6d822adac4
During the first few capture frames, there is no way for the AEC to tell whether there is echo in the capture signal as the echo removal functionality in the AEC has not yet seen any render signal. To avoid initial echo bursts due to this, this CL adds functionality for forcing the echo suppression gain to zero during the first 50 blocks (200 ms) after call start and after a reported echo path change. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2808733002 Cr-Commit-Position: refs/heads/master@{#17624}
121 lines
4.2 KiB
C++
121 lines
4.2 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 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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#include <algorithm>
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#include <memory>
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#include <vector>
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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/erl_estimator.h"
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#include "webrtc/modules/audio_processing/aec3/erle_estimator.h"
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#include "webrtc/modules/audio_processing/aec3/render_buffer.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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AecState();
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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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bool ActiveRender() const { return active_render_blocks_ > 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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// TODO(peah): Make this adaptive.
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float ReverbDecayFactor() const { return 0.f; }
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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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// 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 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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bool echo_leakage_detected);
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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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size_t active_render_blocks_ = 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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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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RTC_DISALLOW_COPY_AND_ASSIGN(AecState);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
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