| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "call/rtp_rtcp_demuxer_helper.h" |
| |
| #include <string.h> |
| |
| #include <cstdio> |
| |
| #include "modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/pli.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/buffer.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| constexpr uint32_t kSsrc = 8374; |
| } // namespace |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ByePacket) { |
| webrtc::rtcp::Bye rtcp_packet; |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_EQ(ssrc, kSsrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, |
| ParseRtcpPacketSenderSsrc_ExtendedReportsPacket) { |
| webrtc::rtcp::ExtendedReports rtcp_packet; |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_EQ(ssrc, kSsrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_PsfbPacket) { |
| webrtc::rtcp::Pli rtcp_packet; // Psfb is abstract; use a subclass. |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_EQ(ssrc, kSsrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ReceiverReportPacket) { |
| webrtc::rtcp::ReceiverReport rtcp_packet; |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_EQ(ssrc, kSsrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_RtpfbPacket) { |
| // Rtpfb is abstract; use a subclass. |
| webrtc::rtcp::RapidResyncRequest rtcp_packet; |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_EQ(ssrc, kSsrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_SenderReportPacket) { |
| webrtc::rtcp::SenderReport rtcp_packet; |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_EQ(ssrc, kSsrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_MalformedRtcpPacket) { |
| uint8_t garbage[100]; |
| memset(&garbage[0], 0, arraysize(garbage)); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(garbage); |
| EXPECT_FALSE(ssrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, |
| ParseRtcpPacketSenderSsrc_RtcpMessageWithoutSenderSsrc) { |
| webrtc::rtcp::ExtendedJitterReport rtcp_packet; // Has no sender SSRC. |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet); |
| EXPECT_FALSE(ssrc); |
| } |
| |
| TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_TruncatedRtcpMessage) { |
| webrtc::rtcp::Bye rtcp_packet; |
| rtcp_packet.SetSenderSsrc(kSsrc); |
| rtc::Buffer raw_packet = rtcp_packet.Build(); |
| |
| constexpr size_t rtcp_length_bytes = 8; |
| ASSERT_EQ(rtcp_length_bytes, raw_packet.size()); |
| |
| absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc( |
| rtc::ArrayView<const uint8_t>(raw_packet.data(), rtcp_length_bytes - 1)); |
| EXPECT_FALSE(ssrc); |
| } |
| |
| } // namespace webrtc |