chromium/third_party/webrtc/modules/audio_processing/agc/agc.cc

/*
 *  Copyright (c) 2012 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/agc/agc.h"

#include <cmath>
#include <cstdlib>
#include <vector>

#include "modules/audio_processing/agc/loudness_histogram.h"
#include "modules/audio_processing/agc/utility.h"
#include "rtc_base/checks.h"

namespace webrtc {
namespace {

constexpr int kDefaultLevelDbfs = …;
constexpr int kNumAnalysisFrames = …;
constexpr double kActivityThreshold = …;
constexpr int kNum10msFramesInOneSecond = …;
constexpr int kMaxSampleRateHz = …;

}  // namespace

Agc::Agc()
    : … { … }

Agc::~Agc() = default;

void Agc::Process(rtc::ArrayView<const int16_t> audio) { … }

bool Agc::GetRmsErrorDb(int* error) { … }

void Agc::Reset() { … }

int Agc::set_target_level_dbfs(int level) { … }

int Agc::target_level_dbfs() const { … }

float Agc::voice_probability() const { … }

}  // namespace webrtc