summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/audio_processing/agc2/speech_level_estimator.cc
blob: 7bf3252116fa836ed6822ae7128dd91588912ca5 (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
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
/*
 *  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) {
  return rtc::SafeClamp<float>(level_estimate_dbfs, -90.0f, 30.0f);
}

// Returns the initial speech level estimate needed to apply the initial gain.
float GetInitialSpeechLevelEstimateDbfs(
    const AudioProcessing::Config::GainController2::AdaptiveDigital& config) {
  return ClampLevelEstimateDbfs(-kSaturationProtectorInitialHeadroomDb -
                                config.initial_gain_db - config.headroom_db);
}

}  // namespace

bool SpeechLevelEstimator::LevelEstimatorState::operator==(
    const SpeechLevelEstimator::LevelEstimatorState& b) const {
  return time_to_confidence_ms == b.time_to_confidence_ms &&
         level_dbfs.numerator == b.level_dbfs.numerator &&
         level_dbfs.denominator == b.level_dbfs.denominator;
}

float SpeechLevelEstimator::LevelEstimatorState::Ratio::GetRatio() const {
  RTC_DCHECK_NE(denominator, 0.f);
  return numerator / denominator;
}

SpeechLevelEstimator::SpeechLevelEstimator(
    ApmDataDumper* apm_data_dumper,
    const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
    int adjacent_speech_frames_threshold)
    : apm_data_dumper_(apm_data_dumper),
      initial_speech_level_dbfs_(GetInitialSpeechLevelEstimateDbfs(config)),
      adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold),
      level_dbfs_(initial_speech_level_dbfs_),
      // TODO(bugs.webrtc.org/7494): Remove init below when AGC2 input volume
      // controller temporal dependency removed.
      is_confident_(false) {
  RTC_DCHECK(apm_data_dumper_);
  RTC_DCHECK_GE(adjacent_speech_frames_threshold_, 1);
  Reset();
}

void SpeechLevelEstimator::Update(float rms_dbfs,
                                  float peak_dbfs,
                                  float speech_probability) {
  RTC_DCHECK_GT(rms_dbfs, -150.0f);
  RTC_DCHECK_LT(rms_dbfs, 50.0f);
  RTC_DCHECK_GT(peak_dbfs, -150.0f);
  RTC_DCHECK_LT(peak_dbfs, 50.0f);
  RTC_DCHECK_GE(speech_probability, 0.0f);
  RTC_DCHECK_LE(speech_probability, 1.0f);
  if (speech_probability < kVadConfidenceThreshold) {
    // Not a speech frame.
    if (adjacent_speech_frames_threshold_ > 1) {
      // When two or more adjacent speech frames are required in order to update
      // the state, we need to decide whether to discard or confirm the updates
      // based on the speech sequence length.
      if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
        // First non-speech frame after a long enough sequence of speech frames.
        // Update the reliable state.
        reliable_state_ = preliminary_state_;
      } else if (num_adjacent_speech_frames_ > 0) {
        // First non-speech frame after a too short sequence of speech frames.
        // Reset to the last reliable state.
        preliminary_state_ = reliable_state_;
      }
    }
    num_adjacent_speech_frames_ = 0;
  } else {
    // Speech frame observed.
    num_adjacent_speech_frames_++;

    // Update preliminary level estimate.
    RTC_DCHECK_GE(preliminary_state_.time_to_confidence_ms, 0);
    const bool buffer_is_full = preliminary_state_.time_to_confidence_ms == 0;
    if (!buffer_is_full) {
      preliminary_state_.time_to_confidence_ms -= kFrameDurationMs;
    }
    // Weighted average of levels with speech probability as weight.
    RTC_DCHECK_GT(speech_probability, 0.0f);
    const float leak_factor = buffer_is_full ? kLevelEstimatorLeakFactor : 1.0f;
    preliminary_state_.level_dbfs.numerator =
        preliminary_state_.level_dbfs.numerator * leak_factor +
        rms_dbfs * speech_probability;
    preliminary_state_.level_dbfs.denominator =
        preliminary_state_.level_dbfs.denominator * leak_factor +
        speech_probability;

    const float level_dbfs = preliminary_state_.level_dbfs.GetRatio();

    if (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_) {
      // `preliminary_state_` is now reliable. Update the last level estimation.
      level_dbfs_ = ClampLevelEstimateDbfs(level_dbfs);
    }
  }
  UpdateIsConfident();
  DumpDebugData();
}

void SpeechLevelEstimator::UpdateIsConfident() {
  if (adjacent_speech_frames_threshold_ == 1) {
    // Ignore `reliable_state_` when a single frame is enough to update the
    // level estimate (because it is not used).
    is_confident_ = preliminary_state_.time_to_confidence_ms == 0;
    return;
  }
  // Once confident, it remains confident.
  RTC_DCHECK(reliable_state_.time_to_confidence_ms != 0 ||
             preliminary_state_.time_to_confidence_ms == 0);
  // During the first long enough speech sequence, `reliable_state_` must be
  // ignored since `preliminary_state_` is used.
  is_confident_ =
      reliable_state_.time_to_confidence_ms == 0 ||
      (num_adjacent_speech_frames_ >= adjacent_speech_frames_threshold_ &&
       preliminary_state_.time_to_confidence_ms == 0);
}

void SpeechLevelEstimator::Reset() {
  ResetLevelEstimatorState(preliminary_state_);
  ResetLevelEstimatorState(reliable_state_);
  level_dbfs_ = initial_speech_level_dbfs_;
  num_adjacent_speech_frames_ = 0;
}

void SpeechLevelEstimator::ResetLevelEstimatorState(
    LevelEstimatorState& state) const {
  state.time_to_confidence_ms = kLevelEstimatorTimeToConfidenceMs;
  state.level_dbfs.numerator = initial_speech_level_dbfs_;
  state.level_dbfs.denominator = 1.0f;
}

void SpeechLevelEstimator::DumpDebugData() const {
  if (!apm_data_dumper_)
    return;
  apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", level_dbfs_);
  apm_data_dumper_->DumpRaw("agc2_speech_level_is_confident", is_confident_);
  apm_data_dumper_->DumpRaw(
      "agc2_adaptive_level_estimator_num_adjacent_speech_frames",
      num_adjacent_speech_frames_);
  apm_data_dumper_->DumpRaw(
      "agc2_adaptive_level_estimator_preliminary_level_estimate_num",
      preliminary_state_.level_dbfs.numerator);
  apm_data_dumper_->DumpRaw(
      "agc2_adaptive_level_estimator_preliminary_level_estimate_den",
      preliminary_state_.level_dbfs.denominator);
  apm_data_dumper_->DumpRaw(
      "agc2_adaptive_level_estimator_preliminary_time_to_confidence_ms",
      preliminary_state_.time_to_confidence_ms);
  apm_data_dumper_->DumpRaw(
      "agc2_adaptive_level_estimator_reliable_time_to_confidence_ms",
      reliable_state_.time_to_confidence_ms);
}

}  // namespace webrtc