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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/extended_jitter_report.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::Ij; | 27 using webrtc::rtcp::ExtendedJitterReport; |
27 using webrtc::rtcp::Nack; | 28 using webrtc::rtcp::Nack; |
28 using webrtc::rtcp::Pli; | 29 using webrtc::rtcp::Pli; |
29 using webrtc::rtcp::Sdes; | 30 using webrtc::rtcp::Sdes; |
30 using webrtc::rtcp::SenderReport; | 31 using webrtc::rtcp::SenderReport; |
31 using webrtc::rtcp::Sli; | 32 using webrtc::rtcp::Sli; |
32 using webrtc::rtcp::RawPacket; | 33 using webrtc::rtcp::RawPacket; |
33 using webrtc::rtcp::ReceiverReport; | 34 using webrtc::rtcp::ReceiverReport; |
34 using webrtc::rtcp::Remb; | 35 using webrtc::rtcp::Remb; |
35 using webrtc::rtcp::ReportBlock; | 36 using webrtc::rtcp::ReportBlock; |
36 using webrtc::rtcp::Rpsi; | 37 using webrtc::rtcp::Rpsi; |
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189 const int kMaxReportBlocks = (1 << 5) - 1; | 190 const int kMaxReportBlocks = (1 << 5) - 1; |
190 ReportBlock rb; | 191 ReportBlock rb; |
191 for (int i = 0; i < kMaxReportBlocks; ++i) { | 192 for (int i = 0; i < kMaxReportBlocks; ++i) { |
192 rb.To(kRemoteSsrc + i); | 193 rb.To(kRemoteSsrc + i); |
193 EXPECT_TRUE(sr.WithReportBlock(rb)); | 194 EXPECT_TRUE(sr.WithReportBlock(rb)); |
194 } | 195 } |
195 rb.To(kRemoteSsrc + kMaxReportBlocks); | 196 rb.To(kRemoteSsrc + kMaxReportBlocks); |
196 EXPECT_FALSE(sr.WithReportBlock(rb)); | 197 EXPECT_FALSE(sr.WithReportBlock(rb)); |
197 } | 198 } |
198 | 199 |
199 TEST(RtcpPacketTest, IjNoItem) { | |
200 Ij ij; | |
201 | |
202 rtc::scoped_ptr<RawPacket> packet(ij.Build()); | |
203 RtcpPacketParser parser; | |
204 parser.Parse(packet->Buffer(), packet->Length()); | |
205 EXPECT_EQ(1, parser.ij()->num_packets()); | |
206 EXPECT_EQ(0, parser.ij_item()->num_packets()); | |
207 } | |
208 | |
209 TEST(RtcpPacketTest, IjOneItem) { | |
210 Ij ij; | |
211 EXPECT_TRUE(ij.WithJitterItem(0x11111111)); | |
212 | |
213 rtc::scoped_ptr<RawPacket> packet(ij.Build()); | |
214 RtcpPacketParser parser; | |
215 parser.Parse(packet->Buffer(), packet->Length()); | |
216 EXPECT_EQ(1, parser.ij()->num_packets()); | |
217 EXPECT_EQ(1, parser.ij_item()->num_packets()); | |
218 EXPECT_EQ(0x11111111U, parser.ij_item()->Jitter()); | |
219 } | |
220 | |
221 TEST(RtcpPacketTest, IjTwoItems) { | |
222 Ij ij; | |
223 EXPECT_TRUE(ij.WithJitterItem(0x11111111)); | |
224 EXPECT_TRUE(ij.WithJitterItem(0x22222222)); | |
225 | |
226 rtc::scoped_ptr<RawPacket> packet(ij.Build()); | |
227 RtcpPacketParser parser; | |
228 parser.Parse(packet->Buffer(), packet->Length()); | |
229 EXPECT_EQ(1, parser.ij()->num_packets()); | |
230 EXPECT_EQ(2, parser.ij_item()->num_packets()); | |
231 EXPECT_EQ(0x22222222U, parser.ij_item()->Jitter()); | |
232 } | |
233 | |
234 TEST(RtcpPacketTest, IjTooManyItems) { | |
235 Ij ij; | |
236 const int kMaxIjItems = (1 << 5) - 1; | |
237 for (int i = 0; i < kMaxIjItems; ++i) { | |
238 EXPECT_TRUE(ij.WithJitterItem(i)); | |
239 } | |
240 EXPECT_FALSE(ij.WithJitterItem(kMaxIjItems)); | |
241 } | |
242 | |
243 TEST(RtcpPacketTest, AppWithNoData) { | 200 TEST(RtcpPacketTest, AppWithNoData) { |
244 App app; | 201 App app; |
245 app.WithSubType(30); | 202 app.WithSubType(30); |
246 uint32_t name = 'n' << 24; | 203 uint32_t name = 'n' << 24; |
247 name += 'a' << 16; | 204 name += 'a' << 16; |
248 name += 'm' << 8; | 205 name += 'm' << 8; |
249 name += 'e'; | 206 name += 'e'; |
250 app.WithName(name); | 207 app.WithName(name); |
251 | 208 |
252 rtc::scoped_ptr<RawPacket> packet(app.Build()); | 209 rtc::scoped_ptr<RawPacket> packet(app.Build()); |
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1079 EXPECT_TRUE(xr.WithDlrr(&dlrr)); | 1036 EXPECT_TRUE(xr.WithDlrr(&dlrr)); |
1080 EXPECT_FALSE(xr.WithDlrr(&dlrr)); | 1037 EXPECT_FALSE(xr.WithDlrr(&dlrr)); |
1081 | 1038 |
1082 VoipMetric voip_metric; | 1039 VoipMetric voip_metric; |
1083 for (int i = 0; i < kMaxBlocks; ++i) | 1040 for (int i = 0; i < kMaxBlocks; ++i) |
1084 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); | 1041 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); |
1085 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); | 1042 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); |
1086 } | 1043 } |
1087 | 1044 |
1088 } // namespace webrtc | 1045 } // namespace webrtc |
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