blob: 6d4493858794548844a26440b2781afcd4b11256 [file] [log] [blame]
* 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 <stddef.h>
#include <type_traits>
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class ApmDataDumper;
// Level estimator for the digital adaptive gain controller.
class AdaptiveModeLevelEstimator {
explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
AdaptiveModeLevelEstimator(const AdaptiveModeLevelEstimator&) = delete;
AdaptiveModeLevelEstimator& operator=(const AdaptiveModeLevelEstimator&) =
AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper,
int adjacent_speech_frames_threshold);
// Updates the level estimation.
void Update(const VadLevelAnalyzer::Result& vad_data);
// Returns the estimated speech plus noise level.
float level_dbfs() const { return level_dbfs_; }
// Returns true if the estimator is confident on its current estimate.
bool IsConfident() const;
void Reset();
// Part of the level estimator state used for check-pointing and restore ops.
struct LevelEstimatorState {
bool operator==(const LevelEstimatorState& s) const;
inline bool operator!=(const LevelEstimatorState& s) const {
return !(*this == s);
struct Ratio {
float numerator;
float denominator;
float GetRatio() const;
// TODO( Remove time_to_confidence_ms if redundant.
int time_to_confidence_ms;
Ratio level_dbfs;
static_assert(std::is_trivially_copyable<LevelEstimatorState>::value, "");
void ResetLevelEstimatorState(LevelEstimatorState& state) const;
void DumpDebugData() const;
ApmDataDumper* const apm_data_dumper_;
const int adjacent_speech_frames_threshold_;
LevelEstimatorState preliminary_state_;
LevelEstimatorState reliable_state_;
float level_dbfs_;
int num_adjacent_speech_frames_;
} // namespace webrtc