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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/base/checks.h" |
| 17 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" | 18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" |
17 #include "webrtc/test/rtcp_packet_parser.h" | 19 #include "webrtc/test/rtcp_packet_parser.h" |
18 | 20 |
19 using ::testing::ElementsAre; | 21 using ::testing::ElementsAre; |
20 | 22 |
21 using webrtc::rtcp::App; | 23 using webrtc::rtcp::App; |
22 using webrtc::rtcp::Bye; | 24 using webrtc::rtcp::Bye; |
23 using webrtc::rtcp::Dlrr; | 25 using webrtc::rtcp::Dlrr; |
24 using webrtc::rtcp::Empty; | 26 using webrtc::rtcp::Empty; |
25 using webrtc::rtcp::Fir; | 27 using webrtc::rtcp::Fir; |
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36 using webrtc::rtcp::Rpsi; | 38 using webrtc::rtcp::Rpsi; |
37 using webrtc::rtcp::Rrtr; | 39 using webrtc::rtcp::Rrtr; |
38 using webrtc::rtcp::SenderReport; | 40 using webrtc::rtcp::SenderReport; |
39 using webrtc::rtcp::Tmmbn; | 41 using webrtc::rtcp::Tmmbn; |
40 using webrtc::rtcp::Tmmbr; | 42 using webrtc::rtcp::Tmmbr; |
41 using webrtc::rtcp::VoipMetric; | 43 using webrtc::rtcp::VoipMetric; |
42 using webrtc::rtcp::Xr; | 44 using webrtc::rtcp::Xr; |
43 using webrtc::test::RtcpPacketParser; | 45 using webrtc::test::RtcpPacketParser; |
44 | 46 |
45 namespace webrtc { | 47 namespace webrtc { |
| 48 namespace rtcp { |
46 | 49 |
47 const uint32_t kSenderSsrc = 0x12345678; | 50 const uint32_t kSenderSsrc = 0x12345678; |
48 const uint32_t kRemoteSsrc = 0x23456789; | 51 const uint32_t kRemoteSsrc = 0x23456789; |
49 | 52 |
| 53 // Override RtcpPacket so we can test protected methods. |
| 54 class RtcpPacketHeaderTest : public ::testing::Test, |
| 55 protected RtcpPacket { |
| 56 public: |
| 57 RtcpPacketHeaderTest() { |
| 58 memset(buffer, 0, kBufferCapacity); |
| 59 } |
| 60 |
| 61 virtual ~RtcpPacketHeaderTest() {} |
| 62 |
| 63 bool Create(uint8_t* packet, |
| 64 size_t* index, |
| 65 size_t max_length, |
| 66 PacketReadyCallback* callback) const override { |
| 67 RTC_NOTREACHED(); |
| 68 return false; |
| 69 } |
| 70 |
| 71 size_t BlockLength() const { return 0; } |
| 72 |
| 73 protected: |
| 74 static const size_t kHeaderSize = 4; |
| 75 static const size_t kBufferCapacity = 40; |
| 76 uint8_t buffer[kBufferCapacity]; |
| 77 RtcpPacket::CommonHeader header; |
| 78 }; |
| 79 |
| 80 TEST_F(RtcpPacketHeaderTest, HeaderSize) { |
| 81 // Buffer needs to be able to hold the header. |
| 82 for (size_t i = 0; i < kHeaderSize; ++i) |
| 83 EXPECT_FALSE(ParseHeader(buffer, 0, i, &header)); |
| 84 } |
| 85 |
| 86 TEST_F(RtcpPacketHeaderTest, Version) { |
| 87 // Version 2 is the only allowed for now. |
| 88 for (int v = 0; v < 4; ++v) { |
| 89 buffer[0] = v << 6; |
| 90 EXPECT_EQ(v == 2, ParseHeader(buffer, 0, kHeaderSize, &header)); |
| 91 } |
| 92 } |
| 93 |
| 94 TEST_F(RtcpPacketHeaderTest, PacketSize) { |
| 95 // Set v = 2, leave p, fmt, pt as 0. |
| 96 buffer[0] = 2 << 6; |
| 97 |
| 98 const size_t kBlockSize = 3; |
| 99 ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize); |
| 100 const size_t kSizeInBytes = (kBlockSize + 1) * 4; |
| 101 |
| 102 EXPECT_FALSE(ParseHeader(buffer, 0, kSizeInBytes - 1, &header)); |
| 103 EXPECT_TRUE(ParseHeader(buffer, 0, kSizeInBytes, &header)); |
| 104 } |
| 105 |
| 106 TEST_F(RtcpPacketHeaderTest, PayloadSize) { |
| 107 // Set v = 2, p = 1, but leave fmt, pt as 0. |
| 108 buffer[0] = (2 << 6) | (1 << 5); |
| 109 |
| 110 const size_t kBlockSize = 3; |
| 111 ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize); |
| 112 const size_t kSizeInBytes = (kBlockSize + 1) * 4; |
| 113 const size_t kPayloadSize = kSizeInBytes - kHeaderSize; |
| 114 |
| 115 // Padding one byte larger than possible. |
| 116 buffer[kSizeInBytes - 1] = kPayloadSize + 1; |
| 117 EXPECT_FALSE(ParseHeader(buffer, 0, kSizeInBytes, &header)); |
| 118 |
| 119 // Pure padding packet? |
| 120 buffer[kSizeInBytes - 1] = kPayloadSize; |
| 121 EXPECT_TRUE(ParseHeader(buffer, 0, kSizeInBytes, &header)); |
| 122 EXPECT_EQ(0u, header.payload_size_bytes); |
| 123 |
| 124 // Single byte of actual data. |
| 125 buffer[kSizeInBytes - 1] = kPayloadSize - 1; |
| 126 EXPECT_TRUE(ParseHeader(buffer, 0, kSizeInBytes, &header)); |
| 127 EXPECT_EQ(1u, header.payload_size_bytes); |
| 128 |
| 129 } |
| 130 |
| 131 TEST_F(RtcpPacketHeaderTest, FormatAndPayloadType) { |
| 132 |
| 133 } |
| 134 |
50 TEST(RtcpPacketTest, Rr) { | 135 TEST(RtcpPacketTest, Rr) { |
51 ReceiverReport rr; | 136 ReceiverReport rr; |
52 rr.From(kSenderSsrc); | 137 rr.From(kSenderSsrc); |
53 | 138 |
54 rtc::scoped_ptr<RawPacket> packet(rr.Build()); | 139 rtc::scoped_ptr<RawPacket> packet(rr.Build()); |
55 RtcpPacketParser parser; | 140 RtcpPacketParser parser; |
56 parser.Parse(packet->Buffer(), packet->Length()); | 141 parser.Parse(packet->Buffer(), packet->Length()); |
57 EXPECT_EQ(1, parser.receiver_report()->num_packets()); | 142 EXPECT_EQ(1, parser.receiver_report()->num_packets()); |
58 EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); | 143 EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); |
59 EXPECT_EQ(0, parser.report_block()->num_packets()); | 144 EXPECT_EQ(0, parser.report_block()->num_packets()); |
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1078 for (int i = 0; i < kMaxBlocks; ++i) | 1163 for (int i = 0; i < kMaxBlocks; ++i) |
1079 EXPECT_TRUE(xr.WithDlrr(&dlrr)); | 1164 EXPECT_TRUE(xr.WithDlrr(&dlrr)); |
1080 EXPECT_FALSE(xr.WithDlrr(&dlrr)); | 1165 EXPECT_FALSE(xr.WithDlrr(&dlrr)); |
1081 | 1166 |
1082 VoipMetric voip_metric; | 1167 VoipMetric voip_metric; |
1083 for (int i = 0; i < kMaxBlocks; ++i) | 1168 for (int i = 0; i < kMaxBlocks; ++i) |
1084 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); | 1169 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); |
1085 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); | 1170 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); |
1086 } | 1171 } |
1087 | 1172 |
| 1173 } // namespace rtcp |
1088 } // namespace webrtc | 1174 } // namespace webrtc |
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