Reland of Create RtcpDemuxer

webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc

Index: webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
diff --git a/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc b/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e51002f6ad2c56592b5c327df14be55815b50c49
--- /dev/null
+++ b/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
@@ -0,0 +1,119 @@
+/*
+ *  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 "webrtc/call/rtp_rtcp_demuxer_helper.h"
+
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "webrtc/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