Chromium Code Reviews| Index: webrtc/modules/audio_processing/level_controller/level_controller.h |
| diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.h b/webrtc/modules/audio_processing/level_controller/level_controller.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..89c3af2b3e23078ecdfab0a9e6d88fec960d8368 |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/level_controller/level_controller.h |
| @@ -0,0 +1,56 @@ |
| +/* |
| + * 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_LEVEL_CONTROLLER_H_ |
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_ |
| + |
| +#include <memory> |
| +#include <vector> |
| + |
| +// TODO(peah): Try to remove. |
| +#include "webrtc/base/constructormagic.h" |
| + |
| +namespace webrtc { |
| + |
| +class ApmDataDumper; |
| +class AudioBuffer; |
| +class GainApplier; |
| +class GainSelector; |
| +class NoiseLevelEstimator; |
| +class PeakLevelEstimator; |
| +class SaturatingGainEstimator; |
| +class SignalClassifier; |
| + |
| +class LevelController { |
| + public: |
| + LevelController(); |
| + ~LevelController(); |
| + |
| + void Initialize(int sample_rate_hz, size_t num_channels_); |
|
hlundin-webrtc
2016/06/27 11:21:15
No trailing underscore on parameter names.
peah-webrtc
2016/06/27 22:51:48
Done.
|
| + void Process(AudioBuffer* audio); |
| + |
| + private: |
| + std::unique_ptr<ApmDataDumper> data_dumper_; |
|
hlundin-webrtc
2016/06/27 11:21:15
Are you using unique_ptrs just to be able to forwa
peah-webrtc
2016/06/27 22:51:48
Good point! That I definitely do. I'll change that
|
| + std::unique_ptr<GainSelector> gain_selector_; |
| + std::unique_ptr<GainApplier> gain_applier_; |
| + std::unique_ptr<SignalClassifier> signal_classifier_; |
| + std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_; |
| + std::unique_ptr<PeakLevelEstimator> peak_level_estimator_; |
| + std::unique_ptr<SaturatingGainEstimator> saturating_gain_estimator_; |
| + int sample_rate_hz_ = 8000; |
| + static int instance_count_; |
| + float dc_level_[2]; |
| + |
| + RTC_DISALLOW_COPY_AND_ASSIGN(LevelController); |
| +}; |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_ |