mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
99 lines
3.0 KiB
C++
99 lines
3.0 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/erl_estimator.h"
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#include <algorithm>
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#include <numeric>
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace {
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constexpr float kMinErl = 0.01f;
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constexpr float kMaxErl = 1000.f;
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} // namespace
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ErlEstimator::ErlEstimator(size_t startup_phase_length_blocks_)
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: startup_phase_length_blocks__(startup_phase_length_blocks_) {
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erl_.fill(kMaxErl);
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hold_counters_.fill(0);
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erl_time_domain_ = kMaxErl;
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hold_counter_time_domain_ = 0;
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}
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ErlEstimator::~ErlEstimator() = default;
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void ErlEstimator::Reset() {
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blocks_since_reset_ = 0;
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}
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void ErlEstimator::Update(bool converged_filter,
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rtc::ArrayView<const float> render_spectrum,
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rtc::ArrayView<const float> capture_spectrum) {
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RTC_DCHECK_EQ(kFftLengthBy2Plus1, render_spectrum.size());
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RTC_DCHECK_EQ(kFftLengthBy2Plus1, capture_spectrum.size());
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const auto& X2 = render_spectrum;
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const auto& Y2 = capture_spectrum;
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// Corresponds to WGN of power -46 dBFS.
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constexpr float kX2Min = 44015068.0f;
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if (++blocks_since_reset_ < startup_phase_length_blocks__ ||
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!converged_filter) {
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return;
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}
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// Update the estimates in a maximum statistics manner.
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for (size_t k = 1; k < kFftLengthBy2; ++k) {
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if (X2[k] > kX2Min) {
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const float new_erl = Y2[k] / X2[k];
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if (new_erl < erl_[k]) {
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hold_counters_[k - 1] = 1000;
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erl_[k] += 0.1f * (new_erl - erl_[k]);
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erl_[k] = std::max(erl_[k], kMinErl);
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}
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}
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}
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std::for_each(hold_counters_.begin(), hold_counters_.end(),
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[](int& a) { --a; });
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std::transform(hold_counters_.begin(), hold_counters_.end(), erl_.begin() + 1,
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erl_.begin() + 1, [](int a, float b) {
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return a > 0 ? b : std::min(kMaxErl, 2.f * b);
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});
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erl_[0] = erl_[1];
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erl_[kFftLengthBy2] = erl_[kFftLengthBy2 - 1];
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// Compute ERL over all frequency bins.
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const float X2_sum = std::accumulate(X2.begin(), X2.end(), 0.0f);
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if (X2_sum > kX2Min * X2.size()) {
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const float Y2_sum = std::accumulate(Y2.begin(), Y2.end(), 0.0f);
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const float new_erl = Y2_sum / X2_sum;
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if (new_erl < erl_time_domain_) {
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hold_counter_time_domain_ = 1000;
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erl_time_domain_ += 0.1f * (new_erl - erl_time_domain_);
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erl_time_domain_ = std::max(erl_time_domain_, kMinErl);
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}
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}
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--hold_counter_time_domain_;
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erl_time_domain_ = (hold_counter_time_domain_ > 0)
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? erl_time_domain_
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: std::min(kMaxErl, 2.f * erl_time_domain_);
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}
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} // namespace webrtc
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