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

/*
 *  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.
 */

#include "modules/audio_processing/agc2/speech_probability_buffer.h"

#include <algorithm>

#include "rtc_base/checks.h"

namespace webrtc {
namespace {

constexpr float kActivityThreshold = …;
constexpr int kNumAnalysisFrames = …;
// We use 12 in AGC2 adaptive digital, but with a slightly different logic.
constexpr int kTransientWidthThreshold = …;

}  // namespace

SpeechProbabilityBuffer::SpeechProbabilityBuffer(
    float low_probability_threshold)
    : … { … }

void SpeechProbabilityBuffer::Update(float probability) { … }

void SpeechProbabilityBuffer::RemoveTransient() { … }

bool SpeechProbabilityBuffer::IsActiveSegment() const { … }

void SpeechProbabilityBuffer::Reset() { … }

}  // namespace webrtc