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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |
13 | 13 |
| 14 #include <memory> |
14 #include <vector> | 15 #include <vector> |
15 | 16 |
16 #include "webrtc/base/criticalsection.h" | 17 #include "webrtc/base/criticalsection.h" |
17 #include "webrtc/base/scoped_ptr.h" | |
18 #include "webrtc/base/thread_annotations.h" | 18 #include "webrtc/base/thread_annotations.h" |
19 #include "webrtc/common_audio/swap_queue.h" | 19 #include "webrtc/common_audio/swap_queue.h" |
20 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 20 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
21 #include "webrtc/modules/audio_processing/processing_component.h" | 21 #include "webrtc/modules/audio_processing/processing_component.h" |
22 | 22 |
23 namespace webrtc { | 23 namespace webrtc { |
24 | 24 |
25 class AudioBuffer; | 25 class AudioBuffer; |
26 | 26 |
27 class GainControlImpl : public GainControl, | 27 class GainControlImpl : public GainControl, |
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89 int analog_capture_level_ GUARDED_BY(crit_capture_); | 89 int analog_capture_level_ GUARDED_BY(crit_capture_); |
90 bool was_analog_level_set_ GUARDED_BY(crit_capture_); | 90 bool was_analog_level_set_ GUARDED_BY(crit_capture_); |
91 bool stream_is_saturated_ GUARDED_BY(crit_capture_); | 91 bool stream_is_saturated_ GUARDED_BY(crit_capture_); |
92 | 92 |
93 size_t render_queue_element_max_size_ GUARDED_BY(crit_render_) | 93 size_t render_queue_element_max_size_ GUARDED_BY(crit_render_) |
94 GUARDED_BY(crit_capture_); | 94 GUARDED_BY(crit_capture_); |
95 std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_); | 95 std::vector<int16_t> render_queue_buffer_ GUARDED_BY(crit_render_); |
96 std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_); | 96 std::vector<int16_t> capture_queue_buffer_ GUARDED_BY(crit_capture_); |
97 | 97 |
98 // Lock protection not needed. | 98 // Lock protection not needed. |
99 rtc::scoped_ptr< | 99 std::unique_ptr< |
100 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> | 100 SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>> |
101 render_signal_queue_; | 101 render_signal_queue_; |
102 }; | 102 }; |
103 } // namespace webrtc | 103 } // namespace webrtc |
104 | 104 |
105 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 105 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |
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