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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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27 | 27 |
28 // This class implements an AudioCaptureModule that can be used to detect if | 28 // This class implements an AudioCaptureModule that can be used to detect if |
29 // audio is being received properly if it is fed by another AudioCaptureModule | 29 // audio is being received properly if it is fed by another AudioCaptureModule |
30 // in some arbitrary audio pipeline where they are connected. It does not play | 30 // in some arbitrary audio pipeline where they are connected. It does not play |
31 // out or record any audio so it does not need access to any hardware and can | 31 // out or record any audio so it does not need access to any hardware and can |
32 // therefore be used in the gtest testing framework. | 32 // therefore be used in the gtest testing framework. |
33 | 33 |
34 // Note P postfix of a function indicates that it should only be called by the | 34 // Note P postfix of a function indicates that it should only be called by the |
35 // processing thread. | 35 // processing thread. |
36 | 36 |
37 #ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ | 37 #ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ |
38 #define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ | 38 #define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ |
39 | 39 |
40 #include "webrtc/base/basictypes.h" | 40 #include "webrtc/base/basictypes.h" |
41 #include "webrtc/base/criticalsection.h" | 41 #include "webrtc/base/criticalsection.h" |
42 #include "webrtc/base/messagehandler.h" | 42 #include "webrtc/base/messagehandler.h" |
43 #include "webrtc/base/scoped_ptr.h" | 43 #include "webrtc/base/scoped_ptr.h" |
44 #include "webrtc/base/scoped_ref_ptr.h" | 44 #include "webrtc/base/scoped_ref_ptr.h" |
45 #include "webrtc/common_types.h" | 45 #include "webrtc/common_types.h" |
46 #include "webrtc/modules/audio_device/include/audio_device.h" | 46 #include "webrtc/modules/audio_device/include/audio_device.h" |
47 | 47 |
48 namespace rtc { | 48 namespace rtc { |
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277 int frames_received_; | 277 int frames_received_; |
278 | 278 |
279 // Protects variables that are accessed from process_thread_ and | 279 // Protects variables that are accessed from process_thread_ and |
280 // the main thread. | 280 // the main thread. |
281 rtc::CriticalSection crit_; | 281 rtc::CriticalSection crit_; |
282 // Protects |audio_callback_| that is accessed from process_thread_ and | 282 // Protects |audio_callback_| that is accessed from process_thread_ and |
283 // the main thread. | 283 // the main thread. |
284 rtc::CriticalSection crit_callback_; | 284 rtc::CriticalSection crit_callback_; |
285 }; | 285 }; |
286 | 286 |
287 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ | 287 #endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ |
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