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

Issue 2449523002: Revert of Fix chromium-style warnings. (Closed)
Patch Set: Created 4 years, 1 month ago
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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"
18 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
19 #include "webrtc/test/gmock.h" 18 #include "webrtc/test/gmock.h"
20 #include "webrtc/test/gtest.h" 19 #include "webrtc/test/gtest.h"
21 #include "webrtc/test/mock_transport.h" 20 #include "webrtc/test/mock_transport.h"
22 #include "webrtc/test/rtcp_packet_parser.h" 21 #include "webrtc/test/rtcp_packet_parser.h"
23 22
24 using ::testing::_; 23 using ::testing::_;
25 using ::testing::ElementsAre; 24 using ::testing::ElementsAre;
26 using ::testing::Invoke; 25 using ::testing::Invoke;
27 using webrtc::RTCPUtility::RtcpCommonHeader; 26 using webrtc::RTCPUtility::RtcpCommonHeader;
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816 rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds()); 815 rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
817 816
818 // Set up XR VoIP metric to be included with BYE 817 // Set up XR VoIP metric to be included with BYE
819 rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound); 818 rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
820 RTCPVoIPMetric metric; 819 RTCPVoIPMetric metric;
821 EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric)); 820 EXPECT_EQ(0, rtcp_sender_->SetRTCPVoIPMetrics(&metric));
822 EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye)); 821 EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpBye));
823 } 822 }
824 823
825 } // namespace webrtc 824 } // namespace webrtc
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