1 /* 2 * Copyright (c) 2018 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 #ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_ 12 #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_ 13 14 #include <memory> 15 16 #include "absl/types/optional.h" 17 #include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h" 18 #include "modules/audio_processing/agc2/noise_level_estimator.h" 19 #include "modules/audio_processing/agc2/saturation_protector.h" 20 #include "modules/audio_processing/agc2/speech_level_estimator.h" 21 #include "modules/audio_processing/include/audio_frame_view.h" 22 #include "modules/audio_processing/include/audio_processing.h" 23 24 namespace webrtc { 25 class ApmDataDumper; 26 27 // Gain controller that adapts and applies a variable digital gain to meet the 28 // target level, which is determined by the given configuration. 29 class AdaptiveDigitalGainController { 30 public: 31 AdaptiveDigitalGainController( 32 ApmDataDumper* apm_data_dumper, 33 const AudioProcessing::Config::GainController2::AdaptiveDigital& config, 34 int sample_rate_hz, 35 int num_channels); 36 AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete; 37 AdaptiveDigitalGainController& operator=( 38 const AdaptiveDigitalGainController&) = delete; 39 ~AdaptiveDigitalGainController(); 40 41 // Detects and handles changes of sample rate and or number of channels. 42 void Initialize(int sample_rate_hz, int num_channels); 43 44 // Analyzes `frame`, adapts the current digital gain and applies it to 45 // `frame`. 46 // TODO(bugs.webrtc.org/7494): Remove `limiter_envelope`. 47 void Process(AudioFrameView<float> frame, 48 float speech_probability, 49 float limiter_envelope); 50 51 // Handles a gain change applied to the input signal (e.g., analog gain). 52 void HandleInputGainChange(); 53 54 // Returns the most recent speech level (dBFs) if the estimator is confident. 55 // Otherwise returns absl::nullopt. 56 absl::optional<float> GetSpeechLevelDbfsIfConfident() const; 57 58 private: 59 SpeechLevelEstimator speech_level_estimator_; 60 AdaptiveDigitalGainApplier gain_controller_; 61 ApmDataDumper* const apm_data_dumper_; 62 std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_; 63 std::unique_ptr<SaturationProtector> saturation_protector_; 64 }; 65 66 } // namespace webrtc 67 68 #endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_ 69