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.
129 lines
4.6 KiB
C++
129 lines
4.6 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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
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#include <math.h>
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#include <stddef.h>
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#include <array>
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#include <vector>
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#include "api/array_view.h"
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#include "api/audio/echo_canceller3_config.h"
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#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/aec3_fft.h"
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#include "modules/audio_processing/aec3/aec_state.h"
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#include "modules/audio_processing/aec3/echo_path_variability.h"
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#include "modules/audio_processing/aec3/main_filter_update_gain.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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#include "modules/audio_processing/aec3/render_signal_analyzer.h"
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#include "modules/audio_processing/aec3/shadow_filter_update_gain.h"
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#include "modules/audio_processing/aec3/subtractor_output.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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// Proves linear echo cancellation functionality
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class Subtractor {
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public:
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Subtractor(const EchoCanceller3Config& config,
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ApmDataDumper* data_dumper,
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Aec3Optimization optimization);
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~Subtractor();
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// Performs the echo subtraction.
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void Process(const RenderBuffer& render_buffer,
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const rtc::ArrayView<const float> capture,
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const RenderSignalAnalyzer& render_signal_analyzer,
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const AecState& aec_state,
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SubtractorOutput* output);
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void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
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// Exits the initial state.
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void ExitInitialState();
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// Returns the block-wise frequency response for the main adaptive filter.
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const std::vector<std::array<float, kFftLengthBy2Plus1>>&
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FilterFrequencyResponse() const {
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return main_filter_.FilterFrequencyResponse();
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}
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// Returns the estimate of the impulse response for the main adaptive filter.
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const std::vector<float>& FilterImpulseResponse() const {
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return main_filter_.FilterImpulseResponse();
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}
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void DumpFilters() {
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main_filter_.DumpFilter("aec3_subtractor_H_main", "aec3_subtractor_h_main");
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shadow_filter_.DumpFilter("aec3_subtractor_H_shadow",
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"aec3_subtractor_h_shadow");
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}
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private:
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class FilterMisadjustmentEstimator {
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public:
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FilterMisadjustmentEstimator() = default;
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~FilterMisadjustmentEstimator() = default;
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// Update the misadjustment estimator.
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void Update(const SubtractorOutput& output);
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// GetMisadjustment() Returns a recommended scale for the filter so the
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// prediction error energy gets closer to the energy that is seen at the
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// microphone input.
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float GetMisadjustment() const {
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RTC_DCHECK_GT(inv_misadjustment_, 0.0f);
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// It is not aiming to adjust all the estimated mismatch. Instead,
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// it adjusts half of that estimated mismatch.
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return 2.f / sqrtf(inv_misadjustment_);
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}
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// Returns true if the prediciton error energy is significantly larger
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// than the microphone signal energy and, therefore, an adjustment is
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// recommended.
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bool IsAdjustmentNeeded() const { return inv_misadjustment_ > 10.f; }
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void Reset();
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void Dump(ApmDataDumper* data_dumper) const;
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private:
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const int n_blocks_ = 4;
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int n_blocks_acum_ = 0;
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float e2_acum_ = 0.f;
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float y2_acum_ = 0.f;
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float inv_misadjustment_ = 0.f;
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int overhang_ = 0.f;
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};
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const Aec3Fft fft_;
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ApmDataDumper* data_dumper_;
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const Aec3Optimization optimization_;
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const EchoCanceller3Config config_;
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const bool adaptation_during_saturation_;
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const bool enable_misadjustment_estimator_;
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const bool enable_agc_gain_change_response_;
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const bool enable_shadow_filter_jumpstart_;
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const bool enable_shadow_filter_boosted_jumpstart_;
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const bool enable_early_shadow_filter_jumpstart_;
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AdaptiveFirFilter main_filter_;
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AdaptiveFirFilter shadow_filter_;
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MainFilterUpdateGain G_main_;
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ShadowFilterUpdateGain G_shadow_;
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FilterMisadjustmentEstimator filter_misadjustment_estimator_;
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size_t poor_shadow_filter_counter_ = 0;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Subtractor);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
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