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

/*
 *  Copyright (c) 2018 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_level_estimator.h"

#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"

namespace webrtc {
namespace {

float ClampLevelEstimateDbfs(float level_estimate_dbfs) { … }

// Returns the initial speech level estimate needed to apply the initial gain.
float GetInitialSpeechLevelEstimateDbfs(
    const AudioProcessing::Config::GainController2::AdaptiveDigital& config) { … }

}  // namespace

bool SpeechLevelEstimator::LevelEstimatorState::operator==(
    const SpeechLevelEstimator::LevelEstimatorState& b) const { … }

float SpeechLevelEstimator::LevelEstimatorState::Ratio::GetRatio() const { … }

SpeechLevelEstimator::SpeechLevelEstimator(
    ApmDataDumper* apm_data_dumper,
    const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
    int adjacent_speech_frames_threshold)
    : … { … }

void SpeechLevelEstimator::Update(float rms_dbfs,
                                  float peak_dbfs,
                                  float speech_probability) { … }

void SpeechLevelEstimator::UpdateIsConfident() { … }

void SpeechLevelEstimator::Reset() { … }

void SpeechLevelEstimator::ResetLevelEstimatorState(
    LevelEstimatorState& state) const { … }

void SpeechLevelEstimator::DumpDebugData() const { … }

}  // namespace webrtc