Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(474)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc

Issue 2400993002: Fix chromium-style warnings. (Closed)
Patch Set: 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 <memory> 11 #include <memory>
12 12
13 #include "webrtc/base/rate_limiter.h" 13 #include "webrtc/base/rate_limiter.h"
14 #include "webrtc/common_types.h" 14 #include "webrtc/common_types.h"
15 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" 15 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_defines_nullimpl.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" 18 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
18 #include "webrtc/test/gmock.h" 19 #include "webrtc/test/gmock.h"
19 #include "webrtc/test/gtest.h" 20 #include "webrtc/test/gtest.h"
20 #include "webrtc/test/mock_transport.h" 21 #include "webrtc/test/mock_transport.h"
21 #include "webrtc/test/rtcp_packet_parser.h" 22 #include "webrtc/test/rtcp_packet_parser.h"
22 23
23 using ::testing::_; 24 using ::testing::_;
24 using ::testing::ElementsAre; 25 using ::testing::ElementsAre;
25 using ::testing::Invoke; 26 using ::testing::Invoke;
26 using webrtc::RTCPUtility::RtcpCommonHeader; 27 using webrtc::RTCPUtility::RtcpCommonHeader;
(...skipping 788 matching lines...) Expand 10 before | Expand all | Expand 10 after
815 rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds()); 816 rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
816 817
817 // Set up XR VoIP metric to be included with BYE 818 // Set up XR VoIP metric to be included with BYE
818 rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound); 819 rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
819 RTCPVoIPMetric metric; 820 RTCPVoIPMetric metric;
820 EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric)); 821 EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric));
821 EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye)); 822 EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
822 } 823 }
823 824
824 } // namespace webrtc 825 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698