| /* | 
 |  *  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 |