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/*
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
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