Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
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/*
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* Copyright (c) 2019 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_NS_SPEECH_PROBABILITY_ESTIMATOR_H_
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#define MODULES_AUDIO_PROCESSING_NS_SPEECH_PROBABILITY_ESTIMATOR_H_
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#include <array>
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#include "api/array_view.h"
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#include "modules/audio_processing/ns/ns_common.h"
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#include "modules/audio_processing/ns/signal_model_estimator.h"
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namespace webrtc {
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// Class for estimating the probability of speech.
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class SpeechProbabilityEstimator {
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public:
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SpeechProbabilityEstimator();
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SpeechProbabilityEstimator(const SpeechProbabilityEstimator&) = delete;
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SpeechProbabilityEstimator& operator=(const SpeechProbabilityEstimator&) =
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delete;
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// Compute speech probability.
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void Update(
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int32_t num_analyzed_frames,
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rtc::ArrayView<const float, kFftSizeBy2Plus1> prior_snr,
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rtc::ArrayView<const float, kFftSizeBy2Plus1> post_snr,
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rtc::ArrayView<const float, kFftSizeBy2Plus1> conservative_noise_spectrum,
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rtc::ArrayView<const float, kFftSizeBy2Plus1> signal_spectrum,
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float signal_spectral_sum,
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float signal_energy);
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float get_prior_probability() const { return prior_speech_prob_; }
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rtc::ArrayView<const float> get_probability() { return speech_probability_; }
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private:
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SignalModelEstimator signal_model_estimator_;
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float prior_speech_prob_ = .5f;
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std::array<float, kFftSizeBy2Plus1> speech_probability_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_NS_SPEECH_PROBABILITY_ESTIMATOR_H_
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