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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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23 explicit GainApplier(ApmDataDumper* data_dumper); | 23 explicit GainApplier(ApmDataDumper* data_dumper); |
24 void Initialize(int sample_rate_hz); | 24 void Initialize(int sample_rate_hz); |
25 | 25 |
26 // Applies the specified gain to the audio frame and returns the resulting | 26 // Applies the specified gain to the audio frame and returns the resulting |
27 // number of saturated sample values. | 27 // number of saturated sample values. |
28 int Process(float new_gain, AudioBuffer* audio); | 28 int Process(float new_gain, AudioBuffer* audio); |
29 | 29 |
30 private: | 30 private: |
31 ApmDataDumper* const data_dumper_; | 31 ApmDataDumper* const data_dumper_; |
32 float old_gain_ = 1.f; | 32 float old_gain_ = 1.f; |
33 float gain_change_step_size_ = 0.f; | 33 float gain_increase_step_size_ = 0.f; |
34 | 34 float gain_normal_decrease_step_size_ = 0.f; |
| 35 float gain_saturated_decrease_step_size_ = 0.f; |
| 36 bool last_frame_was_saturated_; |
35 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier); | 37 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier); |
36 }; | 38 }; |
37 | 39 |
38 } // namespace webrtc | 40 } // namespace webrtc |
39 | 41 |
40 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ | 42 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_ |
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