eladalon | a52722f | 2017-06-26 11:23:54 -0700 | [diff] [blame] | 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 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 13 | #include "call/rtp_rtcp_demuxer_helper.h" |
eladalon | a52722f | 2017-06-26 11:23:54 -0700 | [diff] [blame] | 14 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 15 | #include "modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| 16 | #include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| 17 | #include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| 18 | #include "modules/rtp_rtcp/source/rtcp_packet/pli.h" |
| 19 | #include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" |
| 20 | #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| 21 | #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| 22 | #include "rtc_base/arraysize.h" |
| 23 | #include "rtc_base/basictypes.h" |
| 24 | #include "rtc_base/buffer.h" |
| 25 | #include "test/gtest.h" |
eladalon | a52722f | 2017-06-26 11:23:54 -0700 | [diff] [blame] | 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 |