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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_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
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