 Chromium Code Reviews
 Chromium Code Reviews Issue 2090583002:
  New module for the adaptive level controlling functionality in the audio processing module  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master
    
  
    Issue 2090583002:
  New module for the adaptive level controlling functionality in the audio processing module  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master| Index: webrtc/modules/audio_processing/level_controller/gain_applier.h | 
| diff --git a/webrtc/modules/audio_processing/level_controller/gain_applier.h b/webrtc/modules/audio_processing/level_controller/gain_applier.h | 
| new file mode 100644 | 
| index 0000000000000000000000000000000000000000..9df6572de0e50f50f13fac658fdc0d8b84256b7c | 
| --- /dev/null | 
| +++ b/webrtc/modules/audio_processing/level_controller/gain_applier.h | 
| @@ -0,0 +1,38 @@ | 
| +/* | 
| + * 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" | 
| +#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" | 
| + | 
| +namespace webrtc { | 
| + | 
| +class ApmDataDumper; | 
| 
hlundin-webrtc
2016/06/27 11:21:14
Don't forward declare _and_ #include at the same t
 
peah-webrtc
2016/06/27 22:51:47
Done.
 | 
| +class AudioBuffer; | 
| + | 
| +class GainApplier { | 
| + public: | 
| + explicit GainApplier(ApmDataDumper* data_dumper); | 
| + void Initialize(int sample_rate_hz); | 
| + int Process(float new_gain, AudioBuffer* audio); | 
| + | 
| + private: | 
| + ApmDataDumper* data_dumper_; | 
| + float old_gain_ = 1.f; | 
| + float gain_change_step_size_ = 0.f; | 
| + | 
| + RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier); | 
| +}; | 
| + | 
| +} // namespace webrtc | 
| + | 
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ |