| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_ |
| #define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_ |
| |
| #include <algorithm> |
| #include <array> |
| #include <vector> |
| #include "math.h" |
| |
| #include "modules/audio_processing/aec3/adaptive_fir_filter.h" |
| #include "modules/audio_processing/aec3/aec3_common.h" |
| #include "modules/audio_processing/aec3/aec3_fft.h" |
| #include "modules/audio_processing/aec3/aec_state.h" |
| #include "modules/audio_processing/aec3/echo_path_variability.h" |
| #include "modules/audio_processing/aec3/main_filter_update_gain.h" |
| #include "modules/audio_processing/aec3/render_buffer.h" |
| #include "modules/audio_processing/aec3/shadow_filter_update_gain.h" |
| #include "modules/audio_processing/aec3/subtractor_output.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "modules/audio_processing/utility/ooura_fft.h" |
| #include "rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| |
| // Proves linear echo cancellation functionality |
| class Subtractor { |
| public: |
| Subtractor(const EchoCanceller3Config& config, |
| ApmDataDumper* data_dumper, |
| Aec3Optimization optimization); |
| ~Subtractor(); |
| |
| // Performs the echo subtraction. |
| void Process(const RenderBuffer& render_buffer, |
| const rtc::ArrayView<const float> capture, |
| const RenderSignalAnalyzer& render_signal_analyzer, |
| const AecState& aec_state, |
| SubtractorOutput* output); |
| |
| void HandleEchoPathChange(const EchoPathVariability& echo_path_variability); |
| |
| // Exits the initial state. |
| void ExitInitialState(); |
| |
| // Returns the block-wise frequency response for the main adaptive filter. |
| const std::vector<std::array<float, kFftLengthBy2Plus1>>& |
| FilterFrequencyResponse() const { |
| return main_filter_.FilterFrequencyResponse(); |
| } |
| |
| // Returns the estimate of the impulse response for the main adaptive filter. |
| const std::vector<float>& FilterImpulseResponse() const { |
| return main_filter_.FilterImpulseResponse(); |
| } |
| |
| void DumpFilters() { |
| main_filter_.DumpFilter("aec3_subtractor_H_main", "aec3_subtractor_h_main"); |
| shadow_filter_.DumpFilter("aec3_subtractor_H_shadow", |
| "aec3_subtractor_h_shadow"); |
| } |
| |
| private: |
| class FilterMisadjustmentEstimator { |
| public: |
| FilterMisadjustmentEstimator() = default; |
| ~FilterMisadjustmentEstimator() = default; |
| // Update the misadjustment estimator. |
| void Update(const SubtractorOutput& output); |
| // GetMisadjustment() Returns a recommended scale for the filter so the |
| // prediction error energy gets closer to the energy that is seen at the |
| // microphone input. |
| float GetMisadjustment() const { |
| RTC_DCHECK_GT(inv_misadjustment_, 0.0f); |
| // It is not aiming to adjust all the estimated mismatch. Instead, |
| // it adjusts half of that estimated mismatch. |
| return 2.f / sqrtf(inv_misadjustment_); |
| } |
| // Returns true if the prediciton error energy is significantly larger |
| // than the microphone signal energy and, therefore, an adjustment is |
| // recommended. |
| bool IsAdjustmentNeeded() const { return inv_misadjustment_ > 10.f; } |
| void Reset(); |
| void Dump(ApmDataDumper* data_dumper) const; |
| |
| private: |
| const int n_blocks_ = 4; |
| int n_blocks_acum_ = 0; |
| float e2_acum_ = 0.f; |
| float y2_acum_ = 0.f; |
| float inv_misadjustment_ = 0.f; |
| int overhang_ = 0.f; |
| }; |
| |
| const Aec3Fft fft_; |
| ApmDataDumper* data_dumper_; |
| const Aec3Optimization optimization_; |
| const EchoCanceller3Config config_; |
| const bool adaptation_during_saturation_; |
| const bool enable_misadjustment_estimator_; |
| const bool enable_agc_gain_change_response_; |
| const bool enable_shadow_filter_jumpstart_; |
| const bool enable_shadow_filter_boosted_jumpstart_; |
| const bool enable_early_shadow_filter_jumpstart_; |
| |
| AdaptiveFirFilter main_filter_; |
| AdaptiveFirFilter shadow_filter_; |
| MainFilterUpdateGain G_main_; |
| ShadowFilterUpdateGain G_shadow_; |
| FilterMisadjustmentEstimator filter_misadjustment_estimator_; |
| size_t poor_shadow_filter_counter_ = 0; |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Subtractor); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_ |