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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/test/gmock.h" | |
14 #include "webrtc/test/gtest.h" | |
15 | |
16 #include "webrtc/base/rate_limiter.h" | 13 #include "webrtc/base/rate_limiter.h" |
17 #include "webrtc/common_types.h" | 14 #include "webrtc/common_types.h" |
18 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" | 15 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
20 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" | 17 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" |
| 18 #include "webrtc/test/gmock.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; |
28 | 27 |
29 namespace webrtc { | 28 namespace webrtc { |
30 | 29 |
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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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