chromium/third_party/webrtc/modules/audio_processing/agc2/saturation_protector.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/saturation_protector.h"

#include <memory>

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

namespace webrtc {
namespace {

constexpr int kPeakEnveloperSuperFrameLengthMs = …;
constexpr float kMinMarginDb = …;
constexpr float kMaxMarginDb = …;
constexpr float kAttack = …;
constexpr float kDecay = …;

// Saturation protector state. Defined outside of `SaturationProtectorImpl` to
// implement check-point and restore ops.
struct SaturationProtectorState { … };

// Resets the saturation protector state.
void ResetSaturationProtectorState(float initial_headroom_db,
                                   SaturationProtectorState& state) { … }

// Updates `state` by analyzing the estimated speech level `speech_level_dbfs`
// and the peak level `peak_dbfs` for an observed frame. `state` must not be
// modified without calling this function.
void UpdateSaturationProtectorState(float peak_dbfs,
                                    float speech_level_dbfs,
                                    SaturationProtectorState& state) { … }

// Saturation protector which recommends a headroom based on the recent peaks.
class SaturationProtectorImpl : public SaturationProtector { … };

}  // namespace

std::unique_ptr<SaturationProtector> CreateSaturationProtector(
    float initial_headroom_db,
    int adjacent_speech_frames_threshold,
    ApmDataDumper* apm_data_dumper) { … }

}  // namespace webrtc