|  | /* | 
|  | *  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 |