|  | /* | 
|  | *  Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ | 
|  | #define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ | 
|  |  | 
|  | #include "webrtc/base/constructormagic.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class ApmDataDumper; | 
|  | class AudioBuffer; | 
|  |  | 
|  | class GainApplier { | 
|  | public: | 
|  | explicit GainApplier(ApmDataDumper* data_dumper); | 
|  | void Initialize(int sample_rate_hz); | 
|  |  | 
|  | // Applies the specified gain to the audio frame and returns the resulting | 
|  | // number of saturated sample values. | 
|  | int Process(float new_gain, AudioBuffer* audio); | 
|  |  | 
|  | private: | 
|  | ApmDataDumper* const data_dumper_; | 
|  | float old_gain_ = 1.f; | 
|  | float gain_increase_step_size_ = 0.f; | 
|  | float gain_normal_decrease_step_size_ = 0.f; | 
|  | float gain_saturated_decrease_step_size_ = 0.f; | 
|  | bool last_frame_was_saturated_; | 
|  | RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier); | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ |