b20b93796f
This CL refactors the code in AEC3 that analyzes how well the adaptive filter performs. The purpose of this is both to simplify code that is more complex than needed and also to pave the wave for the upcoming CLs that softens the echo suppression during doubletalk. The main changes are that: -The shadow adaptive filter is now never analyzed. This turned out to never affect the output in the recordings it was tested on. -The convergence analysis was moved to the aec state code. The changes are bitexact on all testcases where they have been tested on. Bug: webrtc:8671 Change-Id: If76b669565325c8eb4d11d1178a7e20306da9a26 Reviewed-on: https://webrtc-review.googlesource.com/87430 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23958}
120 lines
4.3 KiB
C++
120 lines
4.3 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 <algorithm>
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#include <array>
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#include <vector>
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#include "math.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/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 "modules/audio_processing/utility/ooura_fft.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(rtc::ArrayView<const float> e, rtc::ArrayView<const float> y);
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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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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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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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