| /* |
| * 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 <math.h> |
| #include <stddef.h> |
| |
| #include <array> |
| #include <vector> |
| |
| #include "api/array_view.h" |
| #include "api/audio/echo_canceller3_config.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/coarse_filter_update_gain.h" |
| #include "modules/audio_processing/aec3/echo_path_variability.h" |
| #include "modules/audio_processing/aec3/refined_filter_update_gain.h" |
| #include "modules/audio_processing/aec3/render_buffer.h" |
| #include "modules/audio_processing/aec3/render_signal_analyzer.h" |
| #include "modules/audio_processing/aec3/subtractor_output.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| // Proves linear echo cancellation functionality |
| class Subtractor { |
| public: |
| Subtractor(const EchoCanceller3Config& config, |
| size_t num_render_channels, |
| size_t num_capture_channels, |
| ApmDataDumper* data_dumper, |
| Aec3Optimization optimization); |
| ~Subtractor(); |
| Subtractor(const Subtractor&) = delete; |
| Subtractor& operator=(const Subtractor&) = delete; |
| |
| // Performs the echo subtraction. |
| void Process(const RenderBuffer& render_buffer, |
| const std::vector<std::vector<float>>& capture, |
| const RenderSignalAnalyzer& render_signal_analyzer, |
| const AecState& aec_state, |
| rtc::ArrayView<SubtractorOutput> outputs); |
| |
| void HandleEchoPathChange(const EchoPathVariability& echo_path_variability); |
| |
| // Exits the initial state. |
| void ExitInitialState(); |
| |
| // Returns the block-wise frequency responses for the refined adaptive |
| // filters. |
| const std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>& |
| FilterFrequencyResponses() const { |
| return refined_frequency_responses_; |
| } |
| |
| // Returns the estimates of the impulse responses for the refined adaptive |
| // filters. |
| const std::vector<std::vector<float>>& FilterImpulseResponses() const { |
| return refined_impulse_responses_; |
| } |
| |
| void DumpFilters() { |
| data_dumper_->DumpRaw( |
| "aec3_subtractor_h_refined", |
| rtc::ArrayView<const float>( |
| refined_impulse_responses_[0].data(), |
| GetTimeDomainLength( |
| refined_filters_[0]->max_filter_size_partitions()))); |
| if (ApmDataDumper::IsAvailable()) { |
| RTC_DCHECK_GT(coarse_impulse_responses_.size(), 0); |
| data_dumper_->DumpRaw( |
| "aec3_subtractor_h_coarse", |
| rtc::ArrayView<const float>( |
| coarse_impulse_responses_[0].data(), |
| GetTimeDomainLength( |
| coarse_filter_[0]->max_filter_size_partitions()))); |
| } |
| |
| refined_filters_[0]->DumpFilter("aec3_subtractor_H_refined"); |
| coarse_filter_[0]->DumpFilter("aec3_subtractor_H_coarse"); |
| } |
| |
| 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 size_t num_capture_channels_; |
| const bool use_coarse_filter_reset_hangover_; |
| |
| std::vector<std::unique_ptr<AdaptiveFirFilter>> refined_filters_; |
| std::vector<std::unique_ptr<AdaptiveFirFilter>> coarse_filter_; |
| std::vector<std::unique_ptr<RefinedFilterUpdateGain>> refined_gains_; |
| std::vector<std::unique_ptr<CoarseFilterUpdateGain>> coarse_gains_; |
| std::vector<FilterMisadjustmentEstimator> filter_misadjustment_estimators_; |
| std::vector<size_t> poor_coarse_filter_counters_; |
| std::vector<int> coarse_filter_reset_hangover_; |
| std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> |
| refined_frequency_responses_; |
| std::vector<std::vector<float>> refined_impulse_responses_; |
| std::vector<std::vector<float>> coarse_impulse_responses_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_ |