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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; |
(...skipping 10 matching lines...) Expand all Loading... | |
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, protected RtcpPacket { | |
55 public: | |
56 RtcpPacketHeaderTest() { memset(buffer, 0, kBufferCapacity); } | |
57 | |
58 virtual ~RtcpPacketHeaderTest() {} | |
59 | |
60 bool Create(uint8_t* packet, | |
61 size_t* index, | |
62 size_t max_length, | |
63 PacketReadyCallback* callback) const override { | |
64 RTC_NOTREACHED(); | |
65 return false; | |
66 } | |
67 | |
68 size_t BlockLength() const { return 0; } | |
69 | |
70 protected: | |
71 static const size_t kHeaderSize = 4; | |
72 static const size_t kBufferCapacity = 40; | |
åsapersson
2015/09/08 10:20:41
kHeaderSize -> kHeaderSizeInBytes
kBufferCapacity
sprang_webrtc
2015/09/09 09:18:53
Done.
| |
73 uint8_t buffer[kBufferCapacity]; | |
74 RTCPUtility::RtcpCommonHeader header; | |
75 }; | |
76 | |
åsapersson
2015/09/08 10:20:41
I think these parsing tests should be where the pa
sprang_webrtc
2015/09/09 09:18:53
Fair enough. My long (long) term hope is to rewrit
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77 TEST_F(RtcpPacketHeaderTest, HeaderSize) { | |
åsapersson
2015/09/08 10:20:41
Maybe rename to something like TooShortHeaderSize.
sprang_webrtc
2015/09/09 09:18:54
Renamed to TooSmallBuffer.
| |
78 // Buffer needs to be able to hold the header. | |
79 for (size_t i = 0; i < kHeaderSize; ++i) | |
80 EXPECT_FALSE(RtcpParseCommonHeader(buffer, 0, i, &header)); | |
81 } | |
82 | |
83 TEST_F(RtcpPacketHeaderTest, Version) { | |
84 // Version 2 is the only allowed for now. | |
85 for (int v = 0; v < 4; ++v) { | |
86 buffer[0] = v << 6; | |
87 EXPECT_EQ(v == 2, RtcpParseCommonHeader(buffer, 0, kHeaderSize, &header)); | |
88 } | |
89 } | |
90 | |
91 TEST_F(RtcpPacketHeaderTest, PacketSize) { | |
92 // Set v = 2, leave p, fmt, pt as 0. | |
93 buffer[0] = 2 << 6; | |
94 | |
95 const size_t kBlockSize = 3; | |
96 ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize); | |
97 const size_t kSizeInBytes = (kBlockSize + 1) * 4; | |
98 | |
99 EXPECT_FALSE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes - 1, &header)); | |
100 EXPECT_TRUE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); | |
101 } | |
102 | |
103 TEST_F(RtcpPacketHeaderTest, PayloadSize) { | |
104 // Set v = 2, p = 1, but leave fmt, pt as 0. | |
105 buffer[0] = (2 << 6) | (1 << 5); | |
106 | |
107 const size_t kBlockSize = 3; | |
108 ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize); | |
109 const size_t kSizeInBytes = (kBlockSize + 1) * 4; | |
110 const size_t kPayloadSize = kSizeInBytes - kHeaderSize; | |
åsapersson
2015/09/08 10:20:41
kPayloadSize -> kPayloadSizeInBytes
sprang_webrtc
2015/09/09 09:18:53
Done.
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111 | |
112 // Padding one byte larger than possible. | |
113 buffer[kSizeInBytes - 1] = kPayloadSize + 1; | |
114 EXPECT_FALSE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); | |
115 | |
116 // Pure padding packet? | |
117 buffer[kSizeInBytes - 1] = kPayloadSize; | |
118 EXPECT_TRUE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); | |
119 EXPECT_EQ(0u, header.payload_size_bytes); | |
120 | |
121 // Single byte of actual data. | |
122 buffer[kSizeInBytes - 1] = kPayloadSize - 1; | |
123 EXPECT_TRUE(RtcpParseCommonHeader(buffer, 0, kSizeInBytes, &header)); | |
124 EXPECT_EQ(1u, header.payload_size_bytes); | |
125 } | |
126 | |
127 TEST_F(RtcpPacketHeaderTest, FormatAndPayloadType) {} | |
åsapersson
2015/09/08 10:20:41
remove
sprang_webrtc
2015/09/09 09:18:53
Or implement :)
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128 | |
50 TEST(RtcpPacketTest, Rr) { | 129 TEST(RtcpPacketTest, Rr) { |
51 ReceiverReport rr; | 130 ReceiverReport rr; |
52 rr.From(kSenderSsrc); | 131 rr.From(kSenderSsrc); |
53 | 132 |
54 rtc::scoped_ptr<RawPacket> packet(rr.Build()); | 133 rtc::scoped_ptr<RawPacket> packet(rr.Build()); |
55 RtcpPacketParser parser; | 134 RtcpPacketParser parser; |
56 parser.Parse(packet->Buffer(), packet->Length()); | 135 parser.Parse(packet->Buffer(), packet->Length()); |
57 EXPECT_EQ(1, parser.receiver_report()->num_packets()); | 136 EXPECT_EQ(1, parser.receiver_report()->num_packets()); |
58 EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); | 137 EXPECT_EQ(kSenderSsrc, parser.receiver_report()->Ssrc()); |
59 EXPECT_EQ(0, parser.report_block()->num_packets()); | 138 EXPECT_EQ(0, parser.report_block()->num_packets()); |
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1078 for (int i = 0; i < kMaxBlocks; ++i) | 1157 for (int i = 0; i < kMaxBlocks; ++i) |
1079 EXPECT_TRUE(xr.WithDlrr(&dlrr)); | 1158 EXPECT_TRUE(xr.WithDlrr(&dlrr)); |
1080 EXPECT_FALSE(xr.WithDlrr(&dlrr)); | 1159 EXPECT_FALSE(xr.WithDlrr(&dlrr)); |
1081 | 1160 |
1082 VoipMetric voip_metric; | 1161 VoipMetric voip_metric; |
1083 for (int i = 0; i < kMaxBlocks; ++i) | 1162 for (int i = 0; i < kMaxBlocks; ++i) |
1084 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); | 1163 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); |
1085 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); | 1164 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); |
1086 } | 1165 } |
1087 | 1166 |
1167 } // namespace rtcp | |
1088 } // namespace webrtc | 1168 } // namespace webrtc |
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