| /* |
| * 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_ |