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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 <memory> |
| 15 #include <vector> | 15 #include <vector> |
| 16 | 16 |
| 17 #include "webrtc/base/constructormagic.h" |
| 18 #include "webrtc/base/criticalsection.h" |
| 19 #include "webrtc/base/swap_queue.h" |
| 20 #include "webrtc/base/thread_annotations.h" |
| 17 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 21 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 18 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h" | 22 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h" |
| 19 #include "webrtc/rtc_base/constructormagic.h" | |
| 20 #include "webrtc/rtc_base/criticalsection.h" | |
| 21 #include "webrtc/rtc_base/swap_queue.h" | |
| 22 #include "webrtc/rtc_base/thread_annotations.h" | |
| 23 | 23 |
| 24 namespace webrtc { | 24 namespace webrtc { |
| 25 | 25 |
| 26 class ApmDataDumper; | 26 class ApmDataDumper; |
| 27 class AudioBuffer; | 27 class AudioBuffer; |
| 28 | 28 |
| 29 class GainControlImpl : public GainControl { | 29 class GainControlImpl : public GainControl { |
| 30 public: | 30 public: |
| 31 GainControlImpl(rtc::CriticalSection* crit_render, | 31 GainControlImpl(rtc::CriticalSection* crit_render, |
| 32 rtc::CriticalSection* crit_capture); | 32 rtc::CriticalSection* crit_capture); |
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| 88 | 88 |
| 89 rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_); | 89 rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_); |
| 90 rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_); | 90 rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_); |
| 91 | 91 |
| 92 static int instance_counter_; | 92 static int instance_counter_; |
| 93 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl); | 93 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl); |
| 94 }; | 94 }; |
| 95 } // namespace webrtc | 95 } // namespace webrtc |
| 96 | 96 |
| 97 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 97 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |
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