| /* |
| * Copyright (c) 2017 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_GAIN_CONTROLLER2_H_ |
| #define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_ |
| |
| #include <memory> |
| #include <string> |
| |
| #include "modules/audio_processing/agc2/adaptive_agc.h" |
| #include "modules/audio_processing/agc2/cpu_features.h" |
| #include "modules/audio_processing/agc2/gain_applier.h" |
| #include "modules/audio_processing/agc2/limiter.h" |
| #include "modules/audio_processing/agc2/vad_wrapper.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| |
| namespace webrtc { |
| |
| class AudioBuffer; |
| |
| // Gain Controller 2 aims to automatically adjust levels by acting on the |
| // microphone gain and/or applying digital gain. |
| class GainController2 { |
| public: |
| GainController2(const AudioProcessing::Config::GainController2& config, |
| int sample_rate_hz, |
| int num_channels); |
| GainController2(const GainController2&) = delete; |
| GainController2& operator=(const GainController2&) = delete; |
| ~GainController2(); |
| |
| // Detects and handles changes of sample rate and/or number of channels. |
| void Initialize(int sample_rate_hz, int num_channels); |
| |
| // Sets the fixed digital gain. |
| void SetFixedGainDb(float gain_db); |
| |
| // Applies fixed and adaptive digital gains to `audio` and runs a limiter. |
| void Process(AudioBuffer* audio); |
| |
| // Handles analog level changes. |
| void NotifyAnalogLevel(int level); |
| |
| static bool Validate(const AudioProcessing::Config::GainController2& config); |
| |
| private: |
| static int instance_count_; |
| const AvailableCpuFeatures cpu_features_; |
| ApmDataDumper data_dumper_; |
| GainApplier fixed_gain_applier_; |
| std::unique_ptr<VoiceActivityDetectorWrapper> vad_; |
| std::unique_ptr<AdaptiveAgc> adaptive_digital_controller_; |
| Limiter limiter_; |
| int calls_since_last_limiter_log_; |
| int analog_level_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_ |