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

Issue 1430013003: rtcp::Bye packet moved to own file and got a Parse function (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Merged with 'master' Created 5 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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 * This file includes unit tests for the RtcpPacket. 10 * This file includes unit tests for the RtcpPacket.
11 */ 11 */
12 12
13 #include "testing/gmock/include/gmock/gmock.h" 13 #include "testing/gmock/include/gmock/gmock.h"
14 #include "testing/gtest/include/gtest/gtest.h" 14 #include "testing/gtest/include/gtest/gtest.h"
15 15
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
17 #include "webrtc/test/rtcp_packet_parser.h" 18 #include "webrtc/test/rtcp_packet_parser.h"
18 19
19 using ::testing::ElementsAre; 20 using ::testing::ElementsAre;
20 21
21 using webrtc::rtcp::App; 22 using webrtc::rtcp::App;
22 using webrtc::rtcp::Bye; 23 using webrtc::rtcp::Bye;
23 using webrtc::rtcp::Dlrr; 24 using webrtc::rtcp::Dlrr;
24 using webrtc::rtcp::Empty; 25 using webrtc::rtcp::Empty;
25 using webrtc::rtcp::Fir; 26 using webrtc::rtcp::Fir;
26 using webrtc::rtcp::Nack; 27 using webrtc::rtcp::Nack;
(...skipping 1007 matching lines...) Expand 10 before | Expand all | Expand 10 after
1034 EXPECT_TRUE(xr.WithDlrr(&dlrr)); 1035 EXPECT_TRUE(xr.WithDlrr(&dlrr));
1035 EXPECT_FALSE(xr.WithDlrr(&dlrr)); 1036 EXPECT_FALSE(xr.WithDlrr(&dlrr));
1036 1037
1037 VoipMetric voip_metric; 1038 VoipMetric voip_metric;
1038 for (int i = 0; i < kMaxBlocks; ++i) 1039 for (int i = 0; i < kMaxBlocks; ++i)
1039 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); 1040 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric));
1040 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); 1041 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric));
1041 } 1042 }
1042 1043
1043 } // namespace webrtc 1044 } // namespace webrtc
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