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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/app.h" | 17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" |
18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" | 18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" | 19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
20 #include "webrtc/test/rtcp_packet_parser.h" | 20 #include "webrtc/test/rtcp_packet_parser.h" |
21 | 21 |
22 using ::testing::ElementsAre; | 22 using ::testing::ElementsAre; |
23 | 23 |
24 using webrtc::rtcp::App; | 24 using webrtc::rtcp::App; |
25 using webrtc::rtcp::Bye; | 25 using webrtc::rtcp::Bye; |
26 using webrtc::rtcp::Dlrr; | 26 using webrtc::rtcp::Dlrr; |
27 using webrtc::rtcp::Fir; | |
28 using webrtc::rtcp::RawPacket; | 27 using webrtc::rtcp::RawPacket; |
29 using webrtc::rtcp::ReceiverReport; | 28 using webrtc::rtcp::ReceiverReport; |
30 using webrtc::rtcp::ReportBlock; | 29 using webrtc::rtcp::ReportBlock; |
31 using webrtc::rtcp::Rpsi; | 30 using webrtc::rtcp::Rpsi; |
32 using webrtc::rtcp::Rrtr; | 31 using webrtc::rtcp::Rrtr; |
33 using webrtc::rtcp::Sdes; | 32 using webrtc::rtcp::Sdes; |
34 using webrtc::rtcp::SenderReport; | 33 using webrtc::rtcp::SenderReport; |
35 using webrtc::rtcp::VoipMetric; | 34 using webrtc::rtcp::VoipMetric; |
36 using webrtc::rtcp::Xr; | 35 using webrtc::rtcp::Xr; |
37 using webrtc::test::RtcpPacketParser; | 36 using webrtc::test::RtcpPacketParser; |
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280 const uint16_t kNumberOfValidBytes = 10; | 279 const uint16_t kNumberOfValidBytes = 10; |
281 rpsi.WithPictureId(kPictureId); | 280 rpsi.WithPictureId(kPictureId); |
282 | 281 |
283 rtc::scoped_ptr<RawPacket> packet(rpsi.Build()); | 282 rtc::scoped_ptr<RawPacket> packet(rpsi.Build()); |
284 RtcpPacketParser parser; | 283 RtcpPacketParser parser; |
285 parser.Parse(packet->Buffer(), packet->Length()); | 284 parser.Parse(packet->Buffer(), packet->Length()); |
286 EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); | 285 EXPECT_EQ(kNumberOfValidBytes * 8, parser.rpsi()->NumberOfValidBits()); |
287 EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); | 286 EXPECT_EQ(kPictureId, parser.rpsi()->PictureId()); |
288 } | 287 } |
289 | 288 |
290 TEST(RtcpPacketTest, Fir) { | |
291 Fir fir; | |
292 fir.From(kSenderSsrc); | |
293 fir.To(kRemoteSsrc); | |
294 fir.WithCommandSeqNum(123); | |
295 | |
296 rtc::scoped_ptr<RawPacket> packet(fir.Build()); | |
297 RtcpPacketParser parser; | |
298 parser.Parse(packet->Buffer(), packet->Length()); | |
299 EXPECT_EQ(1, parser.fir()->num_packets()); | |
300 EXPECT_EQ(kSenderSsrc, parser.fir()->Ssrc()); | |
301 EXPECT_EQ(1, parser.fir_item()->num_packets()); | |
302 EXPECT_EQ(kRemoteSsrc, parser.fir_item()->Ssrc()); | |
303 EXPECT_EQ(123U, parser.fir_item()->SeqNum()); | |
304 } | |
305 | |
306 TEST(RtcpPacketTest, BuildWithTooSmallBuffer) { | 289 TEST(RtcpPacketTest, BuildWithTooSmallBuffer) { |
307 ReportBlock rb; | 290 ReportBlock rb; |
308 ReceiverReport rr; | 291 ReceiverReport rr; |
309 rr.From(kSenderSsrc); | 292 rr.From(kSenderSsrc); |
310 EXPECT_TRUE(rr.WithReportBlock(rb)); | 293 EXPECT_TRUE(rr.WithReportBlock(rb)); |
311 | 294 |
312 const size_t kRrLength = 8; | 295 const size_t kRrLength = 8; |
313 const size_t kReportBlockLength = 24; | 296 const size_t kReportBlockLength = 24; |
314 | 297 |
315 // No packet. | 298 // No packet. |
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561 EXPECT_TRUE(xr.WithDlrr(&dlrr)); | 544 EXPECT_TRUE(xr.WithDlrr(&dlrr)); |
562 EXPECT_FALSE(xr.WithDlrr(&dlrr)); | 545 EXPECT_FALSE(xr.WithDlrr(&dlrr)); |
563 | 546 |
564 VoipMetric voip_metric; | 547 VoipMetric voip_metric; |
565 for (int i = 0; i < kMaxBlocks; ++i) | 548 for (int i = 0; i < kMaxBlocks; ++i) |
566 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); | 549 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); |
567 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); | 550 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); |
568 } | 551 } |
569 | 552 |
570 } // namespace webrtc | 553 } // namespace webrtc |
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