| Index: webrtc/modules/audio_processing/aec3/aec_state.h
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| diff --git a/webrtc/modules/audio_processing/aec3/aec_state.h b/webrtc/modules/audio_processing/aec3/aec_state.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..e3502b4a4f5b5e42ae0446e0bb305bc98747c285
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| --- /dev/null
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| +++ b/webrtc/modules/audio_processing/aec3/aec_state.h
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| @@ -0,0 +1,127 @@
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| +/*
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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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| +
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| +#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
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| +#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
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| +
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| +#include <algorithm>
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| +#include <memory>
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| +#include <vector>
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| +
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| +#include "webrtc/base/array_view.h"
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| +#include "webrtc/base/constructormagic.h"
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| +#include "webrtc/base/optional.h"
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| +#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
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| +#include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
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| +#include "webrtc/modules/audio_processing/aec3/erle_estimator.h"
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| +#include "webrtc/modules/audio_processing/aec3/erl_estimator.h"
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| +#include "webrtc/modules/audio_processing/aec3/fft_buffer.h"
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| +
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| +namespace webrtc {
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| +
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| +class ApmDataDumper;
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| +
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| +// Handles the state and the conditions for the echo removal functionality.
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| +class AecState {
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| + public:
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| +  AecState();
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| +  ~AecState();
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| +
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| +  // Returns whether the linear filter estimate is usable.
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| +  bool UsableLinearEstimate() const { return usable_linear_estimate_; }
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| +
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| +  // Returns whether there has been echo leakage detected.
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| +  bool EchoLeakageDetected() const { return echo_leakage_detected_; }
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| +
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| +  // Returns whether it is possible at all to use the model based echo removal
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| +  // functionalities.
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| +  bool ModelBasedAecFeasible() const { return model_based_aec_feasible_; }
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| +
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| +  // Returns whether the render signal is currently active.
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| +  bool ActiveRender() const { return active_render_counter_ > 0; }
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| +
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| +  // Returns the ERLE.
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| +  const std::array<float, kFftLengthBy2Plus1>& Erle() const {
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| +    return erle_estimator_.Erle();
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| +  }
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| +
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| +  // Returns the ERL.
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| +  const std::array<float, kFftLengthBy2Plus1>& Erl() const {
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| +    return erl_estimator_.Erl();
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| +  }
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| +
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| +  // Returns the delay estimate based on the linear filter.
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| +  rtc::Optional<size_t> FilterDelay() const { return filter_delay_; }
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| +
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| +  // Returns the externally provided delay.
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| +  rtc::Optional<size_t> ExternalDelay() const { return external_delay_; }
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| +
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| +  // Returns the bands where the linear filter is reliable.
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| +  const std::array<bool, kFftLengthBy2Plus1>& BandsWithReliableFilter() const {
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| +    return bands_with_reliable_filter_;
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| +  }
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| +
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| +  // Reports whether the filter is poorly aligned.
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| +  bool PoorlyAlignedFilter() const {
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| +    return FilterDelay() ? *FilterDelay() > 0.75f * filter_length_ : false;
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| +  }
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| +
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| +  // Returns the strength of the filter.
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| +  const std::array<float, kFftLengthBy2Plus1>& FilterEstimateStrength() const {
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| +    return filter_estimate_strength_;
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| +  }
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| +
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| +  // Returns whether the capture signal is saturated.
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| +  bool SaturatedCapture() const { return capture_signal_saturation_; }
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| +
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| +  // Updates the capture signal saturation.
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| +  void UpdateCaptureSaturation(bool capture_signal_saturation) {
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| +    capture_signal_saturation_ = capture_signal_saturation;
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| +  }
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| +
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| +  // Returns whether a probable headset setup has been detected.
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| +  bool HeadsetDetected() const { return headset_detected_; }
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| +
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| +  // Updates the aec state.
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| +  void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
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| +                  filter_frequency_response,
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| +              const rtc::Optional<size_t>& external_delay_samples,
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| +              const FftBuffer& X_buffer,
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| +              const std::array<float, kFftLengthBy2Plus1>& E2_main,
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| +              const std::array<float, kFftLengthBy2Plus1>& E2_shadow,
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| +              const std::array<float, kFftLengthBy2Plus1>& Y2,
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| +              rtc::ArrayView<const float> x,
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| +              const EchoPathVariability& echo_path_variability,
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| +              bool echo_leakage_detected);
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| +
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| + private:
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| +  static int instance_count_;
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| +  std::unique_ptr<ApmDataDumper> data_dumper_;
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| +  ErlEstimator erl_estimator_;
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| +  ErleEstimator erle_estimator_;
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| +  int echo_path_change_counter_;
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| +  int active_render_counter_;
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| +  bool usable_linear_estimate_ = false;
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| +  bool echo_leakage_detected_ = false;
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| +  bool model_based_aec_feasible_ = false;
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| +  bool capture_signal_saturation_ = false;
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| +  bool headset_detected_ = false;
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| +  rtc::Optional<size_t> filter_delay_;
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| +  rtc::Optional<size_t> external_delay_;
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| +  std::array<bool, kFftLengthBy2Plus1> bands_with_reliable_filter_;
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| +  std::array<float, kFftLengthBy2Plus1> filter_estimate_strength_;
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| +  size_t filter_length_;
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| +
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| +  RTC_DISALLOW_COPY_AND_ASSIGN(AecState);
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| +};
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| +
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| +}  // namespace webrtc
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| +
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| +#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
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| 
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