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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.h

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/*
* 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_RENDER_SIGNAL_ANALYZER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
#include <algorithm>
#include <array>
#include <cstddef>
#include "absl/types/optional.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/render_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
// Provides functionality for analyzing the properties of the render signal.
class RenderSignalAnalyzer {
public:
explicit RenderSignalAnalyzer(const EchoCanceller3Config& config);
~RenderSignalAnalyzer();
// Updates the render signal analysis with the most recent render signal.
void Update(const RenderBuffer& render_buffer,
const absl::optional<size_t>& delay_partitions);
// Returns true if the render signal is poorly exciting.
bool PoorSignalExcitation() const {
RTC_DCHECK_LT(2, narrow_band_counters_.size());
return std::any_of(narrow_band_counters_.begin(),
narrow_band_counters_.end(),
[](size_t a) { return a > 10; });
}
// Zeros the array around regions with narrow bands signal characteristics.
void MaskRegionsAroundNarrowBands(
std::array<float, kFftLengthBy2Plus1>* v) const;
absl::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
private:
const int strong_peak_freeze_duration_;
std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
absl::optional<int> narrow_peak_band_;
size_t narrow_peak_counter_;
RTC_DISALLOW_COPY_AND_ASSIGN(RenderSignalAnalyzer);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_