1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/audio_processing/gain_controller2.h"
12
13 #include "common_audio/include/audio_util.h"
14 #include "modules/audio_processing/audio_buffer.h"
15 #include "modules/audio_processing/include/audio_frame_view.h"
16 #include "modules/audio_processing/logging/apm_data_dumper.h"
17 #include "rtc_base/atomic_ops.h"
18 #include "rtc_base/checks.h"
19 #include "rtc_base/logging.h"
20 #include "rtc_base/strings/string_builder.h"
21
22 namespace webrtc {
23
24 int GainController2::instance_count_ = 0;
25
GainController2()26 GainController2::GainController2()
27 : data_dumper_(rtc::AtomicOps::Increment(&instance_count_)),
28 gain_applier_(/*hard_clip_samples=*/false,
29 /*initial_gain_factor=*/0.f),
30 limiter_(static_cast<size_t>(48000), &data_dumper_, "Agc2"),
31 calls_since_last_limiter_log_(0) {
32 if (config_.adaptive_digital.enabled) {
33 adaptive_agc_ = std::make_unique<AdaptiveAgc>(&data_dumper_);
34 }
35 }
36
37 GainController2::~GainController2() = default;
38
Initialize(int sample_rate_hz)39 void GainController2::Initialize(int sample_rate_hz) {
40 RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
41 sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
42 sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
43 sample_rate_hz == AudioProcessing::kSampleRate48kHz);
44 limiter_.SetSampleRate(sample_rate_hz);
45 data_dumper_.InitiateNewSetOfRecordings();
46 data_dumper_.DumpRaw("sample_rate_hz", sample_rate_hz);
47 calls_since_last_limiter_log_ = 0;
48 }
49
Process(AudioBuffer * audio)50 void GainController2::Process(AudioBuffer* audio) {
51 data_dumper_.DumpRaw("agc2_notified_analog_level", analog_level_);
52 AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
53 audio->num_frames());
54 // Apply fixed gain first, then the adaptive one.
55 gain_applier_.ApplyGain(float_frame);
56 if (adaptive_agc_) {
57 adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
58 }
59 limiter_.Process(float_frame);
60
61 // Log limiter stats every 30 seconds.
62 ++calls_since_last_limiter_log_;
63 if (calls_since_last_limiter_log_ == 3000) {
64 calls_since_last_limiter_log_ = 0;
65 InterpolatedGainCurve::Stats stats = limiter_.GetGainCurveStats();
66 RTC_LOG(LS_INFO) << "AGC2 limiter stats"
67 << " | identity: " << stats.look_ups_identity_region
68 << " | knee: " << stats.look_ups_knee_region
69 << " | limiter: " << stats.look_ups_limiter_region
70 << " | saturation: " << stats.look_ups_saturation_region;
71 }
72 }
73
NotifyAnalogLevel(int level)74 void GainController2::NotifyAnalogLevel(int level) {
75 if (analog_level_ != level && adaptive_agc_) {
76 adaptive_agc_->HandleInputGainChange();
77 }
78 analog_level_ = level;
79 }
80
ApplyConfig(const AudioProcessing::Config::GainController2 & config)81 void GainController2::ApplyConfig(
82 const AudioProcessing::Config::GainController2& config) {
83 RTC_DCHECK(Validate(config));
84
85 config_ = config;
86 if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) {
87 // Reset the limiter to quickly react on abrupt level changes caused by
88 // large changes of the fixed gain.
89 limiter_.Reset();
90 }
91 gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
92 if (config_.adaptive_digital.enabled) {
93 adaptive_agc_ =
94 std::make_unique<AdaptiveAgc>(&data_dumper_, config_.adaptive_digital);
95 } else {
96 adaptive_agc_.reset();
97 }
98 }
99
Validate(const AudioProcessing::Config::GainController2 & config)100 bool GainController2::Validate(
101 const AudioProcessing::Config::GainController2& config) {
102 const auto& fixed = config.fixed_digital;
103 const auto& adaptive = config.adaptive_digital;
104 return fixed.gain_db >= 0.f && fixed.gain_db < 50.f &&
105 adaptive.vad_probability_attack > 0.f &&
106 adaptive.vad_probability_attack <= 1.f &&
107 adaptive.level_estimator_adjacent_speech_frames_threshold >= 1 &&
108 adaptive.initial_saturation_margin_db >= 0.f &&
109 adaptive.initial_saturation_margin_db <= 100.f &&
110 adaptive.extra_saturation_margin_db >= 0.f &&
111 adaptive.extra_saturation_margin_db <= 100.f &&
112 adaptive.gain_applier_adjacent_speech_frames_threshold >= 1 &&
113 adaptive.max_gain_change_db_per_second > 0.f &&
114 adaptive.max_output_noise_level_dbfs <= 0.f;
115 }
116
117 } // namespace webrtc
118