blob: 213fc0f0c8a0b07e18c1e4953329c29107360206 (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
|
/*
* 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
#include <stddef.h>
#include <type_traits>
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/saturation_protector.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 {
public:
explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
AdaptiveModeLevelEstimator(const AdaptiveModeLevelEstimator&) = delete;
AdaptiveModeLevelEstimator& operator=(const AdaptiveModeLevelEstimator&) =
delete;
AdaptiveModeLevelEstimator(
ApmDataDumper* apm_data_dumper,
AudioProcessing::Config::GainController2::LevelEstimator level_estimator,
int adjacent_speech_frames_threshold,
float initial_saturation_margin_db,
float extra_saturation_margin_db);
// 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();
private:
// 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(crbug.com/webrtc/7494): Remove time_to_full_buffer_ms if redundant.
int time_to_full_buffer_ms;
Ratio level_dbfs;
SaturationProtectorState saturation_protector;
};
static_assert(std::is_trivially_copyable<LevelEstimatorState>::value, "");
void ResetLevelEstimatorState(LevelEstimatorState& state) const;
void DumpDebugData() const;
ApmDataDumper* const apm_data_dumper_;
const AudioProcessing::Config::GainController2::LevelEstimator
level_estimator_type_;
const int adjacent_speech_frames_threshold_;
const float initial_saturation_margin_db_;
const float extra_saturation_margin_db_;
LevelEstimatorState preliminary_state_;
LevelEstimatorState reliable_state_;
float level_dbfs_;
int num_adjacent_speech_frames_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
|
