Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(424)

Side by Side Diff: webrtc/call/bitrate_estimator_tests.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
« no previous file with comments | « webrtc/call/bitrate_allocator_unittest.cc ('k') | webrtc/call/call_perf_tests.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 #include <functional> 10 #include <functional>
11 #include <list> 11 #include <list>
12 #include <memory> 12 #include <memory>
13 #include <string> 13 #include <string>
14 14
15 #include "testing/gtest/include/gtest/gtest.h" 15 #include "webrtc/test/gtest.h"
16 16
17 #include "webrtc/api/call/audio_state.h" 17 #include "webrtc/api/call/audio_state.h"
18 #include "webrtc/base/checks.h" 18 #include "webrtc/base/checks.h"
19 #include "webrtc/base/event.h" 19 #include "webrtc/base/event.h"
20 #include "webrtc/base/logging.h" 20 #include "webrtc/base/logging.h"
21 #include "webrtc/base/thread_annotations.h" 21 #include "webrtc/base/thread_annotations.h"
22 #include "webrtc/call.h" 22 #include "webrtc/call.h"
23 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 23 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
24 #include "webrtc/system_wrappers/include/trace.h" 24 #include "webrtc/system_wrappers/include/trace.h"
25 #include "webrtc/test/call_test.h" 25 #include "webrtc/test/call_test.h"
(...skipping 298 matching lines...) Expand 10 before | Expand all | Expand 10 after
324 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); 324 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
325 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); 325 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
326 receiver_log_.PushExpectedLogLine( 326 receiver_log_.PushExpectedLogLine(
327 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); 327 "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
328 streams_.push_back(new Stream(this, false)); 328 streams_.push_back(new Stream(this, false));
329 streams_[0]->StopSending(); 329 streams_[0]->StopSending();
330 streams_[1]->StopSending(); 330 streams_[1]->StopSending();
331 EXPECT_TRUE(receiver_log_.Wait()); 331 EXPECT_TRUE(receiver_log_.Wait());
332 } 332 }
333 } // namespace webrtc 333 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/call/bitrate_allocator_unittest.cc ('k') | webrtc/call/call_perf_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698