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/*
* Copyright (c) 2018 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_OUTPUT_ANALYZER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
#include <vector>
#include "modules/audio_processing/aec3/subtractor_output.h"
namespace webrtc {
// Class for analyzing the properties subtractor output.
class SubtractorOutputAnalyzer {
public:
explicit SubtractorOutputAnalyzer(size_t num_capture_channels);
~SubtractorOutputAnalyzer() = default;
// Analyses the subtractor output.
void Update(rtc::ArrayView<const SubtractorOutput> subtractor_output,
bool* any_filter_converged,
bool* any_coarse_filter_converged,
bool* all_filters_diverged);
const std::vector<bool>& ConvergedFilters() const {
return filters_converged_;
}
// Handle echo path change.
void HandleEchoPathChange();
private:
std::vector<bool> filters_converged_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
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