Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(318)

Side by Side Diff: webrtc/voice_engine/test/auto_test/voe_standard_test.h

Issue 1607353002: Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 25 matching lines...) Expand all
36 #include "webrtc/voice_engine/include/voe_network.h" 36 #include "webrtc/voice_engine/include/voe_network.h"
37 #ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API 37 #ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
38 #include "webrtc/voice_engine/include/voe_video_sync.h" 38 #include "webrtc/voice_engine/include/voe_video_sync.h"
39 #endif 39 #endif
40 #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API 40 #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
41 #include "webrtc/voice_engine/include/voe_volume_control.h" 41 #include "webrtc/voice_engine/include/voe_volume_control.h"
42 #endif 42 #endif
43 43
44 #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API 44 #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
45 namespace webrtc { 45 namespace webrtc {
46 class CriticalSectionWrapper;
47 class VoENetEqStats; 46 class VoENetEqStats;
48 } 47 }
49 #endif 48 #endif
50 49
51 #if defined(WEBRTC_ANDROID) 50 #if defined(WEBRTC_ANDROID)
52 extern char mobileLogMsg[640]; 51 extern char mobileLogMsg[640];
53 #endif 52 #endif
54 53
55 DECLARE_bool(include_timing_dependent_tests); 54 DECLARE_bool(include_timing_dependent_tests);
56 55
(...skipping 140 matching lines...) Expand 10 before | Expand all | Expand 10 after
197 VoEVideoSync* voe_vsync_; 196 VoEVideoSync* voe_vsync_;
198 VoEVolumeControl* voe_volume_control_; 197 VoEVolumeControl* voe_volume_control_;
199 VoEAudioProcessing* voe_apm_; 198 VoEAudioProcessing* voe_apm_;
200 199
201 ResourceManager resource_manager_; 200 ResourceManager resource_manager_;
202 }; 201 };
203 202
204 } // namespace voetest 203 } // namespace voetest
205 204
206 #endif // WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H 205 #endif // WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698