blob: 5fe827494bd3da4b1275f6f1c87a4edb034689de [file] [log] [blame]
/*
* 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 <cstdio>
#include "call/rtp_rtcp_demuxer_helper.h"
#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();
rtc::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();
rtc::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();
rtc::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();
rtc::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();
rtc::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();
rtc::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));
rtc::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();
rtc::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());
rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(
rtc::ArrayView<const uint8_t>(raw_packet.data(), rtcp_length_bytes - 1));
EXPECT_FALSE(ssrc);
}
} // namespace webrtc