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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
156 lines
5.5 KiB
C++
156 lines
5.5 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_AEC_STATE_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
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#include <algorithm>
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#include <memory>
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#include <vector>
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#include "api/array_view.h"
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#include "api/optional.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/echo_path_variability.h"
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#include "modules/audio_processing/aec3/erl_estimator.h"
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#include "modules/audio_processing/aec3/erle_estimator.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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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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explicit AecState(const AudioProcessing::Config::EchoCanceller3& config);
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~AecState();
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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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// Returns whether there has been echo leakage detected.
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bool EchoLeakageDetected() const { return echo_leakage_detected_; }
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// Returns whether the render signal is currently active.
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// TODO(peah): Deprecate this in an upcoming CL.
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bool ActiveRender() const { return blocks_with_filter_adaptation_ > 200; }
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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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// 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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// 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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// Returns the externally provided delay.
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rtc::Optional<size_t> ExternalDelay() const { return external_delay_; }
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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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// Returns whether the echo signal is saturated.
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bool SaturatedEcho() const { return echo_saturation_; }
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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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// Returns whether a probable headset setup has been detected.
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bool HeadsetDetected() const { return headset_detected_; }
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// Takes appropriate action at an echo path change.
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void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
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// Returns the decay factor for the echo reverberation.
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float ReverbDecay() const { return reverb_decay_; }
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// Returns whether the echo suppression gain should be forced to zero.
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bool ForcedZeroGain() const { return force_zero_gain_; }
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// Returns whether the echo in the capture signal is audible.
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bool InaudibleEcho() const { return echo_audibility_.InaudibleEcho(); }
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// Updates the aec state with the AEC output signal.
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void UpdateWithOutput(rtc::ArrayView<const float> e) {
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echo_audibility_.UpdateWithOutput(e);
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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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adaptive_filter_frequency_response,
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const std::array<float, kAdaptiveFilterTimeDomainLength>&
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adaptive_filter_impulse_response,
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const rtc::Optional<size_t>& external_delay_samples,
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const RenderBuffer& render_buffer,
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const std::array<float, kFftLengthBy2Plus1>& E2_main,
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const std::array<float, kFftLengthBy2Plus1>& Y2,
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rtc::ArrayView<const float> x,
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const std::array<float, kBlockSize>& s_main,
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bool echo_leakage_detected);
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private:
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class EchoAudibility {
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public:
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void Update(rtc::ArrayView<const float> x,
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const std::array<float, kBlockSize>& s);
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void UpdateWithOutput(rtc::ArrayView<const float> e);
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bool InaudibleEcho() const { return inaudible_echo_; }
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private:
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float max_nearend_ = 0.f;
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size_t max_nearend_counter_ = 0;
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size_t low_farend_counter_ = 0;
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bool inaudible_echo_ = false;
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};
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void UpdateReverb(const std::array<float, kAdaptiveFilterTimeDomainLength>&
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impulse_response);
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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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size_t blocks_with_filter_adaptation_ = 0;
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bool usable_linear_estimate_ = false;
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bool echo_leakage_detected_ = false;
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bool capture_signal_saturation_ = false;
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bool echo_saturation_ = false;
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bool headset_detected_ = false;
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float previous_max_sample_ = 0.f;
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bool force_zero_gain_ = false;
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bool render_received_ = false;
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size_t force_zero_gain_counter_ = 0;
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rtc::Optional<size_t> filter_delay_;
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rtc::Optional<size_t> external_delay_;
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size_t blocks_since_last_saturation_ = 1000;
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float reverb_decay_to_test_ = 0.9f;
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float reverb_decay_candidate_ = 0.f;
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float reverb_decay_candidate_residual_ = -1.f;
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EchoAudibility echo_audibility_;
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const AudioProcessing::Config::EchoCanceller3 config_;
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float reverb_decay_;
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RTC_DISALLOW_COPY_AND_ASSIGN(AecState);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
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