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| 1 /* |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 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 |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #include <cstdio> |
| 12 |
| 13 #include "webrtc/call/rtp_rtcp_demuxer_helper.h" |
| 14 |
| 15 #include "webrtc/base/arraysize.h" |
| 16 #include "webrtc/base/basictypes.h" |
| 17 #include "webrtc/base/buffer.h" |
| 18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| 19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| 20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| 21 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h" |
| 22 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" |
| 23 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| 25 #include "webrtc/test/gtest.h" |
| 26 |
| 27 namespace webrtc { |
| 28 |
| 29 namespace { |
| 30 constexpr uint32_t kSsrc = 8374; |
| 31 } // namespace |
| 32 |
| 33 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ByePacket) { |
| 34 webrtc::rtcp::Bye rtcp_packet; |
| 35 rtcp_packet.SetSenderSsrc(kSsrc); |
| 36 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 37 |
| 38 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 39 EXPECT_EQ(ssrc, kSsrc); |
| 40 } |
| 41 |
| 42 TEST(RtpRtcpDemuxerHelperTest, |
| 43 ParseRtcpPacketSenderSsrc_ExtendedReportsPacket) { |
| 44 webrtc::rtcp::ExtendedReports rtcp_packet; |
| 45 rtcp_packet.SetSenderSsrc(kSsrc); |
| 46 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 47 |
| 48 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 49 EXPECT_EQ(ssrc, kSsrc); |
| 50 } |
| 51 |
| 52 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_PsfbPacket) { |
| 53 webrtc::rtcp::Pli rtcp_packet; // Psfb is abstract; use a subclass. |
| 54 rtcp_packet.SetSenderSsrc(kSsrc); |
| 55 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 56 |
| 57 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 58 EXPECT_EQ(ssrc, kSsrc); |
| 59 } |
| 60 |
| 61 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ReceiverReportPacket) { |
| 62 webrtc::rtcp::ReceiverReport rtcp_packet; |
| 63 rtcp_packet.SetSenderSsrc(kSsrc); |
| 64 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 65 |
| 66 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 67 EXPECT_EQ(ssrc, kSsrc); |
| 68 } |
| 69 |
| 70 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_RtpfbPacket) { |
| 71 // Rtpfb is abstract; use a subclass. |
| 72 webrtc::rtcp::RapidResyncRequest rtcp_packet; |
| 73 rtcp_packet.SetSenderSsrc(kSsrc); |
| 74 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 75 |
| 76 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 77 EXPECT_EQ(ssrc, kSsrc); |
| 78 } |
| 79 |
| 80 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_SenderReportPacket) { |
| 81 webrtc::rtcp::SenderReport rtcp_packet; |
| 82 rtcp_packet.SetSenderSsrc(kSsrc); |
| 83 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 84 |
| 85 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 86 EXPECT_EQ(ssrc, kSsrc); |
| 87 } |
| 88 |
| 89 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_MalformedRtcpPacket) { |
| 90 uint8_t garbage[100]; |
| 91 memset(&garbage[0], 0, arraysize(garbage)); |
| 92 |
| 93 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(garbage); |
| 94 EXPECT_FALSE(ssrc); |
| 95 } |
| 96 |
| 97 TEST(RtpRtcpDemuxerHelperTest, |
| 98 ParseRtcpPacketSenderSsrc_RtcpMessageWithoutSenderSsrc) { |
| 99 webrtc::rtcp::ExtendedJitterReport rtcp_packet; // Has no sender SSRC. |
| 100 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 101 |
| 102 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| 103 EXPECT_FALSE(ssrc); |
| 104 } |
| 105 |
| 106 TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_TruncatedRtcpMessage) { |
| 107 webrtc::rtcp::Bye rtcp_packet; |
| 108 rtcp_packet.SetSenderSsrc(kSsrc); |
| 109 rtc::Buffer raw_packet = rtcp_packet.Build(); |
| 110 |
| 111 constexpr size_t rtcp_length_bytes = 8; |
| 112 ASSERT_EQ(rtcp_length_bytes, raw_packet.size()); |
| 113 |
| 114 rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc( |
| 115 rtc::ArrayView<const uint8_t>(raw_packet.data(), rtcp_length_bytes - 1)); |
| 116 EXPECT_FALSE(ssrc); |
| 117 } |
| 118 |
| 119 } // namespace webrtc |
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