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

Issue 1575023002: [rtp_rtcp] rtcp::Tmmbr moved into own file (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months 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.
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28 using webrtc::rtcp::Fir; 28 using webrtc::rtcp::Fir;
29 using webrtc::rtcp::RawPacket; 29 using webrtc::rtcp::RawPacket;
30 using webrtc::rtcp::ReceiverReport; 30 using webrtc::rtcp::ReceiverReport;
31 using webrtc::rtcp::Remb; 31 using webrtc::rtcp::Remb;
32 using webrtc::rtcp::ReportBlock; 32 using webrtc::rtcp::ReportBlock;
33 using webrtc::rtcp::Rpsi; 33 using webrtc::rtcp::Rpsi;
34 using webrtc::rtcp::Rrtr; 34 using webrtc::rtcp::Rrtr;
35 using webrtc::rtcp::Sdes; 35 using webrtc::rtcp::Sdes;
36 using webrtc::rtcp::SenderReport; 36 using webrtc::rtcp::SenderReport;
37 using webrtc::rtcp::Sli; 37 using webrtc::rtcp::Sli;
38 using webrtc::rtcp::Tmmbr;
39 using webrtc::rtcp::VoipMetric; 38 using webrtc::rtcp::VoipMetric;
40 using webrtc::rtcp::Xr; 39 using webrtc::rtcp::Xr;
41 using webrtc::test::RtcpPacketParser; 40 using webrtc::test::RtcpPacketParser;
42 41
43 namespace webrtc { 42 namespace webrtc {
44 43
45 const uint32_t kSenderSsrc = 0x12345678; 44 const uint32_t kSenderSsrc = 0x12345678;
46 const uint32_t kRemoteSsrc = 0x23456789; 45 const uint32_t kRemoteSsrc = 0x23456789;
47 46
48 TEST(RtcpPacketTest, Sr) { 47 TEST(RtcpPacketTest, Sr) {
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489 EXPECT_EQ(1, parser.psfb_app()->num_packets()); 488 EXPECT_EQ(1, parser.psfb_app()->num_packets());
490 EXPECT_EQ(kSenderSsrc, parser.psfb_app()->Ssrc()); 489 EXPECT_EQ(kSenderSsrc, parser.psfb_app()->Ssrc());
491 EXPECT_EQ(1, parser.remb_item()->num_packets()); 490 EXPECT_EQ(1, parser.remb_item()->num_packets());
492 EXPECT_EQ(261011, parser.remb_item()->last_bitrate_bps()); 491 EXPECT_EQ(261011, parser.remb_item()->last_bitrate_bps());
493 std::vector<uint32_t> ssrcs = parser.remb_item()->last_ssrc_list(); 492 std::vector<uint32_t> ssrcs = parser.remb_item()->last_ssrc_list();
494 EXPECT_EQ(kRemoteSsrc, ssrcs[0]); 493 EXPECT_EQ(kRemoteSsrc, ssrcs[0]);
495 EXPECT_EQ(kRemoteSsrc + 1, ssrcs[1]); 494 EXPECT_EQ(kRemoteSsrc + 1, ssrcs[1]);
496 EXPECT_EQ(kRemoteSsrc + 2, ssrcs[2]); 495 EXPECT_EQ(kRemoteSsrc + 2, ssrcs[2]);
497 } 496 }
498 497
499 TEST(RtcpPacketTest, Tmmbr) {
500 Tmmbr tmmbr;
501 tmmbr.From(kSenderSsrc);
502 tmmbr.To(kRemoteSsrc);
503 tmmbr.WithBitrateKbps(312);
504 tmmbr.WithOverhead(60);
505
506 rtc::scoped_ptr<RawPacket> packet(tmmbr.Build());
507 RtcpPacketParser parser;
508 parser.Parse(packet->Buffer(), packet->Length());
509 EXPECT_EQ(1, parser.tmmbr()->num_packets());
510 EXPECT_EQ(kSenderSsrc, parser.tmmbr()->Ssrc());
511 EXPECT_EQ(1, parser.tmmbr_item()->num_packets());
512 EXPECT_EQ(312U, parser.tmmbr_item()->BitrateKbps());
513 EXPECT_EQ(60U, parser.tmmbr_item()->Overhead());
514 }
515
516 TEST(RtcpPacketTest, XrWithNoReportBlocks) { 498 TEST(RtcpPacketTest, XrWithNoReportBlocks) {
517 Xr xr; 499 Xr xr;
518 xr.From(kSenderSsrc); 500 xr.From(kSenderSsrc);
519 501
520 rtc::scoped_ptr<RawPacket> packet(xr.Build()); 502 rtc::scoped_ptr<RawPacket> packet(xr.Build());
521 RtcpPacketParser parser; 503 RtcpPacketParser parser;
522 parser.Parse(packet->Buffer(), packet->Length()); 504 parser.Parse(packet->Buffer(), packet->Length());
523 EXPECT_EQ(1, parser.xr_header()->num_packets()); 505 EXPECT_EQ(1, parser.xr_header()->num_packets());
524 EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc()); 506 EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc());
525 } 507 }
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750 EXPECT_TRUE(xr.WithDlrr(&dlrr)); 732 EXPECT_TRUE(xr.WithDlrr(&dlrr));
751 EXPECT_FALSE(xr.WithDlrr(&dlrr)); 733 EXPECT_FALSE(xr.WithDlrr(&dlrr));
752 734
753 VoipMetric voip_metric; 735 VoipMetric voip_metric;
754 for (int i = 0; i < kMaxBlocks; ++i) 736 for (int i = 0; i < kMaxBlocks; ++i)
755 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); 737 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric));
756 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); 738 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric));
757 } 739 }
758 740
759 } // namespace webrtc 741 } // namespace webrtc
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