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/strings/string_builder.h"
20
21 namespace webrtc {
22
23 int GainController2::instance_count_ = 0;
24
GainController2()25 GainController2::GainController2()
26 : data_dumper_(
27 new ApmDataDumper(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_.get(), "Agc2") {
31 if (config_.adaptive_digital.enabled) {
32 adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get()));
33 }
34 }
35
36 GainController2::~GainController2() = default;
37
Initialize(int sample_rate_hz)38 void GainController2::Initialize(int sample_rate_hz) {
39 RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
40 sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
41 sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
42 sample_rate_hz == AudioProcessing::kSampleRate48kHz);
43 limiter_.SetSampleRate(sample_rate_hz);
44 data_dumper_->InitiateNewSetOfRecordings();
45 data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
46 }
47
Process(AudioBuffer * audio)48 void GainController2::Process(AudioBuffer* audio) {
49 AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
50 audio->num_frames());
51 // Apply fixed gain first, then the adaptive one.
52 gain_applier_.ApplyGain(float_frame);
53 if (adaptive_agc_) {
54 adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
55 }
56 limiter_.Process(float_frame);
57 }
58
NotifyAnalogLevel(int level)59 void GainController2::NotifyAnalogLevel(int level) {
60 if (analog_level_ != level && adaptive_agc_) {
61 adaptive_agc_->Reset();
62 }
63 analog_level_ = level;
64 }
65
ApplyConfig(const AudioProcessing::Config::GainController2 & config)66 void GainController2::ApplyConfig(
67 const AudioProcessing::Config::GainController2& config) {
68 RTC_DCHECK(Validate(config));
69
70 config_ = config;
71 if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) {
72 // Reset the limiter to quickly react on abrupt level changes caused by
73 // large changes of the fixed gain.
74 limiter_.Reset();
75 }
76 gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
77 if (config_.adaptive_digital.enabled) {
78 adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
79 } else {
80 adaptive_agc_.reset();
81 }
82 }
83
Validate(const AudioProcessing::Config::GainController2 & config)84 bool GainController2::Validate(
85 const AudioProcessing::Config::GainController2& config) {
86 const auto& fixed = config.fixed_digital;
87 const auto& adaptive = config.adaptive_digital;
88 return fixed.gain_db >= 0.f && fixed.gain_db < 50.f &&
89 adaptive.vad_probability_attack > 0.f &&
90 adaptive.vad_probability_attack <= 1.f &&
91 adaptive.level_estimator_adjacent_speech_frames_threshold >= 1 &&
92 adaptive.initial_saturation_margin_db >= 0.f &&
93 adaptive.initial_saturation_margin_db <= 100.f &&
94 adaptive.extra_saturation_margin_db >= 0.f &&
95 adaptive.extra_saturation_margin_db <= 100.f &&
96 adaptive.gain_applier_adjacent_speech_frames_threshold >= 1 &&
97 adaptive.max_gain_change_db_per_second > 0.f &&
98 adaptive.max_output_noise_level_dbfs <= 0.f;
99 }
100
101 } // namespace webrtc
102