Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(7)

Side by Side Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

Issue 2358993004: Enable the -Wundef warning for clang (Closed)
Patch Set: rebase Created 4 years, 2 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
11 #include <algorithm> 11 #include <algorithm>
12 #include <memory> 12 #include <memory>
13 #include <vector> 13 #include <vector>
14 #include "testing/gtest/include/gtest/gtest.h" 14 #include "webrtc/test/gtest.h"
15 15
16 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" 16 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
17 17
18 #include "webrtc/base/rate_limiter.h" 18 #include "webrtc/base/rate_limiter.h"
19 #include "webrtc/common_types.h" 19 #include "webrtc/common_types.h"
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
22 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" 22 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
23 23
24 namespace webrtc { 24 namespace webrtc {
(...skipping 249 matching lines...) Expand 10 before | Expand all | Expand 10 after
274 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 274 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
275 timeStamp, -1, test, 4, nullptr, 275 timeStamp, -1, test, 4, nullptr,
276 nullptr, nullptr)); 276 nullptr, nullptr));
277 fake_clock.AdvanceTimeMilliseconds(20); 277 fake_clock.AdvanceTimeMilliseconds(20);
278 module1->Process(); 278 module1->Process();
279 } 279 }
280 } 280 }
281 281
282 } // namespace 282 } // namespace
283 } // namespace webrtc 283 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/test/testAPI/test_api.h ('k') | webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698