| /* |
| * 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_DIGITAL_GAIN_CONTROLLER_H_ |
| #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_ |
| |
| #include <memory> |
| |
| #include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h" |
| #include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h" |
| #include "modules/audio_processing/agc2/noise_level_estimator.h" |
| #include "modules/audio_processing/agc2/saturation_protector.h" |
| #include "modules/audio_processing/include/audio_frame_view.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| |
| namespace webrtc { |
| class ApmDataDumper; |
| |
| // Gain controller that adapts and applies a variable digital gain to meet the |
| // target level, which is determined by the given configuration. |
| class AdaptiveDigitalGainController { |
| public: |
| AdaptiveDigitalGainController( |
| ApmDataDumper* apm_data_dumper, |
| const AudioProcessing::Config::GainController2::AdaptiveDigital& config, |
| int sample_rate_hz, |
| int num_channels); |
| AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete; |
| AdaptiveDigitalGainController& operator=( |
| const AdaptiveDigitalGainController&) = delete; |
| ~AdaptiveDigitalGainController(); |
| |
| // Detects and handles changes of sample rate and or number of channels. |
| void Initialize(int sample_rate_hz, int num_channels); |
| |
| // Analyzes `frame`, adapts the current digital gain and applies it to |
| // `frame`. |
| // TODO(bugs.webrtc.org/7494): Remove `limiter_envelope`. |
| void Process(AudioFrameView<float> frame, |
| float speech_probability, |
| float limiter_envelope); |
| |
| // Handles a gain change applied to the input signal (e.g., analog gain). |
| void HandleInputGainChange(); |
| |
| private: |
| AdaptiveModeLevelEstimator speech_level_estimator_; |
| AdaptiveDigitalGainApplier gain_controller_; |
| ApmDataDumper* const apm_data_dumper_; |
| std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_; |
| std::unique_ptr<SaturationProtector> saturation_protector_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_ |