBuffer(const AudioBuffer::Channel& input); + + // Retrieves a buffer of output from the |CircularBuffer| if it contains + // sufficient data. + // + // @param output A channel to hold |num_output_frames| of output data. The + // channel may be greater in length than |num_output_frames|, in this + // case only the first |num_output_frames| will be overwritten. + // @return True if there was sufficient data in the buffer and the output was + // successfully retrieved. + bool RetrieveBuffer(AudioBuffer::Channel* output); + + // Retrieves a buffer of output from the |CircularBuffer| to an offset + // location in an output channel, provided it contains sufficient data. + // + // @param offset Number of samples of offset into the |output| channel. + // @param output A channel to hold |num_output_frames| of output data. The + // channel may be greater in length than |num_output_frames| + |offset|, + // in this case only the first |num_output_frames| after |offset| will be + // overwritten. + // @return True if there was sufficient data in the buffer and the output was + // successfully retrieved. + bool RetrieveBufferWithOffset(size_t offset, AudioBuffer::Channel* output); + + // Returns the number of samples of data currently in the |CircularBuffer|. + // + // @return The number of samples of data currently in the buffer. + size_t GetOccupancy() const { return num_valid_frames_; } + + // Resets the |CircularBuffer|. + void Clear(); + + private: + // Number of input frames to be inserted into the buffer. + const size_t num_input_frames_; + + // Number of output frames to be retrieved from the buffer. + const size_t num_output_frames_; + + // Mono audio buffer to hold the data. + AudioBuffer buffer_; + + // Position at which we are writing into the buffer. + size_t write_cursor_; + + // position at which we are reading from the buffer. + size_t read_cursor_; + + // Number of frames of data currently stored within the buffer. + size_t num_valid_frames_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_CIRCULAR_BUFFER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer_test.cc new file mode 100644 index 000000000..5b53246e6 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/circular_buffer_test.cc @@ -0,0 +1,230 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/circular_buffer.h" + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +// Tests that when the output from the circular buffer is of a smaller length +// than the input, the |CircularBuffer|correctly accepts and rejects inserts +// and retrievals. +TEST(CircularBufferTest, RemainingReadSpaceTest) { + const size_t kInputSize = 5; + const size_t kOutputSize = 4; + const size_t kBufferSize = 9; + CircularBuffer circular_buffer(kBufferSize, kInputSize, kOutputSize); + AudioBuffer input_a(kNumMonoChannels, kInputSize); + input_a[0] = {0.0f, 1.0f, 2.0f, 3.0f, 4.0f}; + AudioBuffer output(kNumMonoChannels, kOutputSize); + output.Clear(); + + EXPECT_TRUE(circular_buffer.InsertBuffer(input_a[0])); + // There should not be enough space to write another buffer. + EXPECT_FALSE(circular_buffer.InsertBuffer(input_a[0])); + + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + // There should not be enough data to read from the buffer. + EXPECT_FALSE(circular_buffer.RetrieveBuffer(&output[0])); + + // The output buffer should contain the first |kOutputSize| entries from the + // input buffer. + for (size_t i = 0; i < kOutputSize; ++i) { + EXPECT_FLOAT_EQ(output[0][i], input_a[0][i]); + } + + AudioBuffer input_b(kNumMonoChannels, kInputSize); + input_b[0] = {5.0f, 6.0f, 7.0f, 8.0f, 9.0f}; + + EXPECT_TRUE(circular_buffer.InsertBuffer(input_b[0])); + // There should not be enough space to write another buffer. + EXPECT_FALSE(circular_buffer.InsertBuffer(input_b[0])); + + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + // There should not be enough data to read from the buffer. + EXPECT_FALSE(circular_buffer.RetrieveBuffer(&output[0])); + + // The output buffer should contain the final entry from the first input + // buffer and then |kOutputSize| - 1 entries from the second. + EXPECT_FLOAT_EQ(output[0][0], input_a[0][kInputSize - 1]); + for (size_t i = 1; i < kOutputSize; ++i) { + EXPECT_FLOAT_EQ(output[0][i], input_b[0][i - 1]); + } +} + +// Tests that when the output from the circular buffer is of a greater length +// than the input, the |CircularBuffer|correctly accepts and rejects inserts +// and retrievals. +TEST(CircularBufferTest, RemainingWriteSpaceTest) { + const size_t kInputSize = 4; + const size_t kOutputSize = 5; + const size_t kBufferSize = 10; + CircularBuffer circular_buffer(kBufferSize, kInputSize, kOutputSize); + AudioBuffer input_a(kNumMonoChannels, kInputSize); + input_a[0] = {0.0f, 1.0f, 2.0f, 3.0f}; + AudioBuffer input_b(kNumMonoChannels, kInputSize); + input_b[0] = {4.0f, 5.0f, 6.0f, 7.0f}; + AudioBuffer input_c(kNumMonoChannels, kInputSize); + input_c[0] = {8.0f, 9.0f, 10.0f, 11.0f}; + AudioBuffer input_d(kNumMonoChannels, kInputSize); + input_d[0] = {12.0f, 13.0f, 14.0f, 15.0f}; + AudioBuffer input_e(kNumMonoChannels, kInputSize); + input_e[0] = {16.0f, 17.0f, 18.0f, 19.0f}; + AudioBuffer output(kNumMonoChannels, kOutputSize); + output.Clear(); + + EXPECT_TRUE(circular_buffer.InsertBuffer(input_a[0])); + EXPECT_TRUE(circular_buffer.InsertBuffer(input_b[0])); + // There should not be enough space to write another buffer. + EXPECT_FALSE(circular_buffer.InsertBuffer(input_c[0])); + + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + // There should not be enough data to read from the buffer. + EXPECT_FALSE(circular_buffer.RetrieveBuffer(&output[0])); + + float value = 0.0f; + for (size_t i = 0; i < kOutputSize; ++i) { + EXPECT_FLOAT_EQ(output[0][i], value); + value += 1.0f; + } + + // Add another 4 samples of input in. + EXPECT_TRUE(circular_buffer.InsertBuffer(input_c[0])); + // There should not be enough space to write another buffer. + EXPECT_FALSE(circular_buffer.InsertBuffer(input_d[0])); + + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + // There should not be enough data to read from the buffer. + EXPECT_FALSE(circular_buffer.RetrieveBuffer(&output[0])); + + for (size_t i = 0; i < kOutputSize; ++i) { + EXPECT_FLOAT_EQ(output[0][i], value); + value += 1.0f; + } + + // Add another 8 samples of input in. + EXPECT_TRUE(circular_buffer.InsertBuffer(input_d[0])); + EXPECT_TRUE(circular_buffer.InsertBuffer(input_e[0])); + // There should not be enough space to write another buffer. + EXPECT_FALSE(circular_buffer.InsertBuffer(input_a[0])); + + // We should be able to get 10 samples of output. + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + for (size_t i = 0; i < kOutputSize; ++i) { + EXPECT_FLOAT_EQ(output[0][i], value); + value += 1.0f; + } + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + for (size_t i = 0; i < kOutputSize; ++i) { + EXPECT_FLOAT_EQ(output[0][i], value); + value += 1.0f; + } + + // There should not be enough data to read from the buffer. + EXPECT_FALSE(circular_buffer.RetrieveBuffer(&output[0])); + + // The buffer should now be completely empty. + EXPECT_EQ(circular_buffer.GetOccupancy(), 0U); +} + +// Tests tha a call to RetrieveBuffer will work when the output buffer is +// oversized, but that only the first kOutputSize samples are filled. +TEST(CircularBufferTest, LongerWriteSpaceTest) { + const size_t kInputSize = 5; + const size_t kOutputSize = 3; + const size_t kBufferSize = 10; + CircularBuffer circular_buffer(kBufferSize, kInputSize, kOutputSize); + AudioBuffer input_a(kNumMonoChannels, kInputSize); + input_a[0] = {1.0f, 2.0f, 3.0f, 4.0f, 5.0f}; + AudioBuffer output(kNumMonoChannels, kBufferSize); + output.Clear(); + + EXPECT_TRUE(circular_buffer.InsertBuffer(input_a[0])); + EXPECT_TRUE(circular_buffer.RetrieveBuffer(&output[0])); + + for (size_t i = 0; i < kBufferSize; ++i) { + if (i < kOutputSize) { + EXPECT_FLOAT_EQ(output[0][i], input_a[0][i]); + } else { + EXPECT_FLOAT_EQ(output[0][i], 0.0f); + } + } +} + +// Tests tha a call to RetrieveBufferOffset will work when the output buffer is +// oversized, but that the first kOutputSize samples after kOffset are filled. +TEST(CircularBufferTest, OffsetRerieveTest) { + const size_t kInputSize = 5; + const size_t kOutputSize = 3; + const size_t kBufferSize = 10; + const size_t kOffset = 2; + CircularBuffer circular_buffer(kBufferSize, kInputSize, kOutputSize); + AudioBuffer input_a(kNumMonoChannels, kInputSize); + input_a[0] = {1.0f, 2.0f, 3.0f, 4.0f, 5.0f}; + AudioBuffer output(kNumMonoChannels, kBufferSize); + output.Clear(); + + EXPECT_TRUE(circular_buffer.InsertBuffer(input_a[0])); + EXPECT_TRUE(circular_buffer.RetrieveBufferWithOffset(kOffset, &output[0])); + + for (size_t i = 0; i < kBufferSize; ++i) { + if (i < kOffset) { + EXPECT_FLOAT_EQ(output[0][i], 0.0f); + } else if (i < kOffset + kOutputSize) { + EXPECT_FLOAT_EQ(output[0][i], input_a[0][i - kOffset]); + } else { + EXPECT_FLOAT_EQ(output[0][i], 0.0f); + } + } +} + +// Tests that the circular buffer handles odd input buffer lengths in the +// manner that it will be passed in the spectral reverb +// (see: dsp/spectral_reverb.cc) +TEST(CircularBufferTest, Strange) { + const size_t kNumRuns = 100; + // An odd non pwer of two input buffer size. + const size_t kOddInputSize = 713; + // Output size is equal to the SpectralReverb's internal buffer size. + const size_t kOutputSize = 1024; + // SpectralReverb's internal forier transform length. + const size_t kFFTSize = 4096; + CircularBuffer input(kFFTSize + kOddInputSize, kOddInputSize, kOutputSize); + CircularBuffer output(kOutputSize + kOddInputSize, kOutputSize, + kOddInputSize); + AudioBuffer in(kNumMonoChannels, kOddInputSize); + AudioBuffer out(kNumMonoChannels, kOutputSize); + + // AudioBuffers will be input and output in the same manner as in + // dsp/spectral_reverb.cc. + EXPECT_TRUE(output.InsertBuffer(out[0])); + for (size_t i = 0; i < kNumRuns; ++i) { + EXPECT_TRUE(input.InsertBuffer(in[0])); + while (input.GetOccupancy() >= kOutputSize) { + EXPECT_TRUE(input.RetrieveBuffer(&out[0])); + EXPECT_TRUE(output.InsertBuffer(out[0])); + } + EXPECT_TRUE(output.RetrieveBuffer(&in[0])); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter.cc new file mode 100644 index 000000000..b448de58b --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter.cc @@ -0,0 +1,180 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/delay_filter.h" + +#include <cmath> + +#include "base/constants_and_types.h" +#include "base/misc_math.h" + + +namespace vraudio { +// The following explains the behaviour of the delay line when a change to the +// delay length is made. Initially |delay| is 2, |frames_per_buffer_| +// is 4, and |delay_line_| is 8 in length. +// +// delay = 2 +// +// input0 [1, 2, 3, 4] output0 [0, 0, 1, 2] +// +// 0 1 2 3 4 5 6 7 +// delay line [1, 2, 3, 4, 0, 0, 0, 0] +// | | +// w r +// ---------------------------------------------------------- +// +// delay = 2 +// +// input1 [5, 6, 7, 8] output1 [3, 4, 5, 6] +// +// 0 1 2 3 4 5 6 7 +// delay line [1, 2, 3, 4, 5, 6, 7, 8] +// | | +// w r +// ---------------------------------------------------------- +// +// delay = 1 +// +// 0 1 2 3 4 5 6 7 +// delay line [1, 2, 3, 4, 5, 6, 7, 8] +// | | +// w r +// ---------------------------------------------------------- +// +// delay = 1 +// +// input2 [9, 10, 11, 12] output2 [8, 9, 10, 11] +// +// 0 1 2 3 4 5 6 7 +// delay line [9, 10, 11, 12, 5, 6, 7, 8] +// | | +// w r +// ---------------------------------------------------------- + +DelayFilter::DelayFilter(size_t max_delay_length, size_t frames_per_buffer) + : frames_per_buffer_(frames_per_buffer), + delay_line_(nullptr), + write_cursor_(0) { + DCHECK_GT(frames_per_buffer_, 0U); + SetMaximumDelay(max_delay_length); +} + +void DelayFilter::SetMaximumDelay(size_t max_delay_length) { + max_delay_length_ = max_delay_length; + const size_t new_buffer_size = frames_per_buffer_ + max_delay_length_; + + if (delay_line_ == nullptr) { + delay_line_.reset(new AudioBuffer(kNumMonoChannels, new_buffer_size)); + delay_line_->Clear(); + return; + } + AudioBuffer::Channel* delay_channel = &(*delay_line_)[0]; + // If |delay_line_| is not large enough, resize. + const size_t current_buffer_size = delay_line_->num_frames(); + if (max_delay_length_ > current_buffer_size - frames_per_buffer_) { + // Allocate |new_delay_line| and populate it with the current |delay_line_| + // data, so that replacing the buffer does not affect the already stored + // samples. + auto new_delay_line = std::unique_ptr<AudioBuffer>( + new AudioBuffer(kNumMonoChannels, new_buffer_size)); + new_delay_line->Clear(); + std::copy(delay_channel->begin() + write_cursor_, delay_channel->end(), + (*new_delay_line)[0].begin()); + if (write_cursor_ > 0) { + // Positive |write_cursor_| means that we still have remaining samples to + // be moved to |new_delay_line_| all of which reside left of the cursor. + std::copy( + delay_channel->begin(), delay_channel->begin() + write_cursor_, + (*new_delay_line)[0].begin() + current_buffer_size - write_cursor_); + write_cursor_ = current_buffer_size; + } + delay_line_ = std::move(new_delay_line); + } +} + +void DelayFilter::InsertData(const AudioBuffer::Channel& input) { + + DCHECK_EQ(input.size(), frames_per_buffer_); + + const size_t delay_buffer_size = delay_line_->num_frames(); + + // Record the remaining space in the |delay_line_| after the write cursor. + const size_t remaining_size_write = delay_buffer_size - write_cursor_; + AudioBuffer::Channel* delay_channel = &(*delay_line_)[0]; + + // Copy the input into the delay line. + if (remaining_size_write >= frames_per_buffer_) { + DCHECK_LE(delay_channel->begin() + write_cursor_ + frames_per_buffer_, + delay_channel->end()); + std::copy(input.begin(), input.end(), + delay_channel->begin() + write_cursor_); + } else { + DCHECK_LE(delay_channel->begin() + write_cursor_ + remaining_size_write, + delay_channel->end()); + DCHECK_LE(input.begin() + remaining_size_write, input.end()); + std::copy(input.begin(), input.begin() + remaining_size_write, + delay_channel->begin() + write_cursor_); + DCHECK_LE(delay_channel->begin() + remaining_size_write, + delay_channel->end()); + std::copy(input.begin() + remaining_size_write, input.end(), + delay_channel->begin()); + } + + write_cursor_ = (write_cursor_ + frames_per_buffer_) % delay_buffer_size; +} + +void DelayFilter::GetDelayedData(size_t delay_samples, + AudioBuffer::Channel* buffer) { + + DCHECK(buffer); + DCHECK_GE(delay_samples, 0U); + DCHECK_LE(delay_samples, max_delay_length_); + + const size_t delay_buffer_size = delay_line_->num_frames(); + // Position in the delay line to begin reading from. + DCHECK_GE(write_cursor_ + delay_buffer_size, + delay_samples + frames_per_buffer_); + const size_t read_cursor = + (write_cursor_ + delay_buffer_size - delay_samples - frames_per_buffer_) % + delay_buffer_size; + // Record the remaining space in the |delay_line_| after the read cursor. + const size_t remaining_size_read = delay_buffer_size - read_cursor; + AudioBuffer::Channel* delay_channel = &(*delay_line_)[0]; + + // Extract a portion of the delay line into the buffer. + if (remaining_size_read >= frames_per_buffer_) { + DCHECK_LE(buffer->begin() + frames_per_buffer_, buffer->end()); + DCHECK_LE(delay_channel->begin() + read_cursor + frames_per_buffer_, + delay_channel->end()); + std::copy(delay_channel->begin() + read_cursor, + delay_channel->begin() + read_cursor + frames_per_buffer_, + buffer->begin()); + } else { + DCHECK_LE(buffer->begin() + delay_channel->size() - read_cursor, + buffer->end()); + std::copy(delay_channel->begin() + read_cursor, delay_channel->end(), + buffer->begin()); + DCHECK_LE(buffer->begin() + frames_per_buffer_, buffer->end()); + DCHECK_LE(delay_channel->begin() + frames_per_buffer_ - remaining_size_read, + delay_channel->end()); + std::copy(delay_channel->begin(), + delay_channel->begin() + frames_per_buffer_ - remaining_size_read, + buffer->begin() + remaining_size_read); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter.h new file mode 100644 index 000000000..be7c49774 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter.h @@ -0,0 +1,85 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_DELAY_FILTER_H_ +#define RESONANCE_AUDIO_DSP_DELAY_FILTER_H_ + +#include <algorithm> +#include <memory> + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Single channel delay line class. Delays input buffer data by a non-negative +// integer number of samples. +// +// This implementation is not thread safe. The ClearBuffer() and InsertData() +// functions should not be called by a seperate thread during GetDelayedData(). +class DelayFilter { + public: + // Constructs a DelayFilter. + // + // @param max_delay_length Maximum number of samples the input should be + // delayed by. + // @param frames_per_buffer Number of frames in each processed buffer. + DelayFilter(size_t max_delay_length, size_t frames_per_buffer); + + // Sets the maximum delay length. It will allocate more space in the + // |delay_line_| if the new |max_delay_length| is more than doubled. + // + // @param max_delay_length New maximum delay in samples. + void SetMaximumDelay(size_t max_delay_length); + + // Returns the current length of the |delay_line_|. + size_t GetDelayBufferLength() const { return delay_line_->num_frames(); } + + // Returns the current maximum delay applicable to input buffers. + size_t GetMaximumDelayLength() const { return max_delay_length_; } + + // Sets all of the |delay_line_| samples to zero. + void ClearBuffer() { delay_line_->Clear(); } + + // Copies an |AudioBuffer::Channel| of data to the delay line. + // + // @param input Input data. + void InsertData(const AudioBuffer::Channel& input); + + // Fills an |AudioBuffer::Channel| with data delayed by a specified amount + // less than or equal to the delay line's set |max_delay_length_|. + // + // @param delay_samples Requested delay to the data extraced from the delay + // line. Must be less than or equal to |max_delay_length_|. + // @param buffer Pointer to the output data, i.e., delayed input data. + void GetDelayedData(size_t delay_samples, AudioBuffer::Channel* buffer); + + private: + // Maximum length of the delay to be applied (in samples). + size_t max_delay_length_; + + // Number of frames in each AudioBuffer input/output. + size_t frames_per_buffer_; + + // The delay line holding all of the delayed samples. + std::unique_ptr<AudioBuffer> delay_line_; + + // Position in the delay line to begin writing to. + size_t write_cursor_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_DELAY_FILTER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter_test.cc new file mode 100644 index 000000000..12a7a8f60 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/delay_filter_test.cc @@ -0,0 +1,197 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/delay_filter.h" + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +// Set of delay lengths to be used in the tests below. +const size_t kInitDelayLength = 4; +const size_t kSecondDelayLength = 5; +const size_t kThirdDelayLength = 2; +const size_t kFourthDelayLength = 11; +const size_t kZeroDelayLength = 0; + +// Frames for buffer for the input and output data buffers below. +const size_t kFramesPerBuffer = 6; + +// Frames per buffer and channel number for the input and output data. +const size_t kFramesPerBuffer2 = 10; + +// Function which passes an AudioBuffer through the delay line followed by all +// zero AudioBuffers to flush the data out. This function then tests that the +// input data has been delayed by the correct amount. +// +// @param delay A pointer to a DelayFilter which will be used in the test. +// @param delay_length An integer delay length to be used in the test. +void IntegerDelayTestHelper(DelayFilter* delay, size_t delay_length) { + std::vector<float> output_collect; + delay->SetMaximumDelay(delay_length); + delay->ClearBuffer(); + + // Initialize mono input buffer and fill with test data. + const std::vector<float> kData = {1.0f, 2.0f, 3.0f, 4.0f, 5.0f, 6.0f}; + AudioBuffer input(kNumMonoChannels, kFramesPerBuffer); + auto* input_channel = &input[0]; + *input_channel = kData; + + // Process input. + delay->InsertData(*input_channel); + delay->GetDelayedData(delay_length, input_channel); + output_collect.insert(output_collect.end(), input_channel->begin(), + input_channel->end()); + + // Keep passing zeros through till we flush all of the input data out. + while (delay_length + kFramesPerBuffer > output_collect.size()) { + // Set the |input| AudioBuffer to have all zero data. + input.Clear(); + // Process input. + delay->InsertData(*input_channel); + delay->GetDelayedData(delay_length, input_channel); + output_collect.insert(output_collect.end(), input_channel->begin(), + input_channel->end()); + } + + // Check that the first GetDelayedData() call yields |delay_length| zeros at + // the beginning of its output. + for (size_t i = 0; i < delay_length; ++i) { + EXPECT_EQ(output_collect[i], 0.0f); + } + // Check that the output is the same data as the input but delayed by + // delay_length samples. + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_EQ(output_collect[i + delay_length], kData[i]); + } +} + +// Tests that the maximum_delay_length_ and delay_buffer_size_ are set to the +// correct values on construction and when SetDelay() is called. +TEST(DelayFilterTest, DelayCorrectTest) { + DelayFilter delay(kInitDelayLength, kFramesPerBuffer); + + EXPECT_EQ(delay.GetMaximumDelayLength(), kInitDelayLength); + EXPECT_EQ(delay.GetDelayBufferLength(), kInitDelayLength + kFramesPerBuffer); + + delay.SetMaximumDelay(kSecondDelayLength); + + EXPECT_EQ(delay.GetMaximumDelayLength(), kSecondDelayLength); + EXPECT_EQ(delay.GetDelayBufferLength(), + kSecondDelayLength + kFramesPerBuffer); + + // In this next case we reduce the maximum length and thus do not expect a + // reallocation. + delay.SetMaximumDelay(kThirdDelayLength); + + EXPECT_EQ(delay.GetMaximumDelayLength(), kThirdDelayLength); + EXPECT_EQ(delay.GetDelayBufferLength(), + kSecondDelayLength + kFramesPerBuffer); + + delay.SetMaximumDelay(kFourthDelayLength); + + EXPECT_EQ(delay.GetMaximumDelayLength(), kFourthDelayLength); + // Now we expect the buffer to have been reallocated. + EXPECT_EQ(delay.GetDelayBufferLength(), + kFourthDelayLength + kFramesPerBuffer); +} + +// Tests whether when setting a different delay on one DelayFilter, +// the output is correct in each case. +TEST(DelayFilterTest, DelayTest) { + DelayFilter delay(kInitDelayLength, kFramesPerBuffer); + + IntegerDelayTestHelper(&delay, kInitDelayLength); + + // Tests the case of an increasing delay. + IntegerDelayTestHelper(&delay, kSecondDelayLength); + + // Tests the case of a decreasing delay. + IntegerDelayTestHelper(&delay, kThirdDelayLength); + + // Tests the case of an increasing delay with allocation of more buffer space. + IntegerDelayTestHelper(&delay, kFourthDelayLength); +} + +// Tests that differently delayed buffers can be extracted from a single delay +// line. +TEST(DelayFilterTest, MultipleDelaysTest) { + DelayFilter delay(kFramesPerBuffer2, kFramesPerBuffer2); + AudioBuffer input(kNumMonoChannels, kFramesPerBuffer2); + for (size_t i = 0; i < kFramesPerBuffer2; ++i) { + input[0][i] = static_cast<float>(i + 1); + } + delay.InsertData(input[0]); + const size_t kDelayOne = 1; + const size_t kDelayTwo = 2; + AudioBuffer buffer_1(kNumMonoChannels, kFramesPerBuffer2); + AudioBuffer buffer_2(kNumMonoChannels, kFramesPerBuffer2); + + delay.GetDelayedData(kDelayOne, &buffer_1[0]); + for (size_t i = 0; i < kFramesPerBuffer2 - kDelayOne; ++i) { + EXPECT_NEAR(buffer_1[0][i + kDelayOne], input[0][i], kEpsilonFloat); + } + + delay.GetDelayedData(kDelayTwo, &buffer_2[0]); + for (size_t i = 0; i < kFramesPerBuffer2 - kDelayTwo; ++i) { + EXPECT_NEAR(buffer_2[0][i + kDelayTwo], input[0][i], kEpsilonFloat); + } +} + +// Tests whether a zero delay length is dealt with correctly, Along with a +// negative delay value (treated as zero delay. +TEST(DelayFilterTest, ZeroDelayTest) { + DelayFilter delay(kZeroDelayLength, kFramesPerBuffer); + IntegerDelayTestHelper(&delay, kZeroDelayLength); +} + +// Tests that output from a delay line that is initally large enough vs one that +// is resized is the same. +TEST(DelayFilterTest, InitialSizeVsResizeTest) { + const size_t kSmallMaxDelay = 2; + const size_t kLargeMaxDelay = 5; + const size_t kActualDelay = 4; + + DelayFilter delay_sufficient(kLargeMaxDelay, kFramesPerBuffer); + DelayFilter delay_insufficient(kSmallMaxDelay, kFramesPerBuffer); + + AudioBuffer input_buffer(kNumMonoChannels, kFramesPerBuffer); + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + input_buffer[0][i] = static_cast<float>(i + 1); + } + + delay_sufficient.InsertData(input_buffer[0]); + delay_insufficient.InsertData(input_buffer[0]); + delay_insufficient.SetMaximumDelay(kLargeMaxDelay); + + AudioBuffer buffer_sufficient(kNumMonoChannels, kFramesPerBuffer); + AudioBuffer buffer_insufficient(kNumMonoChannels, kFramesPerBuffer); + + delay_sufficient.GetDelayedData(kActualDelay, &buffer_sufficient[0]); + delay_insufficient.GetDelayedData(kActualDelay, &buffer_insufficient[0]); + + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR(buffer_sufficient[0][i], buffer_insufficient[0][i], + kEpsilonFloat); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation.cc new file mode 100644 index 000000000..816053a7c --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation.cc @@ -0,0 +1,121 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/distance_attenuation.h" + +#include <algorithm> +#include <cmath> + +#include "base/constants_and_types.h" + +namespace vraudio { + +float ComputeLogarithmicDistanceAttenuation( + const WorldPosition& listener_position, + const WorldPosition& source_position, float min_distance, + float max_distance) { + const float distance = (listener_position - source_position).norm(); + if (distance > max_distance) { + return 0.0f; + } + // Logarithmic attenuation. + const float min_distance_allowed = + std::max(min_distance, kNearFieldThreshold); + if (distance > min_distance_allowed) { + const float attenuation_interval = max_distance - min_distance_allowed; + if (attenuation_interval > kEpsilonFloat) { + // Compute the distance attenuation value by the logarithmic curve + // "1 / (d + 1)" with an offset of |min_distance_allowed|. + const float relative_distance = distance - min_distance_allowed; + const float attenuation = 1.0f / (relative_distance + 1.0f); + // Shift the curve downwards by the attenuation value at |max_distance|, + // and scale the value by the inverse of it in order to keep the curve's + // peak value 1 at |min_distance_allowed|. + const float attenuation_max = 1.0f / (1.0f + attenuation_interval); + return (attenuation - attenuation_max) / (1.0f - attenuation_max); + } + } + return 1.0f; +} + +float ComputeLinearDistanceAttenuation(const WorldPosition& listener_position, + const WorldPosition& source_position, + float min_distance, float max_distance) { + const float distance = (listener_position - source_position).norm(); + if (distance > max_distance) { + return 0.0f; + } + // Linear attenuation. + const float min_distance_allowed = + std::max(min_distance, kNearFieldThreshold); + if (distance > min_distance_allowed) { + const float attenuation_interval = max_distance - min_distance_allowed; + if (attenuation_interval > kEpsilonFloat) { + return (max_distance - distance) / attenuation_interval; + } + } + return 1.0f; +} + +float ComputeNearFieldEffectGain(const WorldPosition& listener_position, + const WorldPosition& source_position) { + const float distance = (listener_position - source_position).norm(); + if (distance < kNearFieldThreshold) { + return (1.0f / std::max(distance, kMinNearFieldDistance)) - 1.0f; + } + return 0.0f; +} + +void UpdateAttenuationParameters(float master_gain, float reflections_gain, + float reverb_gain, + const WorldPosition& listener_position, + SourceParameters* parameters) { + // Compute distance attenuation. + const WorldPosition& source_position = parameters->object_transform.position; + const auto rolloff_model = parameters->distance_rolloff_model; + const float min_distance = parameters->minimum_distance; + const float max_distance = parameters->maximum_distance; + + float distance_attenuation = 0.0f; + switch (rolloff_model) { + case DistanceRolloffModel::kLogarithmic: + distance_attenuation = ComputeLogarithmicDistanceAttenuation( + listener_position, source_position, min_distance, max_distance); + break; + case DistanceRolloffModel::kLinear: + distance_attenuation = ComputeLinearDistanceAttenuation( + listener_position, source_position, min_distance, max_distance); + break; + case DistanceRolloffModel::kNone: + default: + // Distance attenuation is already set by the user. + distance_attenuation = parameters->distance_attenuation; + break; + } + // Update gain attenuations. + const float input_gain = master_gain * parameters->gain; + const float direct_attenuation = input_gain * distance_attenuation; + const float room_effects_attenuation = parameters->room_effects_gain; + + parameters->attenuations[AttenuationType::kInput] = input_gain; + parameters->attenuations[AttenuationType::kDirect] = direct_attenuation; + parameters->attenuations[AttenuationType::kReflections] = + room_effects_attenuation * direct_attenuation * reflections_gain; + parameters->attenuations[AttenuationType::kReverb] = + room_effects_attenuation * input_gain * reverb_gain; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation.h new file mode 100644 index 000000000..2b98ac63c --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation.h @@ -0,0 +1,82 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_DISTANCE_ATTENUATION_H_ +#define RESONANCE_AUDIO_DSP_DISTANCE_ATTENUATION_H_ + +#include "base/misc_math.h" +#include "base/source_parameters.h" + +namespace vraudio { + +// Returns the distance attenuation for |source_position| with respect to +// |listener_position|. The amplitude will decrease by approximately 6 dB every +// time the distance is doubled, i.e., the sound pressure "p" (amplitude) +// approximately falls inversely proportional to the distance "1/r". +// +// @param listener_position World position of the listener. +// @param source_position World position of the source. +// @param min_distance The minimum distance at which distance attenuation is +// applied. +// @param max_distance The maximum distance at which the direct sound has a gain +// of 0.0. +// @return Attenuation (gain) value in range [0.0f, 1.0f]. +float ComputeLogarithmicDistanceAttenuation( + const WorldPosition& listener_position, + const WorldPosition& source_position, float min_distance, + float max_distance); + +// Returns the distance attenuation for |source_position| with respect to +// |listener_position|. The amplitude will decrease linearly between +// |min_distance| and |max_distance| from 1.0 to 0.0. +// +// @param listener_position World position of the listener. +// @param source_position World position of the source. +// @param min_distance The minimum distance at which distance attenuation is +// applied. +// @param max_distance The maximum distance at which the direct sound has a gain +// of 0.0. +// @return Attenuation (gain) value in range [0.0f, 1.0f]. +float ComputeLinearDistanceAttenuation(const WorldPosition& listener_position, + const WorldPosition& source_position, + float min_distance, float max_distance); + +// Calculates the gain to be applied to the near field compensating stereo mix. +// This function will return 0.0f for all sources further away than one meter +// and will return a value between 0.0 and 9.0 for sources as they approach +// the listener's head location. +// +// @param listener_position World position of the listener. +// @param source_position World position of the source. +// @return Gain value in range [0.0f, 9.0f]. +float ComputeNearFieldEffectGain(const WorldPosition& listener_position, + const WorldPosition& source_position); + +// Calculates and updates gain attenuations of the given source |parameters|. +// +// @param master_gain Global gain adjustment in amplitude. +// @param reflections_gain Reflections gain in amplitude. +// @param reverb_gain Reverb gain in amplitude. +// @param listener_position World position of the listener. +// @param parameters Source parameters to apply the gain attenuations into. +void UpdateAttenuationParameters(float master_gain, float reflections_gain, + float reverb_gain, + const WorldPosition& listener_position, + SourceParameters* parameters); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_DISTANCE_ATTENUATION_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation_test.cc new file mode 100644 index 000000000..6bccfcabf --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/distance_attenuation_test.cc @@ -0,0 +1,132 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/distance_attenuation.h" + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "base/misc_math.h" + +namespace vraudio { + +namespace { + +// Tests the logarithmic distance attenuation method against the pre-computed +// results. +TEST(DistanceAttenuationTest, ComputeLogarithmicDistanceAttenuationTest) { + const WorldPosition kListenerPosition(0.0f, 3.0f, 0.0f); + const WorldPosition kSourcePosition(0.0f, 0.0f, -4.0f); + const float kMinDistance_low = 1.0f; + const float kMinDistance_high = 10.0f; + const float kMaxDistance_low = 3.0f; + const float kMaxDistance_high = 500.0f; + const float kExpectedAttenuation_a = 0.1983967f; + const float kExpectedAttenuation_b = 0.0f; + const float kExpectedAttenuation_c = 1.0f; + + float attenuation = ComputeLogarithmicDistanceAttenuation( + kListenerPosition, kSourcePosition, kMinDistance_low, kMaxDistance_high); + EXPECT_NEAR(kExpectedAttenuation_a, attenuation, kEpsilonFloat); + + // Test for the case where the source is beyond the maximum distance. + attenuation = ComputeLogarithmicDistanceAttenuation( + kListenerPosition, kSourcePosition, kMinDistance_low, kMaxDistance_low); + EXPECT_NEAR(kExpectedAttenuation_b, attenuation, kEpsilonFloat); + + // Test for the case where the source is within the minimum distance. + attenuation = ComputeLogarithmicDistanceAttenuation( + kListenerPosition, kSourcePosition, kMinDistance_high, kMaxDistance_high); + EXPECT_NEAR(kExpectedAttenuation_c, attenuation, kEpsilonFloat); +} + +// Tests the linear distance attenuation method against the pre-computed +// results. +TEST(DistanceAttenuationTest, ComputeLinearDistanceAttenuationTest) { + const WorldPosition kListenerPosition(0.0f, 3.0f, 0.0f); + const WorldPosition kSourcePosition(0.0f, 0.0f, -4.0f); + const float kMinDistance_low = 1.0f; + const float kMinDistance_high = 10.0f; + const float kMaxDistance_low = 3.0f; + const float kMaxDistance_high = 8.0f; + const float kExpectedAttenuation_a = 0.4285714f; + const float kExpectedAttenuation_b = 0.0f; + const float kExpectedAttenuation_c = 1.0f; + + float attenuation = ComputeLinearDistanceAttenuation( + kListenerPosition, kSourcePosition, kMinDistance_low, kMaxDistance_high); + EXPECT_NEAR(kExpectedAttenuation_a, attenuation, kEpsilonFloat); + + // Test for the case where the source is beyond the maximum distance. + attenuation = ComputeLinearDistanceAttenuation( + kListenerPosition, kSourcePosition, kMinDistance_low, kMaxDistance_low); + EXPECT_NEAR(kExpectedAttenuation_b, attenuation, kEpsilonFloat); + + // Test for the case where the source is within the minimum distance. + attenuation = ComputeLinearDistanceAttenuation( + kListenerPosition, kSourcePosition, kMinDistance_high, kMaxDistance_high); + EXPECT_NEAR(kExpectedAttenuation_c, attenuation, kEpsilonFloat); +} + +// Tests the gain attenuations update method against the pre-computed results. +TEST(DistanceAttenuationTest, UpdateAttenuationParametersTest) { + const float kMasterGain = 0.5f; + const float kReflectionsGain = 0.5f; + const float kReverbGain = 2.0f; + const WorldPosition kListenerPosition(0.0f, 0.0f, 0.0f); + const WorldPosition kSourcePosition(0.0f, 0.0f, 1.5f); + const float kDistanceAttenuation = 0.2f; + const float kRoomEffectsGain = 0.25f; + + // Initialize new |SourceParameters| with an explicit distance attenuation + // value to avoid extra complexity. + SourceParameters parameters; + parameters.distance_rolloff_model = DistanceRolloffModel::kNone; + parameters.distance_attenuation = kDistanceAttenuation; + parameters.object_transform.position = kSourcePosition; + parameters.room_effects_gain = kRoomEffectsGain; + // Update the gain attenuation parameters. + UpdateAttenuationParameters(kMasterGain, kReflectionsGain, kReverbGain, + kListenerPosition, ¶meters); + // Check the attenuation parameters against the pre-computed values. + const size_t num_attenuations = + static_cast<size_t>(AttenuationType::kNumAttenuationTypes); + const float kExpectedAttenuations[num_attenuations] = {0.5f, 0.1f, 0.0125f, + 0.25f}; + for (size_t i = 0; i < num_attenuations; ++i) { + EXPECT_NEAR(kExpectedAttenuations[i], parameters.attenuations[i], + kEpsilonFloat) + << "Attenuation " << i; + } +} + +// Tests the near field effects gain computation method against the pre-computed +// results. +TEST(NearFieldEffectTest, ComputeNearFieldEffectTest) { + const WorldPosition kListenerPosition(0.0f, 3.0f, 0.0f); + const WorldPosition kSourcePosition_a(0.0f, 0.0f, -4.0f); + const WorldPosition kSourcePosition_b(0.0f, 2.5f, 0.0f); + const float kExpectedGain_a = 0.0f; + const float kExpectedGain_b = 1.0f; + + float gain = ComputeNearFieldEffectGain(kListenerPosition, kSourcePosition_a); + EXPECT_NEAR(kExpectedGain_a, gain, kEpsilonFloat); + gain = ComputeNearFieldEffectGain(kListenerPosition, kSourcePosition_b); + EXPECT_NEAR(kExpectedGain_b, gain, kEpsilonFloat); +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager.cc new file mode 100644 index 000000000..5be470b17 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager.cc @@ -0,0 +1,209 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/fft_manager.h" + +#include <algorithm> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/misc_math.h" +#include "base/simd_utils.h" + + +// Prevent Visual Studio from complaining about std::copy_n. +#if defined(_WIN32) +#define _SCL_SECURE_NO_WARNINGS +#endif + +namespace vraudio { + +namespace { + +// for |fft_size_|s less than 2^14, the stack is used, on the reccomendation of +// the author of the pffft library. +const size_t kPffftMaxStackSize = 16384; + +} // namespace + +// The pffft implementation requires a minimum fft size of 32 samples. +const size_t FftManager::kMinFftSize = 32; + +FftManager::FftManager(size_t frames_per_buffer) + : fft_size_(std::max(NextPowTwo(frames_per_buffer) * 2, kMinFftSize)), + frames_per_buffer_(frames_per_buffer), + inverse_fft_scale_(1.0f / static_cast<float>(fft_size_)), + temp_zeropad_buffer_(kNumMonoChannels, fft_size_), + temp_freq_buffer_(kNumMonoChannels, fft_size_) { + DCHECK_GT(frames_per_buffer, 0U); + DCHECK_GE(fft_size_, kMinFftSize); + DCHECK(!(fft_size_ & (fft_size_ - 1))); // Ensure its a power of two. + // Suggested pffft initialization. + if (fft_size_ > kPffftMaxStackSize) { + // Allocate memory for work space factors etc, Size reccomended by pffft. + const size_t num_bytes = 2 * fft_size_ * sizeof(float); + pffft_workspace_ = + reinterpret_cast<float*>(pffft_aligned_malloc(num_bytes)); + } + + fft_ = pffft_new_setup(static_cast<int>(fft_size_), PFFFT_REAL); + + temp_zeropad_buffer_.Clear(); +} + +FftManager::~FftManager() { + pffft_destroy_setup(fft_); + if (pffft_workspace_ != nullptr) { + pffft_aligned_free(pffft_workspace_); + } +} + +void FftManager::FreqFromTimeDomain(const AudioBuffer::Channel& time_channel, + AudioBuffer::Channel* freq_channel) { + + + DCHECK(freq_channel); + DCHECK_EQ(freq_channel->size(), fft_size_); + DCHECK_LE(time_channel.size(), fft_size_); + + // Perform forward FFT transform. + if (time_channel.size() == fft_size_) { + pffft_transform(fft_, time_channel.begin(), freq_channel->begin(), + pffft_workspace_, PFFFT_FORWARD); + } else { + std::copy_n(time_channel.begin(), frames_per_buffer_, + temp_zeropad_buffer_[0].begin()); + pffft_transform(fft_, temp_zeropad_buffer_[0].begin(), + freq_channel->begin(), pffft_workspace_, PFFFT_FORWARD); + } +} + +void FftManager::TimeFromFreqDomain(const AudioBuffer::Channel& freq_channel, + AudioBuffer::Channel* time_channel) { + + + DCHECK(time_channel); + DCHECK_EQ(freq_channel.size(), fft_size_); + + // Perform reverse FFT transform. + const size_t time_channel_size = time_channel->size(); + if (time_channel_size == fft_size_) { + pffft_transform(fft_, freq_channel.begin(), time_channel->begin(), + pffft_workspace_, PFFFT_BACKWARD); + } else { + DCHECK_EQ(time_channel_size, frames_per_buffer_); + auto& temp_channel = temp_freq_buffer_[0]; + pffft_transform(fft_, freq_channel.begin(), temp_channel.begin(), + pffft_workspace_, PFFFT_BACKWARD); + std::copy_n(temp_channel.begin(), frames_per_buffer_, + time_channel->begin()); + } +} + +void FftManager::ApplyReverseFftScaling(AudioBuffer::Channel* time_channel) { + DCHECK(time_channel->size() == frames_per_buffer_ || + time_channel->size() == fft_size_); + // Normalization must be performed here as we normally do this as part of the + // convolution. + ScalarMultiply(time_channel->size(), inverse_fft_scale_, + time_channel->begin(), time_channel->begin()); +} + +void FftManager::GetCanonicalFormatFreqBuffer(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output) { + DCHECK_EQ(input.size(), fft_size_); + DCHECK_EQ(output->size(), fft_size_); + pffft_zreorder(fft_, input.begin(), output->begin(), PFFFT_FORWARD); +} + +void FftManager::GetPffftFormatFreqBuffer(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output) { + DCHECK_EQ(input.size(), fft_size_); + DCHECK_EQ(output->size(), fft_size_); + pffft_zreorder(fft_, input.begin(), output->begin(), PFFFT_BACKWARD); +} + +void FftManager::MagnitudeFromCanonicalFreqBuffer( + const AudioBuffer::Channel& freq_channel, + AudioBuffer::Channel* magnitude_channel) { + DCHECK(magnitude_channel); + DCHECK_EQ(freq_channel.size(), fft_size_); + DCHECK_EQ(magnitude_channel->size(), frames_per_buffer_ + 1); + + (*magnitude_channel)[0] = std::abs(freq_channel[0]); + ApproxComplexMagnitude(frames_per_buffer_ - 1, freq_channel.begin() + 2, + magnitude_channel->begin() + 1); + (*magnitude_channel)[frames_per_buffer_] = std::abs(freq_channel[1]); +} + +void FftManager::CanonicalFreqBufferFromMagnitudeAndPhase( + const AudioBuffer::Channel& magnitude_channel, + const AudioBuffer::Channel& phase_channel, + AudioBuffer::Channel* canonical_freq_channel) { + DCHECK(canonical_freq_channel); + DCHECK_EQ(magnitude_channel.size(), frames_per_buffer_ + 1); + DCHECK_EQ(phase_channel.size(), frames_per_buffer_ + 1); + DCHECK_EQ(canonical_freq_channel->size(), fft_size_); + + (*canonical_freq_channel)[0] = magnitude_channel[0]; + (*canonical_freq_channel)[1] = -magnitude_channel[frames_per_buffer_]; + for (size_t i = 1, j = 2; i < frames_per_buffer_; ++i, j += 2) { + (*canonical_freq_channel)[j] = + magnitude_channel[i] * std::cos(phase_channel[i]); + (*canonical_freq_channel)[j + 1] = + magnitude_channel[i] * std::sin(phase_channel[i]); + } +} + +void FftManager::CanonicalFreqBufferFromMagnitudeAndSinCosPhase( + size_t phase_offset, const AudioBuffer::Channel& magnitude_channel, + const AudioBuffer::Channel& sin_phase_channel, + const AudioBuffer::Channel& cos_phase_channel, + AudioBuffer::Channel* canonical_freq_channel) { + static const size_t kSimdLength = 4; + DCHECK(canonical_freq_channel); + DCHECK_EQ(magnitude_channel.size(), frames_per_buffer_ + 1); + DCHECK_GE(sin_phase_channel.size() + phase_offset, frames_per_buffer_ + 1); + DCHECK_GE(cos_phase_channel.size() + phase_offset, frames_per_buffer_ + 1); + DCHECK_EQ(canonical_freq_channel->size(), fft_size_); + + (*canonical_freq_channel)[0] = magnitude_channel[0]; + (*canonical_freq_channel)[1] = -magnitude_channel[frames_per_buffer_]; + // Continue on till we can gaurantee alignment in our audio buffer. + for (size_t i = 1, j = 2; i <= kSimdLength; ++i, j += 2) { + (*canonical_freq_channel)[j] = + magnitude_channel[i] * cos_phase_channel[i + phase_offset]; + (*canonical_freq_channel)[j + 1] = + magnitude_channel[i] * sin_phase_channel[i + phase_offset]; + } + ComplexInterleavedFormatFromMagnitudeAndSinCosPhase( + 2 * (frames_per_buffer_ - kSimdLength), &magnitude_channel[kSimdLength], + &cos_phase_channel[kSimdLength + phase_offset], + &sin_phase_channel[kSimdLength + phase_offset], + &(*canonical_freq_channel)[2 * kSimdLength]); +} + +void FftManager::FreqDomainConvolution(const AudioBuffer::Channel& input_a, + const AudioBuffer::Channel& input_b, + AudioBuffer::Channel* scaled_output) { + DCHECK_EQ(input_a.size(), fft_size_); + DCHECK_EQ(input_b.size(), fft_size_); + DCHECK_EQ(scaled_output->size(), fft_size_); + pffft_zconvolve_accumulate(fft_, input_a.begin(), input_b.begin(), + scaled_output->begin(), inverse_fft_scale_); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager.h new file mode 100644 index 000000000..d7869a3d7 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager.h @@ -0,0 +1,182 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_FFT_MANAGER_H_ +#define RESONANCE_AUDIO_DSP_FFT_MANAGER_H_ + +#include "pffft.h" +#include "base/audio_buffer.h" + +namespace vraudio { + +// This class wraps the pffft library and enables reall FFT transformations to +// be performed on aligned float buffers of data. The class also manages all +// necessary data buffers and zero padding. This class is not thread safe. +class FftManager { + public: + // Minimum required FFT size. + static const size_t kMinFftSize; + + // Constructs a FftManager insatnce. One instance of this class can be shared. + // This class is not thread safe. + // + // @param frames_per_buffer System's number of frames per buffer. + explicit FftManager(size_t frames_per_buffer); + + // Destroys a FftManager instance freeing associated aligned memory. + ~FftManager(); + + // Transforms a single channel of time domain input data into a frequency + // domain representation. + // + // @param time_channel Time domain input. If the length is less than + // |fft_size_|, the input is zeropadded. The max length is |fft_size_|. + // @param freq_channel Frequency domain output, |fft_size| samples long. + void FreqFromTimeDomain(const AudioBuffer::Channel& time_channel, + AudioBuffer::Channel* freq_channel); + + // Transforms a single channel of frequency domain input data into a time + // domain representation. Note: The input must be in pffft format see: + // goo.gl/LYbgX7. This method can output to a buffer of either + // |frames_per_buffer_| or |fft_size_| in length. This feature ensures an + // additional copy is not needed where this method is to be used with an + // overlap add. + // + // @param freq_channel Frequency domain input, |fft_size| samples long. + // @param time_channel Time domain output, |frames_per_buffer_| samples long + // OR |fft_size_| samples long. + void TimeFromFreqDomain(const AudioBuffer::Channel& freq_channel, + AudioBuffer::Channel* time_channel); + + // Applies a 1/|fft_size_| scaling to time domain output. NOTE this need not + // be applied where a convolution is taking place as the scaling will be + // included therein. + // + // @param time_channel Time domain data to be scaled. + void ApplyReverseFftScaling(AudioBuffer::Channel* time_channel); + + // Transforms a pffft frequency domain format buffer into canonical format + // with alternating real and imaginary values with increasing frequency. The + // first two entries of |output| are the real part of the DC and Nyquist + // frequencies (imaginary part is zero). The alternating real and imaginary + // parts start from the third entry in |output|. For more info on the pffft + // format see: goo.gl/LYbgX7 + // + // @param input Frequency domain input channel, |fft_size| samples long. + // @param output Frequency domain output channel,|fft_size| samples long. + void GetCanonicalFormatFreqBuffer(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output); + + // Transforms a canonical frequency domain format buffer into pffft format. + // For more info on the pffft format see: goo.gl/LYbgX7 + // + // @param input Frequency domain input channel, |fft_size| samples long. + // @param output Frequency domain output channel, |fft_size| samples long. + void GetPffftFormatFreqBuffer(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output); + + // Genarates a buffer containing the single sided magnitude spectrum of a + // frequency domain buffer. The input must be in Canonical format. The output + // will have DC frequency as it's first entry and the Nyquist as it's last. + // + // @param freq_channel Canonical format frequency domain buffer, + // |fft_size_| samples long. + // @param magnitude_channel Magnitude of the |freq_channel|. + // |frames_per_buffer_| + 1 samples long. + void MagnitudeFromCanonicalFreqBuffer( + const AudioBuffer::Channel& freq_channel, + AudioBuffer::Channel* magnitude_channel); + + // Combines single sided magnitude and phase spectra into a canonical format + // frequency domain buffer. The inputs must have DC frequency as their first + // entry and the Nyquist as their last. + // + // @param magnitude_channel Magnitude of the |frequency_buffer|. + // |frames_per_buffer_| + 1 samples long. + // @param phase_channel Phase of the |frequency_buffer|. + // |frames_per_buffer_| + 1 samples long. + // @param canonical_freq_channel Canonical format frequency domain buffer, + // |fft_size_| samples long. + void CanonicalFreqBufferFromMagnitudeAndPhase( + const AudioBuffer::Channel& magnitude_channel, + const AudioBuffer::Channel& phase_channel, + AudioBuffer::Channel* canonical_freq_channel); + + // Combines single sided magnitude spectrum and the cosine and sine of a phase + // spectrum into a canonical format frequency domain buffer. The inputs must + // have DC frequency as their first entry and the Nyquist as their last. + // The phase spectra channels can be offset by |phase_offset|. This feature + // is specifically for use as an optimization in the |SpectralReverb|. + // + // @param phase_offset An offset into the channels of the phase buffer. + // @param magnitude_channel Magnitude of the |frequency_buffer|. + // |frames_per_buffer_| + 1 samples long. + // @param sin_phase_channel Sine of the phase of the |frequency_buffer|. + // |frames_per_buffer_| + 1 samples long. + // @param cos_phase_channel Cosine of the phase of the |frequency_buffer|. + // |frames_per_buffer_| + 1 samples long. + // @param canonical_freq_channel Canonical format frequency domain buffer, + // |fft_size_| samples long. + void CanonicalFreqBufferFromMagnitudeAndSinCosPhase( + size_t phase_offset, const AudioBuffer::Channel& magnitude_channel, + const AudioBuffer::Channel& sin_phase_channel, + const AudioBuffer::Channel& cos_phase_channel, + AudioBuffer::Channel* canonical_freq_channel); + + // Performs a pointwise complex multiplication of two frequency domain buffers + // and applies tha inverse scaling factor of 1/|fft_size_|. This operation is + // equivalent to a time domain circular convolution. + // + // @param input_a Frequency domain input channel, |fft_size| samples long. + // @param input_b Frequency domain input channel, |fft_size| samples long. + // @param scaled_output Frequency domain output channel, |fft_size| samples + // long. + void FreqDomainConvolution(const AudioBuffer::Channel& input_a, + const AudioBuffer::Channel& input_b, + AudioBuffer::Channel* scaled_output); + + // Returns the number of points in the FFT. + size_t GetFftSize() const { return fft_size_; } + + private: + // FFT size in samples. + const size_t fft_size_; + + // Number of frames in each buffer of input data. + const size_t frames_per_buffer_; + + // Inverse scale to be applied to buffers transformed from frequency to time + // domain. + const float inverse_fft_scale_; + + // Temporary time domain buffer to store zeropadded input. + AudioBuffer temp_zeropad_buffer_; + + // Temporary freq domain buffer to store. + AudioBuffer temp_freq_buffer_; + + // pffft states. + PFFFT_Setup* fft_; + + // Workspace for pffft. This pointer should be set to null for |fft_size_| + // less than 2^14. In which case the stack is used. This is the recommendation + // by the author of the pffft library. + float* pffft_workspace_ = nullptr; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_FFT_MANAGER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager_test.cc new file mode 100644 index 000000000..e96ed231c --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/fft_manager_test.cc @@ -0,0 +1,260 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/fft_manager.h" + +#include <cstdlib> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/audio_buffer.h" +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/misc_math.h" + +namespace vraudio { + +namespace { + +const float kInverseFftEpsilon = 2e-5f; + +// This tests that the |FreqFromTimeDomain| and |TimeFromFreqDomain| +// functions are the inverse of one another for a number of fft sizes and signal +// types. +TEST(FftManagerTest, FftIfftTest) { + // Generate a test signal. + const size_t kNumSignalTypes = 3; + const size_t kNumBufferLengths = 10; + const size_t kBufferLengths[kNumBufferLengths] = {31, 32, 63, 64, 127, + 128, 255, 256, 511, 512}; + + for (size_t length_idx = 0; length_idx < kNumBufferLengths; length_idx++) { + for (size_t type = 0; type < kNumSignalTypes; ++type) { + AudioBuffer time_signal(kNumMonoChannels, kBufferLengths[length_idx]); + for (size_t i = 0; i < kBufferLengths[length_idx]; ++i) { + switch (type) { + case 0: + time_signal[0][i] = static_cast<float>(i) / + static_cast<float>(kBufferLengths[length_idx]); + break; + case 1: + time_signal[0][i] = std::cos(static_cast<float>(i)); + break; + case 2: + time_signal[0][i] = static_cast<float>(i % 2) * -0.5f; + break; + } + } + AudioBuffer freq_signal(kNumMonoChannels, + NextPowTwo(kBufferLengths[length_idx]) * 2); + AudioBuffer output(kNumMonoChannels, kBufferLengths[length_idx]); + output.Clear(); + + FftManager fft_manager(kBufferLengths[length_idx]); + + fft_manager.FreqFromTimeDomain(time_signal[0], &freq_signal[0]); + fft_manager.TimeFromFreqDomain(freq_signal[0], &output[0]); + fft_manager.ApplyReverseFftScaling(&output[0]); + + for (size_t i = 0; i < kBufferLengths[length_idx]; ++i) { + EXPECT_NEAR(output[0][i], time_signal[0][i], kEpsilonFloat); + } + } + } +} + +// Tests that the result from an inverse FFT is the same whether it is written +// into a buffer of |frames_per_buffer_| or |fft_size_| in length. +TEST(FftManagerTest, ReverseFftOutputSizeTest) { + const size_t kFramesPerBuffer = 32; + AudioBuffer freq_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + freq_buffer.Clear(); + AudioBuffer time_buffer_short(kNumMonoChannels, kFramesPerBuffer); + time_buffer_short.Clear(); + AudioBuffer time_buffer_long(kNumMonoChannels, 2 * kFramesPerBuffer); + time_buffer_long.Clear(); + AudioBuffer input_buffer(kNumMonoChannels, kFramesPerBuffer); + + std::srand(0); + for (auto& sample : input_buffer[0]) { + sample = static_cast<float>(std::rand()) / static_cast<float>(RAND_MAX); + } + + FftManager fft_manager(kFramesPerBuffer); + fft_manager.FreqFromTimeDomain(input_buffer[0], &freq_buffer[0]); + fft_manager.TimeFromFreqDomain(freq_buffer[0], &time_buffer_short[0]); + fft_manager.ApplyReverseFftScaling(&time_buffer_short[0]); + fft_manager.TimeFromFreqDomain(freq_buffer[0], &time_buffer_long[0]); + fft_manager.ApplyReverseFftScaling(&time_buffer_long[0]); + + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR(time_buffer_short[0][i], time_buffer_long[0][i], kEpsilonFloat); + EXPECT_NEAR(time_buffer_long[0][i + kFramesPerBuffer], 0.0f, + kInverseFftEpsilon); + } +} + +// Tests that a frequency domain buffer can be transformed into a canonical +// format and back. +TEST(FftManagerTest, PffftFormatToCanonicalFormatTest) { + const size_t kFramesPerBuffer = 32; + AudioBuffer time_buffer(kNumMonoChannels, kFramesPerBuffer); + time_buffer.Clear(); + time_buffer[0][0] = 1.0f; + time_buffer[0][1] = 1.0f; + AudioBuffer freq_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + freq_buffer.Clear(); + AudioBuffer reordered_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + reordered_buffer.Clear(); + AudioBuffer final_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + reordered_buffer.Clear(); + + FftManager fft_manager(kFramesPerBuffer); + fft_manager.FreqFromTimeDomain(time_buffer[0], &freq_buffer[0]); + + fft_manager.GetCanonicalFormatFreqBuffer(freq_buffer[0], + &reordered_buffer[0]); + fft_manager.GetPffftFormatFreqBuffer(reordered_buffer[0], &final_buffer[0]); + + for (size_t i = 0; i < kFramesPerBuffer * 2; ++i) { + EXPECT_NEAR(final_buffer[0][i], freq_buffer[0][i], kEpsilonFloat); + } +} + +// Tests that for a scaled kronecker delta, the magnitude response will be flat +// and equal to the absolute magnitude of the kronecker. +TEST(FftManagerTest, MagnitudeTest) { + const size_t kFramesPerBuffer = 32; + const size_t kMagnitudeLength = kFramesPerBuffer + 1; + FftManager fft_manager(kFramesPerBuffer); + AudioBuffer time_buffer(kNumMonoChannels, kFramesPerBuffer); + time_buffer.Clear(); + AudioBuffer freq_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + AudioBuffer reordered_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + AudioBuffer magnitude_buffer(kNumMonoChannels, kMagnitudeLength); + const std::vector<float> magnitudes = {1.0f, 2.0f, 3.0f, -1.0f, -2.0f, -3.0f}; + + for (auto& magnitude : magnitudes) { + time_buffer[0][0] = magnitude; + fft_manager.FreqFromTimeDomain(time_buffer[0], &freq_buffer[0]); + fft_manager.GetCanonicalFormatFreqBuffer(freq_buffer[0], + &reordered_buffer[0]); + fft_manager.MagnitudeFromCanonicalFreqBuffer(reordered_buffer[0], + &magnitude_buffer[0]); + for (size_t sample = 0; sample < kMagnitudeLength; ++sample) { + // Check its correct to within 0.5%. + const float kErrEpsilon = 5e-3f; + const float expected = std::abs(magnitude); + EXPECT_NEAR(magnitude_buffer[0][sample], expected, + kErrEpsilon * expected); + } + } +} + +// Tests that conversion from Canonical frequency domain data to phase and +// magnitude spectra and back results in an output equal to the input. +TEST(FftManagerTest, FreqFromMagnitudePhase) { + const size_t kFramesPerBuffer = 16; + const size_t kMagnitudePhaseLength = kFramesPerBuffer + 1; + FftManager fft_manager(kFramesPerBuffer); + AudioBuffer time_buffer(kNumMonoChannels, kFramesPerBuffer); + time_buffer.Clear(); + time_buffer[0][0] = 0.5f; + time_buffer[0][1] = 1.0f; + AudioBuffer freq_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + AudioBuffer reordered_buffer(kNumMonoChannels, 2 * kFramesPerBuffer); + AudioBuffer phase_buffer(kNumMonoChannels, kMagnitudePhaseLength); + AudioBuffer magnitude_buffer(kNumMonoChannels, kMagnitudePhaseLength); + fft_manager.FreqFromTimeDomain(time_buffer[0], &freq_buffer[0]); + fft_manager.GetCanonicalFormatFreqBuffer(freq_buffer[0], + &reordered_buffer[0]); + fft_manager.MagnitudeFromCanonicalFreqBuffer(reordered_buffer[0], + &magnitude_buffer[0]); + + // Calculate the phase. + phase_buffer[0][0] = 0.0f; + for (size_t i = 1, j = 2; i < kFramesPerBuffer; ++i, j += 2) { + phase_buffer[0][i] = std::atan2(reordered_buffer[0][j + 1] /*imag*/, + reordered_buffer[0][j] /*real*/); + } + phase_buffer[0][kFramesPerBuffer] = kPi; + + fft_manager.CanonicalFreqBufferFromMagnitudeAndPhase( + magnitude_buffer[0], phase_buffer[0], &freq_buffer[0]); + + for (size_t sample = 0; sample < kFramesPerBuffer * 2; ++sample) { + // Check its correct to within 0.5%. + const float kErrEpsilon = 5e-3f; + EXPECT_NEAR(freq_buffer[0][sample], reordered_buffer[0][sample], + kErrEpsilon * std::abs(reordered_buffer[0][sample])); + } +} + +// Tests that conversion from phase and magnitude spectra and back results in +// an output equal to that from sine and cosine phase, using SIMD on arm. +TEST(FftManagerTest, FMagnitudePhaseAndSineCosinePhase) { + const size_t kFramesPerBuffer = 16; + const size_t kMagnitudePhaseLength = kFramesPerBuffer + 1; + FftManager fft_manager(kFramesPerBuffer); + AudioBuffer freq_buffer_one(kNumMonoChannels, 2 * kFramesPerBuffer); + AudioBuffer freq_buffer_two(kNumMonoChannels, 2 * kFramesPerBuffer); + AudioBuffer phase_buffer(kNumMonoChannels, kMagnitudePhaseLength); + AudioBuffer sin_phase_buffer(kNumMonoChannels, kMagnitudePhaseLength); + AudioBuffer cos_phase_buffer(kNumMonoChannels, kMagnitudePhaseLength); + AudioBuffer magnitude_buffer(kNumMonoChannels, kMagnitudePhaseLength); + + std::fill(magnitude_buffer[0].begin(), magnitude_buffer[0].end(), 2.0f); + phase_buffer[0] = std::vector<float>( + {0.4720f, 1.6100f, -1.9831f, 0.7569f, 0.2799f, -1.1481f, -0.3807f, + 0.3008f, 3.1416f, 2.4314f, -1.1851f, 2.6645f, 0.6369f, -0.0554f, 0.6275f, + -0.1799f, 1.2345f}); + for (size_t i = 0; i < phase_buffer.num_frames(); ++i) { + sin_phase_buffer[0][i] = std::sin(phase_buffer[0][i]); + cos_phase_buffer[0][i] = std::cos(phase_buffer[0][i]); + } + + fft_manager.CanonicalFreqBufferFromMagnitudeAndPhase( + magnitude_buffer[0], phase_buffer[0], &freq_buffer_one[0]); + fft_manager.CanonicalFreqBufferFromMagnitudeAndSinCosPhase( + 0, /* phase_offset */ + magnitude_buffer[0], sin_phase_buffer[0], cos_phase_buffer[0], + &freq_buffer_two[0]); + + for (size_t i = 0; i < kFramesPerBuffer * 2; ++i) { + EXPECT_NEAR(freq_buffer_one[0][i], freq_buffer_two[0][i], kEpsilonFloat); + } +} + +// Tests that the correct scaling factor is applied consistently across a time +// domain buffer. +TEST(FftManagerTest, ReverseScalingTest) { + const size_t kFramesPerBuffer = 128; + const float kExpectedScale = 1.0f / static_cast<float>(2 * kFramesPerBuffer); + + AudioBuffer buffer(kNumMonoChannels, kFramesPerBuffer); + for (auto& sample : buffer[0]) { + sample = 1.0f; + } + FftManager fft_manager(kFramesPerBuffer); + + fft_manager.ApplyReverseFftScaling(&buffer[0]); + for (auto& sample : buffer[0]) { + EXPECT_NEAR(sample, kExpectedScale, kEpsilonFloat); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators.cc new file mode 100644 index 000000000..11d19694d --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators.cc @@ -0,0 +1,165 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/filter_coefficient_generators.h" + +#include <algorithm> +#include <cmath> + +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +// Hard coded 4th order fit calculated from: + +const int kPolynomialOrder = 4; +const float kPolynomialCoefficients[] = { + 2.52730826376129e-06f, 0.00018737263228963f, 0.00618822707412605f, + 0.113947188715447f, 0.999048314740445f}; + +// Threshold frequency for human hearing 20Hz. +const float kTwentyHz = 20.0f; + +// ln(2)/2 for use in bandpass filter. +const float kLn2over2 = 0.346573590279973f; + +} // namespace + +BiquadCoefficients ComputeBandPassBiquadCoefficients(int sample_rate, + float center_frequency, + int bandwidth) { + DCHECK_GT(sample_rate, 0); + DCHECK_GE(center_frequency, 0.0f); + DCHECK_GT(bandwidth, 0); + // Check if an invalid |center_frequency|, greater than or equal to the + // Nyquist rate is passed as input. + CHECK_LT(center_frequency, 0.5f * static_cast<float>(sample_rate)); + + // Frequency of interest (in radians). + const float w0 = kTwoPi * center_frequency / static_cast<float>(sample_rate); + // Intermediate storage. + const float cos_w0 = std::cos(w0); + const float sin_w0 = std::sin(w0); + const float alpha = + sin_w0 * sinhf(kLn2over2 * static_cast<float>(bandwidth) * w0 / sin_w0); + + // The BiquadFilterState constructor is passed (a0, a1, a2, b0, b1, b2). + return BiquadCoefficients(1.0f + alpha, -2.0f * cos_w0, 1.0f - alpha, alpha, + 0.0f, -alpha); +} + +void ComputeDualBandBiquadCoefficients( + int sample_rate, float crossover_frequency, + BiquadCoefficients* low_pass_coefficients, + BiquadCoefficients* high_pass_coefficients) { + DCHECK_GT(sample_rate, 0); + DCHECK_GE(crossover_frequency, 0.0f); + DCHECK_LE(crossover_frequency, static_cast<float>(sample_rate) / 2.0f); + DCHECK(low_pass_coefficients); + DCHECK(high_pass_coefficients); + + const float k = std::tan(static_cast<float>(M_PI) * crossover_frequency / + static_cast<float>(sample_rate)); + const float k_squared = k * k; + const float denominator = k_squared + 2.0f * k + 1.0f; + + // |denominator| must not be near 0. Since |k| is always guaranteed to be in + // the range 0 < k < pi/2, the |denominator| should always be >=1. This is a + // sanity check only. + DCHECK_GT(denominator, kEpsilonDouble); + + // Computes numerator coefficients of the low-pass |low_pass_coefficients| + // bi-quad. + low_pass_coefficients->a[0] = 1.0f; + low_pass_coefficients->a[1] = 2.0f * (k_squared - 1.0f) / denominator; + low_pass_coefficients->a[2] = (k_squared - 2.0f * k + 1.0f) / denominator; + + // Numerator coefficients of the high-pass |high_pass_coefficients| bi-quad + // are the same. + BiquadCoefficients high_pass; + std::copy(low_pass_coefficients->a.begin(), low_pass_coefficients->a.end(), + high_pass_coefficients->a.begin()); + + // Computes denominator coefficients of the low-pass |low_pass_coefficients| + // bi-quad. + low_pass_coefficients->b[0] = k_squared / denominator; + low_pass_coefficients->b[1] = 2.0f * low_pass_coefficients->b[0]; + low_pass_coefficients->b[2] = low_pass_coefficients->b[0]; + + // Computes denominator coefficients of the high-pass |high_pass_coefficients| + // bi-quad. + high_pass_coefficients->b[0] = 1.0f / denominator; + high_pass_coefficients->b[1] = -2.0f * high_pass_coefficients->b[0]; + high_pass_coefficients->b[2] = high_pass_coefficients->b[0]; +} + +BiquadCoefficients ComputeLowPassBiquadCoefficients( + int sample_rate, float specification_frequency, float attenuation) { + DCHECK_GT(sample_rate, 0); + DCHECK_GE(specification_frequency, 0.0f); + DCHECK_LE(specification_frequency, static_cast<float>(sample_rate) / 2.0f); + DCHECK_LT(attenuation, 0.0f); + + // Frequency of interest (in radians). + const float w0 = + kTwoPi * specification_frequency / static_cast<float>(sample_rate); + + // Q is the Q-factor. For more information see "Digital Signal Processing" - + // J. G. Prolakis, D. G. Manolakis - Published by Pearson. + float Q = 0.0f; + + // Variable to handle the growth in power as one extra mult per iteration + // across the polynomial coefficients. + float attenuation_to_a_power = 1.0f; + + // Add in each term in reverse order. + for (int order = kPolynomialOrder; order >= 0; --order) { + Q += kPolynomialCoefficients[order] * attenuation_to_a_power; + attenuation_to_a_power *= attenuation; + } + + // Intermediate storage of commonly used values. + const float alpha = std::sin(w0) / (2.0f * Q); + const float cos_w0 = std::cos(w0); + + // Filter coefficients. + float a0 = 1.0f + alpha; + float a1 = -2.0f * cos_w0; + float a2 = 1.0f - alpha; + // Coefficients b0 and b2 will have the same value in this case. + float b0_b2 = (1.0f - cos_w0) / 2.0f; + float b1 = 1.0f - cos_w0; + + return BiquadCoefficients(a0, a1, a2, b0_b2, b1, b0_b2); +} + +float ComputeLowPassMonoPoleCoefficient(float cuttoff_frequency, + int sample_rate) { + float coefficient = 0.0f; + // Check that the cuttoff_frequency provided is not too low (below human + // threshold, also danger of filter instability). + if (cuttoff_frequency > kTwentyHz) { + const float inverse_time_constant = kTwoPi * cuttoff_frequency; + const float sample_rate_float = static_cast<float>(sample_rate); + coefficient = + sample_rate_float / (inverse_time_constant + sample_rate_float); + } + return coefficient; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators.h new file mode 100644 index 000000000..4c7659a7a --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators.h @@ -0,0 +1,177 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_FILTER_COEFFICIENT_GENERATORS_H_ +#define RESONANCE_AUDIO_DSP_FILTER_COEFFICIENT_GENERATORS_H_ + +#include "dsp/biquad_filter.h" + +// Functions for the generation of filter coefficients for various common tasks. +// Currently supports the following filter types: +// Low pass first-order filter. +// Low pass biquad filter. +// Band pass biquad filter. +// Dual band, matched phase biquad shelf filters. +namespace vraudio { + +// Computes corresponding biquad coefficients for band pass filter with respect +// to the centre frequency and the bandwidth between -3 dB frequencies. +// +// b0 + b1*z^-1 + b2*z^-2 +// H(z) = ------------------------ +// a0 + a1*z^-1 + a2*z^-2 +// +// where: +// w0 = 2*pi*center_frequency/sample_rate +// +// alpha = sin(w0)*sinh( ln(2)/2 * bandwidth * w0/sin(w0) ) +// +// b0 = alpha +// b1 = 0 +// b2 = -alpha +// a0 = 1 + alpha +// a1 = -2*cos(w0) +// a2 = 1 - alpha +// +// @param sample_rate Sampling rate in Hz. +// @param centre_frequency Centre frequency of passband. +// @param bandwidth Bandwidth in octaves between -3 dB frequencies. +// @return Output structure of band-pass BiquadCoefficients. +BiquadCoefficients ComputeBandPassBiquadCoefficients(int sample_rate, + float centre_frequency, + int bandwidth); + +// Computes two sets of transfer function coefficients to be used with a +// pair of generic bi-quad filters. The coefficients are used to implement a +// phase-matched low-pass |low_pass_state| and high-pass |high_pass_state| +// filter pair with a cross-over frequency defined as |crossover_frequency|. +// +// Implementation of the matched bi-quad filter pair as described in: +// http://www.ai.sri.com/ajh/ambisonics/BLaH3.pdf +// +// b0 + b1*z^-1 + b2*z^-2 +// H(z) = ------------------------ +// a0 + a1*z^-1 + a2*z^-2 +// +// where: +// +// a0 = 1 +// +// 2(k^2 − 1) +// a1 = -------------- +// k^2 + 2k + 1 +// +// k^2 - 2k + 1 +// a2 = -------------- +// k^2 + 2k + 1 +// +// low-pass: high-pass: +// +// k^2 1 +// b0 = ------------- b0 = -------------- +// k^2 + 2k + 1 k^2 + 2k + 1 +// +// b1 = 2b0 b1 = -2b0 +// +// b2 = b0 b2 = b0 +// +// and +// +// pi * crossover_frequency +// k = tan -------------------------- +// sample_frequency +// +// @param sample_rate Sampling rate in [Hz] +// @param crossover_frequency Cross-over frequency in [Hz] +// @param low_pass_coefficients Output structure of low-pass bi-quad +// coefficients +// @param high_pass_coefficients Output structure of high-pass bi-quad +// coefficients. +void ComputeDualBandBiquadCoefficients( + int sample_rate, float crossover_frequency, + BiquadCoefficients* low_pass_coefficients, + BiquadCoefficients* high_pass_coefficients); + +// Computes biquad coefficients for low pass filter with respect to the +// specification frequency and the attenuation value at that frequency. +// +// b0 + b1*z^-1 + b2*z^-2 +// H(z) = ------------------------ +// a0 + a1*z^-1 + a2*z^-2 +// +// where: +// Q = 2.5273e-06*attenuation^4 + 0.00018737*attenuation^3 + +// 0.0061882*attenuation^2 + 0.11395*attenuation + 0.99905 +// +// w0 = 2*pi*specification_frequency/sample_rate; +// +// alpha = sin(w0)/(2*Q); +// +// a0 = 1 + alpha; +// a1 = -2*cos(w0); +// a2 = 1 - alpha; +// b0 = (1 - cos(w0))/2; +// b1 = 1 - cos(w0); +// b2 = (1 - cos(w0)/2; +// +// These coefficients were generated after a bilinerar transform of an +// analogue filter H(s) = 1 / (s^2 + s/Q + 1), from: +// www.analog.com/library/analogdialogue/archives/43-09/edch%208%20filter.pdf. +// +// Please note Q has been calculated by fitting a 4th order polynomial to the +// average of Q vs attenuation curves at different f0's from 10kHz to 19kHz. +// Please note that at differing frequencies these graphs had identical shape +// and differed only by an overall offset of at most 0.1 (attenuation) dB from +// the mean. This script can be found at + +// +// @param sample_rate Sampling rate in Hz. +// @param specification_frequency Frequency at which attenuation applies in Hz. +// @param attenuation Attenuation at specification_frequency in negative dB. +// @return low_pass Output structure of low-pass BiquadCoefficients. +BiquadCoefficients ComputeLowPassBiquadCoefficients( + int sample_rate, float specification_frequency, float attenuation); + +// Generates a coefficient for the |MonoPoleFilterClass| based on a 3dB +// bandwidth. +// +// The Laplace transfer function of a first order low pass system is: +// +// 1 +// ------------ where tau is the RC time constant of the system. +// 1 + tau * s +// +// For a discrete moving average filter with input x[n] and output y[n], +// the difference equation is: +// y[n] = a * y[n - 1] + (1 - a) * x[n] +// tau +// a = ---------- where T is the sample period. +// tau + T +// since the -3dB bandwith of a first order system can be +// 1 related to its time constant by +// f3 = -------------- +// 2 * pi * tau we can obtain 'a' from cuttoff_frequency and +// sample rate. +// +// @param cuttoff_frequency The -3dB frequency in Hz of the low pass lobe. +// @param sample_rate System sampling rate. +// @return A |MonoPoleFilterClass| coefficient for the specified bandwidth. +float ComputeLowPassMonoPoleCoefficient(float cuttoff_frequency, + int sample_rate); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_FILTER_COEFFICIENT_GENERATORS_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators_test.cc new file mode 100644 index 000000000..163b6a6d2 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/filter_coefficient_generators_test.cc @@ -0,0 +1,146 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/filter_coefficient_generators.h" + +#include "third_party/googletest/googletest/include/gtest/gtest.h" + +namespace vraudio { + +namespace { + +// Allowable error between filter coefficients. +const float kErrorf = 1e-5f; + +// Input into the ComputeLowPassBiquadCoefficients() function. +const int kSampleRate = 44100; +const float kSpecificationFrequency = 13000.0f; +const float kAttenuation = -15.0f; + +const float kMonoPoleCoefficient = 0.94863985f; + +// These filter coefficients were calculated in MATLAB with a specified Q value +// of 0.17775. The attenuation at the specification frequency was then +// calculated from the MATLAB freqz() calls. +const BiquadCoefficients kIdealCoefficients(3.70224817628938f, + 0.555382147099001f, + -1.70224817628938f, + 0.63884553677475f, 1.2776910735495f, + 0.63884553677475f); + +// These filter coefficients were calculated in MATLAB for a centre_frequency of +// 8kHz and a bandwidth of 1 octave. +const BiquadCoefficients kIdealBandpassCoefficients(1.37364799922092f, -1.0f, + 0.626352000779082f, + 0.373647999220918f, 0.0f, + -0.373647999220918f); + +const float kBandPassCenterFrequency = 8000.0f; +const int kBandPassOctaveBandwidth = 1; + +// Input into the ComputeDualBandBiquadCoefficients() function. +const int kSampleRate48 = 48000; +const float kCrossoverFrequency = 380.0f; + +// Pre-computed coefficients for the phase-matched filter pair (dual-band) with +// the cross-over frequency set at 380Hz and the sample rate 48kHz. +const float kLowPassA[] = {1.0f, -1.9029109f, 0.90526748f}; +const float kLowPassB[] = {0.00058914319f, 0.0011782864f, 0.00058914319f}; +const float kHighPassA[] = {1.0f, -1.9029109f, 0.90526748f}; +const float kHighPassB[] = {0.95204461f, -1.9040892f, 0.95204461f}; + +TEST(FilterCoefficientGeneratorsTest, ComputeMonoPoleLowpassCoefficientTest) { + float coefficient = + ComputeLowPassMonoPoleCoefficient(kCrossoverFrequency, kSampleRate); + EXPECT_NEAR(kMonoPoleCoefficient, coefficient, kErrorf); + + const float twenty_hertz = 20.0f; + coefficient = ComputeLowPassMonoPoleCoefficient(twenty_hertz, kSampleRate); + EXPECT_EQ(0.0f, coefficient); + // New test for the case where a frequency below 20Hz is passed. +} + +// Tests that the BiquadCoefficients generated by +// ComputeBandPassBiquadCoefficients() match those known to be correct as +// directly calculated from MATLAB. +TEST(FilterCoefficientGeneratorsTest, ComputeBandPassBiquadCoefficientsTest) { + // Perform computation. + BiquadCoefficients biquad_filter_coefficients = + ComputeBandPassBiquadCoefficients(kSampleRate48, kBandPassCenterFrequency, + kBandPassOctaveBandwidth); + + // Make sure they are all equal. + for (size_t i = 0; i < biquad_filter_coefficients.a.size(); ++i) { + EXPECT_NEAR(kIdealBandpassCoefficients.a[i], + biquad_filter_coefficients.a[i], kErrorf); + EXPECT_NEAR(kIdealBandpassCoefficients.b[i], + biquad_filter_coefficients.b[i], kErrorf); + } +} + +// Tests that the BiquadCoefficients generated by +// ComputeLowPassBiquadCoefficients() match those known to be correct as +// directly calculated from MATLAB. This also tests the validity of the inherent +// 4th order best fit between attenuation at a specified frequency and Q factor. +TEST(FilterCoefficientGeneratorsTest, ComputeLowPassBiquadCoefficientsTest) { + // Perform computation. + BiquadCoefficients biquad_filter_coefficients = + ComputeLowPassBiquadCoefficients(kSampleRate, kSpecificationFrequency, + kAttenuation); + + // Make sure they are all equal. + for (size_t i = 0; i < biquad_filter_coefficients.a.size(); ++i) { + EXPECT_NEAR(kIdealCoefficients.a[i], biquad_filter_coefficients.a[i], + kErrorf); + EXPECT_NEAR(kIdealCoefficients.b[i], biquad_filter_coefficients.b[i], + kErrorf); + } +} + +// Tests that the BiquadCoefficients generated by +// ComputeDualBandBiquadCoefficients() match those known to be correct as +// directly calculated from MATLAB. These coefficients are as described in: +// http://www.ai.sri.com/ajh/ambisonics/BLaH3.pdf. +TEST(DualBandBiquadCoefficientsTest, ComputeDualBandBiquadCoefficientsTest) { + // Generate default bi-quad coefficients and constructs low- and high-pass + // filter coefficients. + const BiquadCoefficients kDefaultCoefficients; + BiquadCoefficients low_pass_coefficients(kDefaultCoefficients); + BiquadCoefficients high_pass_coefficients(kDefaultCoefficients); + + // Perform computation. + ComputeDualBandBiquadCoefficients(kSampleRate48, kCrossoverFrequency, + &low_pass_coefficients, + &high_pass_coefficients); + + // Check if arrays with a and b coefficients for both filters are of the + // same size. + ASSERT_EQ(low_pass_coefficients.a.size(), low_pass_coefficients.b.size()); + ASSERT_EQ(high_pass_coefficients.a.size(), low_pass_coefficients.a.size()); + ASSERT_EQ(high_pass_coefficients.b.size(), low_pass_coefficients.b.size()); + + // Make sure they are all equal. + for (size_t i = 0; i < low_pass_coefficients.a.size(); ++i) { + EXPECT_NEAR(low_pass_coefficients.a[i], kLowPassA[i], kErrorf); + EXPECT_NEAR(low_pass_coefficients.b[i], kLowPassB[i], kErrorf); + EXPECT_NEAR(high_pass_coefficients.a[i], kHighPassA[i], kErrorf); + EXPECT_NEAR(high_pass_coefficients.b[i], kHighPassB[i], kErrorf); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter.cc new file mode 100644 index 000000000..e77ab71ce --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter.cc @@ -0,0 +1,148 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +// Prevent Visual Studio from complaining about std::copy_n. +#if defined(_WIN32) +#define _SCL_SECURE_NO_WARNINGS +#endif + +#include "dsp/fir_filter.h" + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/simd_macros.h" + +namespace vraudio { + +FirFilter::FirFilter(const AudioBuffer::Channel& filter_coefficients, + size_t frames_per_buffer) { + DCHECK_GE(filter_coefficients.size(), 1U); + // Create the kernel buffer in repeated entry representation from standard + // form FIR representation. + const size_t coefficients_length = filter_coefficients.size(); + filter_length_ = coefficients_length + + ((coefficients_length % SIMD_LENGTH == 0) + ? 0 + : SIMD_LENGTH - (coefficients_length % SIMD_LENGTH)); + num_filter_chunks_ = filter_length_ / SIMD_LENGTH; + DCHECK_EQ(filter_length_ % SIMD_LENGTH, 0); + coefficients_ = AudioBuffer(kNumMonoChannels, filter_length_ * SIMD_LENGTH); + AudioBuffer::Channel* coefficients = &(coefficients_[0]); + // Store the coefficients so that each individual coefficient is repeated + // SIMD_LENGTH times. + for (size_t coeff = 0; coeff < coefficients_length; ++coeff) { + auto coefficient_iter = coefficients->begin() + (coeff * SIMD_LENGTH); + std::fill(coefficient_iter, coefficient_iter + SIMD_LENGTH, + filter_coefficients[coeff]); + } + std::fill(coefficients->begin() + (coefficients_length * SIMD_LENGTH), + coefficients->end(), 0.0f); + // Allocate an aligned buffer with |filter_length_| extra samples to + // store previous input samples. + state_ = AudioBuffer(kNumMonoChannels, frames_per_buffer + filter_length_); + state_.Clear(); +} + +void FirFilter::Process(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output) { + DCHECK(output); + DCHECK_EQ(input.size(), state_.num_frames() - filter_length_); + // In this case we know that SIMD_LENGTH == 1, therefore we don't need to take + // account of it. + const size_t input_length = input.size(); + std::copy_n(state_[0].end() - filter_length_, filter_length_, + state_[0].begin()); + std::copy_n(input.begin(), input_length, state_[0].begin() + filter_length_); +#if defined(SIMD_DISABLED) + AudioBuffer::Channel* coefficients = &(coefficients_[0]); + AudioBuffer::Channel* data = &(state_[0]); + for (size_t frame = 0; frame < input_length; ++frame) { + for (size_t coeff = 0; coeff < filter_length_; ++coeff) { + (*output)[frame] += + (*coefficients)[coeff] * (*data)[filter_length_ - coeff]; + } + } +#else + const SimdVector* cptr_input = + reinterpret_cast<const SimdVector*>(&(state_[0][0])); + const SimdVector* cptr_filter = + reinterpret_cast<const SimdVector*>(&(coefficients_[0][0])); + SimdVector* ptr_output = reinterpret_cast<SimdVector*>(&((*output)[0])); + + const size_t num_input_chunks = input_length / SIMD_LENGTH; + DCHECK_EQ(input_length % SIMD_LENGTH, 0); + + // A pair of |SimdVector|s one for holding the input for each operation, the + // other to store data that needed to be copied as it straddled a four float + // boundary. + SimdVector input_use; + SimdVector input_split; + for (size_t input_position = num_filter_chunks_; + input_position < num_input_chunks + num_filter_chunks_; + ++input_position) { + // Replace these with the indexed pointers to save on copies. + SimdVector* output_now = &ptr_output[input_position - num_filter_chunks_]; + DCHECK_GE(input_position, num_filter_chunks_); + + for (size_t chunk = 0; chunk < num_filter_chunks_; ++chunk) { + const size_t filter_index_offset = chunk * SIMD_LENGTH; + const SimdVector* input_a = &cptr_input[input_position - (chunk + 1)]; + const SimdVector* input_b = &cptr_input[input_position - chunk]; +#if defined(SIMD_SSE) + // Select four floats from input_a and input_b, based on the mask. Here we + // take the latter two entries from input_b followed by the first two + // entries of input_a. + input_split = _mm_shuffle_ps(*input_a, *input_b, _MM_SHUFFLE(1, 0, 3, 2)); + + // All from input_b. + *output_now = SIMD_MULTIPLY_ADD( + *input_b, cptr_filter[filter_index_offset], *output_now); + // One from input_a and the rest (three) from input_b. + input_use = + _mm_shuffle_ps(input_split, *input_b, _MM_SHUFFLE(2, 1, 2, 1)); + *output_now = SIMD_MULTIPLY_ADD( + input_use, cptr_filter[1 + filter_index_offset], *output_now); + // Two from input_a and two from input_b. + *output_now = SIMD_MULTIPLY_ADD( + input_split, cptr_filter[2 + filter_index_offset], *output_now); + // Three from input_a and one from input_b. + input_use = + _mm_shuffle_ps(*input_a, input_split, _MM_SHUFFLE(2, 1, 2, 1)); + *output_now = SIMD_MULTIPLY_ADD( + input_use, cptr_filter[3 + filter_index_offset], *output_now); +#elif defined(SIMD_NEON) + // All from input_b. + *output_now = SIMD_MULTIPLY_ADD( + *input_b, cptr_filter[filter_index_offset], *output_now); + // One from input_a and the rest (three) from input_b. + input_use = vextq_f32(*input_a, *input_b, 3); + *output_now = SIMD_MULTIPLY_ADD( + input_use, cptr_filter[1 + filter_index_offset], *output_now); + // Two from input_a and two from input_b. + input_use = vextq_f32(*input_a, *input_b, 2); + *output_now = SIMD_MULTIPLY_ADD( + input_use, cptr_filter[2 + filter_index_offset], *output_now); + // Three from input_a and one from input_b. + input_use = vextq_f32(*input_a, *input_b, 1); + *output_now = SIMD_MULTIPLY_ADD( + input_use, cptr_filter[3 + filter_index_offset], *output_now); +#endif // SIMD_SSE/SIMD_NEON + } + } +#endif // SIMD_DISABLED +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter.h new file mode 100644 index 000000000..46c1edde5 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter.h @@ -0,0 +1,62 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_FIR_FILTER_H_ +#define RESONANCE_AUDIO_DSP_FIR_FILTER_H_ + +#include "base/audio_buffer.h" + +namespace vraudio { + +class FirFilter { + public: + // Constructs a |FirFilter| from a mono |AudioBuffer| of filter coefficients. + // + // @param filter_coefficients FIR filter coefficients. + // @param frames_per_buffer Number of frames of data in each audio buffer. + FirFilter(const AudioBuffer::Channel& filter_coefficients, + size_t frames_per_buffer); + + // Filters an array of input with a finite impulse response (FIR) filter in + // the time domain. All pointers and lengths passed must be SIMD compatible. + // + // @param input Pointer to an aray of input. + // @param output Pointer to a cleared block of memory of the input length. + void Process(const AudioBuffer::Channel& input, AudioBuffer::Channel* output); + + // Returns the length of the filter kernel after zeropadding. + // + // @return Length of the FIR filter in frames. + size_t filter_length() const { return filter_length_; } + + private: + // Length of the filter kernel in frames after zeropadding. + size_t filter_length_; + + // Number of filter chunks equivalent to the filter length after zeropadding + // divided by the SIMD_LENGTH. + size_t num_filter_chunks_; + + // Coefficients of the filter stored in a repeated format. + AudioBuffer coefficients_; + + // Aligned buffer of previous and current input. + AudioBuffer state_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_FIR_FILTER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter_test.cc new file mode 100644 index 000000000..e64a44f15 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/fir_filter_test.cc @@ -0,0 +1,104 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/fir_filter.h" + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/audio_buffer.h" +#include "base/constants_and_types.h" +#include "base/simd_macros.h" + +namespace vraudio { + +namespace { + +const size_t kFirFramesPerBuffer = 32; +const size_t kFirKernelLength = 8; + +// Tests that the filter is resized to a length compatable with the current +// SIMD_LENGTH whether it is one greater than or one less than a multiple of +// SIMD_LENGTH to begin with. +TEST(FirFilterTest, GetFilterLengthTest) { + const size_t twice_simd_length = (SIMD_LENGTH * 2); + // Divisible by the length of a SIMD vector. + AudioBuffer kernel_buffer1(kNumMonoChannels, twice_simd_length); + FirFilter filter1(kernel_buffer1[0], twice_simd_length); + EXPECT_EQ(twice_simd_length, filter1.filter_length()); + + // Shorter filter. + AudioBuffer kernel_buffer2(kNumMonoChannels, twice_simd_length - 1); + FirFilter filter2(kernel_buffer2[0], twice_simd_length - 1); + EXPECT_EQ(twice_simd_length, filter2.filter_length()); + + // Longer filter. + AudioBuffer kernel_buffer3(kNumMonoChannels, twice_simd_length + 1); + FirFilter filter3(kernel_buffer3[0], twice_simd_length + 1); + EXPECT_EQ(twice_simd_length + SIMD_LENGTH, filter3.filter_length()); +} + +// Tests that the output from a convolution between a double kronecker input and +// a alternating 1, 0 buffer of input yields the correct output buffer. +TEST(FirFilterTest, ProcessWithDoubleKronecker) { + AudioBuffer input_buffer(kNumMonoChannels, kFirFramesPerBuffer); + input_buffer.Clear(); + AudioBuffer output_buffer(1, kFirFramesPerBuffer); + output_buffer.Clear(); + // First create a vector containing the FIR filter coefficients in standard + // format. + AudioBuffer kernel_buffer(kNumMonoChannels, kFirKernelLength); + kernel_buffer.Clear(); + kernel_buffer[0][0] = 1.0f; + kernel_buffer[0][kFirKernelLength / 2] = 1.0f; + // Now we can create the kernel buffer in its repeated entry representation + // from this standard form FIR representation. + FirFilter filter(kernel_buffer[0], kFirFramesPerBuffer); + // Next populate the input buffer. + for (size_t i = 0; i < kFirFramesPerBuffer; i += 2) { + input_buffer[0][i] = 1.0f; + } + + filter.Process(input_buffer[0], &(output_buffer[0])); + + for (size_t i = 0; i < kFirFramesPerBuffer; ++i) { + if (i % 2 != 0) { + EXPECT_NEAR(0.0f, output_buffer[0][i], kEpsilonFloat); + } else if (i <= 3) { + EXPECT_NEAR(1.0f, output_buffer[0][i], kEpsilonFloat); + } else { + EXPECT_NEAR(2.0f, output_buffer[0][i], kEpsilonFloat); + } + } + + // Run again with a cleared buffer to flush out the filter state. + input_buffer.Clear(); + output_buffer.Clear(); + + filter.Process(input_buffer[0], &(output_buffer[0])); + + for (size_t i = 0; i < kFirFramesPerBuffer; ++i) { + if (i % 2 != 0) { + EXPECT_NEAR(0.0f, output_buffer[0][i], kEpsilonFloat); + } else if (i <= 3) { + EXPECT_NEAR(1.0f, output_buffer[0][i], kEpsilonFloat); + } else { + EXPECT_NEAR(0.0f, output_buffer[0][i], kEpsilonFloat); + } + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain.cc new file mode 100644 index 000000000..54e3fc216 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain.cc @@ -0,0 +1,102 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/gain.h" + +#include "base/constants_and_types.h" +#include "base/simd_macros.h" +#include "base/simd_utils.h" + +namespace vraudio { + +float LinearGainRamp(size_t ramp_length, float start_gain, float end_gain, + const AudioBuffer::Channel& input_samples, + AudioBuffer::Channel* output_samples, + bool accumulate_output) { + DCHECK(output_samples); + DCHECK_EQ(input_samples.size(), output_samples->size()); + DCHECK_GT(ramp_length, 0U); + + const size_t process_length = std::min(ramp_length, input_samples.size()); + const float gain_increment_per_sample = + (end_gain - start_gain) / static_cast<float>(ramp_length); + + float current_gain = start_gain; + if (accumulate_output) { + for (size_t frame = 0; frame < process_length; ++frame) { + (*output_samples)[frame] += current_gain * input_samples[frame]; + current_gain += gain_increment_per_sample; + } + } else { + for (size_t frame = 0; frame < process_length; ++frame) { + (*output_samples)[frame] = current_gain * input_samples[frame]; + current_gain += gain_increment_per_sample; + } + } + + return current_gain; +} + +void ConstantGain(size_t offset_index, float gain, + const AudioBuffer::Channel& input_samples, + AudioBuffer::Channel* output_samples, + bool accumulate_output) { + DCHECK(output_samples); + const size_t input_size = input_samples.size(); + DCHECK_EQ(input_size, output_samples->size()); + DCHECK_LT(offset_index, input_size); + + // Apply gain to samples at the beginning, prior to SIMD_LENGTH alignment. + const size_t unaligned_samples = SIMD_LENGTH - (offset_index % SIMD_LENGTH); + const size_t offset_index_simd = + std::min(input_size, offset_index + unaligned_samples); + if (accumulate_output) { + for (size_t i = offset_index; i < offset_index_simd; ++i) { + (*output_samples)[i] += input_samples[i] * gain; + } + } else { + for (size_t i = offset_index; i < offset_index_simd; ++i) { + (*output_samples)[i] = input_samples[i] * gain; + } + } + + if (offset_index_simd == input_size) { + // Return if there are no remaining operations to carry out. + return; + } + + const size_t aligned_length = input_size - offset_index_simd; + const float* aligned_input = &(input_samples[offset_index_simd]); + float* aligned_output = &(*output_samples)[offset_index_simd]; + + // Apply gain via SIMD operations. + if (accumulate_output) { + ScalarMultiplyAndAccumulate(aligned_length, gain, aligned_input, + aligned_output); + } else { + ScalarMultiply(aligned_length, gain, aligned_input, aligned_output); + } +} + +bool IsGainNearZero(float gain) { + return std::abs(gain) < kNegative60dbInAmplitude; +} + +bool IsGainNearUnity(float gain) { + return std::abs(1.0f - gain) < kNegative60dbInAmplitude; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain.h new file mode 100644 index 000000000..500af8c22 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain.h @@ -0,0 +1,67 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_GAIN_H_ +#define RESONANCE_AUDIO_DSP_GAIN_H_ + +#include <cstddef> +#include "base/audio_buffer.h" + +namespace vraudio { + +// Implements a linearly-interpolated application of gain to a buffer channel. +// +// @param ramp_length Length of interpolation ramp in samples. Must be > 0. +// @param start_gain Starting gain value for ramp. +// @param end_gain Finishing gain value for ramp. +// @param input_samples Channel buffer to which interpolated gain is applied. +// @param output_samples Channel buffer to contain scaled output. +// @param accumulate_output True if the processed input should be mixed into the +// output. Otherwise, the output will be replaced by the processed input. +// @return Next gain value to be applied to the buffer channel. +float LinearGainRamp(size_t ramp_length, float start_gain, float end_gain, + const AudioBuffer::Channel& input_samples, + AudioBuffer::Channel* output_samples, + bool accumulate_output); + +// Applies a gain value to a vector of buffer samples starting at some offset. +// +// @param offset_index Starting index for gain application in buffer. +// @param gain Gain value applied to samples. +// @param input_samples Channel buffer to which gain is applied. +// @param output_samples Channel buffer to contain scaled output. +// @param accumulate_output True if the processed input should be mixed into the +// output. Otherwise, the output will be replaced by the processed input. +void ConstantGain(size_t offset_index, float gain, + const AudioBuffer::Channel& input_samples, + AudioBuffer::Channel* output_samples, bool accumulate_output); + +// Checks if the gain factor is close enough to zero (less than -60 decibels). +// +// @param gain Gain value to be tested. +// @return true if the current gain factors are near zero, false otherwise. +bool IsGainNearZero(float gain); + +// Checks if the gain state is close enough to Unity (less than -60 decibels +// below or above). +// +// @param gain Gain value to be tested. +// @return true if the current gain factors are near unity, false otherwise. +bool IsGainNearUnity(float gain); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_GAIN_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer.cc new file mode 100644 index 000000000..06417be11 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer.cc @@ -0,0 +1,115 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/gain_mixer.h" + +#include <cmath> + +#include "base/logging.h" + +namespace vraudio { + +GainMixer::GainMixer(size_t num_channels, size_t frames_per_buffer) + : num_channels_(num_channels), + output_(num_channels_, frames_per_buffer), + is_empty_(false) { + DCHECK_NE(num_channels_, 0U); + Reset(); +} + +void GainMixer::AddInput(const AudioBuffer& input, + const std::vector<float>& gains) { + DCHECK_EQ(gains.size(), num_channels_); + DCHECK_EQ(input.num_channels(), num_channels_); + DCHECK_EQ(input.num_frames(), output_.num_frames()); + + auto* gain_processors = GetOrCreateProcessors(input.source_id()); + // Accumulate the input buffers into the output buffer. + for (size_t i = 0; i < num_channels_; ++i) { + if (input[i].IsEnabled()) { + (*gain_processors)[i].ApplyGain(gains[i], input[i], &output_[i], + true /* accumulate_output */); + } else { + // Make sure the gain processor is initialized. + (*gain_processors)[i].Reset(gains[i]); + } + } + is_empty_ = false; +} + +void GainMixer::AddInputChannel(const AudioBuffer::Channel& input, + SourceId source_id, + const std::vector<float>& gains) { + DCHECK_EQ(gains.size(), num_channels_); + DCHECK_EQ(input.size(), output_.num_frames()); + + auto* gain_processors = GetOrCreateProcessors(source_id); + // Accumulate the input buffers into the output buffer. + for (size_t i = 0; i < num_channels_; ++i) { + if (input.IsEnabled()) { + (*gain_processors)[i].ApplyGain(gains[i], input, &output_[i], + true /* accumulate_output */); + } else { + // Make sure the gain processor is initialized. + (*gain_processors)[i].Reset(gains[i]); + } + } + is_empty_ = false; +} + +const AudioBuffer* GainMixer::GetOutput() const { + if (is_empty_) { + return nullptr; + } + return &output_; +} + +void GainMixer::Reset() { + if (!is_empty_) { + // Delete the processors for sources which no longer exist. + for (auto it = source_gain_processors_.begin(); + it != source_gain_processors_.end(); + /* no increment */) { + if (it->second.processors_active) { + it->second.processors_active = false; + ++it; + } else { + source_gain_processors_.erase(it++); + } + } + // Reset the output buffer. + output_.Clear(); + } + is_empty_ = true; +} + +GainMixer::GainProcessors::GainProcessors(size_t num_channels) + : processors_active(true), processors(num_channels) {} + +std::vector<GainProcessor>* GainMixer::GetOrCreateProcessors( + SourceId source_id) { + // Attempt to find a |ScaleAndAccumulateProcessor| for the given |source_id|, + // if none can be found add one. In either case mark that the processor has + // been used so that it is not later deleted. + if (source_gain_processors_.find(source_id) == + source_gain_processors_.end()) { + source_gain_processors_.insert({source_id, GainProcessors(num_channels_)}); + } + source_gain_processors_.at(source_id).processors_active = true; + return &(source_gain_processors_.at(source_id).processors); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer.h new file mode 100644 index 000000000..18c785c26 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer.h @@ -0,0 +1,96 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_GAIN_MIXER_H_ +#define RESONANCE_AUDIO_DSP_GAIN_MIXER_H_ + +#include <memory> +#include <unordered_map> +#include <vector> + +#include "base/audio_buffer.h" +#include "base/constants_and_types.h" +#include "dsp/gain_processor.h" + +namespace vraudio { + +class GainMixer { + public: + GainMixer(size_t num_channels, size_t frames_per_buffer); + + // Adds a separately scaled version of each channel of the input buffer to the + // output buffer's corresponding channels. + // + // @param input Input buffer to be added. + // @param gains Gains to be applied to the buffers channels. Must be equal in + // length to the number of channels in |input|. + void AddInput(const AudioBuffer& input, const std::vector<float>& gains); + + // Adds a single input channel to each of the output buffer's channels, with + // a separate gain applied to the input channel per output channel. + // + // @param input Input channel to be added. + // @param source_id Identifier corresponding to the input. + // @param gains Gains to be applied to the buffers channels. Must be equal in + // length to the number of channels in the output buffer. + void AddInputChannel(const AudioBuffer::Channel& input, SourceId source_id, + const std::vector<float>& gains); + + // Returns a pointer to the accumulator. + // + // @return Pointer to the processed (mixed) output buffer, or nullptr if no + // input has been added to the accumulator. + const AudioBuffer* GetOutput() const; + + // Resets the state of the accumulator. + void Reset(); + + private: + // Comprises one |GainProcessor| per channel of a source and a boolean to + // denote whether that source is active. + struct GainProcessors { + explicit GainProcessors(size_t num_channels); + + // Bool to signify if a given source is still passing data to a processor. + bool processors_active; + + // Scale and accumulation processors, one per channel for each source. + std::vector<GainProcessor> processors; + }; + + // Returns the |GainProcessor|s associated with a |source_id| (or creates + // one if needed) and sets the corresponding |processors_active| flag to true. + // + // @param source_id Identifier for a given input. + // @return The corresponding |ScalingAccumulators|. + std::vector<GainProcessor>* GetOrCreateProcessors(SourceId source_id); + + // Number of channels. + const size_t num_channels_; + + // Output buffer (accumulator). + AudioBuffer output_; + + // Denotes whether the accumulator has processed any inputs or not. + bool is_empty_; + + // Scale and accumulation processors, one per channel for each source. + std::unordered_map<SourceId, GainProcessors> source_gain_processors_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_GAIN_MIXER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer_test.cc new file mode 100644 index 000000000..70a45a692 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_mixer_test.cc @@ -0,0 +1,188 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/gain_mixer.h" + +#include <iterator> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "utils/planar_interleaved_conversion.h" + +namespace vraudio { + +namespace { + +const int kId1 = 0; +const int kId2 = 1; + +// Number of frames per output buffer. +const size_t kFramesPerBuffer = 3; + +// Sample mono input data, gains, and the corresponding output data. +const float kGain1 = 0.5f; +const float kGain2 = 2.0f; +const float kInput1[kFramesPerBuffer] = {0.2f, 0.4f, 0.6f}; +const float kInput2[kFramesPerBuffer] = {0.5f, 1.0f, 1.5f}; +const float kOutput[kFramesPerBuffer] = {1.1f, 2.2f, 3.3f}; + +// Tests that the gain mixer returns null output if no input has been added. +TEST(MixerTest, EmptyInputTest) { + GainMixer gain_mixer(kNumMonoChannels, kFramesPerBuffer); + + const AudioBuffer* output = gain_mixer.GetOutput(); + EXPECT_TRUE(output == nullptr); +} + +// Mono processing test with two inputs. Tests the mixed output buffer against +// the manually computed output data. +TEST(GainMixerTest, ProcessTest) { + GainMixer gain_mixer(kNumMonoChannels, kFramesPerBuffer); + + // Initialize input buffers. + AudioBuffer input1(kNumMonoChannels, kFramesPerBuffer); + input1.set_source_id(kId1); + FillAudioBuffer(std::begin(kInput1), kFramesPerBuffer, kNumMonoChannels, + &input1); + AudioBuffer input2(kNumMonoChannels, kFramesPerBuffer); + input2.set_source_id(kId2); + FillAudioBuffer(std::begin(kInput2), kFramesPerBuffer, kNumMonoChannels, + &input2); + // Initialize gain vectors. + const std::vector<float> kGainVector1(1, kGain1); + const std::vector<float> kGainVector2(1, kGain2); + + // Add the input buffers to the |GainMixer| (process multiple times so the + // gains have reached steady state. + const AudioBuffer* output = nullptr; + for (size_t iterations = 0; iterations < kUnitRampLength; ++iterations) { + gain_mixer.Reset(); + gain_mixer.AddInput(input1, kGainVector1); + gain_mixer.AddInput(input2, kGainVector2); + // Get the processed output (should contain the pre-computed output data). + output = gain_mixer.GetOutput(); + } + + EXPECT_FALSE(output == nullptr); + EXPECT_EQ(output->num_channels(), kNumMonoChannels); + EXPECT_EQ(output->num_frames(), kFramesPerBuffer); + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR((*output)[0][i], kOutput[i], kEpsilonFloat); + } +} + +// Reset test for the |GainMixer| with single input at a time. First, adds an +// input buffer, resets the |GainMixer|. Then, adds another input buffer and +// tests the final output buffer against the data of the last added input buffer +// to verify if the system has been successfully reset. +TEST(GainMixerTest, ResetTest) { + GainMixer gain_mixer(kNumMonoChannels, kFramesPerBuffer); + + // Initialize input buffers. + AudioBuffer input1(kNumMonoChannels, kFramesPerBuffer); + input1.set_source_id(kId1); + FillAudioBuffer(std::begin(kInput1), kFramesPerBuffer, kNumMonoChannels, + &input1); + AudioBuffer input2(kNumMonoChannels, kFramesPerBuffer); + input2.set_source_id(kId2); + FillAudioBuffer(std::begin(kInput2), kFramesPerBuffer, kNumMonoChannels, + &input2); + // Initialize gain vectors. + const std::vector<float> kGainVector1(1, kGain1); + const std::vector<float> kGainVector2(1, kGain2); + + // Add the first input buffer to the |GainMixer|. + gain_mixer.AddInput(input1, kGainVector1); + // Reset the accumulator (and release the buffer). + gain_mixer.Reset(); + // Add the second input buffers to the |GainMixer|. + gain_mixer.AddInput(input2, kGainVector2); + + // Get the output (should only contain the second input). + const AudioBuffer* output = gain_mixer.GetOutput(); + + EXPECT_FALSE(output == nullptr); + EXPECT_EQ(output->num_channels(), kNumMonoChannels); + EXPECT_EQ(output->num_frames(), kFramesPerBuffer); + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR((*output)[0][i], kInput2[i] * kGain2, kEpsilonFloat); + } +} + +// Tests that gains are correctly applied for multiple channels. +TEST(GainMixerTest, MultiChannelTest) { + const std::vector<float> kGains = {0.1f, -0.2f}; + const std::vector<float> kInputChannel = {10.0f, 10.0f, 10.0f}; + GainMixer gain_mixer(kGains.size(), kFramesPerBuffer); + + // Initialize input buffer. + AudioBuffer input(kGains.size(), kInputChannel.size()); + for (size_t i = 0; i < kGains.size(); ++i) { + input[i] = kInputChannel; + } + + const AudioBuffer* output = nullptr; + for (size_t iterations = 0; iterations < kUnitRampLength; ++iterations) { + gain_mixer.Reset(); + gain_mixer.AddInput(input, kGains); + output = gain_mixer.GetOutput(); + } + + EXPECT_FALSE(output == nullptr); + EXPECT_EQ(output->num_channels(), kGains.size()); + EXPECT_EQ(output->num_frames(), kFramesPerBuffer); + for (size_t channel = 0; channel < kGains.size(); ++channel) { + for (size_t i = 0; i < kInputChannel.size(); ++i) { + EXPECT_NEAR((*output)[channel][i], kInputChannel[i] * kGains[channel], + kEpsilonFloat); + } + } +} + +// Tests that gains are correctly applied for a mono input buffer across a +// multichannel output buffer. +TEST(GainMixerTest, MonoChannelInputTest) { + const std::vector<float> kGains = {2.0f, -3.0f}; + const std::vector<float> kInputChannel = {10.0f, 10.0f, 10.0f}; + GainMixer gain_mixer(kGains.size(), kInputChannel.size()); + + // Create a mono input buffer. + AudioBuffer input(kNumMonoChannels, kInputChannel.size()); + AudioBuffer::Channel* input_channel = &input[0]; + *input_channel = kInputChannel; + + const AudioBuffer* output = nullptr; + for (size_t iterations = 0; iterations < kUnitRampLength; ++iterations) { + gain_mixer.Reset(); + gain_mixer.AddInputChannel(*input_channel, kId1, kGains); + output = gain_mixer.GetOutput(); + } + + EXPECT_FALSE(output == nullptr); + EXPECT_EQ(output->num_channels(), kGains.size()); + EXPECT_EQ(output->num_frames(), kFramesPerBuffer); + for (size_t channel = 0; channel < kGains.size(); ++channel) { + for (size_t i = 0; i < kInputChannel.size(); ++i) { + EXPECT_NEAR((*output)[channel][i], kInputChannel[i] * kGains[channel], + kEpsilonFloat); + } + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor.cc new file mode 100644 index 000000000..f5eaea22d --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor.cc @@ -0,0 +1,94 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/gain_processor.h" + +#include <algorithm> +#include <cmath> + +#include "base/constants_and_types.h" + +#include "dsp/gain.h" + +namespace vraudio { + +// Delegate default constructor. +GainProcessor::GainProcessor() : current_gain_(0.0f), is_initialized_(false) {} + +GainProcessor::GainProcessor(float initial_gain) + : current_gain_(initial_gain), is_initialized_(true) {} + +void GainProcessor::ApplyGain(float target_gain, + const AudioBuffer::Channel& input, + AudioBuffer::Channel* output, + bool accumulate_output) { + + DCHECK(output); + + if (!is_initialized_) { + Reset(target_gain); + } + + // Check buffer length. + const size_t input_length = input.size(); + DCHECK_GT(input_length, 0U); + DCHECK_EQ(input_length, output->size()); + + // Index for where to stop interpolating. + size_t ramp_length = + static_cast<size_t>(std::abs(target_gain - current_gain_) * + static_cast<float>(kUnitRampLength)); + + // Check if there is a new gain value. + if (ramp_length > 0) { + // Apply gain ramp to buffer. + current_gain_ = LinearGainRamp(ramp_length, current_gain_, target_gain, + input, output, accumulate_output); + } else { + // No ramping needed. + current_gain_ = target_gain; + } + + // Apply constant gain to the rest of the buffer. + if (ramp_length < input_length) { + if (IsGainNearZero(current_gain_)) { + // Skip processing if the gain is zero. + if (!accumulate_output) { + // Directly fill the remaining output with zeros. + std::fill(output->begin() + ramp_length, output->end(), 0.0f); + } + return; + } else if (IsGainNearUnity(current_gain_) && !accumulate_output) { + // Skip processing if the gain is unity. + if (&input != output) { + // Directly copy the remaining input samples into output. + std::copy(input.begin() + ramp_length, input.end(), + output->begin() + ramp_length); + } + return; + } + ConstantGain(ramp_length, current_gain_, input, output, accumulate_output); + } +} + +float GainProcessor::GetGain() const { return current_gain_; } + +void GainProcessor::Reset(float gain) { + current_gain_ = gain; + is_initialized_ = true; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor.h new file mode 100644 index 000000000..4baa276ea --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor.h @@ -0,0 +1,74 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_GAIN_PROCESSOR_H_ +#define RESONANCE_AUDIO_DSP_GAIN_PROCESSOR_H_ + +#include <cmath> +#include <cstddef> +#include <vector> + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Processor class which applies a gain to a vector of samples from an audio +// buffer. A short linear ramp is applied to the gain to reduce audible +// artifacts in the output. +class GainProcessor { + public: + // Default constructor keeps the gain state uninitialized. The first call to + // |ApplyGain| sets the internal gain state. + GainProcessor(); + + // Constructs |GainProcessor| with some initial gain value. + // + // @param initial_gain Gain value used as starting point for first processing + // period's gain ramping. + explicit GainProcessor(float initial_gain); + + // Applies gain supplied to the input samples. + // + // @param target_gain Target gain value. + // @param input Samples to which gain will be applied. + // @param output Samples to which gain has been applied. + // @param accumulate_output True if the processed input should be mixed into + // the output. Otherwise, the output will be replaced by the processed + // input. + void ApplyGain(float target_gain, const AudioBuffer::Channel& input, + AudioBuffer::Channel* output, bool accumulate_output); + + // Returns the |current_gain_| value. + // + // @return Current gain applied by the |GainProcessor|. + float GetGain() const; + + // Resets the gain processor to a new gain factor. + // + // @param gain Gain value. + void Reset(float gain); + + private: + // Latest gain value to be applied to buffer values. + float current_gain_; + + // Flag to indiciate if an initial gain has been assigned. + bool is_initialized_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_GAIN_PROCESSOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor_test.cc new file mode 100644 index 000000000..e1077655a --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_processor_test.cc @@ -0,0 +1,181 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/gain_processor.h" + +#include <algorithm> +#include <cmath> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/audio_buffer.h" +#include "base/constants_and_types.h" + +// A set of simple tests to confirm expected behavior when a linear gain ramp is +// applied to a vector of samples. +namespace vraudio { + +namespace { + +// Initial gain value. +const float kInitialGain = 0.1f; + +// Target gain value. +const float kTargetGain = 0.6f; + +// Test input data length. +const size_t kInputLength = 10; + +// Test value of each sample of the output buffer prior to processing. +const float kInitialOutputValue = 2.0f; + +class GainProcessorTest : public ::testing::TestWithParam<bool> {}; + +// This test checks that gain value is applied to the input buffer in a linear +// ramp when the buffer length is less than the ramp length. +TEST_P(GainProcessorTest, ApplyGainOverBufferLengthTest) { + // Expected output data for ramped gain application. + const float kExpectedOutput[kInputLength] = { + 0.1f, 0.1004883f, 0.1009766f, 0.1014648f, 0.1019531f, + 0.1024414f, 0.1029297f, 0.1034180f, 0.1039063f, 0.1043945f}; + const bool accumulate_output = GetParam(); + // Initialize input buffer. + AudioBuffer input(kNumMonoChannels, kInputLength); + std::fill(input[0].begin(), input[0].end(), 1.0f); + // Initialize output buffer. + AudioBuffer output(kNumMonoChannels, kInputLength); + std::fill(output[0].begin(), output[0].end(), kInitialOutputValue); + + // Initialize gain processor with a gain value. + GainProcessor gain_processor(kInitialGain); + // Process buffer samples with the test gain value. + gain_processor.ApplyGain(kTargetGain, input[0], &output[0], + accumulate_output); + + // Check that gain values have been applied correctly to the buffer. + for (size_t i = 0; i < kInputLength; ++i) { + const float expected_value = accumulate_output + ? kInitialOutputValue + kExpectedOutput[i] + : kExpectedOutput[i]; + EXPECT_NEAR(expected_value, output[0][i], kEpsilonFloat); + } +} + +// This test checks that gain value is applied to the buffer samples in a linear +// ramp. The test also confirms that the gain ramping halts at the correct index +// for buffer lengths greater than the ramp length. +TEST_P(GainProcessorTest, ApplyGainLongerThanRampTest) { + const bool accumulate_output = GetParam(); + // Initialize input buffer. + AudioBuffer input(kNumMonoChannels, kUnitRampLength); + std::fill(input[0].begin(), input[0].end(), 1.0f); + // Initialize output buffer. + AudioBuffer output(kNumMonoChannels, kUnitRampLength); + std::fill(output[0].begin(), output[0].end(), kInitialOutputValue); + + // Initialize gain processor with a gain value. + GainProcessor gain_processor(kInitialGain); + // Process an input buffer with the test gain value. + gain_processor.ApplyGain(kTargetGain, input[0], &output[0], + accumulate_output); + + // Generate expected gain ramp. The output should consist of the linearly + // interpolated gain values over the ramp length, and the final (constant) + // gain for the rest of the buffer. + std::vector<float> expected_output(kUnitRampLength); + const size_t ramp_length = + static_cast<size_t>(std::abs(kTargetGain - kInitialGain) * + static_cast<float>(kUnitRampLength)); + const float increment = + (kTargetGain - kInitialGain) / static_cast<float>(ramp_length); + float expected_value = + accumulate_output ? kInitialOutputValue + kInitialGain : kInitialGain; + for (size_t i = 0; i < kUnitRampLength; ++i) { + expected_output[i] = expected_value; + if (i < ramp_length) { + expected_value += increment; + } + } + // Check that gain values have been applied correctly to the buffer. + for (size_t i = 0; i < kUnitRampLength; ++i) { + // Check that ramp was applied. + EXPECT_NEAR(expected_output[i], output[0][i], kEpsilonFloat); + } +} + +// This test checks that gain values are reset to initial gain after a call to +// the |Reset()| method. +TEST_P(GainProcessorTest, ResetGainProcessorTest) { + const bool accumulate_output = GetParam(); + // Initialize input buffer. + AudioBuffer input(kNumMonoChannels, kUnitRampLength); + std::fill(input[0].begin(), input[0].end(), 1.0f); + // Initialize output buffer. + AudioBuffer output(kNumMonoChannels, kUnitRampLength); + std::fill(output[0].begin(), output[0].end(), 0.0f); + + // Initialize gain processor with a gain of 0.0f. + GainProcessor gain_processor(0.0f); + // Reset gain to |kInitialGain|. + gain_processor.Reset(kInitialGain); + // Apply newly-reset gains (no ramp expected). + gain_processor.ApplyGain(kInitialGain, input[0], &output[0], + accumulate_output); + + // Check that uniform gain has been applied correctly to the buffer. + for (size_t i = 0; i < kInputLength; ++i) { + EXPECT_NEAR(kInitialGain, output[0][i], kEpsilonFloat); + } +} + +// Checks that the initial gain is assigned during the first call of |ApplyGain| +// in case the |GainProcessor| instance is constructed via the default +// constructor. +TEST_P(GainProcessorTest, DefaultConstructorTest) { + const bool accumulate_output = GetParam(); + // Initialize input buffer. + AudioBuffer input(kNumMonoChannels, kUnitRampLength); + std::fill(input[0].begin(), input[0].end(), kInitialGain); + // Initialize output buffer. + AudioBuffer output(kNumMonoChannels, kUnitRampLength); + std::fill(output[0].begin(), output[0].end(), 0.0f); + + // Declare gain processor without specifiying a gain value. + GainProcessor gain_processor; + // Apply some new gain value. + gain_processor.ApplyGain(kTargetGain, input[0], &output[0], + accumulate_output); + + // Check that uniform gain has been applied correctly to the buffer. + for (size_t i = 0; i < kInputLength; ++i) { + EXPECT_NEAR(kInitialGain * kTargetGain, output[0][i], kEpsilonFloat); + } +} + +// Tests the |GetGain| method. +TEST(GainProcessorTest, GetGainTest) { + // Test |GainProcessor| with arbitrary gain. + const float kGainValue = -1.5f; + GainProcessor gain_processor(kGainValue); + EXPECT_EQ(gain_processor.GetGain(), kGainValue); +} + +INSTANTIATE_TEST_CASE_P(AccumulateOutput, GainProcessorTest, + ::testing::Values(false, true)); + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_test.cc new file mode 100644 index 000000000..01e4471f4 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/gain_test.cc @@ -0,0 +1,132 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/gain.h" + +#include <algorithm> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/audio_buffer.h" +#include "base/constants_and_types.h" +#include "dsp/utils.h" + +namespace vraudio { + +namespace { + +// Tolerated error margin. +const float kEpsilon = 1e-4f; + +class GainTest : public ::testing::TestWithParam<bool> {}; + +// This test checks that the gain applied using ConstantGain is correct. +TEST_P(GainTest, ConstantGainTest) { + const size_t kMaxInputLength = 32; + const float kGain = 1; + const bool accumulate_output = GetParam(); + + for (size_t length = 1; length < kMaxInputLength; ++length) { + AudioBuffer input_samples(kNumMonoChannels, length); + GenerateUniformNoise( + /*min=*/-1.0f, /*min=*/1.0f, static_cast<unsigned>(length), + &input_samples[0]); + for (size_t offset_index = 0; offset_index < length; ++offset_index) { + // Initialize input buffer. + AudioBuffer input(kNumMonoChannels, length); + input[0] = input_samples[0]; + // Initialize output buffer with the same values. + AudioBuffer output(kNumMonoChannels, length); + output[0] = input_samples[0]; + + // Apply constant gain. + ConstantGain(offset_index, kGain, input[0], &output[0], + accumulate_output); + + // Compute expected values. + AudioBuffer expected_samples_buffer(kNumMonoChannels, length); + expected_samples_buffer[0] = input_samples[0]; + for (size_t i = offset_index; i < length; ++i) { + const float processed_input = kGain * input[0][i]; + if (accumulate_output) { + expected_samples_buffer[0][i] += processed_input; + } else { + expected_samples_buffer[0][i] = processed_input; + } + } + // Check that the output buffer has the expected values per each sample. + for (size_t i = 0; i < length; ++i) { + EXPECT_NEAR(expected_samples_buffer[0][i], output[0][i], kEpsilon) + << " at index=" << i << " with input_length=" << length + << " and offset_index=" << offset_index; + } + } + } +} + +// Test that checks that the gain ramp applied is correct. +TEST_P(GainTest, LinearGainRampTest) { + const float kInitialOutputValue = 2.0f; + const float kStartGain = 0.0f; + const float kEndGain = 1.0f; + const size_t kNumSamples = 10; + const float kPerSampleIncrease = + (kEndGain - kStartGain) / static_cast<float>(kNumSamples); + const bool accumulate_output = GetParam(); + + // Create an input buffer with unity samples. + AudioBuffer input(kNumMonoChannels, kNumSamples); + std::fill(input[0].begin(), input[0].end(), 1.0f); + // Create an output buffer with all samples the same. + AudioBuffer output(kNumMonoChannels, kNumSamples); + std::fill(output[0].begin(), output[0].end(), kInitialOutputValue); + + // Apply linear gain ramp. + LinearGainRamp(kNumSamples, kStartGain, kEndGain, input[0], &(output[0]), + accumulate_output); + + // Check that the output buffer has the expected values per each sample. + float expected_value = accumulate_output ? kInitialOutputValue : 0.0f; + for (size_t i = 0; i < kNumSamples; ++i) { + EXPECT_NEAR(expected_value, output[0][i], kEpsilon); + expected_value += kPerSampleIncrease; + } +} + +TEST(GainTest, GainUtilsTest) { + const float kGainUnity = 1.0f; + const float kGainZero = 0.0f; + const float kGainOther = -1.7f; + + // Test the cases where each should return true. + EXPECT_TRUE(IsGainNearZero(kGainZero)); + EXPECT_TRUE(IsGainNearUnity(kGainUnity)); + + // Test the cases where gain value is non zero, positive and negative. + EXPECT_FALSE(IsGainNearZero(kGainOther)); + EXPECT_FALSE(IsGainNearZero(kGainUnity)); + + // Test the case where gain value is not unity, with alternate value and zero. + EXPECT_FALSE(IsGainNearUnity(kGainOther)); + EXPECT_FALSE(IsGainNearUnity(kGainZero)); +} + +INSTANTIATE_TEST_CASE_P(AccumulateOutput, GainTest, + ::testing::Values(false, true)); + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer.cc new file mode 100644 index 000000000..1b7460428 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer.cc @@ -0,0 +1,56 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/mixer.h" + +#include "base/logging.h" + +namespace vraudio { + +Mixer::Mixer(size_t target_num_channels, size_t frames_per_buffer) + : output_(target_num_channels, frames_per_buffer), is_empty_(false) { + Reset(); +} + +void Mixer::AddInput(const AudioBuffer& input) { + DCHECK_EQ(input.num_frames(), output_.num_frames()); + + // Accumulate the input buffers into the output buffer. + const size_t num_channels = + std::min(input.num_channels(), output_.num_channels()); + for (size_t n = 0; n < num_channels; ++n) { + if (input[n].IsEnabled()) { + output_[n] += input[n]; + } + } + is_empty_ = false; +} + +const AudioBuffer* Mixer::GetOutput() const { + if (is_empty_) { + return nullptr; + } + return &output_; +} + +void Mixer::Reset() { + if (!is_empty_) { + output_.Clear(); + } + is_empty_ = true; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer.h new file mode 100644 index 000000000..b89144799 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer.h @@ -0,0 +1,62 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_MIXER_H_ +#define RESONANCE_AUDIO_DSP_MIXER_H_ + +#include <memory> + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Accepts multiple input buffers and outputs a downmix to a single output +// buffer. All input buffers must have the same number of frames per buffer, the +// output will have the target number of channels regardless of the input number +// of channels. +class Mixer { + public: + // Constructor. + // + // @param target_num_channels Target number of channels in accumulator buffer. + // @param frames_per_buffer Number of frames in accumulator buffer. + Mixer(size_t target_num_channels, size_t frames_per_buffer); + + // Adds an input buffer to the mixer, updates the output buffer accordingly. + // + // @param input Input buffer to be added. + void AddInput(const AudioBuffer& input); + + // Returns a pointer to the accumulator. + // + // @return Pointer to the processed (mixed) output buffer, or nullptr if no + // input has been added to the accumulator. + const AudioBuffer* GetOutput() const; + + // Resets the state of the accumulator. + void Reset(); + + private: + // Output buffer (accumulator). + AudioBuffer output_; + + // Denotes whether the accumulator has processed any inputs or not. + bool is_empty_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_MIXER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer_test.cc new file mode 100644 index 000000000..0cfdf8675 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/mixer_test.cc @@ -0,0 +1,163 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/mixer.h" + +#include <iterator> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "utils/planar_interleaved_conversion.h" + +namespace vraudio { + +namespace { + +// Number of frames per output buffer. +const size_t kFramesPerBuffer = 3; + +// Sample two-channel interleaved input data. +const float kStereoInput1[kNumStereoChannels * kFramesPerBuffer] = { + 0.5f, 0.0f, -0.25f, 0.0f, 0.25f, 0.0f}; +const float kStereoInput2[kNumStereoChannels * kFramesPerBuffer] = { + 0.5f, 1.0f, 0.5f, -1.0f, 0.5f, 1.0f}; + +// Tests that the mixer returns null output if no input has been added. +TEST(MixerTest, EmptyInputTest) { + Mixer mixer(kNumStereoChannels, kFramesPerBuffer); + + const AudioBuffer* output = mixer.GetOutput(); + EXPECT_TRUE(output == nullptr); +} + +// This is a simple two-channel process test with two inputs. Tests the +// mixed output buffer against the manually computed output data. +TEST(MixerTest, ProcessUniformInputChannelsTest) { + Mixer mixer(kNumStereoChannels, kFramesPerBuffer); + + // Initialize the input buffers. + AudioBuffer input1(kNumStereoChannels, kFramesPerBuffer); + FillAudioBuffer(std::begin(kStereoInput1), kFramesPerBuffer, + kNumStereoChannels, &input1); + AudioBuffer input2(kNumStereoChannels, kFramesPerBuffer); + FillAudioBuffer(std::begin(kStereoInput2), kFramesPerBuffer, + kNumStereoChannels, &input2); + + // Add the input buffers to the mixer. + mixer.AddInput(input1); + mixer.AddInput(input2); + // Get the processed output. + const AudioBuffer* output = mixer.GetOutput(); + + // Test that the output channels was accumulated correctly. + EXPECT_FALSE(output == nullptr); + EXPECT_EQ(output->num_channels(), kNumStereoChannels); + EXPECT_EQ(output->num_frames(), kFramesPerBuffer); + for (size_t n = 0; n < kNumStereoChannels; ++n) { + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR((*output)[n][i], input1[n][i] + input2[n][i], kEpsilonFloat); + } + } +} + +// This is a non-uniform process test with two inputs with arbitrary number of +// channels. Tests the mixed output buffer with different target number of +// channels against the manually computed output data. +TEST(MixerTest, ProcessVaryingInputChannelsTest) { + // Initialize the input buffers. + AudioBuffer mono_input(kNumMonoChannels, kFramesPerBuffer); + FillAudioBuffer(std::begin(kStereoInput1), kFramesPerBuffer, kNumMonoChannels, + &mono_input); + AudioBuffer stereo_input(kNumStereoChannels, kFramesPerBuffer); + FillAudioBuffer(std::begin(kStereoInput2), kFramesPerBuffer, + kNumStereoChannels, &stereo_input); + + // Initialize a mono mixer. + Mixer mono_mixer(kNumMonoChannels, kFramesPerBuffer); + // Add the input buffers to the mixer. + mono_mixer.AddInput(mono_input); + mono_mixer.AddInput(stereo_input); + // Get the processed output. + const AudioBuffer* mono_output = mono_mixer.GetOutput(); + // Test that the mono output channel was accumulated correctly. + EXPECT_FALSE(mono_output == nullptr); + EXPECT_EQ(mono_output->num_channels(), kNumMonoChannels); + EXPECT_EQ(mono_output->num_frames(), kFramesPerBuffer); + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR((*mono_output)[0][i], mono_input[0][i] + stereo_input[0][i], + kEpsilonFloat); + } + + // Initialize a stereo mixer. + Mixer stereo_mixer(kNumStereoChannels, kFramesPerBuffer); + // Add the input buffers to the mixer. + stereo_mixer.AddInput(mono_input); + stereo_mixer.AddInput(stereo_input); + // Get the processed output. + const AudioBuffer* stereo_output = stereo_mixer.GetOutput(); + // Test that the stereo output channels were accumulated correctly. + EXPECT_FALSE(stereo_output == nullptr); + EXPECT_EQ(stereo_output->num_channels(), kNumStereoChannels); + EXPECT_EQ(stereo_output->num_frames(), kFramesPerBuffer); + for (size_t n = 0; n < kNumStereoChannels; ++n) { + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + // The second channel should only contain the samples from |stereo_input|. + EXPECT_NEAR( + (*stereo_output)[n][i], + (n == 0) ? mono_input[n][i] + stereo_input[n][i] : stereo_input[n][i], + kEpsilonFloat); + } + } +} + +// This is a two-channel reset test for the mixer with single input at a time. +// First, adds an input buffer, resets the mixer. Then, adds another input +// buffer and tests the final output buffer against the data of the last added +// input buffer to verify if the system has been successfully reset. +TEST(MixerTest, ResetTest) { + Mixer mixer(kNumStereoChannels, kFramesPerBuffer); + + // Initialize the input buffers. + AudioBuffer input1(kNumStereoChannels, kFramesPerBuffer); + FillAudioBuffer(std::begin(kStereoInput1), kFramesPerBuffer, + kNumStereoChannels, &input1); + AudioBuffer input2(kNumStereoChannels, kFramesPerBuffer); + FillAudioBuffer(std::begin(kStereoInput2), kFramesPerBuffer, + kNumStereoChannels, &input2); + + // Add the first input buffer to the mixer. + mixer.AddInput(input1); + // Reset the accumulator. + mixer.Reset(); + // Add the second input buffers to the mixer. + mixer.AddInput(input2); + // Get the processed output. + const AudioBuffer* output = mixer.GetOutput(); + // Test that the output channels contains only the samples from |input2|. + EXPECT_FALSE(output == nullptr); + EXPECT_EQ(output->num_channels(), kNumStereoChannels); + EXPECT_EQ(output->num_frames(), kFramesPerBuffer); + for (size_t n = 0; n < kNumStereoChannels; ++n) { + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + EXPECT_NEAR((*output)[n][i], input2[n][i], kEpsilonFloat); + } + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter.cc new file mode 100644 index 000000000..18c4625d5 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter.cc @@ -0,0 +1,59 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/mono_pole_filter.h" + +#include <algorithm> + +#include "base/constants_and_types.h" + + +namespace vraudio { + +MonoPoleFilter::MonoPoleFilter(float coefficient) : previous_output_(0.0f) { + SetCoefficient(coefficient); +} + +void MonoPoleFilter::SetCoefficient(float coefficient) { + coefficient_ = std::max(std::min(coefficient, 1.0f), 0.0f); +} + +bool MonoPoleFilter::Filter(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output) { + + DCHECK(output); + const size_t num_frames = input.size(); + DCHECK_EQ(num_frames, output->size()); + + // Do not perform processing if the coefficient is zero to avoid wasteful + // "all pass" cases. + if (coefficient_ < kEpsilonFloat) { + previous_output_ = input[num_frames - 1]; + return false; + } + + // The difference equation implemented here is as follows: + // y[n] = a * (y[n-1] - x[n]) + x[n] + // where y[n] is the output and x[n] is the input vector. + for (size_t frame = 0; frame < num_frames; ++frame) { + (*output)[frame] = + coefficient_ * (previous_output_ - input[frame]) + input[frame]; + previous_output_ = (*output)[frame]; + } + return true; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter.h new file mode 100644 index 000000000..4ec3f75dd --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter.h @@ -0,0 +1,57 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_MONO_POLE_FILTER_H_ +#define RESONANCE_AUDIO_DSP_MONO_POLE_FILTER_H_ + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Class representing a mono pole filter. This class also performs filtering. +class MonoPoleFilter { + public: + // Constructs a |MonoPoleFilter| given a single coefficient. + // + // @param coefficient A single coefficient between 0.0f and 1.0f. + explicit MonoPoleFilter(float coefficient); + + // Filter method for use with AudioBuffer::Channel. + // + // @param input |AudioBuffer::Channel| of input to be processed. + // @param output Pointer to output |AudioBuffer::Channel|. + // @return Returns false if the filter has an allpass configuration. This + // helps to avoid copies whenever the output is expected to be identical + // to the input. + bool Filter(const AudioBuffer::Channel& input, AudioBuffer::Channel* output); + + // Sets the filter's coefficent. + // + // @param coefficient A mono pole filter coefficient. + void SetCoefficient(float coefficient); + + private: + // The previous frame computed by the filter. + float previous_output_; + + // Represents and maintains the state of the filter in terms of its + // transfer function coefficient. + float coefficient_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_MONO_POLE_FILTER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter_test.cc new file mode 100644 index 000000000..c93b4fab7 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/mono_pole_filter_test.cc @@ -0,0 +1,59 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/mono_pole_filter.h" + +#include <cmath> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "base/misc_math.h" + +namespace vraudio { + +namespace { + +const float kCoefficient = 0.5f; +const size_t kFramesPerBuffer = 10; + +// Tests that the filter correctly implements the difference equation. +TEST(MonoPoleFilterTest, ImpulseResponseTest) { + MonoPoleFilter filter(kCoefficient); + AudioBuffer buffer(1U, kFramesPerBuffer); + buffer.Clear(); + buffer[0][0] = 1.0f; + EXPECT_TRUE(filter.Filter(buffer[0], &buffer[0])); + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + // The impulse response of this filter (for coefficient a), should be: + // d[n] = (1 - a) * a^n + const float expected = + (1.0f - kCoefficient) * IntegerPow(kCoefficient, static_cast<int>(i)); + EXPECT_NEAR(buffer[0][i], expected, kEpsilonFloat); + } +} + +// Tests that no processing is performed when the filter is allpass. +TEST(MonoPoleFilterTest, AllPassNoProcessingTest) { + MonoPoleFilter filter(0.0f); + AudioBuffer buffer(1U, kFramesPerBuffer); + buffer.Clear(); + buffer[0][0] = 1.0f; + EXPECT_FALSE(filter.Filter(buffer[0], &buffer[0])); +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir.cc new file mode 100644 index 000000000..dc3bbcb8f --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir.cc @@ -0,0 +1,152 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/multi_channel_iir.h" + +#include <algorithm> +#include <limits> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/simd_macros.h" + + +namespace vraudio { + +std::unique_ptr<MultiChannelIir> MultiChannelIir::Create( + size_t num_channels, size_t frames_per_buffer, + const std::vector<std::vector<float>>& numerators, + const std::vector<std::vector<float>>& denominators) { + CHECK_EQ(denominators.size(), numerators.size()); + CHECK_EQ(denominators.size(), num_channels); + CHECK_EQ(num_channels % SIMD_LENGTH, 0U); + CHECK_GT(num_channels, 0U); + for (size_t channel = 0; channel < num_channels; ++channel) { + CHECK_GT(denominators[channel].size(), 0U); + CHECK_GT(std::abs(denominators[channel][0]), + std::numeric_limits<float>::epsilon()); + CHECK_EQ(denominators[channel].size(), numerators[channel].size()); + } + + std::unique_ptr<MultiChannelIir> multi_channel_iir(new MultiChannelIir( + num_channels, frames_per_buffer, numerators[0].size())); + CHECK(multi_channel_iir); + + for (size_t channel = 0; channel < num_channels; ++channel) { + size_t interleaved_index = channel; + for (size_t i = 0; i < numerators[channel].size(); ++i) { + // Make the denominator polynomials monic, (divide all coefficients by the + // the first denominator coefficients). Furthermore negate all + // coefficients beyond the first (see equation in Process method). + // Copy the coefficients into the |numerator_| and |denominator_| buffers. + multi_channel_iir->numerator_[0][interleaved_index] = + numerators[channel][i] / denominators[channel][0]; + multi_channel_iir->denominator_[0][interleaved_index] = + denominators[channel][i] / + ((i > 0) ? -denominators[channel][0] : denominators[channel][0]); + interleaved_index += num_channels; + } + } + + multi_channel_iir->delay_line_.Clear(); + return multi_channel_iir; +} + +void MultiChannelIir::Process(AudioBuffer::Channel* interleaved_buffer) { + + DCHECK(interleaved_buffer); + DCHECK_EQ(interleaved_buffer->size(), num_channels_ * frames_per_buffer_); + const SimdVector* simd_numerator = + reinterpret_cast<SimdVector*>(&(numerator_[0][0])); + const SimdVector* simd_denominator = + reinterpret_cast<SimdVector*>(&(denominator_[0][0])); + SimdVector* simd_delay_line = + reinterpret_cast<SimdVector*>(&(delay_line_[0][0])); + SimdVector* simd_buffer = + reinterpret_cast<SimdVector*>(&((*interleaved_buffer)[0])); + + const size_t num_channel_chunks = num_channels_ / SIMD_LENGTH; + const size_t num_buffer_chunks = interleaved_buffer->size() / SIMD_LENGTH; + const size_t delay_length_in_chunks = num_channel_chunks * num_coefficients_; + + for (size_t current_frame = 0, individual_frame = 0; + current_frame < num_buffer_chunks; + current_frame += num_channel_chunks, individual_frame += num_channels_) { + DCHECK_LT(individual_frame, interleaved_buffer->size()); + // Copy the current sample into the delay line at the very start. + // {x[n], w[n-1], w[n-2], . . .} where each x, w represents all channels. + std::copy_n(interleaved_buffer->begin() + individual_frame, num_channels_, + delay_line_[0].begin() + (delay_line_front_ * SIMD_LENGTH)); + // Using A Direct Form II implementation difference equation: + // Source: Digital Signal Processing Principles Algorithms & Applications + // Fourth Edition. John G. Prolakis and Dimitris G. Manolakis - Chap 9 + // w[n] = x[n] - (a1/a0)*w[n-1] - (a2/a0)*w[n-2] - . . . + // y(n) = (b0/a0)*w[n] + (b1/a0)*w[n-1] + (b2/a0)*w[n-2] + . . . + // where x[n] is input, w[n] is storage and y[n] is output. + // The division by a0 has been performed in the constructor along with the + // negation of the denominator coefficients beyond the first. Note also + // that each term here refers to a set of channels. + for (size_t channel_chunk = 0; channel_chunk < num_channel_chunks; + ++channel_chunk) { + // First zero out the relevant section of the buffer before accumulation. + // Zero constant used for loading zeros into a neon simd array, as the + // |vld1q_dup_f32| neon intrinsic requires an lvalue parameter. + const float kZerof = 0.0f; + simd_buffer[current_frame + channel_chunk] = SIMD_LOAD_ONE_FLOAT(kZerof); + for (size_t coeff_offset = num_channel_chunks; + coeff_offset < delay_length_in_chunks; + coeff_offset += num_channel_chunks) { + // Denominator part. + const size_t multiplication_index = channel_chunk + coeff_offset; + const size_t delay_multiplication_index = + (multiplication_index + delay_line_front_) % delay_length_in_chunks; + const size_t delay_write_index = channel_chunk + delay_line_front_; + simd_delay_line[delay_write_index] = + SIMD_MULTIPLY_ADD(simd_denominator[multiplication_index], + simd_delay_line[delay_multiplication_index], + simd_delay_line[delay_write_index]); + } + for (size_t coeff_offset = 0; coeff_offset < delay_length_in_chunks; + coeff_offset += num_channel_chunks) { + // Numerator part. + const size_t multiplication_index = channel_chunk + coeff_offset; + const size_t write_index = current_frame + channel_chunk; + const size_t delay_multiplication_index = + (multiplication_index + delay_line_front_) % delay_length_in_chunks; + simd_buffer[write_index] = + SIMD_MULTIPLY_ADD(simd_numerator[multiplication_index], + simd_delay_line[delay_multiplication_index], + simd_buffer[write_index]); + } + } + // Update the index to the wrapped around 'front' of the delay line. + delay_line_front_ = ((static_cast<int>(delay_line_front_) - + num_channel_chunks + delay_length_in_chunks) % + delay_length_in_chunks); + } +} + +MultiChannelIir::MultiChannelIir(size_t num_channels, size_t frames_per_buffer, + size_t num_coefficients) + : num_channels_(num_channels), + frames_per_buffer_(frames_per_buffer), + num_coefficients_(num_coefficients), + delay_line_front_(0), + numerator_(kNumMonoChannels, num_coefficients_ * num_channels_), + denominator_(kNumMonoChannels, num_coefficients_ * num_channels_), + delay_line_(kNumMonoChannels, num_coefficients_ * num_channels_) {} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir.h new file mode 100644 index 000000000..fd7efe72e --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir.h @@ -0,0 +1,88 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_MULTI_CHANNEL_IIR_H_ +#define RESONANCE_AUDIO_DSP_MULTI_CHANNEL_IIR_H_ + +#include <memory> +#include <vector> + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Class that performs IIR filtering on interleaved data. This class can be used +// to implement any order of IIR filter on multichannel data where the number of +// channels is a multiple of the SIMD vector length. +class MultiChannelIir { + public: + // Returns a |MultiChannelIir| given valid parameters. + // + // @param num_channels Number of channels in each input buffer. The number of + // channels must be divisible by |SIMD_LENGTH|. + // @param frames_per_buffer Number of frames in each input buffer. + // @param numerators Numerator coefficients, one set per channel. + // @param denominators Denominator coefficients, should be equal in length to + // the |numerator| vector, one set per channel. + // @return A |MultiChannelIir| instance. + static std::unique_ptr<MultiChannelIir> Create( + size_t num_channels, size_t frames_per_buffer, + const std::vector<std::vector<float>>& numerators, + const std::vector<std::vector<float>>& denominators); + + // Processes an interleaved buffer of input data with the given IIR filter. + // + // @param interleaved_buffer A single channel of data containing input in + // interleaved format, this will contain output data in interleaved format + // on return. + void Process(AudioBuffer::Channel* interleaved_buffer); + + private: + // Constructs a |MultiChannelIir|. + // + // @param num_channels Number of channels in each input buffer. The number of + // channels must be divisible by |SIMD_LENGTH|. + // @param frames_per_buffer Number of frames in each input buffer. + // @param num_coefficients Number of coefficients in the numerator, which + // equals the number in the denominator. + MultiChannelIir(size_t num_channels, size_t frames_per_buffer, + size_t num_coefficients); + + // Number of channels in each input buffer. + const size_t num_channels_; + + // Number of frames in each input buffer. + const size_t frames_per_buffer_; + + // Number of coefficients in the numerator and denominator polynomials. + const size_t num_coefficients_; + + // Current front of the delay line which is circularly indexed. + size_t delay_line_front_; + + // Stores numerator coefficients in repeated fashion. + AudioBuffer numerator_; + + // Stores denominator coefficients in repeated fashion. + AudioBuffer denominator_; + + // Holds previous data computed from the numerator section. + AudioBuffer delay_line_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_MULTI_CHANNEL_IIR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir_test.cc new file mode 100644 index 000000000..3c291196e --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/multi_channel_iir_test.cc @@ -0,0 +1,117 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/multi_channel_iir.h" + +#include <memory> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "base/logging.h" + +namespace vraudio { + +namespace { + +const size_t kFramesPerBuffer = 4; + +// Fills a channel buffer with each of the values passed |num_channels| times. +void FillChannels(const std::vector<float>& values, size_t num_channels, + AudioBuffer::Channel* buffer) { + DCHECK_EQ(values.size() * num_channels, buffer->size()); + size_t offset = 0; + for (auto value : values) { + std::fill_n(buffer->begin() + offset, num_channels, value); + offset += num_channels; + } +} + +// Compares the contents of the |input| to the corresponding section of the +// |expected| vector. +void CompareMultiChannelOutput(const std::vector<std::vector<float>>& expected, + size_t offset, size_t num_channels, + const AudioBuffer::Channel& input) { + DCHECK_GE(expected[0].size(), kFramesPerBuffer + offset); + for (size_t channel = 0; channel < num_channels; ++channel) { + for (size_t frame = 0; frame < kFramesPerBuffer; ++frame) { + EXPECT_NEAR(expected[channel][frame + offset], + input[frame * num_channels + channel], kEpsilonFloat); + } + } +} + +// Tests that the |MultiChannelIir| can filter a single channel of input +// simultaneously with four different biquad filters. +TEST(MultiChannelIirTest, MultipleFilterSetProcessTest) { + const std::vector<std::vector<float>> numerators({{1.0f, 3.0f, 5.0f}, + {2.0f, 2.0f, 4.0f}, + {1.0f, 2.0f, 2.0f}, + {2.0f, 4.0f, 2.0f}}); + const std::vector<std::vector<float>> denominators({{1.0f, 1.0f, 0.0f}, + {2.0f, 2.0f, 0.0f}, + {1.0f, 1.0f, 0.0f}, + {2.0f, 2.0f, 0.0f}}); + + const std::vector<float> initial_input({1.0f, 4.0f, 6.0f, 0.0f}); + const std::vector<float> zero_input({0.0f, 0.0f, 0.0f, 0.0f}); + // These values have been determined through the MATLAB commands: + // filter([1 3 5], [1 1 0], [1 4 6 0, 0 0 0 0, 0 0 0 0]) + // filter([2 2 4], [2 2 0], [1 4 6 0, 0 0 0 0, 0 0 0 0]) + // filter([1 2 2], [1 1 0], [1 4 6 0, 0 0 0 0, 0 0 0 0]) + // filter([2 4 2], [2 2 0], [1 4 6 0, 0 0 0 0, 0 0 0 0]) + const std::vector<std::vector<float>> kExpectedOutputs = { + {1.0f, 6.0f, 17.0f, 21.0f, 9.0f, -9.0f, 9.0f, -9.0f, 9.0f, -9.0f, 9.0f, + -9.0f}, + {1.0f, 4.0f, 8.0f, 6.0f, 6.0f, -6.0f, 6.0f, -6.0f, 6.0f, -6.0f, 6.0f, + -6.0f}, + {1.0f, 5.0f, 11.0f, 9.0f, 3.0f, -3.0f, 3.0f, -3.0f, 3.0f, -3.0f, 3.0f, + -3.0f}, + {1.0f, 5.0f, 10.0f, 6.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, + 0.0f}}; + + const size_t num_channels = numerators.size(); + + AudioBuffer interleaved_buffer(kNumMonoChannels, + num_channels * kFramesPerBuffer); + std::unique_ptr<MultiChannelIir> filter = MultiChannelIir::Create( + num_channels, kFramesPerBuffer, numerators, denominators); + + // Filter the initial buffer [1 4 6 0], and compare the output to the first + // kFramesPerBuffer entries in |kExpectedOutputs|. + FillChannels(initial_input, num_channels, &interleaved_buffer[0]); + filter->Process(&(interleaved_buffer[0])); + CompareMultiChannelOutput(kExpectedOutputs, /*offset*/ 0, num_channels, + interleaved_buffer[0]); + + // Filter zeros [0 0 0 0], and compare the output to the next kFramesPerBuffer + // entries in |kExpectedOutputs|. + FillChannels(zero_input, num_channels, &interleaved_buffer[0]); + filter->Process(&(interleaved_buffer[0])); + CompareMultiChannelOutput(kExpectedOutputs, kFramesPerBuffer, num_channels, + interleaved_buffer[0]); + + // Filter zeros [0 0 0 0], and compare the output to the next kFramesPerBuffer + // entries in |kExpectedOutputs|. + FillChannels(zero_input, num_channels, &interleaved_buffer[0]); + filter->Process(&(interleaved_buffer[0])); + CompareMultiChannelOutput(kExpectedOutputs, 2 * kFramesPerBuffer, + num_channels, interleaved_buffer[0]); +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/near_field_processor.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/near_field_processor.cc new file mode 100644 index 000000000..90afcc472 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/near_field_processor.cc @@ -0,0 +1,97 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/near_field_processor.h" + +#include "base/constants_and_types.h" +#include "dsp/filter_coefficient_generators.h" +#include "dsp/gain.h" + +namespace vraudio { + +namespace { + +// Cross-over frequency of the band-splitting filter. +const float kCrossOverFrequencyHz = 1000.0f; + +// +6dB bass boost factor converted to linear scale. +const float kBassBoost = 2.0f; + +// Average group delay of the HRTF filters in seconds. Please see +// [internal ref] +const float kMeanHrtfGroupDelaySeconds = 0.00066667f; + +// Average group delay of the shelf-filter in samples. +const size_t kMeanShelfFilterGroupDelaySamples = 1; + +} // namespace + +NearFieldProcessor::NearFieldProcessor(int sample_rate, + size_t frames_per_buffer) + : frames_per_buffer_(frames_per_buffer), + delay_compensation_(static_cast<size_t>(kMeanHrtfGroupDelaySeconds * + static_cast<float>(sample_rate)) - + kMeanShelfFilterGroupDelaySamples), + lo_pass_filter_(BiquadCoefficients(), frames_per_buffer_), + hi_pass_filter_(BiquadCoefficients(), frames_per_buffer_), + low_passed_buffer_(kNumMonoChannels, frames_per_buffer_), + delay_filter_(delay_compensation_, frames_per_buffer_) { + DCHECK_GT(sample_rate, 0); + DCHECK_GT(frames_per_buffer, 0); + DCHECK_LT(kCrossOverFrequencyHz, 0.5f * static_cast<float>(sample_rate)); + + // Generate biquad coefficients and construct low- and high-pass filter + // states. + BiquadCoefficients lo_pass_coefficients; + BiquadCoefficients hi_pass_coefficients; + ComputeDualBandBiquadCoefficients(sample_rate, kCrossOverFrequencyHz, + &lo_pass_coefficients, + &hi_pass_coefficients); + + // Create two biquad filters initialized with the above filter coefficients. + lo_pass_filter_.SetCoefficients(lo_pass_coefficients); + hi_pass_filter_.SetCoefficients(hi_pass_coefficients); +} + +void NearFieldProcessor::Process(const AudioBuffer::Channel& input, + AudioBuffer::Channel* output, + bool enable_hrtf) { + + DCHECK(output); + DCHECK_EQ(input.size(), frames_per_buffer_); + DCHECK_EQ(output->size(), frames_per_buffer_); + + // Low-pass filter the input and put it in the temporary low-passed buffer. + auto* low_passed_channel = &low_passed_buffer_[0]; + lo_pass_filter_.Filter(input, low_passed_channel); + + // High-pass filter the input and put it in the output channel (unmodified). + hi_pass_filter_.Filter(input, output); + // Iterate through all the samples in the |low_passed_buffer_| and apply + // the bass boost. Then, combine with the high-passed part in order to form + // the shelf-filtered output. Note: phase flip of the low-passed signal is + // required to form the correct filtered output. + ConstantGain(/*offset_index=*/0, -kBassBoost, *low_passed_channel, output, + /*accumulate_output=*/true); + + if (enable_hrtf) { + // Delay the output to compensate for the average HRTF group delay. + delay_filter_.InsertData(*output); + delay_filter_.GetDelayedData(delay_compensation_, output); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/near_field_processor.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/near_field_processor.h new file mode 100644 index 000000000..86a3093d5 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/near_field_processor.h @@ -0,0 +1,75 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_NEAR_FIELD_PROCESSOR_H_ +#define RESONANCE_AUDIO_DSP_NEAR_FIELD_PROCESSOR_H_ + +#include "base/audio_buffer.h" +#include "dsp/biquad_filter.h" +#include "dsp/delay_filter.h" + +namespace vraudio { + +// Class which applies an approximate near field effect to a source mono input. +// The effect consists of a +6dB bass boost (shelf-filter) as well as phase +// correction (HRTF group delay compensation) when the signal is to be combined +// with the binaural output. +// +// For more information: [internal ref] +class NearFieldProcessor { + public: + // Constructor of the |NearFieldProcessor|, uses the following parameters: + // + // @param sample_rate Sampling rate in [Hz]. + // @param frames_per_buffer Number of frames per buffer in the input/output + // signal. + NearFieldProcessor(int sample_rate, size_t frames_per_buffer); + + // Returns currently used delay compensation in samples. + size_t GetDelayCompensation() const { return delay_compensation_; } + + // Applies approximate near field effect to the source mono input signal. + // + // @param input Mono input channel. + // @param output Pointer to mono output channel. + // @param enable_hrtf Whether to enable delay compensation for HRTF filtering. + void Process(const AudioBuffer::Channel& input, AudioBuffer::Channel* output, + bool enable_hrtf); + + private: + // Number of frames per buffer. + const size_t frames_per_buffer_; + + // Delay compensation computed as average group delay of the HRTF filter + // minus average group delay of the shelf-filter. Should be disabled when + // using with stereo-panned sound sources. + const size_t delay_compensation_; + + // Biquad filters that apply frequency splitting of the input mono signal. + BiquadFilter lo_pass_filter_; + BiquadFilter hi_pass_filter_; + + // Buffer for the low-passed signal. We do not modify the high-passed signal + // so we can write it directly to the output channel. + AudioBuffer low_passed_buffer_; + + // Delay filter used to delay the incoming input mono buffer. + DelayFilter delay_filter_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_NEAR_FIELD_PROCESSOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator.cc new file mode 100644 index 000000000..9c8d92c1b --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator.cc @@ -0,0 +1,52 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/occlusion_calculator.h" + +#include <cmath> + +#include "base/logging.h" +#include "base/misc_math.h" + +namespace vraudio { + +float CalculateDirectivity(float alpha, float order, + const SphericalAngle& spherical_angle) { + // Clamp alpha weighting. + const float alpha_clamped = std::min(std::max(alpha, 0.0f), 1.0f); + + // Check for zero-valued alpha (omnidirectional). + if (alpha_clamped < std::numeric_limits<float>::epsilon()) { + return 1.0f; + } else { + const float gain = (1.0f - alpha_clamped) + + alpha_clamped * (std::cos(spherical_angle.azimuth()) * + std::cos(spherical_angle.elevation())); + + return std::pow(std::abs(gain), std::max(order, 1.0f)); + } +} + +float CalculateOcclusionFilterCoefficient(float directivity, + float occlusion_intensity) { + DCHECK_GE(occlusion_intensity, 0.0f); + + const float occlusion_factor = + 1.0f / IntegerPow(occlusion_intensity + 1.0f, 4); + return std::max(0.0f, 1.0f - directivity * occlusion_factor); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator.h new file mode 100644 index 000000000..924da9068 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator.h @@ -0,0 +1,54 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_OCCLUSION_CALCULATOR_H_ +#define RESONANCE_AUDIO_DSP_OCCLUSION_CALCULATOR_H_ + +#include "base/spherical_angle.h" + +namespace vraudio { + +// Calculates directivity gain value for supplied source and listener +// parameters. +// +// @param alpha Balance between dipole pattern and omnidirectional pattern for +// source emission. By varying this value, differing directivity patterns +// can be formed. Value in range [0, 1]. 2D visualization for several +// values: http://goo.gl/GhKvoc. +// @param order Order of directivity function. Higher values will result in +// increased directivity. Value in range [1, +inf]. 2D visualization for +// several orders: +// http://goo.gl/sNrm1a. +// @param spherical_angle Spherical angle of the listener relative to the +// audio source which is being shaped. +// @return Gain value in range [0, 1]. +float CalculateDirectivity(float alpha, float order, + const SphericalAngle& spherical_angle); + +// This function calculates a |MonoPoleFilter| coefficient based upon the +// directivity and occlusion values. The coefficient calculation was designed +// via empirical methods. +// +// @param directivity Gain value calculated based upon the directivity. +// @param occlusion_intensity Gain value calculated based upon the degree of +// occlusion. +// @return Filter coefficient for a mono pole low pass filter. +float CalculateOcclusionFilterCoefficient(float directivity, + float occlusion_intensity); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_OCCLUSION_CALCULATOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator_test.cc new file mode 100644 index 000000000..4de2d07ad --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/occlusion_calculator_test.cc @@ -0,0 +1,139 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/occlusion_calculator.h" + +#include <algorithm> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "base/misc_math.h" + +namespace vraudio { + +namespace { + +// Spherical coordinates of listener relative to source. +const float kListenerAheadRads[2] = {0.0f, 0.0f}; +const float kListenerBesideRads[2] = {0.0f, static_cast<float>(M_PI_2)}; +const float kListenerAboveAndAheadRads[2] = {static_cast<float>(M_PI_4), 0.0f}; + +// Test case data structure. +struct DirectivityTestParams { + // Directivity function's alpha weighting. + const float alpha; + // Directivity function's order. + const float order; + // Spherical angle of the listener relative to the source. + const float* relative_spherical_angle; + // Expected value produced by the directivity function for the given input + // parameters. + const float expected_directivity; +}; + +// This test runs through a series of parametrized test cases and checks that +// the directivity value calculated according to those parameters is correct. +class DirectivityCalculatorParametrizedTest + : public testing::Test, + public testing::WithParamInterface<DirectivityTestParams> {}; + +// Parametrized test which verifies that the directivity value calculated from +// the test parameters matches the expected value supplied. +TEST_P(DirectivityCalculatorParametrizedTest, CalculateDirectivityTest) { + // Test parameters. + const DirectivityTestParams test_params = GetParam(); + + // Construct test angle. + const float elevation_rad = test_params.relative_spherical_angle[0]; + const float azimuth_rad = test_params.relative_spherical_angle[1]; + SphericalAngle test_angle(elevation_rad, azimuth_rad); + + // Calculate directivity. + const float directivity = + CalculateDirectivity(test_params.alpha, test_params.order, test_angle); + + // Check calculated directivity. + EXPECT_NEAR(test_params.expected_directivity, directivity, kEpsilonFloat); +} + +// Test parameters, according to struct |DirectivityParams|. +DirectivityTestParams test_cases[] = { + // Omnidirectional. + {0.0f, 1.0f, &(kListenerAheadRads[0]), 1.0f}, + {0.0f, 1.0f, &(kListenerBesideRads[0]), 1.0f}, + {0.0f, 1.0f, &(kListenerAboveAndAheadRads[0]), 1.0f}, + {0.0f, 2.0f, &(kListenerAboveAndAheadRads[0]), 1.0f}, + {0.0f, 0.5f, &(kListenerAboveAndAheadRads[0]), 1.0f}, + // Hypocardioid. + {0.25f, 1.0f, &(kListenerAheadRads[0]), 1.0f}, + {0.25f, 1.0f, &(kListenerBesideRads[0]), 0.75f}, + {0.25f, 1.0f, &(kListenerAboveAndAheadRads[0]), 0.926777f}, + {0.25f, 2.0f, &(kListenerAboveAndAheadRads[0]), 0.858915f}, + {0.25f, 0.5f, &(kListenerAboveAndAheadRads[0]), 0.926777f}, + // Cardioid. + {0.5f, 1.0f, &(kListenerAheadRads[0]), 1.0f}, + {0.5f, 1.0f, &(kListenerBesideRads[0]), 0.5f}, + {0.5f, 1.0f, &(kListenerAboveAndAheadRads[0]), 0.853553f}, + {0.5f, 2.0f, &(kListenerAboveAndAheadRads[0]), 0.728553f}, + // Hypercardioid. + {0.75f, 1.0f, &(kListenerAheadRads[0]), 1.0f}, + {0.75f, 1.0f, &(kListenerBesideRads[0]), 0.25f}, + {0.75f, 1.0f, &(kListenerAboveAndAheadRads[0]), 0.780330f}, + {0.75f, 2.0f, &(kListenerAboveAndAheadRads[0]), 0.608915f}, + {0.75f, 0.5f, &(kListenerAboveAndAheadRads[0]), 0.780330f}, + // Dipole. + {1.0f, 1.0f, &(kListenerAheadRads[0]), 1.0f}, + {1.0f, 1.0f, &(kListenerBesideRads[0]), 0.0f}, + {1.0f, 1.0f, &(kListenerAboveAndAheadRads[0]), 0.707107f}, + {1.0f, 2.0f, &(kListenerAboveAndAheadRads[0]), 0.5f}, + {1.0f, 0.5f, &(kListenerAboveAndAheadRads[0]), 0.707107f}}; + +INSTANTIATE_TEST_CASE_P(Instance, DirectivityCalculatorParametrizedTest, + testing::ValuesIn(test_cases)); + +TEST(DirectivityTest, CalculateOcclusionFilterCoefficientTest) { + // When there is no occlusion (occlusion == 0) expect the filter coefficient + // to be 1 - directivity. + const std::vector<float> directivities = {0.0f, 0.3f, 0.5f, 0.7f, 0.9f, 1.5f}; + for (size_t i = 0; i < directivities.size() - 1; ++i) { + const float coefficient = + CalculateOcclusionFilterCoefficient(directivities[i], 0.0f); + EXPECT_EQ(1.0f - directivities[i], coefficient); + } + // Ensure the minimum value returned is 0. + const float coefficient = + CalculateOcclusionFilterCoefficient(directivities.back(), 0.0f); + EXPECT_EQ(0.0f, coefficient); +} + +TEST(OcclusionTest, CalculateOcclusionFilterCoefficientTest) { + // When there is no effect on directivity, expect the filter coefficient + // to be 1 / (x + 1)^4 where x is the occlusion intensity. + const std::vector<float> occlusions = {0.01f, 0.1f, 1.0f, 10.0f, 100.0f}; + std::vector<float> expected_coefficients = occlusions; + // Calculate 1 / (x + 1)^4 where x is the occlusion intensity. + std::for_each(expected_coefficients.begin(), expected_coefficients.end(), + [](float& n) { n = 1.0f - 1.0f / IntegerPow(n + 1.0f, 4); }); + for (size_t i = 0; i < expected_coefficients.size(); ++i) { + const float coefficient = + CalculateOcclusionFilterCoefficient(1.0f, occlusions[i]); + EXPECT_EQ(expected_coefficients[i], coefficient); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter.cc new file mode 100644 index 000000000..4a2f071f8 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter.cc @@ -0,0 +1,267 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +// Prevent Visual Studio from complaining about std::copy_n. +#if defined(_WIN32) +#define _SCL_SECURE_NO_WARNINGS +#endif + +#include "dsp/partitioned_fft_filter.h" + +#include <algorithm> + +#include "pffft.h" +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/misc_math.h" +#include "base/simd_utils.h" + +#include "dsp/utils.h" + +namespace vraudio { + +PartitionedFftFilter::PartitionedFftFilter(size_t filter_size, + size_t frames_per_buffer, + FftManager* fft_manager) + : PartitionedFftFilter(filter_size, frames_per_buffer, filter_size, + fft_manager) {} + +PartitionedFftFilter::PartitionedFftFilter(size_t filter_size, + size_t frames_per_buffer, + size_t max_filter_size, + FftManager* fft_manager) + : fft_manager_(fft_manager), + fft_size_(fft_manager_->GetFftSize()), + chunk_size_(fft_size_ / 2), + frames_per_buffer_(frames_per_buffer), + max_filter_size_( + CeilToMultipleOfFramesPerBuffer(max_filter_size, frames_per_buffer_)), + max_num_partitions_(max_filter_size_ / frames_per_buffer_), + filter_size_( + CeilToMultipleOfFramesPerBuffer(filter_size, frames_per_buffer_)), + num_partitions_(filter_size_ / frames_per_buffer_), + kernel_freq_domain_buffer_(max_num_partitions_, fft_size_), + buffer_selector_(0), + curr_front_buffer_(0), + freq_domain_buffer_(max_num_partitions_, fft_size_), + filtered_time_domain_buffers_(kNumStereoChannels, fft_size_), + freq_domain_accumulator_(kNumMonoChannels, fft_size_), + temp_zeropad_buffer_(kNumMonoChannels, chunk_size_), + temp_kernel_chunk_buffer_(kNumMonoChannels, frames_per_buffer_) { + // Ensure that |frames_per_buffer_| is less than or equal to the final + // partition |chunk_size_|. + CHECK(fft_manager_); + CHECK_LE(frames_per_buffer_, chunk_size_); + CHECK_GE(filter_size_, filter_size); + CHECK_GE(max_filter_size_, max_filter_size); + // Ensure that |filter_size_| does not exceed |max_filter_size_|. + CHECK_LE(filter_size, max_filter_size_); + // Make sure all partitions have the same size. + CHECK_EQ(num_partitions_ * frames_per_buffer_, filter_size_); + CHECK_EQ(max_num_partitions_ * frames_per_buffer_, max_filter_size_); + + Clear(); +} + +void PartitionedFftFilter::Clear() { + // Reset valid part of the filter |FreqDomainBuffer|s to zero. + for (size_t i = 0; i < num_partitions_; ++i) { + kernel_freq_domain_buffer_[i].Clear(); + freq_domain_buffer_[i].Clear(); + } + // Reset filter state to zero. + filtered_time_domain_buffers_.Clear(); +} + +void PartitionedFftFilter::ResetFreqDomainBuffers(size_t new_filter_size) { + + // Update the filter size. + DCHECK_GT(new_filter_size, 0U); + filter_size_ = + CeilToMultipleOfFramesPerBuffer(new_filter_size, frames_per_buffer_); + DCHECK_LE(filter_size_, max_filter_size_); + + const size_t old_num_partitions = num_partitions_; + num_partitions_ = filter_size_ / frames_per_buffer_; + const size_t min_num_partitions = + std::min(old_num_partitions, num_partitions_); + + if (curr_front_buffer_ > 0) { + + FreqDomainBuffer temp_freq_domain_buffer(min_num_partitions, fft_size_); + // Copy in |min_num_partitions| to |temp_freq_domain_buffer|, starting with + // the partition at |curr_front_buffer_| to be moved back to the beginning + // of |freq_domain_buffer| . + for (size_t i = 0; i < min_num_partitions; ++i) { + temp_freq_domain_buffer[i] = + freq_domain_buffer_[(curr_front_buffer_ + i) % old_num_partitions]; + } + // Replace the partitions. + for (size_t i = 0; i < min_num_partitions; ++i) { + freq_domain_buffer_[i] = temp_freq_domain_buffer[i]; + } + curr_front_buffer_ = 0; + } + // Clear out the remaining partitions in case the filter size grew. + for (size_t i = old_num_partitions; i < num_partitions_; ++i) { + freq_domain_buffer_[i].Clear(); + } +} + +void PartitionedFftFilter::ReplacePartition( + size_t partition_index, const AudioBuffer::Channel& kernel_chunk) { + DCHECK_GE(partition_index, 0U); + DCHECK_LT(partition_index, num_partitions_); + DCHECK_EQ(kernel_chunk.size(), frames_per_buffer_); + + fft_manager_->FreqFromTimeDomain( + kernel_chunk, &kernel_freq_domain_buffer_[partition_index]); +} + +void PartitionedFftFilter::SetFilterLength(size_t new_filter_size) { + DCHECK_GT(new_filter_size, 0U); + new_filter_size = + CeilToMultipleOfFramesPerBuffer(new_filter_size, frames_per_buffer_); + DCHECK_LE(new_filter_size, max_filter_size_); + + const size_t new_num_partitions = new_filter_size / frames_per_buffer_; + DCHECK_LE(new_num_partitions, max_num_partitions_); + + // Clear out the remaining partitions in case the filter size grew. + for (size_t i = num_partitions_; i < new_num_partitions; ++i) { + kernel_freq_domain_buffer_[i].Clear(); + } + // Call |ResetFreqDomainBuffers| to make sure that the input buffers are also + // correctly resized. + ResetFreqDomainBuffers(new_filter_size); +} + +void PartitionedFftFilter::SetTimeDomainKernel( + const AudioBuffer::Channel& kernel) { + + + // Precomputes a set of floor(|filter_size_|/(|fft_size|/2)) frequency domain + // kernels, one for each partition of the |kernel|. This allows to reduce + // computational complexity if a fixed set of filter kernels is needed. The + // size of the |kernel| can be arbitrarily long but must not be less than + // |fft_size_|/2. Filter lengths which are multiples of |fft_size_|/2 are most + // efficient. + + // Calculate the new number of partitions as filter length may have changed. + // This number is set to 1 if the filter length is smaller than + // |frames_per_buffer_|. + const size_t new_num_partitions = + CeilToMultipleOfFramesPerBuffer(kernel.size(), frames_per_buffer_) / + frames_per_buffer_; + + auto& padded_channel = temp_kernel_chunk_buffer_[0]; + // Break up time domain filter into chunks and FFT each of these separately. + for (size_t partition = 0; partition < new_num_partitions; ++partition) { + DCHECK_LE(partition * frames_per_buffer_, kernel.size()); + const float* chunk_begin_itr = + kernel.begin() + partition * frames_per_buffer_; + const size_t num_frames_to_copy = + std::min<size_t>(frames_per_buffer_, kernel.end() - chunk_begin_itr); + + std::copy_n(chunk_begin_itr, num_frames_to_copy, padded_channel.begin()); + // This fill only occurs on the very last partition. + std::fill(padded_channel.begin() + num_frames_to_copy, padded_channel.end(), + 0.0f); + fft_manager_->FreqFromTimeDomain(padded_channel, + &kernel_freq_domain_buffer_[partition]); + } + + if (new_num_partitions != num_partitions_) { + const size_t new_filter_size = new_num_partitions * frames_per_buffer_; + ResetFreqDomainBuffers(new_filter_size); + } +} + +void PartitionedFftFilter::SetFreqDomainKernel(const FreqDomainBuffer& kernel) { + DCHECK_LE(kernel.num_channels(), max_num_partitions_); + DCHECK_EQ(kernel.num_frames(), fft_size_); + + const size_t new_num_partitions = kernel.num_channels(); + for (size_t i = 0; i < new_num_partitions; ++i) { + kernel_freq_domain_buffer_[i] = kernel[i]; + } + if (new_num_partitions != num_partitions_) { + const size_t new_filter_size = new_num_partitions * frames_per_buffer_; + ResetFreqDomainBuffers(new_filter_size); + } +} + +void PartitionedFftFilter::Filter(const FreqDomainBuffer::Channel& input) { + + + DCHECK_EQ(input.size(), fft_size_); + std::copy_n(input.begin(), fft_size_, + freq_domain_buffer_[curr_front_buffer_].begin()); + buffer_selector_ = !buffer_selector_; + freq_domain_accumulator_.Clear(); + auto* accumulator_channel = &freq_domain_accumulator_[0]; + + for (size_t i = 0; i < num_partitions_; ++i) { + // Complex vector product in frequency domain with filter kernel. + const size_t modulo_index = (curr_front_buffer_ + i) % num_partitions_; + + // Perform inverse scaling along with accumulation of last fft buffer. + fft_manager_->FreqDomainConvolution(freq_domain_buffer_[modulo_index], + kernel_freq_domain_buffer_[i], + accumulator_channel); + } + // Our modulo based index. + curr_front_buffer_ = + (curr_front_buffer_ + num_partitions_ - 1) % num_partitions_; + // Perform inverse FFT transform of |freq_domain_buffer_| and store the + // result back in |filtered_time_domain_buffers_|. + fft_manager_->TimeFromFreqDomain( + *accumulator_channel, &filtered_time_domain_buffers_[buffer_selector_]); +} + +void PartitionedFftFilter::GetFilteredSignal(AudioBuffer::Channel* output) { + + DCHECK(output); + DCHECK_EQ(output->size(), frames_per_buffer_); + + const size_t curr_buffer = buffer_selector_; + const size_t prev_buffer = !buffer_selector_; + + // Overlap add. + if (frames_per_buffer_ == chunk_size_) { + AddPointwise(chunk_size_, &(filtered_time_domain_buffers_[curr_buffer][0]), + &filtered_time_domain_buffers_[prev_buffer][chunk_size_], + &((*output)[0])); + } else { + // If we have a non power of two |frames_per_buffer| we will have to perform + // the overlap add and then a copy, as buffer lengths are not a multiple of + // |chunk_size_|. NOTE: Indexing into |filtered_time_domain_buffers_| for a + // non power of two |frames_per_buffer_| means that the |input_b| parameter + // of |AddPointwiseOutOfPlace| may not be aligned in memory and thus we will + // not be able to use SIMD for the overlap add operation. + const auto& first_channel = filtered_time_domain_buffers_[curr_buffer]; + const auto& second_channel = filtered_time_domain_buffers_[prev_buffer]; + auto& output_channel = temp_zeropad_buffer_[0]; + for (size_t i = 0; i < frames_per_buffer_; ++i) { + output_channel[i] = + first_channel[i] + second_channel[i + frames_per_buffer_]; + } + std::copy_n(temp_zeropad_buffer_[0].begin(), frames_per_buffer_, + output->begin()); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter.h new file mode 100644 index 000000000..f8399922a --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter.h @@ -0,0 +1,167 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_PARTITIONED_FFT_FILTER_H_ +#define RESONANCE_AUDIO_DSP_PARTITIONED_FFT_FILTER_H_ + +#include <vector> + +#include "base/audio_buffer.h" +#include "base/misc_math.h" +#include "dsp/fft_manager.h" + +struct PFFFT_Setup; + +namespace vraudio { + +// Class performing a FFT-based overlap and add FIR convolution. +// Given an FFT size N and a filter size M > N/2; the filter is broken in to +// floor(M/(N/2)) partitions in the time domain. This results in a set of +// frequency domain "filters" of length (N/2)+1. +class PartitionedFftFilter { + public: + // Typedef declares the data type for storing frequency domain buffers. Each + // channel stores the kernel for a partition. + typedef AudioBuffer FreqDomainBuffer; + + // Constructor preallocates memory based on the |filter_size|. This can be + // used for simplicity if the filter size will be constant after creation. + // + // @param filter_size Length of the time domain filter in samples. This will + // be increased such that it becomes a multiple of |chunk_size_|. + // @param frames_per_buffer Number of points in each time domain input buffer. + // @param fft_manager Pointer to a manager to perform FFT transformations. + PartitionedFftFilter(size_t filter_size, size_t frames_per_buffer, + FftManager* fft_manager); + + // Constructor preallocates memory based on the |max_filter_size|. The + // |fft_size_| will be twice |frames_per_buffer| if this is a power of two and + // twice the next larger power of two if it is not. + // + // @param filter_size Length of the time domain filter in samples. This will + // be increased such that it becomes a multiple of |chunk_size_|. + // @param frames_per_buffer Number of points in each time domain input buffer. + // @param max_filter_size Maximum length that |filter_size| can get. + // @param fft_manager Pointer to a manager for all fft related functionality. + PartitionedFftFilter(size_t filter_size, size_t frames_per_buffer, + size_t max_filter_size, FftManager* fft_manager); + + // Initializes the FIR filter from a time domain kernel. + // + // @parem kernel Time domain filter to be used for processing. + void SetTimeDomainKernel(const AudioBuffer::Channel& kernel); + + // Initializes the FIR filter from a precomputed frequency domain kernel. + // + // @param kernel Frequency domain filter to be used for processing. + void SetFreqDomainKernel(const FreqDomainBuffer& kernel); + + // Replaces a partition indicated by the |partition_index| with + // |kernel_chunk|'s frequency domain equivalent. + // + // @param partition_index Location (partition) of the time domain filter we + // wish to replace. + // @param kernel_chunk |fft_size_|/2 length chunk of a filter used to + // replace the |partition_index|th partition. + void ReplacePartition(size_t partition_index, + const AudioBuffer::Channel& kernel_chunk); + + // Alters the filter length by adding or removing partitions in the frequency + // domain. If |new_filter_size| is not a multiple of |chunk_size_| (i.e. + // frames per buffer), then the time domain filter kernel will be zeropadded + // to a multiple of |chunk_size_|. + // + // @param new_filter_size New length of the time domain filter kernel. + void SetFilterLength(size_t new_filter_size); + + // Processes a block of frequency domain samples. The size of the input + // block must be |fft_size_|. + // + // @param Frequency domain input buffer. + void Filter(const FreqDomainBuffer::Channel& input); + + // Returns block of filtered signal output of size |fft_size_|/2. + // + // @param output Time domain block filtered with the given kernel. + void GetFilteredSignal(AudioBuffer::Channel* output); + + // Resets the filter state. + void Clear(); + + private: + friend class PartitionedFftFilterFrequencyBufferTest; + + // Adjusts the |num_partitions_| and the size of |freq_domain_buffers_| for + // use with a new time domain filter kernel greater in length than the + // previous kernel. + // + // @param new_kernel_size Length of the time domain filter kernel. + void ResetFreqDomainBuffers(size_t new_kernel_size); + + // Manager for all FFT related functionality (not owned). + FftManager* const fft_manager_; + + // Number of points in the |fft_manager_|s FFT. + const size_t fft_size_; + + // Size of each partition of the filter in time domain. + const size_t chunk_size_; + + // Number of frames in each buffer of input data. + const size_t frames_per_buffer_; + + // Maximum filter size in samples. + const size_t max_filter_size_; + + // Maximum partition count. + const size_t max_num_partitions_; + + // Filter size in samples. + size_t filter_size_; + + // Partition Count. + size_t num_partitions_; + + // Kernel buffer in frequency domain. + FreqDomainBuffer kernel_freq_domain_buffer_; + + // Buffer selector to switch between two filtered signal buffers. + size_t buffer_selector_; + + // The freq_domain_buffer we will write new incoming audio into. + size_t curr_front_buffer_; + + // Frequency domain buffer used to perform filtering. + FreqDomainBuffer freq_domain_buffer_; + + // Two buffers that are consecutively filled with filtered signal output. + AudioBuffer filtered_time_domain_buffers_; + + // Accumulator for the outputs from each convolution partition + FreqDomainBuffer freq_domain_accumulator_; + + // Temporary time domain buffer to store output when zero padding has been + // applied due to non power of two input buffer lengths. + AudioBuffer temp_zeropad_buffer_; + + // Temporary time domain buffer to hold time domain kernel chunks during + // conversion of a kernel from time to frequency domain. + AudioBuffer temp_kernel_chunk_buffer_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_PARTITIONED_FFT_FILTER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter_test.cc new file mode 100644 index 000000000..230ec2544 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/partitioned_fft_filter_test.cc @@ -0,0 +1,761 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +// Prevent Visual Studio from complaining about std::copy_n. +#if defined(_WIN32) +#define _SCL_SECURE_NO_WARNINGS +#endif + +#include "dsp/partitioned_fft_filter.h" + +#include <cmath> +#include <iostream> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/audio_buffer.h" +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "dsp/fft_manager.h" + +namespace vraudio { + +namespace { + +// Permitted error in the output relative to the expected output. This value is +// 1e-3 as expected output is rounded for readability. +const float kFftEpsilon = 1e-3f; + +// Length of input buffer for use in LongerShorterKernelTest. +const size_t kLength = 32; + +// Helper function for use in LongerShorterKernelTest. Passes a dirac of length +// kLength through a filter followed by |zeros_iteration| zero vectors of the +// same length. +void ProcessFilterWithImpulseSignal(PartitionedFftFilter* filter, + FftManager* fft_manager, + size_t zeros_iteration, + std::vector<float>* output_signal) { + AudioBuffer signal_buffer(kNumMonoChannels, kLength); + signal_buffer.Clear(); + signal_buffer[0][0] = 1.0f; + AudioBuffer output_buffer(kNumMonoChannels, kLength); + + // Begin filtering with the original (small) kernel. + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer(kNumMonoChannels, + kLength * 2); + fft_manager->FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + filter->Filter(freq_domain_buffer[0]); + filter->GetFilteredSignal(&output_buffer[0]); + output_signal->insert(output_signal->end(), output_buffer[0].begin(), + output_buffer[0].end()); + // Filter zeros to flush out internal buffers. + std::fill(signal_buffer[0].begin(), signal_buffer[0].end(), 0.0f); + for (size_t i = 0; i < zeros_iteration; ++i) { + fft_manager->FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + filter->Filter(freq_domain_buffer[0]); + filter->GetFilteredSignal(&output_buffer[0]); + output_signal->insert(output_signal->end(), output_buffer[0].begin(), + output_buffer[0].end()); + } +} + +// Tests that convolution will work correctly when the input buffer length is +// not a power of two. +TEST(PartitionedFftFilterTest, NonPow2Test) { + // Non power of two input buffer length. + const size_t kMinNonPowTwo = 13; + const size_t kMaxNonPowTwo = 41; + for (size_t buffer_length = kMinNonPowTwo; buffer_length <= kMaxNonPowTwo; + buffer_length += 2) { + for (size_t filter_length = kLength; filter_length <= 3 * kLength; + filter_length += kLength) { + // Use a filter length that is a power of two. + AudioBuffer kernel_buffer(kNumMonoChannels, filter_length); + // Place to collect all of the output. + std::vector<float> output_signal; + // First set the kernel to a linear ramp. + for (size_t i = 0; i < filter_length; ++i) { + kernel_buffer[0][i] = static_cast<float>(i) / 4.0f; + } + FftManager fft_manager(buffer_length); + PartitionedFftFilter filter(filter_length, buffer_length, &fft_manager); + filter.SetTimeDomainKernel(kernel_buffer[0]); + + // Kronecker delta signal. + AudioBuffer signal_buffer(kNumMonoChannels, buffer_length); + signal_buffer.Clear(); + signal_buffer[0][0] = 1.0f; + AudioBuffer output_buffer(kNumMonoChannels, buffer_length); + // Create a freq domain buffer which should be fft_size + // (i.e. NextPowTwo(buffer_length) * 2). + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer( + kNumMonoChannels, NextPowTwo(buffer_length) * 2); + freq_domain_buffer.Clear(); + fft_manager.FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + // Perform convolution. + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + while (output_signal.size() < kernel_buffer.num_frames()) { + // Flush with zeros. + freq_domain_buffer.Clear(); + output_buffer.Clear(); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + } + // Ensure the output is identical to the input buffer. (large epsilon + // needed due to large values in filter). + for (size_t i = 0; i < kernel_buffer.num_frames(); ++i) { + EXPECT_NEAR(kernel_buffer[0][i], output_signal[i], + kEpsilonFloat * 4.0f); + } + } + } +} + +// Tests that the outputs from the convolution are correct based on a +// precomputed vector from MATLAB. +TEST(PartitionedFftFilterTest, CorrectNonPowTwoOutputTest) { + const size_t kBufferSize = 15; + + // Create an arbirary vector of size 16 for filter and 15 for input signal. + const std::vector<float> kernel = {1.0f, 3.0f, 0.0f, 2.0f, 5.0f, 1.0f, + 3.0f, 2.0f, 0.0f, 4.0f, 1.0f, 3.0f, + 0.0f, 2.0f, 1.0f, 2.0f}; + const std::vector<float> signal = {2.0f, 3.0f, 3.0f, 4.0f, 0.0f, + 0.0f, 2.0f, 1.0f, 2.0f, 1.0f, + 3.0f, 2.0f, 4.0f, 0.0f, 2.0f}; + // Ideal output vector verified with MATLAB. + const std::vector<float> ideal_output = { + 2.0f, 9.0f, 12.0f, 17.0f, 28.0f, 23.0f, 34.0f, 43.0f, 24.0f, + 37.0f, 40.0f, 43.0f, 57.0f, 49.0f, 50.0f, 57.0f, 65.0f, 57.0f, + 60.0f, 61.0f, 47.0f, 74.0f, 59.0f, 55.0f, 48.0f, 62.0f, 51.0f, + 69.0f, 51.0f, 54.0f, 55.0f, 56.0f, 45.0f, 43.0f, 33.0f, 24.0f, + 40.0f, 16.0f, 31.0f, 11.0f, 22.0f, 8.0f, 12.0f, 2.0f, 4.0f}; + + AudioBuffer kernel_buffer(kNumMonoChannels, kernel.size()); + kernel_buffer[0] = kernel; + + AudioBuffer signal_buffer(kNumMonoChannels, signal.size()); + signal_buffer[0] = signal; + + AudioBuffer output_buffer(kNumMonoChannels, signal.size()); + + std::vector<float> output_signal; + output_signal.reserve(kernel.size() + signal.size() * 3); + + FftManager fft_manager(kBufferSize); + PartitionedFftFilter filter(kernel.size(), kBufferSize, &fft_manager); + filter.SetTimeDomainKernel(kernel_buffer[0]); + + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer( + 1, NextPowTwo(kBufferSize) * 2); + fft_manager.FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + + // Filter again with the same input. + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + + // Filter zeros to flush out internal buffers. + signal_buffer.Clear(); + fft_manager.FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + + for (size_t sample = 0; sample < ideal_output.size(); ++sample) { + EXPECT_NEAR(output_signal[sample], ideal_output[sample], kFftEpsilon); + } +} + +// Tests that we can switch to a time domain filter kernel of greater +// length than the original kernal provided at instantiation. It then tests that +// we can switch to a time domain filter kernel of lesser length than the +// original kernel. +TEST(PartitionedFftFilterTest, LongerShorterTimeDomainKernelTest) { + AudioBuffer small_kernel_buffer(kNumMonoChannels, kLength); + AudioBuffer big_kernel_buffer(kNumMonoChannels, kLength * 2); + + // Place to collect all of the output. + std::vector<float> total_output_signal; + + // First set the kernels to linear ramps of differing lengths. + for (size_t i = 0; i < kLength * 2; ++i) { + if (i < kLength) { + small_kernel_buffer[0][i] = static_cast<float>(i) / 4.0f; + } + big_kernel_buffer[0][i] = static_cast<float>(i) / 4.0f; + } + + FftManager fft_manager(kLength); + PartitionedFftFilter filter(small_kernel_buffer.num_frames(), kLength, + big_kernel_buffer.num_frames(), &fft_manager); + + filter.SetTimeDomainKernel(small_kernel_buffer[0]); + ProcessFilterWithImpulseSignal(&filter, &fft_manager, 2, + &total_output_signal); + + filter.SetTimeDomainKernel(big_kernel_buffer[0]); + ProcessFilterWithImpulseSignal(&filter, &fft_manager, 3, + &total_output_signal); + + filter.SetTimeDomainKernel(small_kernel_buffer[0]); + ProcessFilterWithImpulseSignal(&filter, &fft_manager, 2, + &total_output_signal); + + // Test to see if output from both kernels is present kLength * 2 zeros in + // between. + for (size_t i = 0; i < kLength; ++i) { + EXPECT_NEAR(static_cast<float>(i) / 4.0f, total_output_signal[i], + kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + kLength], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 2 * kLength], kFftEpsilon); + EXPECT_NEAR(static_cast<float>(i) / 4.0f, + total_output_signal[i + 3 * kLength], kFftEpsilon); + EXPECT_NEAR(static_cast<float>(i + kLength) / 4.0f, + total_output_signal[i + 4 * kLength], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 5 * kLength], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 6 * kLength], kFftEpsilon); + EXPECT_NEAR(static_cast<float>(i) / 4.0f, + total_output_signal[i + 7 * kLength], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 8 * kLength], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 9 * kLength], kFftEpsilon); + } +} + +// Tests that the outputs from the same convolution performed with different FFT +// sizes will all return equal results, including when the FFT size is equal to +// the kernel size. +TEST(PartitionedFftFilterTest, PartitionSizeInvarianceTest) { + const std::vector<size_t> kFftSizes = {32, 64, 128}; + + // Create an arbirary vector of size 128 for both filter and input signal. + const size_t max_fft_size = kFftSizes[kFftSizes.size() - 1]; + AudioBuffer kernel_buffer(kNumMonoChannels, max_fft_size); + AudioBuffer signal_buffer(kNumMonoChannels, max_fft_size); + for (size_t i = 0; i < max_fft_size; ++i) { + kernel_buffer[0][i] = static_cast<float>(static_cast<int>(i) % 13 - 7); + signal_buffer[0][i] = static_cast<float>(static_cast<int>(i) % 17 - 9); + } + + std::vector<std::vector<float>> output_signal( + kFftSizes.size(), std::vector<float>(max_fft_size * 2)); + + // Iterate over 3 fft sizes. + for (size_t fft_idx = 0; fft_idx < kFftSizes.size(); ++fft_idx) { + const size_t chunk_size = kFftSizes[fft_idx] / 2; + FftManager fft_manager(chunk_size); + PartitionedFftFilter filter(max_fft_size, chunk_size, &fft_manager); + filter.SetTimeDomainKernel(kernel_buffer[0]); + + AudioBuffer input_chunk(kNumMonoChannels, chunk_size); + AudioBuffer output_chunk(kNumMonoChannels, chunk_size); + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer( + 1, kFftSizes[fft_idx]); + + // Break the input signal into chunks of fft size / 2. + for (size_t chunk = 0; chunk < (max_fft_size / chunk_size); ++chunk) { + AudioBuffer signal_block(kNumMonoChannels, chunk_size); + std::copy_n(signal_buffer[0].begin() + (chunk * chunk_size), chunk_size, + signal_block[0].begin()); + + fft_manager.FreqFromTimeDomain(signal_block[0], &freq_domain_buffer[0]); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_chunk[0]); + output_signal[fft_idx].insert(output_signal[fft_idx].end(), + output_chunk[0].begin(), + output_chunk[0].end()); + } + } + // Now test the outputs are pretty much equal to one another (to the first). + for (size_t i = 1; i < kFftSizes.size(); ++i) { + for (size_t sample = 0; sample < output_signal[0].size(); ++sample) { + EXPECT_NEAR(output_signal[0][sample], output_signal[i][sample], + kFftEpsilon); + } + } +} + +// Tests that the outputs from the convolution are correct based on a +// precomputed vector from MATLAB. +TEST(PartitionedFftFilterTest, CorrectOutputTest) { + const size_t kBufferSize = 32; + + // Create an arbirary vector of size 32 for both filter and input signal. + const std::vector<float> kernel = { + 1.0f, 3.0f, 0.0f, 2.0f, 5.0f, 1.0f, 3.0f, 2.0f, 0.0f, 4.0f, 1.0f, + 3.0f, 0.0f, 2.0f, 1.0f, 2.0f, 2.0f, 1.0f, 0.0f, 3.0f, 5.0f, 2.0f, + 3.0f, 0.0f, 1.0f, 4.0f, 2.0f, 0.0f, 1.0f, 0.0f, 2.0f, 1.0f}; + const std::vector<float> signal = { + 2.0f, 1.0f, 3.0f, 3.0f, 2.0f, 4.0f, 2.0f, 1.0f, 3.0f, 4.0f, 5.0f, + 3.0f, 2.0f, 2.0f, 5.0f, 4.0f, 5.0f, 3.0f, 3.0f, 4.0f, 0.0f, 0.0f, + 2.0f, 1.0f, 2.0f, 1.0f, 3.0f, 2.0f, 4.0f, 0.0f, 2.0f, 1.0f}; + + AudioBuffer kernel_buffer(kNumMonoChannels, kernel.size()); + kernel_buffer[0] = kernel; + + AudioBuffer signal_buffer(kNumMonoChannels, signal.size()); + signal_buffer[0] = signal; + + AudioBuffer output_buffer(kNumMonoChannels, kernel.size()); + + std::vector<float> output_signal; + output_signal.reserve(kernel.size() + signal.size()); + + FftManager fft_manager(kBufferSize); + PartitionedFftFilter filter(kernel.size(), kBufferSize, &fft_manager); + filter.SetTimeDomainKernel(kernel_buffer[0]); + + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer(kNumMonoChannels, + kBufferSize * 2); + fft_manager.FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + + // Filter zeros to flush out internal buffers. + signal_buffer.Clear(); + fft_manager.FreqFromTimeDomain(signal_buffer[0], &freq_domain_buffer[0]); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_buffer[0]); + output_signal.insert(output_signal.end(), output_buffer[0].begin(), + output_buffer[0].end()); + + // Ideal output vector verified with MATLAB. + const std::vector<float> kIdeal = { + 2.0f, 7.0f, 6.0f, 16.0f, 23.0f, 23.0f, 42.0f, 36.0f, 38.0f, + 62.0f, 51.0f, 66.0f, 67.0f, 72.0f, 88.0f, 90.0f, 90.0f, 95.0f, + 104.0f, 118.0f, 127.0f, 113.0f, 123.0f, 131.0f, 116.0f, 144.0f, 119.0f, + 126.0f, 138.0f, 138.0f, 153.0f, 142.0f, 130.0f, 125.0f, 137.0f, 142.0f, + 121.0f, 113.0f, 99.0f, 112.0f, 84.0f, 89.0f, 68.0f, 66.0f, 74.0f, + 54.0f, 54.0f, 59.0f, 53.0f, 42.0f, 42.0f, 26.0f, 32.0f, 28.0f, + 18.0f, 15.0f, 19.0f, 9.0f, 12.0f, 5.0f, 4.0f, 4.0f, 1.0f}; + for (size_t sample = 0; sample < kIdeal.size(); ++sample) { + EXPECT_NEAR(output_signal[sample], kIdeal[sample], kFftEpsilon); + } +} + +// Tests that the outputs from the convolution are equal to zero when the inputs +// are all zero. +TEST(PartitionedFftFilterTest, ZeroInputZeroOutputTest) { + const size_t kChunkSize = 16; + + // Create an arbirary vector of size 16 for filter. + const std::vector<float> kernel = {1.0f, 3.0f, 0.0f, 2.0f, 5.0f, 1.0f, + 3.0f, 2.0f, 0.0f, 4.0f, 1.0f, 3.0f, + 0.0f, 2.0f, 1.0f, 2.0f}; + + AudioBuffer kernel_buffer(kNumMonoChannels, kernel.size()); + kernel_buffer[0] = kernel; + + std::vector<float> output_signal; + output_signal.reserve(kernel.size() * 2); + + FftManager fft_manager(kChunkSize); + PartitionedFftFilter filter(kernel_buffer[0].size(), kChunkSize, + &fft_manager); + filter.SetTimeDomainKernel(kernel_buffer[0]); + + AudioBuffer output_chunk(kNumMonoChannels, kChunkSize); + + // Filter zeros to flush out internal buffers. + AudioBuffer zero_signal(kNumMonoChannels, kChunkSize); + zero_signal.Clear(); + + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer(kNumMonoChannels, + kChunkSize * 2); + for (size_t j = 0; j < 2 * kernel.size() / kChunkSize; ++j) { + fft_manager.FreqFromTimeDomain(zero_signal[0], &freq_domain_buffer[0]); + filter.Filter(freq_domain_buffer[0]); + filter.GetFilteredSignal(&output_chunk[0]); + output_signal.insert(output_signal.end(), output_chunk[0].begin(), + output_chunk[0].end()); + } + + // Check that all output samples are practically 0.0. + for (size_t sample = 0; sample < output_signal.size(); ++sample) { + EXPECT_EQ(output_signal[sample], 0.0f); + } +} + +// This test uses a simple shifted dirac impulse as kernel and checks the +// resulting signal delay. +TEST(PartitionedFftFilterTest, DiracImpulseTest) { + const size_t kFilterSize = 32; + const size_t kNumBlocks = 4; + const size_t kSignalSize = kFilterSize * kNumBlocks; + + // Generate a saw tooth signal. + AudioBuffer test_signal(kNumMonoChannels, kSignalSize); + for (size_t i = 0; i < kSignalSize; ++i) { + test_signal[0][i] = static_cast<float>(i % 5); + } + + FftManager fft_manager(kFilterSize); + PartitionedFftFilter fft_filter(kFilterSize, kFilterSize, &fft_manager); + + AudioBuffer kernel(kNumMonoChannels, kFilterSize); + // Construct dirac impulse response. This kernel should result in a delay of + // length "|kFilterSize| / 2". + kernel.Clear(); + kernel[0][kFilterSize / 2] = 1.0f; + fft_filter.SetTimeDomainKernel(kernel[0]); + + std::vector<float> filtered_signal; + filtered_signal.reserve(kSignalSize); + + AudioBuffer filtered_block(kNumMonoChannels, kFilterSize); + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer(kNumMonoChannels, + kFilterSize * 2); + + for (size_t b = 0; b < kNumBlocks; ++b) { + AudioBuffer signal_block(kNumMonoChannels, kFilterSize); + std::copy_n(test_signal[0].begin() + b * kFilterSize, kFilterSize, + signal_block[0].begin()); + + fft_manager.FreqFromTimeDomain(signal_block[0], &freq_domain_buffer[0]); + fft_filter.Filter(freq_domain_buffer[0]); + fft_filter.GetFilteredSignal(&filtered_block[0]); + filtered_signal.insert(filtered_signal.end(), filtered_block[0].begin(), + filtered_block[0].end()); + } + + for (size_t i = 0; i < kFilterSize / 2; ++i) { + // First "filter_size / 2" samples should be zero padded due to the applied + // delay. + EXPECT_NEAR(filtered_signal[i], 0.0f, 1e-5f); + } + for (size_t i = kFilterSize / 2; i < kSignalSize; ++i) { + // Test if signal is delayed by exactly |kFilterSize| / 2 samples. + EXPECT_NEAR(filtered_signal[i], test_signal[0][i - kFilterSize / 2], 1e-5f); + } +} + +} // namespace + +class PartitionedFftFilterFrequencyBufferTest : public ::testing::Test { + protected: + PartitionedFftFilterFrequencyBufferTest() {} + // Virtual methods from ::testing::Test + ~PartitionedFftFilterFrequencyBufferTest() override {} + void SetUp() override {} + void TearDown() override {} + + void SetFreqDomainBuffer(PartitionedFftFilter* filter, + FftManager* fft_manager) { + std::vector<std::vector<float>> values_vectors( + filter->num_partitions_, + std::vector<float>(fft_manager->GetFftSize(), 0.0f)); + for (size_t i = 0; i < filter->num_partitions_; ++i) { + values_vectors[i][0] = static_cast<float>(i + 1); + filter->freq_domain_buffer_[i] = values_vectors[i]; + } + } + + void SetFreqDomainKernel(PartitionedFftFilter* filter) { + // Put a Kronecker delta in the first partition, a kronecker delta times 2 + // in the second, a kronecker delta times 3 in the third, etc... + AudioBuffer kernel(kNumMonoChannels, filter->filter_size_); + for (size_t i = 0; i < filter->num_partitions_; ++i) { + for (size_t sample = 0; sample < filter->chunk_size_; ++sample) { + kernel[0][i * filter->chunk_size_ + sample] = static_cast<float>(i + 1); + } + } + filter->SetTimeDomainKernel(kernel[0]); + } + + void TestFrequencyBufferReset(size_t initial_size, size_t bigger_size, + size_t smaller_size, + PartitionedFftFilter* filter, + FftManager* fft_manager) { + const size_t initial_num_partitions = 2 * initial_size; + const size_t bigger_num_partitions = 2 * bigger_size; + const size_t smaller_num_partitions = 2 * smaller_size; + // Set the |freq_domain_buffer_| inside |filter| such that it has known + // values. + SetFreqDomainBuffer(filter, fft_manager); + AudioBuffer initial_freq_domain_buffer; + initial_freq_domain_buffer = filter->freq_domain_buffer_; + + // Set the current front buffer to something other than zero and less than + // the number of partitions. If |num_partitions_| is 1 it is set to 0. + filter->curr_front_buffer_ = filter->num_partitions_ / 2; + + // Reset the |freq_domain_buffers_| for a new bigger kernel size. + filter->ResetFreqDomainBuffers(bigger_size * fft_manager->GetFftSize()); + AudioBuffer bigger_freq_domain_buffer; + bigger_freq_domain_buffer = filter->freq_domain_buffer_; + // Verify that the input index has been reset. + EXPECT_EQ(filter->curr_front_buffer_, 0U); + + // Set the current front buffer to something other than zero and less than + // the number of partitions. If |num_partitions_| is 1 it is set to 0. + filter->curr_front_buffer_ = filter->num_partitions_ / 2; + + // Reset the |freq_domain_buffers_| for a new smaller kernel size. + filter->ResetFreqDomainBuffers(smaller_size * fft_manager->GetFftSize()); + AudioBuffer smaller_freq_domain_buffer; + smaller_freq_domain_buffer = filter->freq_domain_buffer_; + // Verify that the input index has been reset. + EXPECT_EQ(filter->curr_front_buffer_, 0U); + + // Expect the following to have happened: + // Initially there are 16 partitions with the first element in each channel + // of the |freq_domain_buffers_| in |filter| being 1, 2, ..., 16. + for (size_t i = 0; i < initial_num_partitions; ++i) { + EXPECT_EQ(initial_freq_domain_buffer[i][0], static_cast<float>(i + 1)); + } + + // |curr_front_buffer_| is set to 8. After a reset with a larger kernel we + // expect all of the buffers to have been copied into the new + // |freq_domain_buffers_| starting from |curr_front_buffer_| and wrapping + // round with the remaining channels set to zero. + for (size_t i = 0; i < initial_num_partitions; ++i) { + EXPECT_EQ(bigger_freq_domain_buffer[i][0], + initial_freq_domain_buffer[(initial_size + i) % + initial_num_partitions][0]); + } + for (size_t i = initial_num_partitions; i < bigger_num_partitions; ++i) { + EXPECT_EQ(bigger_freq_domain_buffer[i][0], 0.0f); + } + + // |curr_front_buffer_| is then set to 5. After a reset with a smaller + // kernel ammounting to 10 partitions we expect just the first 10 buffers to + // have been copied into the new |freq_domain_buffers_| starting from + // |curr_front_buffer_| and wrapping round. + for (size_t i = 0; i < smaller_num_partitions; ++i) { + EXPECT_EQ(smaller_freq_domain_buffer[i][0], + bigger_freq_domain_buffer[(bigger_size + i) % + bigger_num_partitions][0]); + } + } + + void TestLongerShorterFrequencyDomainKernel(size_t buffer_size) { + AudioBuffer small_kernel_buffer(kNumMonoChannels, buffer_size); + AudioBuffer big_kernel_buffer(kNumMonoChannels, 2 * buffer_size); + + // Place to collect all of the output. + std::vector<float> total_output_signal; + + // First set the kernels to linear ramps of differing lengths. + for (size_t i = 0; i < 2 * buffer_size; ++i) { + if (i < buffer_size) { + small_kernel_buffer[0][i] = static_cast<float>(i) / 4.0f; + } + big_kernel_buffer[0][i] = static_cast<float>(i) / 4.0f; + } + + FftManager fft_manager(buffer_size); + PartitionedFftFilter filter(buffer_size, buffer_size, 2 * buffer_size, + &fft_manager); + + // Generate the small and large frequency domain buffers. + filter.SetTimeDomainKernel(small_kernel_buffer[0]); + const size_t small_num_partitions = filter.num_partitions_; + PartitionedFftFilter::FreqDomainBuffer small_freq_domain_kernel = + PartitionedFftFilter::FreqDomainBuffer(small_num_partitions, + fft_manager.GetFftSize()); + for (size_t i = 0; i < small_num_partitions; ++i) { + small_freq_domain_kernel[i] = filter.kernel_freq_domain_buffer_[i]; + } + + filter.SetTimeDomainKernel(big_kernel_buffer[0]); + const size_t big_num_partitions = filter.num_partitions_; + PartitionedFftFilter::FreqDomainBuffer big_freq_domain_kernel = + PartitionedFftFilter::FreqDomainBuffer(big_num_partitions, + fft_manager.GetFftSize()); + for (size_t i = 0; i < big_num_partitions; ++i) { + big_freq_domain_kernel[i] = filter.kernel_freq_domain_buffer_[i]; + } + + filter.SetFreqDomainKernel(small_freq_domain_kernel); + ProcessFilterWithImpulseSignal(&filter, &fft_manager, 2, + &total_output_signal); + + filter.SetFreqDomainKernel(big_freq_domain_kernel); + ProcessFilterWithImpulseSignal(&filter, &fft_manager, 3, + &total_output_signal); + + filter.SetFreqDomainKernel(small_freq_domain_kernel); + ProcessFilterWithImpulseSignal(&filter, &fft_manager, 2, + &total_output_signal); + + // Test to see if output from both kernels is present |big_size| zeros in + // between. + for (size_t i = 0; i < buffer_size; ++i) { + EXPECT_NEAR(static_cast<float>(i) / 4.0f, total_output_signal[i], + kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + buffer_size], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 2 * buffer_size], kFftEpsilon); + EXPECT_NEAR(static_cast<float>(i) / 4.0f, + total_output_signal[i + 3 * buffer_size], kFftEpsilon); + EXPECT_NEAR(static_cast<float>(i + buffer_size) / 4.0f, + total_output_signal[i + 4 * buffer_size], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 5 * buffer_size], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 6 * buffer_size], kFftEpsilon); + EXPECT_NEAR(static_cast<float>(i) / 4.0f, + total_output_signal[i + 7 * buffer_size], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 8 * buffer_size], kFftEpsilon); + EXPECT_NEAR(0.0f, total_output_signal[i + 9 * buffer_size], kFftEpsilon); + } + } + + void TestPartitionReplacement(PartitionedFftFilter* filter, + FftManager* fft_manager) { + // Fill the first partition with a kronecker delta in the first partition, a + // kronecker delta times 2 in the second, a kronecker delta times 3 in the + // third, etc... + SetFreqDomainKernel(filter); + // Get a copy of the kernel. + AudioBuffer initial_kernel; + initial_kernel = filter->kernel_freq_domain_buffer_; + + // Create a freq domain kernel chunk with the same values as the final + // partition. + AudioBuffer kernel_chunk(kNumMonoChannels, filter->chunk_size_); + kernel_chunk.Clear(); + AudioBuffer::Channel& kernel_chunk_channel = kernel_chunk[0]; + for (size_t sample = 0; sample < filter->chunk_size_; ++sample) { + kernel_chunk_channel[sample] = + static_cast<float>(filter->num_partitions_); + } + // The partition we will replace. + const size_t kReplacePartitionIndex = 2; + // Replace a partition with a new time domain kernel chunk (all zeros). + filter->ReplacePartition(kReplacePartitionIndex, kernel_chunk_channel); + + // Get a copy of the kernel after replacing the 3rd freq domain partition. + AudioBuffer replaced_kernel; + replaced_kernel = filter->kernel_freq_domain_buffer_; + + for (size_t i = 0; i < filter->num_partitions_; ++i) { + for (size_t sample = 0; sample < fft_manager->GetFftSize(); ++sample) { + // Expect that all of the other partitions are unchanged. + if (i == kReplacePartitionIndex) { + // Check if the chosen partition has been altered. + EXPECT_EQ(initial_kernel[filter->num_partitions_ - 1][sample], + replaced_kernel[i][sample]); + } else { + // Check if all of the other partitions are unchanged. + EXPECT_EQ(initial_kernel[i][sample], replaced_kernel[i][sample]); + } + } + } + } + + void TestFilterLengthSetter(PartitionedFftFilter* filter, + FftManager* fft_manager) { + // Set the |freq_domain_buffer_| inside |filter| such that it has known + // values (In the time domain, all value 1, then 2 etc.. per partition). + SetFreqDomainKernel(filter); + + // Get a copy of the kernel and number of partitions before we set the + // filter length. + AudioBuffer initial_kernel; + initial_kernel = filter->kernel_freq_domain_buffer_; + const size_t initial_num_partitions = filter->num_partitions_; + + filter->SetFilterLength(filter->filter_size_ / 2); + + // Get a copy after we half the filter length. + AudioBuffer half_kernel; + half_kernel = filter->kernel_freq_domain_buffer_; + const size_t half_num_partitions = filter->num_partitions_; + EXPECT_EQ(initial_num_partitions, 2 * half_num_partitions); + + filter->SetFilterLength(filter->filter_size_ * 2); + + // Get a copy after we re-double the filter length. + AudioBuffer redouble_kernel; + redouble_kernel = filter->kernel_freq_domain_buffer_; + EXPECT_EQ(initial_num_partitions, filter->num_partitions_); + + for (size_t i = 0; i < half_num_partitions; ++i) { + for (size_t sample = 0; sample < fft_manager->GetFftSize(); ++sample) { + // Check that the first two partitions in all 3 cases are the same. + EXPECT_EQ(initial_kernel[i][sample], half_kernel[i][sample]); + EXPECT_EQ(initial_kernel[i][sample], redouble_kernel[i][sample]); + // Check that the final two partitions, after resizing the filters + // frequency domain buffers, contain only zeros. + EXPECT_EQ(redouble_kernel[i + half_num_partitions][sample], 0.0f); + } + } + } +}; + +// Tests whether the reordering of the channels of |freq_domain_buffers_| is as +// expected after calling |ResetFreqDomainBuffers| with both longer and shorter +// filters. +TEST_F(PartitionedFftFilterFrequencyBufferTest, FrequencyBufferResetTest) { + const size_t kBufferSize = 32; + const size_t kInitialFilterSizeFactor = 8; + const size_t kBiggerFilterSizeFactor = 10; + const size_t kSmallerFilterSizeFactor = 5; + // Initially there will be 16 partitions, then 20, then 10. + FftManager fft_manager(kBufferSize); + PartitionedFftFilter filter( + kInitialFilterSizeFactor * kBufferSize * 2, kBufferSize, + kBiggerFilterSizeFactor * kBufferSize * 2, &fft_manager); + TestFrequencyBufferReset(kInitialFilterSizeFactor, kBiggerFilterSizeFactor, + kSmallerFilterSizeFactor, &filter, &fft_manager); +} + +// Tests that we can switch to a frequency domain filter kernel with a greater +// number of partitions than the original kernel provided at instantiation. +// Ensures that we can switch to a frequency domain filter kernel with less +// partitions than the original kernel. +TEST_F(PartitionedFftFilterFrequencyBufferTest, + LongerShorterFrequencyDomainKernelTest) { + TestLongerShorterFrequencyDomainKernel(kLength); +} + +// Tests whether replacing an individual partition works correctly. +TEST_F(PartitionedFftFilterFrequencyBufferTest, ReplacePartitionTest) { + const size_t kBufferSize = 32; + const size_t kFilterSizeFactor = 5; + // A filter with 5 partitions. + FftManager fft_manager(kBufferSize); + PartitionedFftFilter filter(kFilterSizeFactor * kBufferSize * 2, kBufferSize, + &fft_manager); + TestPartitionReplacement(&filter, &fft_manager); +} + +// Tests whether setting the length of the filter kernel, and thus the number +// of partitions gives the expected results. +TEST_F(PartitionedFftFilterFrequencyBufferTest, SetFilterLengthTest) { + const size_t kBufferSize = 32; + const size_t kFilterSizeFactor = 5; + // A filter with 5 partitions. + FftManager fft_manager(kBufferSize); + PartitionedFftFilter filter(kFilterSizeFactor * kBufferSize * 2, kBufferSize, + kFilterSizeFactor * kBufferSize * 4, + &fft_manager); + TestFilterLengthSetter(&filter, &fft_manager); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reflection.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflection.h new file mode 100644 index 000000000..4b134f6f2 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflection.h @@ -0,0 +1,34 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_REFLECTION_H_ +#define RESONANCE_AUDIO_DSP_REFLECTION_H_ + +namespace vraudio { + +// Describes a room reflection that contains information on its time of arrival +// and magnitude. +struct Reflection { + // Time of arrival of the reflection in seconds. + float delay_time_seconds = 0.0f; + + // Magnitude of the reflection. + float magnitude = 0.0f; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_REFLECTION_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor.cc new file mode 100644 index 000000000..9a9174cdc --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor.cc @@ -0,0 +1,177 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/reflections_processor.h" + +#include <algorithm> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "dsp/filter_coefficient_generators.h" +#include "dsp/gain.h" +#include "dsp/shoe_box_room.h" + +namespace vraudio { + +namespace { + +// Maximum allowed delay time for a reflection. Above 2s, the effective output +// level of a reflection will fall below -60dB and thus perceived dynamic +// changes should be negligible. +const size_t kMaxDelayTimeSeconds = 2; + +// Returns the maximum delay time in the given set of reflections. +float FindMaxReflectionDelayTime(const std::vector<Reflection>& reflections) { + float max_delay_time = 0.0f; + for (const auto& reflection : reflections) { + max_delay_time = std::max(max_delay_time, reflection.delay_time_seconds); + } + return max_delay_time; +} + +} // namespace + +ReflectionsProcessor::ReflectionsProcessor(int sample_rate, + size_t frames_per_buffer) + : sample_rate_(sample_rate), + frames_per_buffer_(frames_per_buffer), + max_delay_samples_(kMaxDelayTimeSeconds * sample_rate_), + low_pass_filter_(0.0f), + temp_mono_buffer_(kNumMonoChannels, frames_per_buffer_), + current_reflection_buffer_(kNumFirstOrderAmbisonicChannels, + frames_per_buffer), + target_reflection_buffer_(kNumFirstOrderAmbisonicChannels, + frames_per_buffer), + target_reflections_(kNumRoomSurfaces), + crossfade_(false), + crossfader_(frames_per_buffer_), + num_frames_to_process_on_empty_input_(0), + delays_(kNumRoomSurfaces), + delay_filter_(max_delay_samples_, frames_per_buffer), + delay_buffer_(kNumRoomSurfaces, frames_per_buffer), + gains_(kNumRoomSurfaces), + gain_processors_(kNumRoomSurfaces) { + DCHECK_GT(sample_rate_, 0); + DCHECK_GT(frames_per_buffer_, 0U); +} + +void ReflectionsProcessor::Update( + const ReflectionProperties& reflection_properties, + const WorldPosition& listener_position) { + + // Initialize the low-pass filter. + const float low_pass_coefficient = ComputeLowPassMonoPoleCoefficient( + reflection_properties.cutoff_frequency, sample_rate_); + low_pass_filter_.SetCoefficient(low_pass_coefficient); + // Update the target reflections. + WorldPosition relative_listener_position; + GetRelativeDirection( + WorldPosition(reflection_properties.room_position), + WorldRotation(reflection_properties.room_rotation).conjugate(), + listener_position, &relative_listener_position); + ComputeReflections(relative_listener_position, + WorldPosition(reflection_properties.room_dimensions), + reflection_properties.coefficients, &target_reflections_); + // Additional |frames_per_buffer_| to process needed to compensate the + // crossfade between the current and target reflections. + num_frames_to_process_on_empty_input_ = + frames_per_buffer_ + + static_cast<size_t>(FindMaxReflectionDelayTime(target_reflections_) * + static_cast<float>(sample_rate_)); + // Reflections have been updated so crossfade is required. + crossfade_ = true; +} + +void ReflectionsProcessor::Process(const AudioBuffer& input, + AudioBuffer* output) { + DCHECK_EQ(input.num_channels(), kNumMonoChannels); + DCHECK_EQ(input.num_frames(), frames_per_buffer_); + DCHECK(output); + DCHECK_GE(output->num_channels(), kNumFirstOrderAmbisonicChannels); + DCHECK_EQ(output->num_frames(), frames_per_buffer_); + // Prefilter mono input. + const AudioBuffer::Channel& input_channel = input[0]; + AudioBuffer::Channel* temp_channel = &temp_mono_buffer_[0]; + const bool filter_success = + low_pass_filter_.Filter(input_channel, temp_channel); + const AudioBuffer::Channel& low_pass_channel = + filter_success ? *temp_channel : input_channel; + delay_filter_.InsertData(low_pass_channel); + // Process reflections. + if (crossfade_) { + ApplyReflections(¤t_reflection_buffer_); + UpdateGainsAndDelays(); + ApplyReflections(&target_reflection_buffer_); + crossfader_.ApplyLinearCrossfade(target_reflection_buffer_, + current_reflection_buffer_, output); + crossfade_ = false; + } else { + ApplyReflections(output); + } +} + +void ReflectionsProcessor::UpdateGainsAndDelays() { + for (size_t i = 0; i < kNumRoomSurfaces; ++i) { + delays_[i] = + std::min(max_delay_samples_, + static_cast<size_t>(target_reflections_[i].delay_time_seconds * + static_cast<float>(sample_rate_))); + gains_[i] = target_reflections_[i].magnitude; + } +} + +void ReflectionsProcessor::ApplyReflections(AudioBuffer* output) { + DCHECK(output); + DCHECK_GE(output->num_channels(), kNumFirstOrderAmbisonicChannels); + (*output).Clear(); + for (size_t i = 0; i < kNumRoomSurfaces; ++i) { + auto* delay_channel = &delay_buffer_[i]; + delay_filter_.GetDelayedData(delays_[i], delay_channel); + const bool zero_gain = IsGainNearZero(gains_[i]) && + IsGainNearZero(gain_processors_[i].GetGain()); + if (!zero_gain) { + gain_processors_[i].ApplyGain(gains_[i], *delay_channel, delay_channel, + false /* accumulate_output */); + // Applies fast Ambisonic reflections encoding. + (*output)[0] += *delay_channel; + switch (i) { + case 0: /* left wall reflection */ + (*output)[1] += *delay_channel; + break; + case 1: /* right wall reflection */ + (*output)[1] -= *delay_channel; + break; + case 2: /* floor reflection */ + (*output)[2] -= *delay_channel; + break; + case 3: /* ceiling reflection */ + (*output)[2] += *delay_channel; + break; + case 4: /* front wall reflection */ + (*output)[3] += *delay_channel; + break; + case 5: /* back wall reflection */ + (*output)[3] -= *delay_channel; + break; + } + } else { + // Make sure the gain processor is initialized. + gain_processors_[i].Reset(0.0f); + } + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor.h new file mode 100644 index 000000000..1b9765b0f --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor.h @@ -0,0 +1,128 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_REFLECTIONS_PROCESSOR_H_ +#define RESONANCE_AUDIO_DSP_REFLECTIONS_PROCESSOR_H_ + +#include <utility> +#include <vector> + +#include "api/resonance_audio_api.h" +#include "base/audio_buffer.h" +#include "base/misc_math.h" +#include "dsp/delay_filter.h" +#include "dsp/gain_processor.h" +#include "dsp/mono_pole_filter.h" +#include "dsp/reflection.h" +#include "utils/buffer_crossfader.h" + +namespace vraudio { + +// Class that accepts single mono buffer as input and outputs a first order +// ambisonic buffer of the mix of all the rooms early reflections computed for +// the buffer. +// +// The input consists of a mono mix of all the sound objects in the system. The +// reflections are assumed to be aligned with the aabb axes and thus the first +// order ambisonic axes. +class ReflectionsProcessor { + public: + // Constructs a |ReflectionsProcessor|. + // + // @param sample_rate System sampling rate. + // @param frames_per_buffer System frames per buffer. + ReflectionsProcessor(int sample_rate, size_t frames_per_buffer); + + // Updates reflections according to the new |ReflectionProperties|. + // + // @param reflection_properties New reflection properties. + // @param listener_position New listener position. + void Update(const ReflectionProperties& reflection_properties, + const WorldPosition& listener_position); + + // Processes a mono |AudioBuffer| into an ambisonic |AudioBuffer|. + // + // @param input Mono input buffer. + // @param output Ambisonic output buffer. + void Process(const AudioBuffer& input, AudioBuffer* output); + + // Returns the number of frames required to keep processing on empty input + // signal. This value can be used to avoid any potential artifacts on the + // final output signal when the input signal stream is empty. + size_t num_frames_to_process_on_empty_input() const { + return num_frames_to_process_on_empty_input_; + } + + private: + // Updates |gains_| and |delays_| vectors of the |ReflectionsProcessor|. + void UpdateGainsAndDelays(); + + // Applies |target_reflections_| and fast-encodes them into first order + // ambisonics. + // + // @param output Ambisonic output buffer. + void ApplyReflections(AudioBuffer* output); + + // System sampling rate. + const int sample_rate_; + + // System number of frames per buffer. + const size_t frames_per_buffer_; + + // Maximum allowed delay time for a reflection (in samples). + const size_t max_delay_samples_; + + // Low pass filter to be applied to input signal. + MonoPoleFilter low_pass_filter_; + + // Audio buffer to store mono low pass filtered buffers during processing. + AudioBuffer temp_mono_buffer_; + + // Audio buffers to store FOA reflections buffers during crossfading. + AudioBuffer current_reflection_buffer_; + AudioBuffer target_reflection_buffer_; + + // Target reflections filled with new data when |Update| is called. + std::vector<Reflection> target_reflections_; + + // Indicates whether relfections have been updated and a crossfade is needed. + bool crossfade_; + + // Buffer crossfader to apply linear crossfade during reflections update. + BufferCrossfader crossfader_; + + // Number of frames needed to keep processing on empty input signal. + size_t num_frames_to_process_on_empty_input_; + + // Number of samples of delay to be applied for each reflection. + std::vector<size_t> delays_; + + // Delay filter to delay the incoming buffer. + DelayFilter delay_filter_; + + // Delay buffer used to store delayed reflections before scaling and encoding. + AudioBuffer delay_buffer_; + + // Gains to be applied for each reflection. + std::vector<float> gains_; + + // |GainProcessor|s to apply |gains_|. + std::vector<GainProcessor> gain_processors_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_REFLECTIONS_PROCESSOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor_test.cc new file mode 100644 index 000000000..ab1688e51 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reflections_processor_test.cc @@ -0,0 +1,142 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/reflections_processor.h" + +#include <algorithm> +#include <utility> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "api/resonance_audio_api.h" +#include "base/constants_and_types.h" +#include "utils/test_util.h" + +namespace vraudio { + +namespace { + +const size_t kFramesPerBuffer = 512; +const int kSampleRate = 48000; +const float kRoomDimensions[3] = {2.0f, 2.0f, 2.0f}; +const size_t kExpectedDelaySamples = 279; + +} // namespace + +class ReflectionsProcessorTest : public ::testing::Test { + protected: + void SetUp() override { + const float kReflectionCoefficients[kNumRoomSurfaces] = {0.5f, 0.5f, 0.5f, + 0.5f, 0.5f, 0.5f}; + const WorldPosition kListenerPosition(0.0f, 0.0f, 0.0f); + std::copy(std::begin(kReflectionCoefficients), + std::end(kReflectionCoefficients), + std::begin(reflection_properties_.coefficients)); + std::copy(std::begin(kRoomDimensions), std::end(kRoomDimensions), + std::begin(reflection_properties_.room_dimensions)); + listener_position_ = kListenerPosition; + } + + ReflectionProperties reflection_properties_; + WorldPosition listener_position_; +}; // namespace vraudio + +// Tests that the processed output is delayed, filtered, and scaled as expected. +TEST_F(ReflectionsProcessorTest, ProcessTest) { + ReflectionsProcessor reflections_processor(kSampleRate, kFramesPerBuffer); + reflections_processor.Update(reflection_properties_, listener_position_); + + AudioBuffer input(kNumMonoChannels, kFramesPerBuffer); + AudioBuffer output(kNumFirstOrderAmbisonicChannels, kFramesPerBuffer); + // Process until we have transitioned to the specified Gain. + for (size_t i = 0; i < kUnitRampLength; i += kFramesPerBuffer) { + input.Clear(); + output.Clear(); + GenerateDiracImpulseFilter(0, &input[0]); + reflections_processor.Process(input, &output); + } + + // The reflections calculated should be as follows: + // delay = 280 samples, magnitude = 0.25, direction = {azim 90, elev 0}. + // delay = 280 samples, magnitude = 0.25, direction = {azim -90, elev 0}. + // delay = 280 samples, magnitude = 0.25, direction = {azim 0, elev -90}. + // delay = 280 samples, magnitude = 0.25, direction = {azim 0, elev 90}. + // delay = 280 samples, magnitude = 0.25, direction = {azim 0, elev 0}. + // delay = 280 samples, magnitude = 0.25, direction = {azim 180, elev 0}. + const float kExpectedMagnitude = 0.25f; + // We expect the following ambisonic encoding coefficients: + // {azim 90, elev 0} = {1 1 0 0}. + // {azim -90, elev 0} = {1 -1 0 0}. + // {azim 0, elev -90} = {1 0 -1 0}. + // {azim 0, elev 90} = {1 0 1 0}. + // {azim 0, elev 0} = {1 0 0 1}. + // {azim 180, elev 0} = {1 0 0 -1}. + const std::vector<std::vector<float>> kEncodingCoefficients = { + {1.0f, 1.0f, 0.0f, 0.0f}, {1.0f, -1.0f, 0.0f, 0.0f}, + {1.0f, 0.0f, -1.0f, 0.0f}, {1.0f, 0.0f, 1.0f, 0.0f}, + {1.0f, 0.0f, 0.0f, 1.0f}, {1.0f, 0.0f, 0.0f, -1.0f}}; + + for (size_t j = kExpectedDelaySamples; j < kFramesPerBuffer; ++j) { + for (size_t k = 0; k < kNumFirstOrderAmbisonicChannels; ++k) { + float coefficient_sum = 0.0f; + for (size_t i = 0; i < kEncodingCoefficients.size(); ++i) { + coefficient_sum += kEncodingCoefficients[i][k]; + } + const float expected_sample = input[0][j - kExpectedDelaySamples] * + kExpectedMagnitude * coefficient_sum; + EXPECT_NEAR(expected_sample, output[k][j], kEpsilonFloat); + } + } +} + +// Tests that when transitioning from a reflection with zero magnitude to one +// with non-zero magnitude, there will be a steady incremental increase in +// the "fade in". +TEST_F(ReflectionsProcessorTest, CrossFadeTest) { + ReflectionsProcessor reflections_processor(kSampleRate, kFramesPerBuffer); + + AudioBuffer input(kNumMonoChannels, kFramesPerBuffer); + AudioBuffer output(kNumFirstOrderAmbisonicChannels, kFramesPerBuffer); + + input.Clear(); + output.Clear(); + for (size_t i = 0; i < kFramesPerBuffer; ++i) { + input[0][i] = 1.0f; + } + + reflections_processor.Process(input, &output); + reflections_processor.Update(reflection_properties_, listener_position_); + reflections_processor.Process(input, &output); + + // All the reflections are expected to arrive at the listener at the same + // time. That is why their directional contributions are expected to cancel + // out. Only in first channel we will see a steady increase. + for (size_t channel = 0; channel < kNumFirstOrderAmbisonicChannels; + ++channel) { + if (channel < kNumMonoChannels) { + for (size_t frame = kExpectedDelaySamples; frame < kFramesPerBuffer; + ++frame) { + EXPECT_TRUE(output[channel][frame] > output[channel][frame - 1]); + } + } else { + for (size_t frame = 0; frame < kFramesPerBuffer; ++frame) { + EXPECT_NEAR(output[channel][frame], 0.0f, kEpsilonFloat); + } + } + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler.cc new file mode 100644 index 000000000..e135afcdf --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler.cc @@ -0,0 +1,335 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +// Prevent Visual Studio from complaining about std::copy_n. +#if defined(_WIN32) +#define _SCL_SECURE_NO_WARNINGS +#endif + +#include "dsp/resampler.h" + +#include <functional> +#include <numeric> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/misc_math.h" +#include "base/simd_utils.h" + +#include "dsp/utils.h" + +namespace vraudio { + +namespace { + +// (kTransitionBandwidthRatio / 2) * (sample_rate / cutoff_frequency) +// = filter_length. +// The value below was chosen empirically as a tradeoff between execution time +// and filter rolloff wrt. cutoff frequency. +const size_t kTransitionBandwidthRatio = 13; + +// Maximum number of channels internally based upon the maximum supported +// ambisonic order. +const size_t kMaxNumChannels = + (kMaxSupportedAmbisonicOrder + 1) * (kMaxSupportedAmbisonicOrder + 1); + +} // namespace + +Resampler::Resampler() + : up_rate_(0), + down_rate_(0), + time_modulo_up_rate_(0), + last_processed_sample_(0), + num_channels_(0), + coeffs_per_phase_(0), + transposed_filter_coeffs_(kNumMonoChannels, kMaxSupportedNumFrames), + temporary_filter_coeffs_(kNumMonoChannels, kMaxSupportedNumFrames), + state_(kMaxNumChannels, kMaxSupportedNumFrames) { + state_.Clear(); +} + +void Resampler::Process(const AudioBuffer& input, AudioBuffer* output) { + + // See "Digital Signal Processing, 4th Edition, Prolakis and Manolakis, + // Pearson, Chapter 11 (specificaly Figures 11.5.10 and 11.5.13). + DCHECK_EQ(input.num_channels(), num_channels_); + const size_t input_length = input.num_frames(); + DCHECK_GE(output->num_frames(), GetNextOutputLength(input_length)); + DCHECK_LE(output->num_frames(), GetMaxOutputLength(input_length)); + DCHECK_EQ(output->num_channels(), num_channels_); + output->Clear(); + + if (up_rate_ == down_rate_) { + *output = input; + return; + } + + // |input_sample| is the last processed sample. + size_t input_sample = last_processed_sample_; + // |output_sample| is the output. + size_t output_sample = 0; + + const auto& filter_coefficients = transposed_filter_coeffs_[0]; + + while (input_sample < input_length) { + size_t filter_index = time_modulo_up_rate_ * coeffs_per_phase_; + size_t offset_input_index = input_sample - coeffs_per_phase_ + 1; + const int offset = -static_cast<int>(offset_input_index); + + if (offset > 0) { + // We will need to draw data from the |state_| buffer. + const int state_num_frames = static_cast<int>(coeffs_per_phase_ - 1); + int state_index = state_num_frames - offset; + while (state_index < state_num_frames) { + for (size_t channel = 0; channel < num_channels_; ++channel) { + (*output)[channel][output_sample] += + state_[channel][state_index] * filter_coefficients[filter_index]; + } + state_index++; + filter_index++; + } + // Move along by |offset| samples up as far as |input|. + offset_input_index += offset; + } + + // We now move back to where |input_sample| "points". + while (offset_input_index <= input_sample) { + for (size_t channel = 0; channel < num_channels_; ++channel) { + (*output)[channel][output_sample] += + input[channel][offset_input_index] * + filter_coefficients[filter_index]; + } + offset_input_index++; + filter_index++; + } + output_sample++; + + time_modulo_up_rate_ += down_rate_; + // Advance the input pointer. + input_sample += time_modulo_up_rate_ / up_rate_; + // Decide which phase of the polyphase filter to use next. + time_modulo_up_rate_ %= up_rate_; + } + DCHECK_GE(input_sample, input_length); + last_processed_sample_ = input_sample - input_length; + + // Take care of the |state_| buffer. + const int samples_left_in_input = + static_cast<int>(coeffs_per_phase_) - 1 - static_cast<int>(input_length); + if (samples_left_in_input > 0) { + for (size_t channel = 0; channel < num_channels_; ++channel) { + // Copy end of the |state_| buffer to the beginning. + auto& state_channel = state_[channel]; + DCHECK_GE(static_cast<int>(state_channel.size()), samples_left_in_input); + std::copy_n(state_channel.end() - samples_left_in_input, + samples_left_in_input, state_channel.begin()); + // Then copy input to the end of the buffer. + std::copy_n(input[channel].begin(), input_length, + state_channel.end() - input_length); + } + } else { + for (size_t channel = 0; channel < num_channels_; ++channel) { + // Copy the last of the |input| samples into the |state_| buffer. + DCHECK_GT(coeffs_per_phase_, 0U); + DCHECK_GT(input[channel].size(), coeffs_per_phase_ - 1); + std::copy_n(input[channel].end() - (coeffs_per_phase_ - 1), + coeffs_per_phase_ - 1, state_[channel].begin()); + } + } +} + +size_t Resampler::GetMaxOutputLength(size_t input_length) const { + if (up_rate_ == down_rate_) { + return input_length; + } + DCHECK_GT(down_rate_, 0U); + // The + 1 takes care of the case where: + // (time_modulo_up_rate_ + up_rate_ * last_processed_sample_) < + // ((input_length * up_rate_) % down_rate_) + // The output length will be equal to the return value or the return value -1. + return (input_length * up_rate_) / down_rate_ + 1; +} + +size_t Resampler::GetNextOutputLength(size_t input_length) const { + if (up_rate_ == down_rate_) { + return input_length; + } + const size_t max_length = GetMaxOutputLength(input_length); + if ((time_modulo_up_rate_ + up_rate_ * last_processed_sample_) >= + ((input_length * up_rate_) % down_rate_)) { + return max_length - 1; + } + return max_length; +} + +void Resampler::SetRateAndNumChannels(int source_frequency, + int destination_frequency, + size_t num_channels) { + + // Convert sampling rates to be relatively prime. + DCHECK_GT(source_frequency, 0); + DCHECK_GT(destination_frequency, 0); + DCHECK_GT(num_channels, 0U); + const int greatest_common_divisor = + FindGcd(destination_frequency, source_frequency); + const size_t destination = + static_cast<size_t>(destination_frequency / greatest_common_divisor); + const size_t source = + static_cast<size_t>(source_frequency / greatest_common_divisor); + + // Obtain the size of the |state_| before |coeffs_per_phase_| is updated in + // |GenerateInterpolatingFilter()|. + const size_t old_state_size = + coeffs_per_phase_ > 0 ? coeffs_per_phase_ - 1 : 0; + if ((destination != up_rate_) || (source != down_rate_)) { + up_rate_ = destination; + down_rate_ = source; + if (up_rate_ == down_rate_) { + return; + } + // Create transposed multirate filters from sincs. + GenerateInterpolatingFilter(source_frequency); + // Reset the time variable as it may be longer than the new filter length if + // we switched from upsampling to downsampling via a call to SetRate(). + time_modulo_up_rate_ = 0; + } + + // Update the |state_| buffer. + if (num_channels_ != num_channels) { + num_channels_ = num_channels; + InitializeStateBuffer(old_state_size); + } +} + +bool Resampler::AreSampleRatesSupported(int source, int destination) { + DCHECK_GT(source, 0); + DCHECK_GT(destination, 0); + // Determines whether sample rates are supported based upon whether our + // maximul filter lenhgth is big enough to hold the corresponding + // interpolation filter. + const int max_rate = + std::max(source, destination) / FindGcd(source, destination); + size_t filter_length = max_rate * kTransitionBandwidthRatio; + filter_length += filter_length % 2; + return filter_length <= kMaxSupportedNumFrames; +} + +void Resampler::ResetState() { + + time_modulo_up_rate_ = 0; + last_processed_sample_ = 0; + state_.Clear(); +} + +void Resampler::InitializeStateBuffer(size_t old_state_num_frames) { + // Update the |state_| buffer if it is null or if the number of coefficients + // per phase in the polyphase filter has changed. + if (up_rate_ == down_rate_ || num_channels_ == 0) { + return; + } + // If the |state_| buffer is to be kept. For example in the case of a change + // in either source or destination sampling rate, maintaining the old |state_| + // buffers contents allows a glitch free transition. + const size_t new_state_num_frames = + coeffs_per_phase_ > 0 ? coeffs_per_phase_ - 1 : 0; + if (old_state_num_frames != new_state_num_frames) { + const size_t min_size = + std::min(new_state_num_frames, old_state_num_frames); + const size_t max_size = + std::max(new_state_num_frames, old_state_num_frames); + for (size_t channel = 0; channel < num_channels_; ++channel) { + auto& state_channel = state_[channel]; + DCHECK_LT(state_channel.begin() + max_size, state_channel.end()); + std::fill(state_channel.begin() + min_size, + state_channel.begin() + max_size, 0.0f); + } + } +} + +void Resampler::GenerateInterpolatingFilter(int sample_rate) { + // See "Digital Signal Processing, 4th Edition, Prolakis and Manolakis, + // Pearson, Chapter 11 (specificaly Figures 11.5.10 and 11.5.13). + const size_t max_rate = std::max(up_rate_, down_rate_); + const float cutoff_frequency = + static_cast<float>(sample_rate) / static_cast<float>(2 * max_rate); + size_t filter_length = max_rate * kTransitionBandwidthRatio; + filter_length += filter_length % 2; + auto* filter_channel = &temporary_filter_coeffs_[0]; + filter_channel->Clear(); + GenerateSincFilter(cutoff_frequency, static_cast<float>(sample_rate), + filter_length, filter_channel); + + // Pad out the filter length so that it can be arranged in polyphase fashion. + const size_t transposed_length = + filter_length + max_rate - (filter_length % max_rate); + coeffs_per_phase_ = transposed_length / max_rate; + ArrangeFilterAsPolyphase(filter_length, *filter_channel); +} + +void Resampler::ArrangeFilterAsPolyphase(size_t filter_length, + const AudioBuffer::Channel& filter) { + // Coefficients are transposed and flipped. + // Suppose |up_rate_| is 3, and the input number of coefficients is 10, + // h[0], ..., h[9]. + // Then the |transposed_filter_coeffs_| buffer will look like this: + // h[9], h[6], h[3], h[0], flipped phase 0 coefs. + // 0, h[7], h[4], h[1], flipped phase 1 coefs (zero-padded). + // 0, h[8], h[5], h[2], flipped phase 2 coefs (zero-padded). + transposed_filter_coeffs_.Clear(); + auto& transposed_coefficients_channel = transposed_filter_coeffs_[0]; + for (size_t i = 0; i < up_rate_; ++i) { + for (size_t j = 0; j < coeffs_per_phase_; ++j) { + if (j * up_rate_ + i < filter_length) { + const size_t coeff_index = + (coeffs_per_phase_ - 1 - j) + i * coeffs_per_phase_; + transposed_coefficients_channel[coeff_index] = filter[j * up_rate_ + i]; + } + } + } +} + +void Resampler::GenerateSincFilter(float cutoff_frequency, float sample_rate, + size_t filter_length, + AudioBuffer::Channel* buffer) { + + DCHECK_GT(sample_rate, 0.0f); + const float angular_cutoff_frequency = + kTwoPi * cutoff_frequency / sample_rate; + + const size_t half_filter_length = filter_length / 2; + GenerateHannWindow(true /* Full Window */, filter_length, buffer); + auto* buffer_channel = &buffer[0]; + + for (size_t i = 0; i < filter_length; ++i) { + if (i == half_filter_length) { + (*buffer_channel)[half_filter_length] *= angular_cutoff_frequency; + } else { + const float denominator = + static_cast<float>(i) - (static_cast<float>(filter_length) / 2.0f); + DCHECK_GT(std::abs(denominator), kEpsilonFloat); + (*buffer_channel)[i] *= + std::sin(angular_cutoff_frequency * denominator) / denominator; + } + } + // Normalize. + const float normalizing_factor = + static_cast<float>(up_rate_) / + std::accumulate(buffer_channel->begin(), buffer_channel->end(), 0.0f); + ScalarMultiply(filter_length, normalizing_factor, buffer_channel->begin(), + buffer_channel->begin()); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler.h new file mode 100644 index 000000000..c2c079a80 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler.h @@ -0,0 +1,135 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_RESAMPLER_H_ +#define RESONANCE_AUDIO_DSP_RESAMPLER_H_ + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Class that provides rational resampling of audio data. +class Resampler { + public: + Resampler(); + + // Resamples an |AudioBuffer| of input data sampled at |source_frequency| to + // |destination_frequency|. + // + // @param input Input data to be resampled. + // @param output Resampled output data. + void Process(const AudioBuffer& input, AudioBuffer* output); + + // Returns the maximum length which the output buffer will be, given the + // current source and destination frequencies and input length. The actual + // output length will either be this or one less. + // + // @param input_length Length of the input. + // @return Maximum length of the output. + size_t GetMaxOutputLength(size_t input_length) const; + + // Returns the next length which the output buffer will be, given the + // current source and destination frequencies and input length. + // + // @param input_length Length of the input. + // @return Next length of the output. + size_t GetNextOutputLength(size_t input_length) const; + + // Sets the source and destination sampling rate as well as the number of + // channels. Note this method only resets the filter state number of channel + // changes. + // + // @param source_frequency Sampling rate of input data. + // @param destination_frequency Desired output sampling rate. + // @param num_channels Number of channels to process. + void SetRateAndNumChannels(int source_frequency, int destination_frequency, + size_t num_channels); + + // Returns whether the sampling rates provided are supported by the resampler. + // + // @param source Source sampling rate. + // @param destination Destination sampling rate. + // @return True if the sampling rate pair are supported. + static bool AreSampleRatesSupported(int source, int destination); + + // Resets the inner state of the |Resampler| allowing its use repeatedly on + // different data streams. + void ResetState(); + + private: + friend class PolyphaseFilterTest; + // Initializes the |state_| buffer. Called when sampling rate is changed or + // the state is reset. + // + // @param size_t old_state_num_frames Number of frames in the |state_| buffer + // previous to the most recent call to |GenerateInterpolatingFilter|. + void InitializeStateBuffer(size_t old_state_num_frames); + + // Generates a windowed sinc to act as the interpolating/anti-aliasing filter. + // + // @param sample_rate The system sampling rate. + void GenerateInterpolatingFilter(int sample_rate); + + // Arranges the anti aliasing filter coefficients in polyphase filter format. + // + // @param filter_length Number of frames in |filter| containing filter + // coefficients. + // @param filter Vector of filter coefficients. + void ArrangeFilterAsPolyphase(size_t filter_length, + const AudioBuffer::Channel& filter); + + // Generates Hann windowed sinc function anti aliasing filters. + // + // @param cutoff_frequency Transition band (-3dB) frequency of the filter. + // @param sample_rate The system sampling rate. + // @param filter_length Number of frames in |buffer| containing filter + // coefficients. + // @param buffer |AudioBuffer::Channel| to contain the filter coefficients. + void GenerateSincFilter(float cutoff_frequency, float sample_rate, + size_t filter_length, AudioBuffer::Channel* buffer); + + // Rate of the interpolator section of the rational sampling rate converter. + size_t up_rate_; + + // Rate of the decimator section of the rational sampling rate convereter. + size_t down_rate_; + + // Time variable for the polyphase filter. + size_t time_modulo_up_rate_; + + // Marks the last processed sample of the input. + size_t last_processed_sample_; + + // Number of channels in the |AudioBuffer|s processed. + size_t num_channels_; + + // Number of filter coefficients in each phase of the polyphase filter. + size_t coeffs_per_phase_; + + // Filter coefficients stored in polyphase form. + AudioBuffer transposed_filter_coeffs_; + + // Filter coefficients in planar form, used for calculating the transposed + // filter. + AudioBuffer temporary_filter_coeffs_; + + // Buffer holding the samples of input required between input buffers. + AudioBuffer state_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_RESAMPLER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler_test.cc new file mode 100644 index 000000000..5b23de58f --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/resampler_test.cc @@ -0,0 +1,325 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/resampler.h" + +#include <numeric> +#include <utility> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "utils/test_util.h" + +namespace vraudio { + +namespace { + +const int kToneFrequency = 1000; + +const size_t kInputBufferLength = 441; + +const int kSourceDataSampleRate = 44100; +const int kUpDestinationDataSampleRate = 48000; +const int kDownDestinationDataSampleRate = 24000; + +TEST(ResamplerTest, UpSampleTest) { + // Create input buffer with 1000Hz sine wave at 44100Hz, 441 samples long. + // This should be ten periods of a sine wave. + AudioBuffer input(kNumMonoChannels, kInputBufferLength); + GenerateSineWave(kToneFrequency, kSourceDataSampleRate, &input[0]); + + Resampler resampler; + resampler.SetRateAndNumChannels( + kSourceDataSampleRate, kUpDestinationDataSampleRate, kNumMonoChannels); + AudioBuffer output(kNumMonoChannels, + resampler.GetNextOutputLength(kInputBufferLength)); + resampler.Process(input, &output); + + // Ensure the length of the output is as expected. + const size_t input_length_proportion = + kSourceDataSampleRate / kInputBufferLength; + const size_t expected_output_length = + kUpDestinationDataSampleRate / input_length_proportion; + EXPECT_EQ(expected_output_length, output.num_frames()); + + // Ensure the output also contains 10 periods of a sine wave, confirming it is + // still at 1000Hz given the new sampling rate. + AudioBuffer thresholded_output(kNumMonoChannels, output.num_frames()); + for (size_t i = 0; i < output.num_frames(); ++i) { + thresholded_output[0][i] = std::abs(output[0][i]) < 0.5f ? 0.0f : 1.0f; + } + size_t num_square_wave_corners = 1; + // We have now generated a square wave with twice the number of corners as + // there are zero crossings. + for (size_t i = 0; i < output.num_frames() - 1; ++i) { + if (thresholded_output[0][i] != thresholded_output[0][i + 1]) { + num_square_wave_corners++; + } + } + const size_t zerocross_count = num_square_wave_corners / 2; + + const size_t num_expected_zero_crossings = + 2 * kToneFrequency / (kSourceDataSampleRate / kInputBufferLength); + EXPECT_EQ(num_expected_zero_crossings, zerocross_count); +} + +TEST(ResamplerTest, DownSampleTest) { + // Create input buffer with 1000Hz sine wave at 44100Hz, 441 samples long. + // This should be ten periods of a sine wave. + AudioBuffer input(kNumMonoChannels, kInputBufferLength); + GenerateSineWave(kToneFrequency, kSourceDataSampleRate, &input[0]); + + Resampler resampler; + resampler.SetRateAndNumChannels( + kSourceDataSampleRate, kDownDestinationDataSampleRate, kNumMonoChannels); + AudioBuffer output(kNumMonoChannels, + resampler.GetNextOutputLength(kInputBufferLength)); + resampler.Process(input, &output); + + // Ensure the length of the output is as expected. + const size_t input_length_proportion = + kSourceDataSampleRate / kInputBufferLength; + const size_t expected_output_length = + kDownDestinationDataSampleRate / input_length_proportion; + EXPECT_EQ(expected_output_length, output.num_frames()); + + // Ensure the output also contains one period of a sine wave, confirming it is + // still at 100Hz given the new sampling rate. + AudioBuffer thresholded_output(kNumMonoChannels, output.num_frames()); + for (size_t i = 0; i < output.num_frames(); ++i) { + thresholded_output[0][i] = std::abs(output[0][i]) < 0.5f ? 0.0f : 1.0f; + } + int zero_cross_count = 1; + for (size_t i = 0; i < output.num_frames() - 1; ++i) { + if (thresholded_output[0][i] != thresholded_output[0][i + 1]) { + zero_cross_count++; + } + } + const int num_expected_zero_crossings = + 4 * kToneFrequency / (kSourceDataSampleRate / kInputBufferLength); + EXPECT_EQ(num_expected_zero_crossings, zero_cross_count); +} + +TEST(ResamplerTest, ResetStateTest) { + // Create input buffer with 1000Hz sine wave at 44100Hz, 441 samples long. + // This should be ten periods of a sine wave. + AudioBuffer input(kNumMonoChannels, kInputBufferLength); + GenerateSineWave(kToneFrequency, kSourceDataSampleRate, &input[0]); + + Resampler resampler; + resampler.SetRateAndNumChannels( + kSourceDataSampleRate, kUpDestinationDataSampleRate, kNumMonoChannels); + AudioBuffer output_1(kNumMonoChannels, + resampler.GetNextOutputLength(kInputBufferLength)); + resampler.Process(input, &output_1); + + resampler.ResetState(); + AudioBuffer output_2(kNumMonoChannels, + resampler.GetNextOutputLength(kInputBufferLength)); + resampler.Process(input, &output_2); + + // If the clearing of the resampler worked properly there should be no + // internal state between process calls and thus both outputs should be + // identical. + for (size_t sample = 0; sample < output_1.num_frames(); ++sample) { + EXPECT_NEAR(output_1[0][sample], output_2[0][sample], kEpsilonFloat); + } +} + +TEST(ResamplerTest, TwoChannelTest) { + AudioBuffer input(kNumStereoChannels, kInputBufferLength); + GenerateSineWave(kToneFrequency, kSourceDataSampleRate, &input[0]); + GenerateSineWave(2 * kToneFrequency, kSourceDataSampleRate, &input[1]); + + Resampler resampler; + resampler.SetRateAndNumChannels( + kSourceDataSampleRate, kUpDestinationDataSampleRate, kNumStereoChannels); + AudioBuffer output(kNumStereoChannels, + resampler.GetNextOutputLength(kInputBufferLength)); + resampler.Process(input, &output); + + AudioBuffer thresholded_output(kNumStereoChannels, output.num_frames()); + for (size_t i = 0; i < output.num_frames(); ++i) { + thresholded_output[0][i] = std::abs(output[0][i]) < 0.5f ? 0.0f : 1.0f; + thresholded_output[1][i] = std::abs(output[1][i]) < 0.5f ? 0.0f : 1.0f; + } + int channel_0_zero_cross_count = 1; + int channel_1_zero_cross_count = 1; + for (size_t i = 0; i < output.num_frames() - 1; ++i) { + if (thresholded_output[0][i] != thresholded_output[0][i + 1]) { + channel_0_zero_cross_count++; + } + if (thresholded_output[1][i] != thresholded_output[1][i + 1]) { + channel_1_zero_cross_count++; + } + } + const int channel_0_expected_zero_crossings = + 4 * kToneFrequency / (kSourceDataSampleRate / kInputBufferLength); + const int channel_1_expected_zero_crossings = + 8 * kToneFrequency / (kSourceDataSampleRate / kInputBufferLength); + EXPECT_EQ(channel_0_zero_cross_count, channel_0_expected_zero_crossings); + EXPECT_EQ(channel_1_zero_cross_count, channel_1_expected_zero_crossings); +} + +TEST(ResamplerTest, DiracImpulseUpsampleTest) { + const size_t kInputSignalSize = 128; + AudioBuffer input_signal(kNumMonoChannels, kInputSignalSize); + GenerateDiracImpulseFilter(kInputSignalSize / 2, &input_signal[0]); + + const int kSourceSamplingRate = 100; + const int kDestinationSamplingRate = 200; + + Resampler resampler; + resampler.SetRateAndNumChannels(kSourceSamplingRate, kDestinationSamplingRate, + kNumMonoChannels); + + AudioBuffer output_signal(kNumMonoChannels, + resampler.GetNextOutputLength(kInputSignalSize)); + const int resampling_factor = kDestinationSamplingRate / kSourceSamplingRate; + EXPECT_EQ(kInputSignalSize * resampling_factor, output_signal.num_frames()); + // Perform resampling. Dirac impulse position should shift according to + // "resampling_factor". + resampler.Process(input_signal, &output_signal); + EXPECT_EQ(kInputSignalSize * resampling_factor, output_signal.num_frames()); + + DelayCompare(input_signal[0], output_signal[0], kInputSignalSize / 2, + kEpsilonFloat); +} + +TEST(ResamplerTest, GetNextOutputLengthTest) { + const size_t input_length = 10; + Resampler resampler; + resampler.SetRateAndNumChannels(kSourceDataSampleRate, + kSourceDataSampleRate / 2, kNumMonoChannels); + // In this case the source rate is twice the destination rate, we can expect + // the output length to be half the input_length. + + resampler.SetRateAndNumChannels(kSourceDataSampleRate, + kSourceDataSampleRate * 2, kNumMonoChannels); + // In this case the source rate is half the destination rate, we can + // expect the output length to be twice the input_length. + EXPECT_EQ(input_length * 2, resampler.GetNextOutputLength(input_length)); +} + +TEST(Resampler, AreSampleRatesSupportedTest) { + const size_t kNumPairs = 6; + const int kSourceRates[kNumPairs] = {4000, 16000, 44100, 96000, 41999, 48000}; + const int kDestRates[kNumPairs] = {96000, 44100, 48000, 32000, 44100, 43210}; + const bool kExpectedResults[kNumPairs] = {true, true, true, + true, false, false}; + + // We generally support only rates with relatively high GCD. + for (size_t i = 0; i < kNumPairs; ++i) { + const bool result = + Resampler::AreSampleRatesSupported(kSourceRates[i], kDestRates[i]); + EXPECT_EQ(result, kExpectedResults[i]); + } +} + +class OutputLengthTest : public ::testing::TestWithParam<std::pair<int, int>> { +}; + +// Tests that the lengths returned by the |Process| method are always equal to +// the length returned by |GetMaxOutputLength|, or one sample less. +TEST_P(OutputLengthTest, OutputLengthTest) { + const size_t kInputSize = 512; + Resampler resampler; + const auto rates = GetParam(); + resampler.SetRateAndNumChannels(rates.first, rates.second, kNumMonoChannels); + AudioBuffer input(kNumMonoChannels, kInputSize); + const size_t max_output_length = resampler.GetMaxOutputLength(kInputSize); + AudioBuffer output(kNumMonoChannels, max_output_length); + + // Process 100 times and expect the lengts to be maximum length or one less. + const size_t kTimesToProcess = 100; + for (size_t i = 0; i < kTimesToProcess; ++i) { + const size_t next_output_length = resampler.GetNextOutputLength(kInputSize); + resampler.Process(input, &output); + EXPECT_LE(next_output_length, max_output_length); + EXPECT_LE(max_output_length - next_output_length, 1U); + } +} + +INSTANTIATE_TEST_CASE_P(RatePairs, OutputLengthTest, + ::testing::Values(std::make_pair(44100, 48000), + std::make_pair(48000, 44100))); + +} // namespace + +class PolyphaseFilterTest : public ::testing::Test { + protected: + PolyphaseFilterTest() {} + // Virtual methods from ::testing::Test + ~PolyphaseFilterTest() override {} + void SetUp() override {} + void TearDown() override {} + + std::unique_ptr<AudioBuffer> GetPolyphaseFilter( + const AudioBuffer::Channel& filter_coefficients, int coeffs_per_phase, + int up_rate) { + // Create a resampler with arbitrary source and destination sampling rates. + Resampler resampler; + resampler.SetRateAndNumChannels(kSourceDataSampleRate, + kSourceDataSampleRate, kNumMonoChannels); + resampler.up_rate_ = up_rate; + resampler.transposed_filter_coeffs_.Clear(); + resampler.coeffs_per_phase_ = coeffs_per_phase; + resampler.ArrangeFilterAsPolyphase(filter_coefficients.size(), + filter_coefficients); + std::unique_ptr<AudioBuffer> polyphase_filter(new AudioBuffer()); + *polyphase_filter = resampler.transposed_filter_coeffs_; + return polyphase_filter; + } +}; + +TEST_F(PolyphaseFilterTest, CorrectPolyphaseFilterTest) { + // Choose an uprate which is a factor of the filters length. + const int filter_length = 24; + const int uprate = 4; + const int phase_length = filter_length / uprate; + + // Create a vector of ascending numbers (1:24). + AudioBuffer ascending(kNumMonoChannels, filter_length); + ascending.Clear(); + std::iota(ascending[0].begin(), ascending[0].end(), 1.0f); + + std::unique_ptr<AudioBuffer> output = + GetPolyphaseFilter(ascending[0], phase_length, uprate); + + // In polyphase format, the filter coefficients can be thought of as a matrix + // with uprate columns. The last uprate coeffecients are in the first row and + // the first uprate coefficients are in the last row. This is stored in a + // vector, column by column. + // + // 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 + // becomes: + // 21 22 23 24 + // 17 18 19 20 + // 13 14 15 16 + // 9 10 11 12 + // 5 6 7 8 + // 1 2 3 4 + size_t index = 0; + for (int phase = uprate; phase > 0; --phase) { + for (int value = filter_length - phase + 1; value > uprate - phase; + value -= uprate) { + EXPECT_EQ(static_cast<float>(value), (*output)[0][index++]); + } + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_compensator.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_compensator.cc new file mode 100644 index 000000000..2996d3799 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_compensator.cc @@ -0,0 +1,214 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/reverb_onset_compensator.h" + +#include <algorithm> +#include <cmath> +#include <iterator> + +#include "base/constants_and_types.h" +#include "dsp/spectral_reverb_constants_and_tables.h" +#include "dsp/utils.h" + +namespace vraudio { + +namespace { + +// Number of reverb updaters. Twelve were chosen as this represents one update +// per buffer length at 24kHz, a number we are very unlikely to exceed. +const size_t kNumReverbUpdaters = 12; + +} // namespace + +ReverbOnsetCompensator::ReverbOnsetCompensator(int sampling_rate, + size_t frames_per_buffer, + FftManager* fft_manager) + : fft_manager_(fft_manager), + sampling_rate_(sampling_rate), + frames_per_buffer_(frames_per_buffer), + base_curves_(kNumStereoChannels, kCorrectionCurveLength), + adder_curves_(kNumStereoChannels, kCorrectionCurveLength), + left_filter_(CeilToMultipleOfFramesPerBuffer(kCorrectionCurveLength, + frames_per_buffer_), + frames_per_buffer_, fft_manager_), + right_filter_(CeilToMultipleOfFramesPerBuffer(kCorrectionCurveLength, + frames_per_buffer_), + frames_per_buffer_, fft_manager_), + delay_filter_(CeilToMultipleOfFramesPerBuffer(kCorrectionCurveLength, + frames_per_buffer_), + frames_per_buffer_), + num_active_processors_(0), + temp_kernel_buffer_(kNumStereoChannels, frames_per_buffer_), + temp_freq_buffer_(kNumMonoChannels, fft_manager_->GetFftSize()) { + CHECK(fft_manager_); + DCHECK_GT(sampling_rate_, 0); + DCHECK_GT(frames_per_buffer_, 0U); + + temp_kernel_buffer_.Clear(); + temp_freq_buffer_.Clear(); + + GenerateNoiseVectors(); + GenerateCorrectionCurves(); + + // Insert reverb updaters. + for (size_t i = 0; i < kNumReverbUpdaters; ++i) { + update_processors_.emplace_front(new ReverbOnsetUpdateProcessor( + frames_per_buffer_, sampling_rate_, &base_curves_, &adder_curves_)); + } +} + +void ReverbOnsetCompensator::Process(const AudioBuffer& input, + AudioBuffer* output) { + DCHECK(output); + DCHECK_EQ(kNumMonoChannels, input.num_channels()); + DCHECK_EQ(frames_per_buffer_, input.num_frames()); + DCHECK_EQ(kNumStereoChannels, output->num_channels()); + DCHECK_EQ(frames_per_buffer_, output->num_frames()); + + delay_filter_.InsertData(input[0]); + delay_filter_.GetDelayedData(kCompensationOnsetLength, &(*output)[0]); + + // Process reverb updates. + AudioBuffer::Channel* kernel_channel_left = &temp_kernel_buffer_[0]; + AudioBuffer::Channel* kernel_channel_right = &temp_kernel_buffer_[1]; + + size_t processor_index = 0; + while (processor_index < num_active_processors_) { + auto current_processor = update_processors_.begin(); + std::advance(current_processor, processor_index); + const size_t partition_index = + (*current_processor)->GetCurrentPartitionIndex(); + if ((*current_processor) + ->Process(bandpassed_noise_left_, bandpassed_noise_right_, + kernel_channel_left, kernel_channel_right)) { + left_filter_.ReplacePartition(partition_index, *kernel_channel_left); + right_filter_.ReplacePartition(partition_index, *kernel_channel_right); + ++processor_index; + } else { + // Update of the |current_processor| is finished, move it to the end of + // the list and reduce the number of active processors. + update_processors_.splice(update_processors_.end(), update_processors_, + current_processor); + --num_active_processors_; + } + } + + // Filter the input (Using the output buffer due to the delay operation). + fft_manager_->FreqFromTimeDomain((*output)[0], &temp_freq_buffer_[0]); + + left_filter_.Filter(temp_freq_buffer_[0]); + right_filter_.Filter(temp_freq_buffer_[0]); + + left_filter_.GetFilteredSignal(&(*output)[0]); + right_filter_.GetFilteredSignal(&(*output)[1]); +} + +void ReverbOnsetCompensator::Update(const float* rt60_values, float gain) { + DCHECK(rt60_values); + // Reset a reverb update processor from the end of the list and place it at + // the front. If the list is full, rotate the list and reuse the oldest active + // processor. + std::list<std::unique_ptr<ReverbOnsetUpdateProcessor>>::iterator + new_processor; + if (num_active_processors_ < kNumReverbUpdaters) { + new_processor = update_processors_.end(); + std::advance(new_processor, -1); + } else { + new_processor = update_processors_.begin(); + } + + (*new_processor)->SetReverbTimes(rt60_values); + (*new_processor)->SetGain(gain); + + if (new_processor != update_processors_.begin()) { + auto list_item = update_processors_.begin(); + std::advance(list_item, num_active_processors_); + if (list_item != new_processor) { + update_processors_.splice(list_item, update_processors_, new_processor, + std::next(new_processor)); + } + ++num_active_processors_; + } else { + std::rotate(update_processors_.begin(), + std::next(update_processors_.begin()), + update_processors_.end()); + } +} + +void ReverbOnsetCompensator::GenerateCorrectionCurves() { + // Copy into the adder curves such that the memory is aligned. + std::copy(kLowCorrectionCurve, kLowCorrectionCurve + kCorrectionCurveLength, + adder_curves_[0].begin()); + std::copy(kHighCorrectionCurve, kHighCorrectionCurve + kCorrectionCurveLength, + adder_curves_[1].begin()); + + // Evaluate the polynomials to generate the base curves. Here the 'low' and + // 'high' names refer to the reverberation times. + AudioBuffer::Channel* low_channel = &base_curves_[0]; + AudioBuffer::Channel* high_channel = &base_curves_[1]; + for (size_t i = 0; i < kCorrectionCurveLength; ++i) { + // Scaled independent variable (Allowed better conditioning). + const float conditioning_scalar = + (static_cast<float>(i) - kCurveOffset) * kCurveScale; + (*low_channel)[i] = kLowReverberationCorrectionCurve[0]; + (*high_channel)[i] = kHighReverberationCorrectionCurve[0]; + float power = conditioning_scalar; + for (size_t k = 1; k < kCurvePolynomialLength; ++k) { + (*low_channel)[i] += power * kLowReverberationCorrectionCurve[k]; + (*high_channel)[i] += power * kHighReverberationCorrectionCurve[k]; + power *= conditioning_scalar; + } + (*low_channel)[i] = std::max((*low_channel)[i], 0.0f); + (*high_channel)[i] = std::max((*high_channel)[i], 0.0f); + } +} + +void ReverbOnsetCompensator::GenerateNoiseVectors() { + const size_t num_octave_bands = GetNumReverbOctaveBands(sampling_rate_); + const size_t noise_length = CeilToMultipleOfFramesPerBuffer( + kCorrectionCurveLength, frames_per_buffer_); + for (size_t band = 0; band < num_octave_bands; ++band) { + // Generate preset tail. + bandpassed_noise_left_.emplace_back(kNumMonoChannels, noise_length); + GenerateBandLimitedGaussianNoise(kOctaveBandCentres[band], sampling_rate_, + /*seed=*/1U, + &bandpassed_noise_left_[band]); + bandpassed_noise_right_.emplace_back(kNumMonoChannels, noise_length); + GenerateBandLimitedGaussianNoise(kOctaveBandCentres[band], sampling_rate_, + /*seed=*/2U, + &bandpassed_noise_right_[band]); + + auto min_max = std::minmax_element(bandpassed_noise_left_[band][0].begin(), + bandpassed_noise_left_[band][0].end()); + const float left_scale = + std::max(std::fabs(*min_max.first), std::fabs(*min_max.second)); + min_max = std::minmax_element(bandpassed_noise_right_[band][0].begin(), + bandpassed_noise_right_[band][0].end()); + const float right_scale = + std::max(std::fabs(*min_max.first), std::fabs(*min_max.second)); + + const float scale = std::max(left_scale, right_scale); + + ScalarMultiply(noise_length, scale, bandpassed_noise_left_[band][0].begin(), + bandpassed_noise_left_[band][0].begin()); + ScalarMultiply(noise_length, scale, + bandpassed_noise_right_[band][0].begin(), + bandpassed_noise_right_[band][0].begin()); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_compensator.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_compensator.h new file mode 100644 index 000000000..4bf76f8f2 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_compensator.h @@ -0,0 +1,110 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_REVERB_ONSET_COMPENSATOR_H_ +#define RESONANCE_AUDIO_DSP_REVERB_ONSET_COMPENSATOR_H_ + +#include <list> +#include <vector> + +#include "base/audio_buffer.h" +#include "dsp/delay_filter.h" +#include "dsp/fft_manager.h" +#include "dsp/partitioned_fft_filter.h" +#include "dsp/reverb_onset_update_processor.h" + +namespace vraudio { + +// Implements a convolutional compensator for the spectral reverb onset curve. +class ReverbOnsetCompensator { + public: + // Constructs a |ReverbOnsetCompensator|. + // + // @param sampling_rate The sampling rate in Hz. + // @param frames_per_buffer The number of frames per buffer in the system. + // @param fft_manager Pointer to a FftManager to perform FFT transformations. + ReverbOnsetCompensator(int sampling_rate, size_t frames_per_buffer, + FftManager* fft_manager); + + // Resets the reverb with a new set of reverberation times. The new tail is + // generated by replacing the current tail buffer by buffer. + // + // @param rt60_values |kNumReverbOctaveBands| values denoting the + // reverberation decay time to -60dB in octave bands starting at + // |kLowestOctaveBand|. + // @param gain Gain to be applied across all frequencies. + void Update(const float* rt60_values, float gain); + + // Processes a mono |AudioBuffer| with a reverberant tail. + // + // @param input A mono |AudioBuffer| of input data. + // @param output Pointer to stereo output buffer. + void Process(const AudioBuffer& input, AudioBuffer* output); + + private: + // Generates the constituent curves which are combined to make up the + // correction curve envelopes. These envelopes ensure the initial part of the + // specral reverb's impulse response, which exhibits 'growing' behaviour, + // follows the desired exponential decay. + void GenerateCorrectionCurves(); + + // Generates a pair of |kNumOctaveBands| band, octave filtered, noise buffers. + void GenerateNoiseVectors(); + + // Manager for all FFT related functionality (not owned). + FftManager* const fft_manager_; + + // The system sampling rate. + const int sampling_rate_; + + // The system number of frames per buffer. + const size_t frames_per_buffer_; + + // Pre-generated band-passed noise to be used as a base for the reverb tail. + std::vector<AudioBuffer> bandpassed_noise_left_; + std::vector<AudioBuffer> bandpassed_noise_right_; + + // The constituent curves used to generate the onset compensation envelopes. + AudioBuffer base_curves_; + AudioBuffer adder_curves_; + + // Filter for processing the left reverberant channel. + PartitionedFftFilter left_filter_; + + // Filter for processing the right reverberant channel. + PartitionedFftFilter right_filter_; + + // Delay filter used to ensure the compensation curve starts at the same point + // as the spectral reverb. + DelayFilter delay_filter_; + + // Number of active update processors. + size_t num_active_processors_; + + // Active reverb update processors to replace the corresponding filter + // partitions of the reverb tail within each process call. + std::list<std::unique_ptr<ReverbOnsetUpdateProcessor>> update_processors_; + + // Temporary buffer used to process filter kernel partitions. + AudioBuffer temp_kernel_buffer_; + + // Temporary buffer to hold FFT frequency domain output. + PartitionedFftFilter::FreqDomainBuffer temp_freq_buffer_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_REVERB_ONSET_COMPENSATOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_update_processor.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_update_processor.cc new file mode 100644 index 000000000..3a5dbcf02 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_update_processor.cc @@ -0,0 +1,184 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/reverb_onset_update_processor.h" + +#include <algorithm> + +#include "base/constants_and_types.h" +#include "base/simd_utils.h" +#include "dsp/spectral_reverb_constants_and_tables.h" +#include "dsp/utils.h" + +namespace vraudio { + +namespace { + +// Find the absolute difference between two size_t values. +inline size_t absdiff(size_t lhs, size_t rhs) { + return lhs > rhs ? lhs - rhs : rhs - lhs; +} + +} // namespace + +ReverbOnsetUpdateProcessor::ReverbOnsetUpdateProcessor( + size_t frames_per_buffer, int sampling_rate, AudioBuffer* base_curves, + AudioBuffer* adder_curves) + : sampling_rate_(sampling_rate), + tail_update_cursor_(0), + tail_length_(CeilToMultipleOfFramesPerBuffer(kCorrectionCurveLength, + frames_per_buffer)), + gain_(1.0f), + curve_indices_(GetNumReverbOctaveBands(sampling_rate_), kInvalidIndex), + pure_decay_coefficients_(curve_indices_.size(), 0.0f), + pure_decay_exponents_(curve_indices_.size(), 0.0f), + band_buffer_(kNumStereoChannels, frames_per_buffer), + envelope_buffer_(kNumMonoChannels, frames_per_buffer), + base_curves_(base_curves), + adder_curves_(adder_curves) {} + +void ReverbOnsetUpdateProcessor::SetReverbTimes(const float* rt60_values) { + DCHECK(rt60_values); + const size_t num_octave_bands = curve_indices_.size(); + const float sampling_rate_float = static_cast<float>(sampling_rate_); + tail_update_cursor_ = 0; + // Choose curves for each band. + for (size_t band = 0; band < num_octave_bands; ++band) { + curve_indices_[band] = + GetFeedbackIndexFromRt60(rt60_values[band], sampling_rate_float); + // Deal with the case where only the convolution is needed. + if (curve_indices_[band] == kInvalidIndex) { + const float min_reverb_time = + kMinReverbTimeForFeedback48kHz * + (sampling_rate_float / kDefaultSpectralReverbSampleRate); + const float effective_rt = + rt60_values[band] <= min_reverb_time ? rt60_values[band] : 0.0f; + pure_decay_exponents_[band] = + std::abs(effective_rt) > kEpsilonFloat + ? std::exp(kNegativeLog1000 / + (sampling_rate_float * effective_rt)) + : 0.0f; + pure_decay_coefficients_[band] = pure_decay_exponents_[band]; + } + } +} + +bool ReverbOnsetUpdateProcessor::Process( + const std::vector<AudioBuffer>& bandpassed_noise_left, + const std::vector<AudioBuffer>& bandpassed_noise_right, + AudioBuffer::Channel* kernel_channel_left, + AudioBuffer::Channel* kernel_channel_right) { + if (tail_update_cursor_ >= tail_length_) { + // Processing the reverb tail is finished. + tail_update_cursor_ = 0; + return false; + } + const size_t frames_per_buffer = band_buffer_.num_frames(); + DCHECK(kernel_channel_left); + DCHECK(kernel_channel_right); + DCHECK_EQ(bandpassed_noise_left.size(), curve_indices_.size()); + DCHECK_EQ(bandpassed_noise_right.size(), curve_indices_.size()); + DCHECK_EQ(bandpassed_noise_left[0].num_frames(), tail_length_); + DCHECK_EQ(bandpassed_noise_right[0].num_frames(), tail_length_); + DCHECK_GE(tail_length_, kCorrectionCurveLength); + DCHECK_EQ(kernel_channel_left->size(), frames_per_buffer); + DCHECK_EQ(kernel_channel_right->size(), frames_per_buffer); + + // Clear for accumulation per frequency band. + kernel_channel_left->Clear(); + kernel_channel_right->Clear(); + + AudioBuffer::Channel& band_channel_left = band_buffer_[0]; + AudioBuffer::Channel& band_channel_right = band_buffer_[1]; + // Define the number of samples we are still able to copy from the multiplier + // and adder curves. + const size_t copy_length = + frames_per_buffer + tail_update_cursor_ <= kCorrectionCurveLength + ? frames_per_buffer + : absdiff(kCorrectionCurveLength, tail_update_cursor_); + AudioBuffer::Channel* envelope_channel = &envelope_buffer_[0]; + // Compute the band buffer for each band response. + for (size_t band = 0; band < curve_indices_.size(); ++band) { + const AudioBuffer::Channel& noise_channel_left = + bandpassed_noise_left[band][0]; + const AudioBuffer::Channel& noise_channel_right = + bandpassed_noise_right[band][0]; + // Fill the band buffer with the next noise buffer and apply gain. + ScalarMultiply(frames_per_buffer, gain_, + noise_channel_left.begin() + tail_update_cursor_, + band_channel_left.begin()); + ScalarMultiply(frames_per_buffer, gain_, + noise_channel_right.begin() + tail_update_cursor_, + band_channel_right.begin()); + // Skip the band if we have an invalid index + const int curve_index = curve_indices_[band]; + if (curve_index != kInvalidIndex) { + // Apply the correct compensation curve to the buffer. + const float scale = kCurveCorrectionMultipliers[curve_index]; + AudioBuffer::Channel* adder_curve_channel; + if (tail_update_cursor_ < kCorrectionCurveLength) { + // Use either the high frequency or low frequency curve. + if (static_cast<size_t>(curve_index) >= kCurveChangeoverIndex) { + adder_curve_channel = &(*adder_curves_)[1]; + std::copy_n((*base_curves_)[1].begin() + tail_update_cursor_, + copy_length, envelope_channel->begin()); + } else { + adder_curve_channel = &(*adder_curves_)[0]; + std::copy_n((*base_curves_)[0].begin() + tail_update_cursor_, + copy_length, envelope_channel->begin()); + } + // Construct the correct envelope (chunk thereof). + ScalarMultiplyAndAccumulate( + copy_length, scale, + adder_curve_channel->begin() + tail_update_cursor_, + envelope_channel->begin()); + // Ensure the end part of the envelope does not contain spurious data. + std::fill(envelope_channel->begin() + copy_length, + envelope_channel->end(), 0.0f); + } else { + // If we have moved past the length of the correction curve, fill the + // envelope chunk with zeros. + envelope_channel->Clear(); + } + + // Apply that envelope to the given band and accumulate into the output. + MultiplyAndAccumulatePointwise( + frames_per_buffer, envelope_channel->begin(), + band_channel_left.begin(), kernel_channel_left->begin()); + MultiplyAndAccumulatePointwise( + frames_per_buffer, envelope_channel->begin(), + band_channel_right.begin(), kernel_channel_right->begin()); + } else { + // If the decay time is too short for the spectral reverb to make a + // contribution (0.15s @48kHz), the compensation filter will consist of + // the entire tail. + for (size_t frame = 0; frame < frames_per_buffer; ++frame) { + (*kernel_channel_left)[frame] += + pure_decay_coefficients_[band] * band_channel_left[frame]; + (*kernel_channel_right)[frame] += + pure_decay_coefficients_[band] * band_channel_right[frame]; + // Update the decay coefficient. + pure_decay_coefficients_[band] *= pure_decay_exponents_[band]; + } + } + } + // Update the cursor. + tail_update_cursor_ += frames_per_buffer; + + return true; +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_update_processor.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_update_processor.h new file mode 100644 index 000000000..19fee4b42 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/reverb_onset_update_processor.h @@ -0,0 +1,113 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_REVERB_ONSET_UPDATE_PROCESSOR_H_ +#define RESONANCE_AUDIO_DSP_REVERB_ONSET_UPDATE_PROCESSOR_H_ + +#include <vector> + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Updater for the |ReverbOnsetCompensator|. +class ReverbOnsetUpdateProcessor { + public: + // Constructs a update processor for the sprectral reverb onset compensator. + // + // @param frames_per_buffer_ System buffer length in frames. + // @param sampling_rate System sample rate. + // @param base_curves Constituent curve used for envelope generation. + // @param ader_curves Constituent curve used for envelope generation. + ReverbOnsetUpdateProcessor(size_t frames_per_buffer, int sampling_rate, + AudioBuffer* base_curves, + AudioBuffer* adder_curves); + + // Sets reverberation times in different frequency bands. + // + // @param rt60_values |kNumReverbOctaveBands| values denoting the + // reverberation decay time to -60dB in octave bands starting at + // |kLowestOctaveBand|. + void SetReverbTimes(const float* rt60_values); + + // Sets the gain applied to the overall compensation envelope. + // + // @param gain Gain applied to overall envelope. + void SetGain(float gain) { gain_ = gain; } + + // Processes the next tail update. + // + // @param bandpassed_noise_left Pre-computed bandpassed noise buffer. + // @param bandpassed_noise_right Pre-computed bandpassed noise buffer. + // @param kernel_channel_left Kernel channel to fill in the processed output. + // @param kernel_channel_right Kernel channel to fill in the processed output. + // @return True if the tail update continues. + bool Process(const std::vector<AudioBuffer>& bandpassed_noise_left, + const std::vector<AudioBuffer>& bandpassed_noise_right, + AudioBuffer::Channel* kernel_channel_left, + AudioBuffer::Channel* kernel_channel_right); + + // Returns the partition index of the current update state. + // + // @return Current partition index. + size_t GetCurrentPartitionIndex() const { + const size_t frames_per_buffer = band_buffer_.num_frames(); + DCHECK_NE(frames_per_buffer, 0U); + return tail_update_cursor_ / frames_per_buffer; + } + + // Disable copy and assignment operator. + ReverbOnsetUpdateProcessor(ReverbOnsetUpdateProcessor const&) = delete; + void operator=(ReverbOnsetUpdateProcessor const&) = delete; + + private: + // System sample rate. + int sampling_rate_; + + // Current frame position of the reverb tail to be updated. + size_t tail_update_cursor_; + + // Length of the new reverb tail to be replaced in frames. + size_t tail_length_; + + // Gain applied to the reverb compensation. + float gain_; + + // Indices of the multiplication factor to be used to create the onset + // compensation curve at each frequency band. + std::vector<int> curve_indices_; + + // Decay coefficients per each band of the reverb tail, used below 0.15s + // @48kHz. + std::vector<float> pure_decay_coefficients_; + + // Decay exponential per each band of the reverb tail, used below 0.15s + // @48kHz. + std::vector<float> pure_decay_exponents_; + + // Temporary buffers used to process the decayed noise per each band. + AudioBuffer band_buffer_; + AudioBuffer envelope_buffer_; + + // Pointers to audio buffers owned by the |ReverbOnsetCompensator| storing the + // constituent curves used to generate the onset compensation envelopes. + AudioBuffer* base_curves_; + AudioBuffer* adder_curves_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_REVERB_ONSET_UPDATE_PROCESSOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/sh_hrir_creator.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/sh_hrir_creator.cc new file mode 100644 index 000000000..11226a6fc --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/sh_hrir_creator.cc @@ -0,0 +1,72 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/sh_hrir_creator.h" + +#include "third_party/SADIE_hrtf_database/generated/hrtf_assets.h" +#include "ambisonics/utils.h" +#include "base/logging.h" +#include "dsp/resampler.h" +#include "utils/planar_interleaved_conversion.h" + +namespace vraudio { + +std::unique_ptr<AudioBuffer> CreateShHrirsFromWav(const Wav& wav, + int target_sample_rate_hz, + Resampler* resampler) { + DCHECK(resampler); + const size_t num_channels = wav.GetNumChannels(); + CHECK(IsValidAmbisonicOrder(num_channels)); + + const size_t sh_hrir_length = wav.interleaved_samples().size() / num_channels; + std::unique_ptr<AudioBuffer> sh_hrirs( + new AudioBuffer(num_channels, sh_hrir_length)); + FillAudioBuffer(wav.interleaved_samples(), num_channels, sh_hrirs.get()); + + const int wav_sample_rate_hz = wav.GetSampleRateHz(); + CHECK_GT(wav_sample_rate_hz, 0); + CHECK_GT(target_sample_rate_hz, 0); + if (wav_sample_rate_hz != target_sample_rate_hz) { + if (!Resampler::AreSampleRatesSupported(wav_sample_rate_hz, + target_sample_rate_hz)) { + LOG(FATAL) << "Unsupported sampling rates for loading HRIRs: " + << wav_sample_rate_hz << ", " << target_sample_rate_hz; + } + resampler->ResetState(); + // Resample the SH HRIRs if necessary. + resampler->SetRateAndNumChannels(wav_sample_rate_hz, target_sample_rate_hz, + num_channels); + std::unique_ptr<AudioBuffer> resampled_sh_hrirs(new AudioBuffer( + num_channels, resampler->GetNextOutputLength(sh_hrir_length))); + resampler->Process(*sh_hrirs, resampled_sh_hrirs.get()); + return resampled_sh_hrirs; + } + return sh_hrirs; +} + +std::unique_ptr<AudioBuffer> CreateShHrirsFromAssets( + const std::string& filename, int target_sample_rate_hz, + Resampler* resampler) { + // Read SH HRIR from asset store. + sadie::HrtfAssets hrtf_assets; + std::unique_ptr<std::string> sh_hrir_data = hrtf_assets.GetFile(filename); + CHECK_NOTNULL(sh_hrir_data.get()); + std::istringstream wav_data_stream(*sh_hrir_data); + std::unique_ptr<const Wav> wav = Wav::CreateOrNull(&wav_data_stream); + return CreateShHrirsFromWav(*wav, target_sample_rate_hz, resampler); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/sh_hrir_creator.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/sh_hrir_creator.h new file mode 100644 index 000000000..5a11926f5 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/sh_hrir_creator.h @@ -0,0 +1,54 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_SH_HRIR_CREATOR_H_ +#define RESONANCE_AUDIO_DSP_SH_HRIR_CREATOR_H_ + +#include "base/audio_buffer.h" +#include "dsp/resampler.h" +#include "utils/wav.h" + +namespace vraudio { + +// Creates a multichannel audio buffer of Spherical Harmonic-encoded Head +// Related Impulse Responses (SH HRIRs) from Wav SH HRIR assets. It also +// checks if the channel count of the SH HRIR file is correct and resamples the +// SH HRIRs if necessary to match the system (target) sampling rate. +// +// @param wav |Wav| instance that contains SH HRIRs. +// @param target_sample_rate_hz Target sampling rate in Hertz. +// @param resampler Pointer to a resampler used to convert HRIRs to the system +// rate, +// (This resampler's internal state will be reset on each function call). +// @return Unique pointer to |AudioBuffer| where the SH HRIRs will be written. +std::unique_ptr<AudioBuffer> CreateShHrirsFromWav(const Wav& wav, + int target_sample_rate_hz, + Resampler* resampler); + +// Creates a SH HRIR multichannel audio buffer from assets. +// +// @param filename Name of the Wav file that contains SH HRIRs. +// @param target_sample_rate_hz Target sampling rate in Hertz. +// @param resampler Pointer to a resampler used to convert HRIRs to the system +// rate. +// @return Unique pointer to |AudioBuffer| where the SH HRIRs will be written. +std::unique_ptr<AudioBuffer> CreateShHrirsFromAssets( + const std::string& filename, int target_sample_rate_hz, + Resampler* resampler); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_SH_HRIR_CREATOR_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room.cc new file mode 100644 index 000000000..769166248 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room.cc @@ -0,0 +1,61 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/shoe_box_room.h" + +#include "base/constants_and_types.h" + +namespace vraudio { + +void ComputeReflections(const WorldPosition& relative_listener_position, + const WorldPosition& room_dimensions, + const float* reflection_coefficients, + std::vector<Reflection>* reflections) { + DCHECK(reflection_coefficients); + DCHECK(reflections); + DCHECK_EQ(reflections->size(), kNumRoomSurfaces); + const WorldPosition kOrigin(0.0f, 0.0f, 0.0f); + if (!IsPositionInAabb(relative_listener_position, kOrigin, room_dimensions)) { + // Listener is outside the room, skip computation. + std::fill(reflections->begin(), reflections->end(), Reflection()); + return; + } + // Calculate the distance of the listener to each wall. + // Since all the sources are 'attached' to the listener in the computation + // of reflections, the distance travelled is arbitrary. So, we add 1.0f to + // the computed distance in order to avoid delay time approaching 0 and the + // magnitude approaching +inf. + const WorldPosition offsets = 0.5f * room_dimensions; + const float distances_travelled[kNumRoomSurfaces] = { + offsets[0] + relative_listener_position[0] + 1.0f, + offsets[0] - relative_listener_position[0] + 1.0f, + offsets[1] + relative_listener_position[1] + 1.0f, + offsets[1] - relative_listener_position[1] + 1.0f, + offsets[2] + relative_listener_position[2] + 1.0f, + offsets[2] - relative_listener_position[2] + 1.0f}; + + for (size_t i = 0; i < kNumRoomSurfaces; ++i) { + // Convert distances to time delays in seconds. + (*reflections)[i].delay_time_seconds = + distances_travelled[i] / kSpeedOfSound; + // Division by distance is performed here as we don't want this applied more + // than once. + (*reflections)[i].magnitude = + reflection_coefficients[i] / distances_travelled[i]; + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room.h new file mode 100644 index 000000000..eab514922 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room.h @@ -0,0 +1,44 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_SHOE_BOX_ROOM_H_ +#define RESONANCE_AUDIO_DSP_SHOE_BOX_ROOM_H_ + +#include <vector> + +#include "base/misc_math.h" +#include "dsp/reflection.h" + +namespace vraudio { + +// Computes a set of reflections from each surface of a shoe-box room model. +// Uses a simplified calculation method which assumes that all the sources are +// 'attached' to the listener. Also, assumes that the listener is inside the +// shoe-box room. +// +// @param relative_listener_position Relative listener position to the center of +// the room. +// @param room_dimensions Dimensions of the room. +// @param reflection_coefficients Reflection coefficients. +// @return List of computed reflections by the image source method. +void ComputeReflections(const WorldPosition& relative_listener_position, + const WorldPosition& room_dimensions, + const float* reflection_coefficients, + std::vector<Reflection>* reflections); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_SHOE_BOX_ROOM_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room_test.cc new file mode 100644 index 000000000..78393604f --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/shoe_box_room_test.cc @@ -0,0 +1,96 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/shoe_box_room.h" + +#include <algorithm> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +// Expected reflection delay times in seconds. +const float kExpectedDelays[kNumRoomSurfaces] = { + 0.011661f, 0.005830f, 0.016035f, 0.004373f, 0.020408f, 0.002915f}; + +// Expected reflection magnitudes. +const float kExpectedMagnitudes[kNumRoomSurfaces] = { + 0.25f, 0.5f, 0.181818f, 0.666666f, 0.142857f, 1.0f}; + +// Tests that a set of single reflections arrive in from the correct directions. +TEST(ShoeBoxRoomTest, SingleReflectionsTest) { + const WorldPosition kListenerPosition(1.0f, 2.0f, 3.0f); + const WorldPosition kRoomDimensions(4.0f, 5.0f, 6.0f); + + // Perform the simplified image source method with the wall at the given index + // having a reflection coefficient of 1 and all of the other walls having a + // reflection coefficient of 0. Thus we expect one reflection in the output + // and we should be able to predict its magnitude, delay and direction of + // arrival in terms of azimuth and elevation. + float reflection_coefficients[kNumRoomSurfaces]; + std::fill(std::begin(reflection_coefficients), + std::end(reflection_coefficients), 0.0f); + std::vector<Reflection> reflections(kNumRoomSurfaces); + for (size_t index = 0; index < kNumRoomSurfaces; ++index) { + reflection_coefficients[index] = 1.0f; + + ComputeReflections(kListenerPosition, kRoomDimensions, + reflection_coefficients, &reflections); + EXPECT_EQ(kNumRoomSurfaces, reflections.size()); + + // Check that the correct reflection is returned for the given call. + size_t num_returned = 0; + for (size_t i = 0; i < kNumRoomSurfaces; ++i) { + if (reflections[i].magnitude > 0.0f) { + EXPECT_NEAR(kExpectedDelays[index], reflections[i].delay_time_seconds, + kEpsilonFloat); + EXPECT_NEAR(kExpectedMagnitudes[index], reflections[i].magnitude, + kEpsilonFloat); + ++num_returned; + } + } + EXPECT_EQ(1U, num_returned); + + // Reset so that all reflection reflection_coefficients are 0.0f again. + reflection_coefficients[index] = 0.0f; + } +} + +// Tests that no reflections arrive when the listener is outside the room. +TEST(ShoeBoxRoomTest, ReflectionsOutsideRoomTest) { + const WorldPosition kListenerPosition(4.0f, 5.0f, 6.0f); + const WorldPosition kRoomDimensions(1.0f, 2.0f, 3.0f); + const float kReflectionCoefficients[] = {1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f}; + + std::vector<Reflection> reflections(kNumRoomSurfaces); + ComputeReflections(kListenerPosition, kRoomDimensions, + kReflectionCoefficients, &reflections); + EXPECT_EQ(kNumRoomSurfaces, reflections.size()); + + // Check that all the reflections have zeros. + for (size_t i = 0; i < kNumRoomSurfaces; ++i) { + EXPECT_NEAR(0.0f, reflections[i].delay_time_seconds, kEpsilonFloat); + EXPECT_NEAR(0.0f, reflections[i].magnitude, kEpsilonFloat); + } +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb.cc new file mode 100644 index 000000000..e03d34ccd --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb.cc @@ -0,0 +1,350 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +// Prevent Visual Studio from complaining about std::copy_n. +#if defined(_WIN32) +#define _SCL_SECURE_NO_WARNINGS +#endif + +#include "dsp/spectral_reverb.h" + +#include <algorithm> +#include <cstdlib> +#include <numeric> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/simd_utils.h" +#include "dsp/spectral_reverb_constants_and_tables.h" +#include "dsp/utils.h" + +namespace vraudio { + +namespace { + +// FFT length and length of time domain data processed. +const size_t kFftSize = 4096; + +// Length of magnitude and phase buffers. +const size_t kMagnitudeLength = kFftSize / 2 + 1; + +// Number of overlaps per kFftSize time domain chunk of input. +const size_t kNumOverlap = 4; + +// Number of samples per overlap. +const size_t kOverlapLength = kFftSize / kNumOverlap; + +// Three quarters FFT size, used for copying chunks of input. +const size_t kThreeQuarterFftSize = kFftSize - kOverlapLength; + +// Number of channels needed to overlap add output data. +const size_t kNumQuadChannels = 4; + +// Number of buffers of delay applies to the magnitude spectrum. +const size_t kMagnitudeDelay = 3; + +// Length of a buffer of noise used to provide random phase. +const size_t kNoiseLength = 16384; + +// Length of the noise buffer from which a phase buffer can start. +const size_t kAvailableNoiselength = kNoiseLength - kMagnitudeLength; + +// Returns a random integer in the range provided. Used to index into the random +// buffer for phase values. +inline size_t GetRandomIntegerInRange(size_t min, size_t max) { + DCHECK_GE(max, min); + return min + static_cast<size_t>(std::rand()) % (max - min); +} + +} // namespace + +SpectralReverb::SpectralReverb(int sample_rate, size_t frames_per_buffer) + : sample_rate_(sample_rate), + frames_per_buffer_(frames_per_buffer), + magnitude_delay_index_(0), + overlap_add_index_(0), + fft_manager_(kFftSize / 2), + sin_cos_random_phase_buffer_(kNumStereoChannels, kNoiseLength), + unscaled_window_(kNumMonoChannels, kFftSize), + window_(kNumMonoChannels, kFftSize), + feedback_(kNumMonoChannels, kMagnitudeLength), + magnitude_compensation_(kNumMonoChannels, kMagnitudeLength), + magnitude_delay_(kMagnitudeDelay, kMagnitudeLength), + fft_size_input_(kNumMonoChannels, kFftSize), + input_circular_buffer_(kFftSize + frames_per_buffer_, frames_per_buffer_, + kOverlapLength), + output_circular_buffers_(kNumStereoChannels), + out_time_buffer_(kNumQuadChannels, kFftSize), + temp_freq_buffer_(kNumStereoChannels, kFftSize), + scaled_magnitude_buffer_(kNumMonoChannels, kMagnitudeLength), + temp_magnitude_buffer_(kNumMonoChannels, kMagnitudeLength), + temp_phase_buffer_(kNumStereoChannels, kMagnitudeLength), + output_accumulator_(kNumStereoChannels), + is_gain_near_zero_(false), + is_feedback_near_zero_(false) { + DCHECK_GT(sample_rate, 0); + DCHECK_GT(frames_per_buffer_, 0U); + + // Seed std::rand, used for phase selection. + std::srand(1); + GenerateRandomPhaseBuffer(); + GenerateAnalysisWindow(); + InitializeCircularBuffersAndAccumulators(); + fft_size_input_.Clear(); + magnitude_compensation_.Clear(); +} + +void SpectralReverb::SetGain(float gain) { + DCHECK_GE(gain, 0.0f); + ScalarMultiply(window_.num_frames(), gain, &unscaled_window_[0][0], + &window_[0][0]); + // If the gain is less than -60dB we will bypass all processing. + is_gain_near_zero_ = gain <= kNegative60dbInAmplitude; + // If we are to bypass processing we clear the circular buffer so that we + // don't process old input when the spectral reverb is restarted. + if (is_gain_near_zero_ || is_feedback_near_zero_) { + input_circular_buffer_.Clear(); + } +} + +void SpectralReverb::SetRt60PerOctaveBand(const float* rt60_values) { + DCHECK(rt60_values); + const float sample_rate_float = static_cast<float>(sample_rate_); + // Fill the entries in the feedback channel with the feedback values up to + // that frequency. The feedback channels entries are per frequency in the same + // way as the magnitude vectors. Also fill the magnitude compensation vector + // such that the peak value at each frequency for any given reverberation time + // will be |kDefaultReverbGain|. + AudioBuffer::Channel* feedback_channel = &feedback_[0]; + feedback_channel->Clear(); + AudioBuffer::Channel* magnitude_compensation_channel = + &magnitude_compensation_[0]; + magnitude_compensation_channel->Clear(); + const float frequency_step = sample_rate_float / static_cast<float>(kFftSize); + int index = GetFeedbackIndexFromRt60(rt60_values[0], sample_rate_float); + float current_feedback = + index == kInvalidIndex ? 0.0f : kSpectralReverbFeedback[index]; + float magnitude_compensation_value = + index == kInvalidIndex ? 0.0f : kMagnitudeCompensation[index]; + const size_t max_frequency_bin = + std::min(static_cast<size_t>( + (kOctaveBandCentres[kNumReverbOctaveBands - 1] * kSqrtTwo) / + frequency_step), + feedback_channel->size()); + // The upper edge of the octave band is sqrt(2) * centre_frequency : + // https://en.wikipedia.org/wiki/Octave_band#Octave_Bands + float upper_octave_band_edge = kOctaveBandCentres[0] * kSqrtTwo; + for (size_t i = 0, j = 0; i < max_frequency_bin; ++i) { + const float current_freq = static_cast<float>(i) * frequency_step; + if (current_freq > upper_octave_band_edge) { + ++j; + DCHECK_LT(j, kNumReverbOctaveBands); + upper_octave_band_edge = kOctaveBandCentres[j] * kSqrtTwo; + index = GetFeedbackIndexFromRt60(rt60_values[j], sample_rate_float); + current_feedback = + index == kInvalidIndex ? 0.0f : kSpectralReverbFeedback[index]; + magnitude_compensation_value = + index == kInvalidIndex ? 0.0f : kMagnitudeCompensation[index]; + } + (*feedback_channel)[i] = current_feedback; + (*magnitude_compensation_channel)[i] = magnitude_compensation_value; + } + // If the sum of all feedback values is below the minimum feedback value, it + // is safe to assume we can bypass the spectral reverb entirely. + is_feedback_near_zero_ = + std::accumulate(feedback_channel->begin(), feedback_channel->end(), + 0.0f) < kSpectralReverbFeedback[0]; + // If we are to bypass processing we clear the circular buffer so that we + // don't process old input when the spectral reverb is restarted. + if (is_gain_near_zero_ || is_feedback_near_zero_) { + input_circular_buffer_.Clear(); + } +} + +void SpectralReverb::Process(const AudioBuffer::Channel& input, + AudioBuffer::Channel* left_out, + AudioBuffer::Channel* right_out) { + DCHECK_EQ(input.size(), left_out->size()); + DCHECK_EQ(input.size(), right_out->size()); + DCHECK_EQ(input.size(), frames_per_buffer_); + + + if (is_gain_near_zero_ || is_feedback_near_zero_) { + left_out->Clear(); + right_out->Clear(); + return; + } + + // Here we insert |frames_per_buffer_| samples on each function call. Then, + // once there are |kOverlapLength| samples in the input circular buffer we + // retrieve |kOverlapLength| samples from it and process |kFftSize| samples of + // input at a time, sliding along by |kOverlapLength| samples. We then place + // |kOverlapLength| samples into the output buffer and we will extract + // |frames_per_buffer_| samples from the output buffer on each function call. + input_circular_buffer_.InsertBuffer(input); + while (input_circular_buffer_.GetOccupancy() >= kOverlapLength) { + std::copy_n(&fft_size_input_[0][kOverlapLength], kThreeQuarterFftSize, + &fft_size_input_[0][0]); + input_circular_buffer_.RetrieveBufferWithOffset(kThreeQuarterFftSize, + &fft_size_input_[0]); + fft_manager_.FreqFromTimeDomain(fft_size_input_[0], &temp_freq_buffer_[0]); + fft_manager_.GetCanonicalFormatFreqBuffer(temp_freq_buffer_[0], + &temp_freq_buffer_[1]); + fft_manager_.MagnitudeFromCanonicalFreqBuffer(temp_freq_buffer_[1], + &scaled_magnitude_buffer_[0]); + // Apply the magnitude compensation to the input magnitude spectrum before + // feedback is applied. + MultiplyPointwise(kMagnitudeLength, magnitude_compensation_[0].begin(), + scaled_magnitude_buffer_[0].begin(), + scaled_magnitude_buffer_[0].begin()); + // Generate time domain reverb blocks. + GetNextReverbBlock(magnitude_delay_index_, &out_time_buffer_[0], + &out_time_buffer_[1]); + magnitude_delay_index_ = (magnitude_delay_index_ + 1) % kMagnitudeDelay; + GetNextReverbBlock(magnitude_delay_index_, &out_time_buffer_[2], + &out_time_buffer_[3]); + + // Combine the reverb blocks for both left and right output. + AddPointwise(kFftSize, out_time_buffer_[0].begin(), + out_time_buffer_[2].begin(), out_time_buffer_[0].begin()); + AddPointwise(kFftSize, out_time_buffer_[1].begin(), + out_time_buffer_[3].begin(), out_time_buffer_[1].begin()); + + // Window the left and right output (While applying inverse FFT scaling). + MultiplyPointwise(kFftSize, out_time_buffer_[0].begin(), window_[0].begin(), + out_time_buffer_[0].begin()); + MultiplyPointwise(kFftSize, out_time_buffer_[1].begin(), window_[0].begin(), + out_time_buffer_[1].begin()); + + // Next perform the addition and the submission into the output buffer. + AccumulateOverlap(0 /*channel*/, out_time_buffer_[0]); + AccumulateOverlap(1 /*channel*/, out_time_buffer_[1]); + overlap_add_index_ = (overlap_add_index_ + 1) % kNumOverlap; + } + output_circular_buffers_[0]->RetrieveBuffer(left_out); + output_circular_buffers_[1]->RetrieveBuffer(right_out); +} + +void SpectralReverb::AccumulateOverlap(size_t channel_index, + const AudioBuffer::Channel& buffer) { + // Use a modulo indexed multi channel audio buffer with each channel of length + // |kOverlapLength| to perform an overlap add. + for (size_t i = 0, index = overlap_add_index_; i < kNumOverlap; + ++i, index = (index + 1) % kNumOverlap) { + float* accumulator_start_point = + output_accumulator_[channel_index][index].begin(); + AddPointwise(kOverlapLength, buffer.begin() + i * kOverlapLength, + accumulator_start_point, accumulator_start_point); + } + output_circular_buffers_[channel_index]->InsertBuffer( + output_accumulator_[channel_index][overlap_add_index_]); + output_accumulator_[channel_index][overlap_add_index_].Clear(); +} + +void SpectralReverb::GenerateAnalysisWindow() { + // Genarate a pseudo tukey window from three overlapping hann windows, scaled + // by the inverse fft scale. + AudioBuffer::Channel* window_channel = &window_[0]; + // Use the unscaled window buffer as temporary storage. + GenerateHannWindow(true /* full */, kMagnitudeLength, &unscaled_window_[0]); + float* hann_window = &unscaled_window_[0][0]; + // Scale the hann window such that the sum of three will have unity peak. + const float kThreeQuarters = 0.75f; + ScalarMultiply(kMagnitudeLength, kThreeQuarters, hann_window, hann_window); + for (size_t offset = 0; offset < kThreeQuarterFftSize; + offset += kOverlapLength) { + float* tripple_hann_window = &(*window_channel)[offset]; + AddPointwise(kMagnitudeLength, hann_window, tripple_hann_window, + tripple_hann_window); + } + fft_manager_.ApplyReverseFftScaling(window_channel); + unscaled_window_[0] = *window_channel; +} + +void SpectralReverb::GenerateRandomPhaseBuffer() { + AudioBuffer::Channel* sin_phase_channel = &sin_cos_random_phase_buffer_[0]; + AudioBuffer::Channel* cos_phase_channel = &sin_cos_random_phase_buffer_[1]; + // Initially use the sin channel to store the random data before taking sine + // and cosine. + GenerateUniformNoise(/*min=*/0.0f, /*max=*/kPi, /*seed=*/1U, + sin_phase_channel); + for (size_t i = 0; i < sin_cos_random_phase_buffer_.num_frames(); ++i) { + + (*cos_phase_channel)[i] = std::cos((*sin_phase_channel)[i]); + (*sin_phase_channel)[i] = std::sin((*sin_phase_channel)[i]); + } +} + +void SpectralReverb::GetNextReverbBlock(size_t delay_index, + AudioBuffer::Channel* left_channel, + AudioBuffer::Channel* right_channel) { + DCHECK(left_channel); + DCHECK(right_channel); + + // Generate reverb magnitude by combining the delayed magnitude with the + // current magnitude. + AudioBuffer::Channel* temp_magnitude_channel = &temp_magnitude_buffer_[0]; + *temp_magnitude_channel = scaled_magnitude_buffer_[0]; + MultiplyAndAccumulatePointwise( + kMagnitudeLength, magnitude_delay_[delay_index].begin(), + feedback_[0].begin(), temp_magnitude_channel->begin()); + // Reinsert this new reverb magnitude into the delay buffer. + magnitude_delay_[delay_index] = *temp_magnitude_channel; + + for (size_t i = 0; i < kNumStereoChannels; ++i) { + // Extract a random phase buffer. + const size_t random_offset = + GetRandomIntegerInRange(0, kAvailableNoiselength); + // We gaurantee an aligned offset as when SSE is used we need it. + const size_t phase_offset = FindNextAlignedArrayIndex( + random_offset, sizeof(float), kMemoryAlignmentBytes); + // Convert from magnitude and phase to a time domain output. + fft_manager_.CanonicalFreqBufferFromMagnitudeAndSinCosPhase( + phase_offset, (*temp_magnitude_channel), + sin_cos_random_phase_buffer_[0], sin_cos_random_phase_buffer_[1], + &temp_freq_buffer_[0]); + + AudioBuffer::Channel* out_channel = i == 0 ? left_channel : right_channel; + fft_manager_.GetPffftFormatFreqBuffer(temp_freq_buffer_[0], + &temp_freq_buffer_[1]); + fft_manager_.TimeFromFreqDomain(temp_freq_buffer_[1], out_channel); + } +} + +void SpectralReverb::InitializeCircularBuffersAndAccumulators() { + AudioBuffer zeros(kNumMonoChannels, kOverlapLength); + zeros.Clear(); + for (size_t channel = 0; channel < kNumStereoChannels; ++channel) { + // Prefill the |output_circular_buffers_| with |kOverlapLength| / + // |frames_per_buffer_| calls worth of zeros. + output_circular_buffers_[channel].reset( + new CircularBuffer(kOverlapLength + frames_per_buffer_, kOverlapLength, + frames_per_buffer_)); + // Due to differences in the |frames_per_buffer_| used for input and output + // and |kOverlapLength| used for processing, a certain number of buffers of + // zeros must be inserted into the output buffers such that enough input can + // build up to process |kOverlapLength| worth, and enough output will build + // up to return |frames_per_buffer_| worth. + const size_t zeroed_buffers_of_output = kOverlapLength / frames_per_buffer_; + for (size_t i = 0; i < zeroed_buffers_of_output; ++i) { + output_circular_buffers_[channel]->InsertBuffer(zeros[0]); + } + output_accumulator_[channel] = AudioBuffer(kNumOverlap, kOverlapLength); + output_accumulator_[channel].Clear(); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb.h new file mode 100644 index 000000000..0734772e0 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb.h @@ -0,0 +1,173 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_H_ +#define RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_H_ + +#include <memory> +#include <vector> + +#include "base/audio_buffer.h" +#include "dsp/circular_buffer.h" +#include "dsp/fft_manager.h" + +namespace vraudio { + +// Implements a spectral reverb producing a decorrelated stereo output. See: +// [1] E. Vickers, J-L Wu, P.G. Krishnan, R. N. K. Sadanandam, "Frequency Domain +// Artificial Reverberation using Spectral Magnitude Decay", +// https://goo.gl/hv1pdJ. +class SpectralReverb { + public: + // Constructs a spectral reverb. + // + // @param sample_rate The system sample rate. + // @param frames_per_buffer System frames per buffer of input and output. + // Note that this class expects power of two buffers of input and output. + SpectralReverb(int sample_rate, size_t frames_per_buffer); + + // Sets the overall gain to be applied to the output of the reverb. + // + // @param gain Gain to be applied to the reverb output, min value 0.0f. + void SetGain(float gain); + + // Sets the |SpectralReverb|'s reverberation times in different frequency + // bands. Supports times between: + // (0.15 * 48000 / |sample_rate|)s and (25 * 48000 / |sample_rate|)s. + // + // @param rt60_values |kNumReverbOctaveBands| values denoting the + // reverberation decay time to -60dB in octave bands starting at + // |kLowestOctaveBand|. + void SetRt60PerOctaveBand(const float* rt60_values); + + // Applies reverb to an input channel of audio data and produces a stereo + // output. + // + // @param input Mono inpu data. + // @param left_out Left channel of reverberated output. + // @param right_out Right channel of reverberated output. + void Process(const AudioBuffer::Channel& input, + AudioBuffer::Channel* left_out, AudioBuffer::Channel* right_out); + + private: + // Uses an AudioBuffer with four channels to overlap add and insert the final + // reverb into the output circular buffers. + // + // @param channel_index Denotes the (left or right) channel to output to. + // @param buffer The buffer to be added onto the pre-existing reverb output. + void AccumulateOverlap(size_t channel_index, + const AudioBuffer::Channel& buffer); + + // Generates a window function which is a normalized sum of three overlapping + // (50%) hann windows of length (|kFftSize| / 2) that also incorporates the + // inverse fft scaling. + void GenerateAnalysisWindow(); + + // Generates a large buffer of sines and cosines of random noise between 0 and + // pi to be randomly indexed into in order to cheaply generate highly + // decorrelated phase buffers, + void GenerateRandomPhaseBuffer(); + + // Obtains the next stero pair of time domain reverb blocks which can then be + // summed together in an overlap add fashion to provide the reverb output. + // + // @param delay_index An index into the frequency domain magnitude ring + // buffer. + // @param left_channel Channel to contain the left partial reverb output. + // @param right_channel Channel to contain the right partial reverb output. + void GetNextReverbBlock(size_t delay_index, + AudioBuffer::Channel* left_channel, + AudioBuffer::Channel* right_channel); + + // Initializes the output circular buffers such that they contain zeros if the + // value of |frames_per_buffer_| is sufficiently smaller than that of + // |kOverlapLength| that buffering of input will be required prior to + // processing. Also allocates memory for the output accumulators. + void InitializeCircularBuffersAndAccumulators(); + + // System sample rate. + const int sample_rate_; + + // System frames per buffer. + const size_t frames_per_buffer_; + + // Indices into the magnitude and overlap add delay lines, modulo of their + // respective lengths. + size_t magnitude_delay_index_; + size_t overlap_add_index_; + + // Manages the time-frequency transforms and phase/magnitude-frequency + // transforms. + FftManager fft_manager_; + + // Buffer containing sines and cosines of random values between 0 and pi to be + // used for phase. + AudioBuffer sin_cos_random_phase_buffer_; + + // Buffer containing a triple overlapping hann window for windowing time + // domain data. + AudioBuffer unscaled_window_; + + // Buffer containing a triple overlapping hann window for windowing time + // domain data, this window has been scaled by the output gain factor. + AudioBuffer window_; + + // Buffer containing RT60 tuned feedback values. + AudioBuffer feedback_; + + // Buffer used to store scaling values which account for the different initial + // peak magnitudes for different RT60s. + AudioBuffer magnitude_compensation_; + + // Buffer that acts as the frequency domain magnitde delay. + AudioBuffer magnitude_delay_; + + // Buffer to contain a linear |kFftSize| chunk of input data. + AudioBuffer fft_size_input_; + + // Circular buffers to sit at the input and output of the |Process()| method + // to allow |frames_per_buffer_| to differ from |kFftSize|. + CircularBuffer input_circular_buffer_; + std::vector<std::unique_ptr<CircularBuffer>> output_circular_buffers_; + + // Time domain buffer used to store reverb before the overlap add operation. + AudioBuffer out_time_buffer_; + + // Temporary frequency domain buffer, used to store frequency domain data when + // transforming between Pffft and Canonical format frequency domain data. + AudioBuffer temp_freq_buffer_; + + // Buffer used to store feedback scaled magnitude values. + AudioBuffer scaled_magnitude_buffer_; + + // Buffer used for the accumulation of scaled magnitude buffers. + AudioBuffer temp_magnitude_buffer_; + + // Buffer used to store randomized phase. + AudioBuffer temp_phase_buffer_; + + // Buffers used to calculate the overlap add at the output. + std::vector<AudioBuffer> output_accumulator_; + + // Processing of the spectral reverb is bypassed when the feedback values are + // all approximately zero OR when the gain is set to near zero. + bool is_gain_near_zero_; + bool is_feedback_near_zero_; +}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb_constants_and_tables.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb_constants_and_tables.h new file mode 100644 index 000000000..84329f329 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb_constants_and_tables.h @@ -0,0 +1,6508 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_CONSTANTS_AND_TABLES_H_ +#define RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_CONSTANTS_AND_TABLES_H_ + +#include <algorithm> +#include <cstddef> + +namespace vraudio { + +// Sampling rate for which the spectral reverb feedback coefficients are +// calculated. +static const float kDefaultSpectralReverbSampleRate = 48000.0f; + +// Minimum reverb time supported by the feedback table below (seconds). +static const float kMinReverbTimeForFeedback48kHz = 0.15f; + +// Step size between each reverberation time supported by the feedback table +// (seconds). +static const float kTimeStepSizeForFeedback48kHz = 0.01f; + +// Number of entries in the feedback table below. +static const size_t kNumFeedbackValues = 2485; + +// Value denoting an invalid index into the feedback and magnitude correction +// tables. The feedback and magnitude correction factors will be 0.0 when this +// value is returned from the index lookup method. +const int kInvalidIndex = -1; + +// Number of frames of zeros before the reverb onset curve. +const size_t kCompensationOnsetLength = 1024; + +// Length in samples of the spectral reverb onset correction curves. +const size_t kCorrectionCurveLength = 6144; + +// Index of the feedback factor and curve multipliers. below which the low +// frequency correction curves must be used. +const size_t kCurveChangeoverIndex = 51; + +// Offset and scaling values for the compensation curve polynomials. Used to +// ensure the polynomial fitting was well conditioned. +const float kCurveOffset = 3584.5f; +const float kCurveScale = 0.00027897893709025f; + +// Number of coefficients in the curve generation polynomials. +const size_t kCurvePolynomialLength = 5; + +// Coefficients of the curves making up the first part of the compensation +// envelopes, split into low and high frequency part. +const float kHighReverberationCorrectionCurve[kCurvePolynomialLength] = { + 0.207891278205479f, -0.32101050261694f, 0.124608132159297f, + 0.0119620847734548f, -0.0093840571415877f}; + +const float kLowReverberationCorrectionCurve[kCurvePolynomialLength] = { + 0.0277040197264401f, -0.0394421854392145f, 0.0519110103856154f, + -0.0778183422829366f, 0.0343221444553963f}; + +// Offset from zero for indexing into the feedback array. +const size_t kReverbLookupOffset = static_cast<size_t>( + kMinReverbTimeForFeedback48kHz / kTimeStepSizeForFeedback48kHz); + +// Returns an index to the feedback factor corresponding to the given +// reverberation time. Supports times between: +// (0.15 * 48000 / |sample_rate|)s and (25 * 48000 / |sample_rate|)s. +// +// @param reverberation_time RT60 in seconds. +// @return Index to the closest spectral reverb feedback factor in +// |kSpectralReverbFeedback|, or |kInvalidIndex| if |reverberation_time| +// is too small. +static inline int GetFeedbackIndexFromRt60(float reverberation_time, + float sample_rate) { + // Scaling factor on reverb times applied due to current |sample_rate|. All + // feedback factors calculated correspond to 48kHz. + const float scale_factor = sample_rate / kDefaultSpectralReverbSampleRate; + if (reverberation_time * scale_factor < kMinReverbTimeForFeedback48kHz) { + return kInvalidIndex; + } + return static_cast<int>( + std::min(kNumFeedbackValues - 1, + static_cast<size_t>((reverberation_time * scale_factor) / + kTimeStepSizeForFeedback48kHz) - + kReverbLookupOffset)); +} + +// Mapping between reverberation times at a given frequency and the +// corresponding feedback value required in the specral reverb. This table +// covers reverberation times between kMinReverbTimeForFeedback and +// kNumFeedbackValues * kTimeStepSizeForFeedback, i.e. 0.15s and 25s @48kHz. +static const float kSpectralReverbFeedback[kNumFeedbackValues] = { + 0.231626448720861f, 0.231626448720861f, 0.271965742827169f, + 0.291599040391330f, 0.313450848869688f, 0.332864905289138f, + 0.350285532738638f, 0.366049684043223f, 0.383143567924904f, + 0.398574170268545f, 0.427500186399158f, 0.441038795823222f, + 0.453430889363735f, 0.466397294153677f, 0.478272702354267f, + 0.490519904155818f, 0.501757394995813f, 0.512125955040727f, + 0.521739757418252f, 0.548135448526531f, 0.548135448526531f, + 0.563990087718933f, 0.571497329505391f, 0.578784680803617f, + 0.585892965998708f, 0.592858971592466f, 0.599714945736390f, + 0.606488097764666f, 0.613200097727167f, 0.619866575922435f, + 0.626496622430680f, 0.633092286646765f, 0.639648076813205f, + 0.646150459553150f, 0.652577359403386f, 0.658897658347319f, + 0.665070695347976f, 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0.991284387345243f, 0.991289243787817f, + 0.991289243787817f, 0.991294099841006f, 0.991298955504871f, + 0.991303810779470f, 0.991303810779470f, 0.991308665664861f, + 0.991313520161105f, 0.991318374268260f, 0.991318374268260f, + 0.991323227986385f, 0.991328081315538f, 0.991332934255781f, + 0.991332934255781f, 0.991337786807170f, 0.991342638969767f, + 0.991347490743627f, 0.991347490743627f, 0.991352342128812f, + 0.991357193125380f, 0.991357193125380f, 0.991362043733390f, + 0.991366893952901f, 0.991371743783973f, 0.991371743783973f, + 0.991376593226661f, 0.991381442281029f, 0.991386290947134f, + 0.991386290947134f, 0.991391139225033f, 0.991395987114787f, + 0.991400834616454f, 0.991400834616454f, 0.991405681730093f, + 0.991410528455763f, 0.991410528455763f, 0.991415374793522f, + 0.991420220743429f, 0.991425066305544f, 0.991425066305544f, + 0.991429911479924f, 0.991434756266630f, 0.991439600665718f, + 0.991439600665718f, 0.991444444677248f, 0.991449288301280f, + 0.991449288301280f}; + +// Mapping between reverberation times at a given frequency and the +// corresponding magnitude compensation value required in the specral reverb to +// achieve 1.0 magnitude for any decay rate when combined with the convolution +// compensator. This table covers reverberation times between +// kMinReverbTimeForFeedback and kNumFeedbackValues * kTimeStepSizeForFeedback, +// i.e. 0.15s and 25s @48kHz. +static const float kMagnitudeCompensation[kNumFeedbackValues] = { + 17.3996715487709f, 17.4034178858004f, 16.6795907475848f, 16.3502965109599f, + 15.9879028507326f, 15.6718483944106f, 15.4042251167111f, 15.1677088170684f, + 14.9119678952536f, 14.6906933374371f, 14.2839187463763f, 14.0999475816121f, + 13.9338828421032f, 13.7652114598195f, 13.6133888181873f, 13.4638764218013f, + 13.3247278048121f, 13.1976499042192f, 13.0833841919256f, 12.7669049046296f, + 12.7761019806869f, 12.7855596748707f, 12.5979372062571f, 12.5244917829334f, + 12.4453918694998f, 12.3672592847373f, 12.3020244979211f, 12.2257529341565f, + 12.1547662449381f, 12.0845455398564f, 12.0231689748723f, 11.9586881174297f, + 11.8906434508319f, 11.8264759328308f, 11.7652619217413f, 11.7033346037227f, + 11.6379964417238f, 11.5862703081324f, 11.5299758955061f, 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time, by which the +// |kHighCorrectionCurve| and |kLowCorrectionCurve| curves are multiplied in +// order to generate the correct onset compensation curves. +static const float kCurveCorrectionMultipliers[kNumFeedbackValues] = { + 0.000249027612622225f, 0.0475827382701585f, 0.0982856973675534f, + 0.144077780284872f, 0.187710331328442f, 0.229214552437128f, + 0.268761703298196f, 0.306384317687497f, 0.342116205684862f, + 0.376122154314109f, 0.408019265118341f, 0.438826303187204f, + 0.468213758592444f, 0.496269855121223f, 0.523103160344453f, + 0.548734562484123f, 0.573191744554386f, 0.596776538466997f, + 0.619556090502133f, 0.646894092669191f, 0.661870435136273f, + 0.675494733319630f, 0.701821797958321f, 0.713358543560036f, + 0.731743894313465f, 0.749694785147017f, 0.758967096040535f, + 0.775801399270244f, 0.792218438223182f, 0.808206900102085f, + 0.815142557606016f, 0.830420164791760f, 0.845380788487056f, + 0.859865023229817f, 0.865296767216337f, 0.879331838452952f, + 0.893145291831122f, 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generated by the |kHighReverberationCorrectionCurve| polynomial, for +// reverberation times which result in a feedback factor index greater than +// |kCurveChangeoverIndex|. +static const float kHighCorrectionCurve[kCorrectionCurveLength] = { + 0.00905424704513036f, 0.00926599025691877f, 0.00947778313262726f, + 0.00968962562016418f, 0.00990151766744984f, 0.0101134592224174f, + 0.0103254502330116f, 0.0105374906471899f, 0.0107495804129221f, + 0.0109617194781898f, 0.0111739077909878f, 0.0113861452993219f, + 0.0115984319512115f, 0.0118107676946871f, 0.0120231524777923f, + 0.0122355862485828f, 0.0124480689551263f, 0.0126606005455030f, + 0.0128731809678055f, 0.0130858101701389f, 0.0132984881006193f, + 0.0135112147073766f, 0.0137239899385524f, 0.0139368137423006f, + 0.0141496860667869f, 0.0143626068601902f, 0.0145755760707011f, + 0.0147885936465225f, 0.0150016595358697f, 0.0152147736869704f, + 0.0154279360480643f, 0.0156411465674036f, 0.0158544051932527f, + 0.0160677118738880f, 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float kLowCorrectionCurve[kCorrectionCurveLength] = { + 0.0355835132858260f, 0.0361471820669677f, 0.0367104001412021f, + 0.0372731677107544f, 0.0378354849778072f, 0.0383973521445001f, + 0.0389587694129315f, 0.0395197369851564f, 0.0400802550631881f, + 0.0406403238489973f, 0.0411999435445122f, 0.0417591143516188f, + 0.0423178364721608f, 0.0428761101079397f, 0.0434339354607135f, + 0.0439913127321995f, 0.0445482421240714f, 0.0451047238379613f, + 0.0456607580754581f, 0.0462163450381097f, 0.0467714849274197f, + 0.0473261779448504f, 0.0478804242918223f, 0.0484342241697123f, + 0.0489875777798562f, 0.0495404853235463f, 0.0500929470020328f, + 0.0506449630165243f, 0.0511965335681860f, 0.0517476588581411f, + 0.0522983390874708f, 0.0528485744572133f, 0.0533983651683647f, + 0.0539477114218787f, 0.0544966134186670f, 0.0550450713595981f, + 0.0555930854454990f, 0.0561406558771536f, 0.0566877828553037f, + 0.0572344665806487f, 0.0577807072538459f, 0.0583265050755101f, + 0.0588718602462131f, 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0.00156310403017101f, 0.00150587338740016f, + 0.00144865561216939f, 0.00139145070158553f, 0.00133425865275616f, + 0.00127707946278949f, 0.00121991312879444f, 0.00116275964788048f, + 0.00110561901715772f, 0.00104849123373701f, 0.000991376294729840f, + 0.000934274197248231f, 0.000877184938404996f, 0.000820108515313556f, + 0.000763044925087941f, 0.000705994164842849f, 0.000648956231693587f, + 0.000591931122756240f, 0.000534918835147447f, 0.000477919365984458f, + 0.000420932712385191f, 0.000363958871468284f, 0.000306997840353040f, + 0.000250049616159320f, 0.000193114196007482f, 0.000136191577018996f, + 7.92817563154968e-05f, 2.23847310195091e-05f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f, + 0.000000000000000000f, 0.000000000000000000f, 0.000000000000000000f}; + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_SPECTRAL_REVERB_CONSTANTS_AND_TABLES_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb_test.cc new file mode 100644 index 000000000..015159fa1 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/spectral_reverb_test.cc @@ -0,0 +1,324 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/spectral_reverb.h" + +#include <algorithm> +#include <cmath> +#include <numeric> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "dsp/biquad_filter.h" +#include "dsp/fft_manager.h" +#include "dsp/filter_coefficient_generators.h" +#include "utils/test_util.h" + +namespace vraudio { + +namespace { + +const size_t kFramesPerBuffer512 = 512; +const size_t kFramesPerBuffer2048 = 2048; +const size_t kFramesPerBuffer713 = 713; +const size_t kNumReverbOverlap = 4; +const size_t kReverbFftSize = 4096; +const int kSampleFrequency16 = 16000; +const int kSampleFrequency24 = 24000; + +// A set of 9 RT60 values for each of the octave bands with center frequencies +// between 31.25Hz and 8kHz. +const float kRt60s[kNumReverbOctaveBands] = {0.8f, 0.8f, 0.7f, 0.7f, 0.65f, + 0.65f, 0.6f, 0.6f, 0.5f}; + +void ImpulseResponse(float length_sec, int sample_frequency, + size_t frames_per_buffer, SpectralReverb* reverb, + std::vector<float>* left_response, + std::vector<float>* right_response) { + AudioBuffer input(kNumMonoChannels, frames_per_buffer); + input.Clear(); + input[0][0] = 1.0f; + AudioBuffer output(kNumStereoChannels, frames_per_buffer); + + reverb->Process(input[0], &output[0], &output[1]); + + // The number of iterations required so that we will have no more than a few + // zeros following each of our impulse responses. + const float tail_length_samples = + kReverbFftSize + length_sec * static_cast<float>(sample_frequency); + const size_t num_iterations = + 100 + 1 + + static_cast<size_t>(std::ceil(tail_length_samples / + static_cast<float>(frames_per_buffer))); + + for (size_t i = 0; i < num_iterations; ++i) { + left_response->insert(left_response->end(), output[0].begin(), + output[0].end()); + right_response->insert(right_response->end(), output[1].begin(), + output[1].end()); + + input.Clear(); + reverb->Process(input[0], &output[0], &output[1]); + } +} + +void StereoOutputTestHelper(int sample_frequency, size_t frames_per_buffer) { + SpectralReverb reverb(sample_frequency, frames_per_buffer); + reverb.SetRt60PerOctaveBand(kRt60s); + + std::vector<float> output_collect_left; + std::vector<float> output_collect_right; + ImpulseResponse(1.0f /*length [sec]*/, sample_frequency, frames_per_buffer, + &reverb, &output_collect_left, &output_collect_right); + + // First test that there are kReverbFftSize / kNumReverbOverlap zeros at the + // beginning of the system impulse responses ONLY if this value is greater + // than |frames_per_buffer|, otherwise expect no onset zeros. + const size_t kOverlapLength = kReverbFftSize / kNumReverbOverlap; + const size_t onset = kOverlapLength > frames_per_buffer ? kOverlapLength : 0; + for (size_t i = 0; i < onset; ++i) { + EXPECT_NEAR(0.0f, output_collect_left[i], kEpsilonFloat); + EXPECT_NEAR(0.0f, output_collect_right[i], kEpsilonFloat); + } + // Test the sample five samples later is non zero. i.e. the tail has begun. + EXPECT_NE(0.0f, output_collect_left[onset + 5]); + EXPECT_NE(0.0f, output_collect_right[onset + 5]); +} + +void ZeroRt60TestHelper(int sample_frequency, float rt_60) { + SpectralReverb reverb(sample_frequency, kFramesPerBuffer512); + reverb.SetRt60PerOctaveBand(kRt60s); + + std::vector<float> output_collect_left; + std::vector<float> output_collect_right; + ImpulseResponse(1.0f /*length [sec]*/, sample_frequency, kFramesPerBuffer512, + &reverb, &output_collect_left, &output_collect_right); + + // Test that all the frames of the tail buffer are zeros. + for (size_t i = 0; i < kFramesPerBuffer512; ++i) { + EXPECT_NEAR(0.0f, output_collect_left[i], kEpsilonFloat); + EXPECT_NEAR(0.0f, output_collect_right[i], kEpsilonFloat); + } +} + +void TailDecayTestHelper(int sample_frequency, size_t frames_per_buffer) { + // Butterworth Lowpass filter with cutoff frequency at 3Hz. + BiquadCoefficients low_pass_coefficients( + 1.0f, -1.999444639647f, 0.999444793816755f, 1e-5f, 2e-5f, 1e-5f); + BiquadFilter low_pass_filter(low_pass_coefficients, frames_per_buffer); + const std::vector<float> kUniformRt60s(kNumReverbOctaveBands, kRt60s[0]); + SpectralReverb reverb(sample_frequency, frames_per_buffer); + reverb.SetRt60PerOctaveBand(kUniformRt60s.data()); + + std::vector<float> output_collect_left; + std::vector<float> output_collect_right; + ImpulseResponse(2.0f /*length [sec]*/, sample_frequency, frames_per_buffer, + &reverb, &output_collect_left, &output_collect_right); + + for (size_t i = 0; i < output_collect_left.size(); ++i) { + output_collect_left[i] = std::abs(output_collect_left[i]); + output_collect_right[i] = std::abs(output_collect_right[i]); + } + + const size_t response_length = output_collect_left.size(); + + AudioBuffer test_buffer(kNumMonoChannels, response_length); + test_buffer[0] = output_collect_left; + + // Very low frequency content of tail. This should essentially just preserve + // the decay. + low_pass_filter.Filter(test_buffer[0], &test_buffer[0]); + + const size_t max_location = static_cast<size_t>( + std::max_element(test_buffer[0].begin(), test_buffer[0].end()) - + test_buffer[0].begin()); + + // Stop before the very end of the tail as it goes to zero. + const size_t end_point = max_location + kReverbFftSize; + const size_t step_size = (end_point - max_location) / 20; + + // Test for decay. + for (size_t i = max_location + step_size; i < end_point; i += step_size) { + EXPECT_GT(std::abs(test_buffer[0][i - step_size]), + std::abs(test_buffer[0][i])); + } +} + +void DecorrelatedTailsTestHelper(int sample_frequency, + size_t frames_per_buffer) { + // This value has been found empirically in MATLAB. + const float kMaxCrossCorrelation = 12.0f; + const std::vector<float> kUniformRt60s(kNumReverbOctaveBands, 0.7f); + SpectralReverb reverb(sample_frequency, frames_per_buffer); + reverb.SetRt60PerOctaveBand(kUniformRt60s.data()); + + std::vector<float> output_collect_left; + std::vector<float> output_collect_right; + ImpulseResponse(0.7f /*length [sec]*/, sample_frequency, frames_per_buffer, + &reverb, &output_collect_left, &output_collect_right); + + // Find the absolute maximum elements of each vector. + auto min_max_left = std::minmax_element(output_collect_left.begin(), + output_collect_left.end()); + size_t left_max_index = + std::abs(*min_max_left.first) > std::abs(*min_max_left.second) + ? (min_max_left.first - output_collect_left.begin()) + : (min_max_left.second - output_collect_left.begin()); + auto min_max_right = std::minmax_element(output_collect_right.begin(), + output_collect_right.end()); + size_t right_max_index = + std::abs(*min_max_right.first) > std::abs(*min_max_right.second) + ? (min_max_right.first - output_collect_right.begin()) + : (min_max_right.second - output_collect_right.begin()); + + // Take a sample of the tails for cross correlation. + AudioBuffer pair(kNumStereoChannels, kReverbFftSize); + for (size_t i = 0; i < kReverbFftSize; ++i) { + pair[0][i] = output_collect_left[i + left_max_index] / + output_collect_left[left_max_index]; + pair[1][i] = output_collect_right[i + right_max_index] / + output_collect_right[right_max_index]; + } + + // The cross correlation is not normalized. Thus we can expect a very small + // value. Naturally, if the RT60 inputs are changed the expected value would + // thus be different. + const float max_xcorr = MaxCrossCorrelation(pair[0], pair[1]); + EXPECT_LT(max_xcorr, kMaxCrossCorrelation); +} + +} // namespace + +// Tests that the stereo output from the Reverbs Process fuction has the +// expected properties of predelay and length. +TEST(SpectralReverbTest, StereoOutputTest) { + StereoOutputTestHelper(kSampleFrequency24, kFramesPerBuffer512); + StereoOutputTestHelper(kSampleFrequency24, kFramesPerBuffer2048); + StereoOutputTestHelper(kSampleFrequency24, kFramesPerBuffer713); + StereoOutputTestHelper(kSampleFrequency16, kFramesPerBuffer512); + StereoOutputTestHelper(kSampleFrequency16, kFramesPerBuffer2048); + StereoOutputTestHelper(kSampleFrequency16, kFramesPerBuffer713); +} + +// Tests that the stereo output from the Reverb's Process function has the +// output of all zeros when the RT60 values are all zero. +TEST(SpectralReverbTest, ZeroRt60Test) { + const float kZeroRt = 0.0f; + const float kBelowMinRt = 0.12f; + ZeroRt60TestHelper(kSampleFrequency24, kZeroRt); + ZeroRt60TestHelper(kSampleFrequency16, kBelowMinRt); + ZeroRt60TestHelper(kSampleFrequency16, kZeroRt); + ZeroRt60TestHelper(kSampleFrequency16, kBelowMinRt); +} + +// Tests that the tail is decaying over time. +TEST(SpectralReverbTest, TailDecayTest) { + TailDecayTestHelper(kSampleFrequency24, kFramesPerBuffer512); + TailDecayTestHelper(kSampleFrequency24, kFramesPerBuffer2048); + TailDecayTestHelper(kSampleFrequency16, kFramesPerBuffer512); + TailDecayTestHelper(kSampleFrequency16, kFramesPerBuffer2048); +} + +// Tests that the stereo tail pairs are highy decorrelated. +TEST(SpectralReverbTest, DecorrelatedTailsTest) { + DecorrelatedTailsTestHelper(kSampleFrequency24, kFramesPerBuffer512); + DecorrelatedTailsTestHelper(kSampleFrequency24, kFramesPerBuffer2048); + DecorrelatedTailsTestHelper(kSampleFrequency24, kFramesPerBuffer713); + DecorrelatedTailsTestHelper(kSampleFrequency16, kFramesPerBuffer512); + DecorrelatedTailsTestHelper(kSampleFrequency16, kFramesPerBuffer2048); + DecorrelatedTailsTestHelper(kSampleFrequency16, kFramesPerBuffer713); +} + +// Tests that the gain parameter behaves as expected. +TEST(SpecralReverbTest, GainTest) { + const float kReverbLength = 0.5f; + const float kGain = 100.0f; + const float kGainEpsilon = 0.32f; + const std::vector<float> kUniformRt60s(kNumReverbOctaveBands, kReverbLength); + SpectralReverb reverb(kSampleFrequency24, kFramesPerBuffer512); + reverb.SetRt60PerOctaveBand(kUniformRt60s.data()); + + // Calculate scaled and unscaled impulse responses. + std::vector<float> output_left; + std::vector<float> output_right; + ImpulseResponse(kReverbLength, kSampleFrequency24, kFramesPerBuffer512, + &reverb, &output_left, &output_right); + std::vector<float> output_left_scaled; + reverb.SetGain(kGain); + ImpulseResponse(kReverbLength, kSampleFrequency24, kFramesPerBuffer512, + &reverb, &output_left_scaled, &output_right); + + // Determine the max absolute entry in each impulse response. + std::transform(output_left.begin(), output_left.end(), output_left.begin(), + static_cast<float (*)(float)>(&std::abs)); + std::transform(output_left_scaled.begin(), output_left_scaled.end(), + output_left_scaled.begin(), + static_cast<float (*)(float)>(&std::abs)); + const float max_unscaled = + *std::max_element(output_left.begin(), output_left.end()); + const float max_scaled = + *std::max_element(output_left_scaled.begin(), output_left_scaled.end()); + EXPECT_GT(max_scaled, max_unscaled); + EXPECT_NEAR((max_unscaled / max_scaled) / (1.0f / kGain), 1.0f, kGainEpsilon); +} + +// Tests that when the feedback values are all ~0.0f, no processing is +// performed (output is all zero). Also tests that if even one of the rt60s +// result in a non zero feedback that the result will be non zero. +TEST(SpectralReverbTest, DisabledProcessingTest) { + const float kReverbLength = 0.1f; + const std::vector<float> kUniformRt60s(kNumReverbOctaveBands, kReverbLength); + SpectralReverb reverb(kSampleFrequency24, kFramesPerBuffer512); + reverb.SetRt60PerOctaveBand(kUniformRt60s.data()); + std::vector<float> output_left; + std::vector<float> output_right; + ImpulseResponse(kReverbLength, kSampleFrequency24, kFramesPerBuffer512, + &reverb, &output_left, &output_right); + for (size_t i = 0; i < output_left.size(); ++i) { + EXPECT_FLOAT_EQ(output_left[i], 0.0f); + EXPECT_FLOAT_EQ(output_right[i], 0.0f); + } + + // Test a non zero case. + const float kLongerReverbLength = 0.4f; + std::vector<float> rt60s(kNumReverbOctaveBands, 0.0f); + rt60s[0] = kLongerReverbLength; + output_left.resize(0); + output_right.resize(0); + reverb.SetRt60PerOctaveBand(rt60s.data()); + ImpulseResponse(kReverbLength, kSampleFrequency24, kFramesPerBuffer512, + &reverb, &output_left, &output_right); + const float sum_left = + std::accumulate(output_left.begin(), output_left.end(), 0.0f); + EXPECT_NE(sum_left, 0.0f); + const float sum_right = + std::accumulate(output_right.begin(), output_right.end(), 0.0f); + EXPECT_NE(sum_right, 0.0f); + + // Set gain to zero and test again. + output_left.resize(0); + output_right.resize(0); + reverb.SetGain(0.0f); + ImpulseResponse(kReverbLength, kSampleFrequency24, kFramesPerBuffer512, + &reverb, &output_left, &output_right); + for (size_t i = 0; i < output_left.size(); ++i) { + EXPECT_FLOAT_EQ(output_left[i], 0.0f); + EXPECT_FLOAT_EQ(output_right[i], 0.0f); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner.cc new file mode 100644 index 000000000..373db9989 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner.cc @@ -0,0 +1,46 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/stereo_panner.h" + +#include <cmath> + +#include "base/constants_and_types.h" + +namespace vraudio { + +const float kStereoLeftRadians = kRadiansFromDegrees * kStereoLeftDegrees; +const float kStereoRightRadians = kRadiansFromDegrees * kStereoRightDegrees; + +void CalculateStereoPanGains(const SphericalAngle& source_direction, + std::vector<float>* stereo_gains) { + // Note this applies the same panning law as was applied by the ambisonic + // equivalent panner to ensure consistency. + DCHECK(stereo_gains); + stereo_gains->resize(kNumStereoChannels); + + const float cos_direction_elevation = std::cos(source_direction.elevation()); + + (*stereo_gains)[0] = + 0.5f * (1.0f + std::cos(kStereoLeftRadians - source_direction.azimuth()) * + cos_direction_elevation); + (*stereo_gains)[1] = + 0.5f * + (1.0f + std::cos(kStereoRightRadians - source_direction.azimuth()) * + cos_direction_elevation); +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner.h new file mode 100644 index 000000000..bf4cf1708 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner.h @@ -0,0 +1,35 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_STEREO_PANNER_H_ +#define RESONANCE_AUDIO_DSP_STEREO_PANNER_H_ + +#include <vector> + +#include "base/spherical_angle.h" + +namespace vraudio { + +// Computes a pair of stereo panner gains based on the |source_direction|. +// +// @param source_direction Azimuth and elevation of the sound source. +// @param stereo_gains A pointer to vector of stereo loudspeaker gains. +void CalculateStereoPanGains(const SphericalAngle& source_direction, + std::vector<float>* stereo_gains); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_STEREO_PANNER_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner_test.cc new file mode 100644 index 000000000..d24230241 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/stereo_panner_test.cc @@ -0,0 +1,88 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/stereo_panner.h" + +#include <cmath> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" + +namespace vraudio { + +namespace { + +const float kMinusThreeDecibels = kInverseSqrtTwo; + +// Tests that the |CalculateStereoPanGains| method will generate correct stereo +// pan gains. +TEST(StereoPannerTest, StereoTest) { + std::vector<float> speaker_gains; + + SphericalAngle source_direction = SphericalAngle::FromDegrees(-90.0f, 0.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(0.0f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(1.0f, speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(-45.0f, 0.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(kMinusThreeDecibels, speaker_gains[1] - speaker_gains[0], + kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(0.0f, 0.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(speaker_gains[0], speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(45.0f, 0.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(kMinusThreeDecibels, speaker_gains[0] - speaker_gains[1], + kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(90.0f, 0.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(1.0f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(0.0f, speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(0.0f, 45.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(0.5f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(0.5f, speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(0.0f, -60.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(0.5f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(0.5f, speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(0.0f, 90.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(0.5f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(0.5f, speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(0.0f, -90.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(0.5f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(0.5f, speaker_gains[1], kEpsilonFloat); + + source_direction = SphericalAngle::FromDegrees(45.0f, 45.0f); + CalculateStereoPanGains(source_direction, &speaker_gains); + EXPECT_NEAR(0.75f, speaker_gains[0], kEpsilonFloat); + EXPECT_NEAR(0.25f, speaker_gains[1], kEpsilonFloat); +} + +} // namespace + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/utils.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/utils.cc new file mode 100644 index 000000000..e1d8396a0 --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/utils.cc @@ -0,0 +1,190 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/utils.h" + +#include <algorithm> +#include <cmath> +#include <limits> +#include <random> +#include <vector> + +#include "base/constants_and_types.h" +#include "base/logging.h" +#include "base/misc_math.h" + +#include "dsp/biquad_filter.h" +#include "dsp/filter_coefficient_generators.h" + +namespace { + +// The mean and standard deviation of the normal distribution for bandlimited +// Gaussian noise. +const float kMean = 0.0f; +const float kStandardDeviation = 1.0f; + +// Maximum group delay in seconds for each filter. In order to avoid audible +// distortion, the maximum phase shift of a re-combined stereo sequence should +// not exceed 5ms at high frequencies. That is why, maximum phase shift of +// each filter is set to 1/2 of that value. +const float kMaxGroupDelaySeconds = 0.0025f; + +// Phase modulation depth, chosen so that for a given max group delay filters +// provide the lowest cross-correlation coefficient. +const float kPhaseModulationDepth = 1.18f; + +// Constants used in the generation of uniform random number distributions. +// https://en.wikipedia.org/wiki/Linear_congruential_generator +const uint64 kMultiplier = 1664525L; +const uint64 kIncrement = 1013904223L; +const float kInt32ToFloat = + 1.0f / static_cast<float>(std::numeric_limits<uint32>::max()); + +} // namespace + +namespace vraudio { + +void GenerateGaussianNoise(float mean, float std_deviation, unsigned seed, + AudioBuffer::Channel* noise_channel) { + DCHECK(noise_channel); + // First generate uniform noise. + GenerateUniformNoise(0.0f, 1.0f, seed, noise_channel); + const size_t length = noise_channel->size(); + + // Gaussian distribution with mean and standard deviation in pairs via the + // box-muller transform + // https://en.wikipedia.org/wiki/Box%E2%80%93Muller_transform. + for (size_t i = 0; i < length - 1; i += 2) { + const float part_one = std::sqrt(-2.0f * std::log((*noise_channel)[i])); + const float part_two = kTwoPi * (*noise_channel)[i + 1]; + const float z0 = part_one * std::cos(part_two); + const float z1 = part_one * std::sin(part_two); + (*noise_channel)[i] = std_deviation * z0 + mean; + (*noise_channel)[i + 1] = std_deviation * z1 + mean; + } + // Handle the odd buffer length case cheaply. + if (length % 2 > 0) { + (*noise_channel)[length - 1] = (*noise_channel)[0]; + } +} + +void GenerateUniformNoise(float min, float max, unsigned seed, + AudioBuffer::Channel* noise_channel) { + // Simple random generator to avoid the use of std::uniform_real_distribution + // affected by https://gcc.gnu.org/bugzilla/show_bug.cgi?id=56202 + DCHECK(noise_channel); + DCHECK_LT(min, max); + const float scaled_conversion_factor = kInt32ToFloat * (max - min); + uint32 state = static_cast<uint32>(seed); + for (float& sample : *noise_channel) { + state = static_cast<uint32>(state * kMultiplier + kIncrement); + sample = min + static_cast<float>(state) * scaled_conversion_factor; + } +} + +void GenerateBandLimitedGaussianNoise(float center_frequency, int sampling_rate, + unsigned seed, + AudioBuffer* noise_buffer) { + + + DCHECK(noise_buffer); + DCHECK_GT(sampling_rate, 0); + DCHECK_LT(center_frequency, static_cast<float>(sampling_rate) / 2.0f); + const size_t num_frames = noise_buffer->num_frames(); + + BiquadCoefficients bandpass_coefficients = ComputeBandPassBiquadCoefficients( + sampling_rate, center_frequency, /*bandwidth=*/1); + BiquadFilter bandpass_filter(bandpass_coefficients, num_frames); + + for (auto& channel : *noise_buffer) { + GenerateGaussianNoise(kMean, kStandardDeviation, seed, &channel); + bandpass_filter.Filter(channel, &channel); + bandpass_filter.Clear(); + } +} + +std::unique_ptr<AudioBuffer> GenerateDecorrelationFilters(int sampling_rate) { + + const int kMaxGroupDelaySamples = static_cast<int>( + roundf(kMaxGroupDelaySeconds * static_cast<float>(sampling_rate))); + + // Filter coefficients according to: + // [1] F. Zotter, M. Frank, "Efficient Phantom Source Widening", Archives of + // Acoustics, Vol. 38, No. 1, pp. 27–37 (2013). + const float g0 = 1.0f - 0.25f * IntegerPow(kPhaseModulationDepth, 2); + const float g1 = 0.5f * kPhaseModulationDepth - + 0.0625f * IntegerPow(kPhaseModulationDepth, 3); + const float g2 = 0.1250f * IntegerPow(kPhaseModulationDepth, 2); + std::vector<float> filter1_coefficients{g2, g1, g0, -g1, g2}; + std::vector<float> filter2_coefficients{g2, -g1, g0, g1, g2}; + + const size_t filter_length = + filter1_coefficients.size() * kMaxGroupDelaySamples; + std::unique_ptr<AudioBuffer> decorrelation_filters( + new AudioBuffer(kNumStereoChannels, filter_length)); + decorrelation_filters->Clear(); + + for (size_t coefficient = 0; coefficient < filter1_coefficients.size(); + ++coefficient) { + (*decorrelation_filters)[0][coefficient * kMaxGroupDelaySamples] = + filter1_coefficients[coefficient]; + (*decorrelation_filters)[1][coefficient * kMaxGroupDelaySamples] = + filter2_coefficients[coefficient]; + } + + return decorrelation_filters; +} + +size_t GetNumReverbOctaveBands(int sampling_rate) { + DCHECK_GT(sampling_rate, 0); + + const float max_band = + log2f(0.5f * static_cast<float>(sampling_rate) / kLowestOctaveBandHz); + return std::min(kNumReverbOctaveBands, static_cast<size_t>(roundf(max_band))); +} + +size_t GetNumSamplesFromMilliseconds(float milliseconds, int sampling_rate) { + DCHECK_GE(milliseconds, 0.0f); + DCHECK_GT(sampling_rate, 0); + return static_cast<size_t>(milliseconds * kSecondsFromMilliseconds * + static_cast<float>(sampling_rate)); +} + +size_t CeilToMultipleOfFramesPerBuffer(size_t size, size_t frames_per_buffer) { + DCHECK_NE(frames_per_buffer, 0U); + const size_t remainder = size % frames_per_buffer; + return remainder == 0 ? std::max(size, frames_per_buffer) + : size + frames_per_buffer - remainder; +} + +void GenerateHannWindow(bool full_window, size_t window_length, + AudioBuffer::Channel* buffer) { + + DCHECK(buffer); + DCHECK_LE(window_length, buffer->size()); + const float full_window_scaling_factor = + kTwoPi / (static_cast<float>(window_length) - 1.0f); + const float half_window_scaling_factor = + kTwoPi / (2.0f * static_cast<float>(window_length) - 1.0f); + const float scaling_factor = + (full_window) ? full_window_scaling_factor : half_window_scaling_factor; + for (size_t i = 0; i < window_length; ++i) { + (*buffer)[i] = + 0.5f * (1.0f - std::cos(scaling_factor * static_cast<float>(i))); + } +} + +} // namespace vraudio diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/utils.h b/src/3rdparty/resonance-audio/resonance_audio/dsp/utils.h new file mode 100644 index 000000000..ceb3060de --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/utils.h @@ -0,0 +1,93 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#ifndef RESONANCE_AUDIO_DSP_UTILS_H_ +#define RESONANCE_AUDIO_DSP_UTILS_H_ + +#include "base/audio_buffer.h" + +namespace vraudio { + +// Generates a gaussian white noise signal. +// +// @param mean The mean distribution parameter. +// @param std_deviation The standard deviation distribution parameter. +// @param seed A seed for the random generator. +// @param noise_channel Buffer channel in which to store the noise. +void GenerateGaussianNoise(float mean, float std_deviation, unsigned seed, + AudioBuffer::Channel* noise_channel); + +// Generates a gaussian white noise signal. +// +// @param min The lowest value in the distribution. +// @param max The highest value in te distribution, must be > min. +// @param seed A seed for the random generator. +// @param noise_channel Buffer channel in which to store the noise. +void GenerateUniformNoise(float min, float max, unsigned seed, + AudioBuffer::Channel* noise_channel); + +// Generates a band limited gaussian white noise signal, one octave band wide. +// +// @param center_frequency Center frequency of the given octave band in Hz. +// @param sampling_rate System sampling rate in Hz. +// @param seed A seed for the random generator. +// @param noise_buffer Buffer in which to store the band limited noise. +void GenerateBandLimitedGaussianNoise(float center_frequency, int sampling_rate, + unsigned seed, AudioBuffer* noise_buffer); + +// Genarates a pair of decorrelation filters (for use in low quality/high +// effiency mode reverb). +// +// @param sampling_rate System sampling rate in Hz. +// @return Buffer containing the stereo filters. +std::unique_ptr<AudioBuffer> GenerateDecorrelationFilters(int sampling_rate); + +// Returns the number of octave bands necessary for the given |sampling_rate|. +// +// @param sampling_rate Sampling rate in Hertz. +// @return Number of reverb octave bands. +size_t GetNumReverbOctaveBands(int sampling_rate); + +// Converts the given |milliseconds| to number of samples with the given +// |sampling_rate|. This method should *not* be used when more precise +// (double-precission) value is desired. +// +// @param milliseconds Milliseconds in single-precission floating point. +// @param sampling_rate Sampling rate in Hertz. +// @return Number of samples. +size_t GetNumSamplesFromMilliseconds(float milliseconds, int sampling_rate); + +// Ceils the given |size| to the next multiple of given |frames_per_buffer|. +// +// @param size Input size in frames. +// @param frames_per_buffer Frames per buffer. +// @return Ceiled size in frames. +size_t CeilToMultipleOfFramesPerBuffer(size_t size, size_t frames_per_buffer); + +// Generates a Hann window (used for smooth onset and tapering of the generated +// reverb response tails). +// +// @param full_window True to generate a full window, false to generate a half. +// @param window_length Length of the window to be generated. Must be less than +// or equal to the number of frames in the |buffer|. +// @param buffer AudioBuffer::Channel to which the window is written, the number +// of frames will be the length in samples of the generated Hann window. +void GenerateHannWindow(bool full_window, size_t window_length, + AudioBuffer::Channel* buffer); + +} // namespace vraudio + +#endif // RESONANCE_AUDIO_DSP_UTILS_H_ diff --git a/src/3rdparty/resonance-audio/resonance_audio/dsp/utils_test.cc b/src/3rdparty/resonance-audio/resonance_audio/dsp/utils_test.cc new file mode 100644 index 000000000..dffe991ec --- /dev/null +++ b/src/3rdparty/resonance-audio/resonance_audio/dsp/utils_test.cc @@ -0,0 +1,194 @@ +/* +Copyright 2018 Google Inc. All Rights Reserved. + +Licensed under the Apache License, Version 2.0 (the "License"); +you may not use this file except in compliance with the License. +You may obtain a copy of the License at + + http://www.apache.org/licenses/LICENSE-2.0 + +Unless required by applicable law or agreed to in writing, software +distributed under the License is distributed on an "AS-IS" BASIS, +WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +See the License for the specific language governing permissions and +limitations under the License. +*/ + +#include "dsp/utils.h" + +#include <cmath> +#include <limits> +#include <memory> +#include <vector> + +#include "third_party/googletest/googletest/include/gtest/gtest.h" +#include "base/constants_and_types.h" +#include "dsp/fft_manager.h" +#include "dsp/partitioned_fft_filter.h" +#include "utils/test_util.h" + +namespace vraudio { + +namespace { + +const int kSamplingRate = 48000; + +const size_t kHalfHannWindowLength = 8; +const float kExpectedHalfHannWindow[] = {0.0000000f, 0.04322727f, 0.1654347f, + 0.3454915f, 0.55226423f, 0.7500000f, + 0.9045085f, 0.98907380f}; + +const size_t kHannWindowLength = 15; +const float kExpectedHannWindow[] = { + 0.0000000f, 0.0495156f, 0.1882551f, 0.3887395f, 0.6112605f, + 0.8117449f, 0.9504844f, 1.0000000f, 0.9504844f, 0.8117449f, + 0.6112605f, 0.3887395f, 0.1882551f, 0.0495156f, 0.0000000f}; + +// Test that the noise generation functions create noise vectors with the +// expected means etc. +TEST(DspUtilsTest, NoiseTest) { + const size_t kNoiseBufferLength = 1e5; + // A high epsilon is used to determine if mean, std dev etc. are acurate as it + // would be infeasible to have enough samples to take advantage of the central + // limit theorem. + const float kEpsilon = 1e-1f; + + AudioBuffer gaussian(kNumMonoChannels, kNoiseBufferLength); + AudioBuffer uniform(kNumMonoChannels, kNoiseBufferLength); + + // Generate Gaussian Noise with mean 0 and std deviation 2. + const float kGaussianMean = 0.0f; + const float kStdDeviation = 2.0f; + GenerateGaussianNoise(kGaussianMean, kStdDeviation, /*seed=*/1U, + &gaussian[0]); + + // Calculate the mean and compare to that specified. + float mean = 0.0f; + for (auto& sample : gaussian[0]) { + mean += sample; + } + mean /= static_cast<float>(gaussian.num_frames()); + EXPECT_NEAR(mean, kGaussianMean, kEpsilon); + + // Calculate the std deviation and compare to that specified. + float std_dev = 0.0f; + for (auto& sample : gaussian[0]) { + std_dev += (sample - mean) * (sample - mean); + } + std_dev /= static_cast<float>(gaussian.num_frames()); + std_dev = std::sqrt(std_dev); + EXPECT_NEAR(std_dev, kStdDeviation, kEpsilon); + + // Genarate uniformly distributed noise min -1, max 1 and thus mean 0. + const float kMin = -1.0f; + const float kMax = 1.0f; + GenerateUniformNoise(kMin, kMax, /*seed=*/1U, &uniform[0]); + // Calculate the mean and min/max values, compare to expected values. + mean = 0.0f; + float min = std::numeric_limits<float>::max(); + float max = std::numeric_limits<float>::min(); + for (auto& sample : uniform[0]) { + mean += sample; + min = sample < min ? sample : min; + max = sample > max ? sample : max; + } + mean /= static_cast<float>(uniform.num_frames()); + EXPECT_NEAR(mean, (kMax + kMin) / 2.0f, kEpsilon); + EXPECT_GE(kMax, max); + EXPECT_LE(kMin, min); +} + +// Tests that the ceiled input size in frames matches the expected multiple of +// frames per buffer for arbitrary inputs. +TEST(DspUtilsTest, CeilToMultipleOfFramesPerBufferTest) { + const size_t kFramesPerBuffer = 512; + const std::vector<size_t> kInput = {0, 100, 512, 1000, 5000, 10240}; + const std::vector<size_t> kExpectedOutput = {512, 512, 512, + 1024, 5120, 10240}; + + for (size_t i = 0; i < kInput.size(); ++i) { + EXPECT_EQ(kExpectedOutput[i], + CeilToMultipleOfFramesPerBuffer(kInput[i], kFramesPerBuffer)); + } +} + +// Tests that on filtering a noise sample with a pair of decorrelation filters, +// the correlation between those outputs is less than the result of an +// autocorrelation. +TEST(DspUtilsTest, GenerateDecorrelationFiltersTest) { + // Size of FFT to be used in |GenerateDecorrelationFiltersTest|. + const size_t kBufferSize = 512; + // Centre frequency for noise used in |GenerateDecorrelationFiltersTest|. + const float kNoiseCenter = 1000.0f; + std::unique_ptr<AudioBuffer> kernels = + GenerateDecorrelationFilters(kSamplingRate); + AudioBuffer noise(kNumMonoChannels, kBufferSize); + GenerateBandLimitedGaussianNoise(kNoiseCenter, kSamplingRate, /*seed=*/1U, + &noise); + FftManager fft_manager(kBufferSize); + PartitionedFftFilter fft_filter(kernels->num_frames(), kBufferSize, + &fft_manager); + fft_filter.SetTimeDomainKernel((*kernels)[0]); + AudioBuffer output_one(kNumMonoChannels, kBufferSize); + + PartitionedFftFilter::FreqDomainBuffer freq_domain_buffer(kNumMonoChannels, + kBufferSize * 2); + fft_manager.FreqFromTimeDomain(noise[0], &freq_domain_buffer[0]); + fft_filter.Filter(freq_domain_buffer[0]); + fft_filter.GetFilteredSignal(&output_one[0]); + fft_filter.SetTimeDomainKernel((*kernels)[1]); + AudioBuffer output_two(kNumMonoChannels, kBufferSize); + fft_manager.FreqFromTimeDomain(noise[0], &freq_domain_buffer[0]); + fft_filter.Filter(freq_domain_buffer[0]); + fft_filter.GetFilteredSignal(&output_two[0]); + const float auto_correlation = MaxCrossCorrelation(noise[0], noise[0]); + const float decorrelated = MaxCrossCorrelation(output_one[0], output_two[0]); + EXPECT_LT(decorrelated, auto_correlation); +} + +// Tests half-Hann window calculation against values returned by MATLAB's hann() +// function. +TEST(DspUtilsTest, GenerateHalfHannWindowTest) { + AudioBuffer half_hann_window(kNumMonoChannels, kHalfHannWindowLength); + GenerateHannWindow(false, kHalfHannWindowLength, &half_hann_window[0]); + for (size_t i = 0; i < kHalfHannWindowLength; ++i) { + EXPECT_NEAR(half_hann_window[0][i], kExpectedHalfHannWindow[i], + kEpsilonFloat); + } +} + +// Tests Hann window generation for odd window lengths. +TEST(DspUtilsTest, GenerateHannWindowOddLengthTest) { + AudioBuffer hann_window(kNumMonoChannels, kHannWindowLength); + GenerateHannWindow(true, kHannWindowLength, &hann_window[0]); + for (size_t i = 0; i < kHannWindowLength; ++i) { + EXPECT_NEAR(hann_window[0][i], kExpectedHannWindow[i], kEpsilonFloat); + } +} + +// Tests that the calculated number of reverb octave bands matches the +// pre-computed results with arbitrary sampling rates. +TEST(DspUtilsTest, GetNumReverbOctaveBandsTest) { + const std::vector<int> kSamplingRates = {8000, 22050, 44100, 48000, 96000}; + const std::vector<size_t> kExpectedOutput = {7, 8, 9, 9, 9}; + + for (size_t i = 0; i < kSamplingRates.size(); ++i) { + EXPECT_EQ(kExpectedOutput[i], GetNumReverbOctaveBands(kSamplingRates[i])); + } +} + +// Tests that the calculated number of samples for arbitrary milliseconds values +// matches the pre-computed results with a specific sampling rate. +TEST(DspUtilsTest, GetNumSamplesFromMillisecondsTest) { + const std::vector<float> kInput = {0.0f, 2.5f, 50.0f, 123.45f, 1000.0f}; + const std::vector<size_t> kExpectedOutput = {0, 120, 2400, 5925, 48000}; + + for (size_t i = 0; i < kInput.size(); ++i) { + EXPECT_EQ(kExpectedOutput[i], + GetNumSamplesFromMilliseconds(kInput[i], kSamplingRate)); + } +} + +} // namespace + +} // namespace vraudio |