/* * 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 #include #include "modules/audio_processing/agc2/adaptive_digital_gain_controller.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 vad_; std::unique_ptr adaptive_digital_controller_; Limiter limiter_; int calls_since_last_limiter_log_; int analog_level_; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_