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.
103 lines
3.5 KiB
C++
103 lines
3.5 KiB
C++
/*
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* Copyright (c) 2018 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/agc2/saturation_protector.h"
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#include <algorithm>
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#include <iterator>
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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namespace {
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void ShiftBuffer(std::array<float, kPeakEnveloperBufferSize>* buffer_) {
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// Move everything one element back.
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std::copy(buffer_->begin() + 1, buffer_->end(), buffer_->begin());
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}
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} // namespace
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SaturationProtector::PeakEnveloper::PeakEnveloper() = default;
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void SaturationProtector::PeakEnveloper::Process(float frame_peak_dbfs) {
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// Update the delayed buffer and the current superframe peak.
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current_superframe_peak_dbfs_ =
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std::max(current_superframe_peak_dbfs_, frame_peak_dbfs);
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speech_time_in_estimate_ms_ += kFrameDurationMs;
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if (speech_time_in_estimate_ms_ > kPeakEnveloperSuperFrameLengthMs) {
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speech_time_in_estimate_ms_ = 0;
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const bool buffer_full = elements_in_buffer_ == kPeakEnveloperBufferSize;
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if (buffer_full) {
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ShiftBuffer(&peak_delay_buffer_);
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*peak_delay_buffer_.rbegin() = current_superframe_peak_dbfs_;
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} else {
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peak_delay_buffer_[elements_in_buffer_] = current_superframe_peak_dbfs_;
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elements_in_buffer_++;
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}
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current_superframe_peak_dbfs_ = -90.f;
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}
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}
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float SaturationProtector::PeakEnveloper::Query() const {
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float result;
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if (elements_in_buffer_ > 0) {
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result = peak_delay_buffer_[0];
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} else {
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result = current_superframe_peak_dbfs_;
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}
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return result;
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}
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SaturationProtector::SaturationProtector(ApmDataDumper* apm_data_dumper)
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: SaturationProtector(apm_data_dumper, GetExtraSaturationMarginOffsetDb()) {
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}
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SaturationProtector::SaturationProtector(ApmDataDumper* apm_data_dumper,
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float extra_saturation_margin_db)
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: apm_data_dumper_(apm_data_dumper),
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last_margin_(GetInitialSaturationMarginDb()),
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extra_saturation_margin_db_(extra_saturation_margin_db) {}
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void SaturationProtector::UpdateMargin(
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const VadWithLevel::LevelAndProbability& vad_data,
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float last_speech_level_estimate) {
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peak_enveloper_.Process(vad_data.speech_peak_dbfs);
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const float delayed_peak_dbfs = peak_enveloper_.Query();
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const float difference_db = delayed_peak_dbfs - last_speech_level_estimate;
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if (last_margin_ < difference_db) {
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last_margin_ = last_margin_ * kSaturationProtectorAttackConstant +
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difference_db * (1.f - kSaturationProtectorAttackConstant);
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} else {
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last_margin_ = last_margin_ * kSaturationProtectorDecayConstant +
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difference_db * (1.f - kSaturationProtectorDecayConstant);
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}
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last_margin_ = rtc::SafeClamp<float>(last_margin_, 12.f, 25.f);
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}
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float SaturationProtector::LastMargin() const {
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return last_margin_ + extra_saturation_margin_db_;
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}
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void SaturationProtector::Reset() {
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peak_enveloper_ = PeakEnveloper();
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}
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void SaturationProtector::DebugDumpEstimate() const {
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apm_data_dumper_->DumpRaw(
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"agc2_adaptive_saturation_protector_delayed_peak_dbfs",
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peak_enveloper_.Query());
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apm_data_dumper_->DumpRaw("agc2_adaptive_saturation_margin_db", last_margin_);
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}
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} // namespace webrtc
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