|  | /* | 
|  | *  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_AGC2_GAIN_CONTROLLER2_H_ | 
|  | #define MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_ | 
|  |  | 
|  | #include <memory> | 
|  | #include <string> | 
|  |  | 
|  | #include "modules/audio_processing/include/audio_processing.h" | 
|  | #include "rtc_base/constructormagic.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class ApmDataDumper; | 
|  | class AudioBuffer; | 
|  |  | 
|  | // Gain Controller 2 aims to automatically adjust levels by acting on the | 
|  | // microphone gain and/or applying digital gain. | 
|  | // | 
|  | // Temporarily implements a fixed gain mode with hard-clipping. | 
|  | class GainController2 { | 
|  | public: | 
|  | GainController2(); | 
|  | ~GainController2(); | 
|  |  | 
|  | void Initialize(int sample_rate_hz); | 
|  | void Process(AudioBuffer* audio); | 
|  |  | 
|  | void ApplyConfig(const AudioProcessing::Config::GainController2& config); | 
|  | static bool Validate(const AudioProcessing::Config::GainController2& config); | 
|  | static std::string ToString( | 
|  | const AudioProcessing::Config::GainController2& config); | 
|  |  | 
|  | private: | 
|  | static int instance_count_; | 
|  | std::unique_ptr<ApmDataDumper> data_dumper_; | 
|  | int sample_rate_hz_; | 
|  | float fixed_gain_; | 
|  |  | 
|  | RTC_DISALLOW_COPY_AND_ASSIGN(GainController2); | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_ |