chromium/third_party/webrtc/modules/audio_processing/agc2/speech_probability_buffer.h

/*
 *  Copyright (c) 2022 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_AGC2_SPEECH_PROBABILITY_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_

#include <vector>

#include "rtc_base/gtest_prod_util.h"

namespace webrtc {

// This class implements a circular buffer that stores speech probabilities
// for a speech segment and estimates speech activity for that segment.
class SpeechProbabilityBuffer { … };

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_SPEECH_PROBABILITY_BUFFER_H_