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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ | 10 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ |
11 #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ | 11 #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ |
12 | 12 |
| 13 #include <memory> |
13 #include <string> | 14 #include <string> |
14 | 15 |
15 #include "webrtc/base/criticalsection.h" | 16 #include "webrtc/base/criticalsection.h" |
16 #include "webrtc/base/platform_thread.h" | 17 #include "webrtc/base/platform_thread.h" |
17 #include "webrtc/base/scoped_ptr.h" | 18 #include "webrtc/base/scoped_ptr.h" |
18 #include "webrtc/modules/audio_device/include/fake_audio_device.h" | 19 #include "webrtc/modules/audio_device/include/fake_audio_device.h" |
19 #include "webrtc/test/drifting_clock.h" | 20 #include "webrtc/test/drifting_clock.h" |
20 #include "webrtc/typedefs.h" | 21 #include "webrtc/typedefs.h" |
21 | 22 |
22 namespace webrtc { | 23 namespace webrtc { |
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52 static const size_t kBufferSizeBytes = 2 * kFrequencyHz; | 53 static const size_t kBufferSizeBytes = 2 * kFrequencyHz; |
53 | 54 |
54 AudioTransport* audio_callback_; | 55 AudioTransport* audio_callback_; |
55 bool capturing_; | 56 bool capturing_; |
56 int8_t captured_audio_[kBufferSizeBytes]; | 57 int8_t captured_audio_[kBufferSizeBytes]; |
57 int8_t playout_buffer_[kBufferSizeBytes]; | 58 int8_t playout_buffer_[kBufferSizeBytes]; |
58 const float speed_; | 59 const float speed_; |
59 int64_t last_playout_ms_; | 60 int64_t last_playout_ms_; |
60 | 61 |
61 DriftingClock clock_; | 62 DriftingClock clock_; |
62 rtc::scoped_ptr<EventTimerWrapper> tick_; | 63 std::unique_ptr<EventTimerWrapper> tick_; |
63 rtc::CriticalSection lock_; | 64 rtc::CriticalSection lock_; |
64 rtc::PlatformThread thread_; | 65 rtc::PlatformThread thread_; |
65 rtc::scoped_ptr<ModuleFileUtility> file_utility_; | 66 std::unique_ptr<ModuleFileUtility> file_utility_; |
66 rtc::scoped_ptr<FileWrapper> input_stream_; | 67 std::unique_ptr<FileWrapper> input_stream_; |
67 }; | 68 }; |
68 } // namespace test | 69 } // namespace test |
69 } // namespace webrtc | 70 } // namespace webrtc |
70 | 71 |
71 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ | 72 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ |
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