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..01628b5a1dfe67b837db3481a0e222ed119f8a59
--- /dev/null
+++ b/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
@@ -0,0 +1,120 @@
+/*
+ *  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/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/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_CorrectOutputForBye) {
+  webrtc::rtcp::Bye rtcp_packet;
+  rtcp_packet.SetSenderSsrc(kSsrc);
+
+  rtc::Buffer buffer = rtcp_packet.Build();
+  rtc::ArrayView<const uint8_t> array =
+      rtc::ArrayView<const uint8_t>(buffer.data(), buffer.size());
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(array);
+  EXPECT_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_CorrectOutputForPsfb) {
+  webrtc::rtcp::Pli rtcp_packet;  // Psfb is abstract; use a subclass.
+  rtcp_packet.SetSenderSsrc(kSsrc);
+
+  rtc::Buffer buffer = rtcp_packet.Build();
+  rtc::ArrayView<const uint8_t> array =
+      rtc::ArrayView<const uint8_t>(buffer.data(), buffer.size());
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(array);
+  EXPECT_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest,
+     ParseRtcpPacketSenderSsrc_CorrectOutputForReceiverReport) {
+  webrtc::rtcp::ReceiverReport rtcp_packet;
+  rtcp_packet.SetSenderSsrc(kSsrc);
+
+  rtc::Buffer buffer = rtcp_packet.Build();
+  rtc::ArrayView<const uint8_t> array =
+      rtc::ArrayView<const uint8_t>(buffer.data(), buffer.size());
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(array);
+  EXPECT_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest,
+     ParseRtcpPacketSenderSsrc_CorrectOutputForRtpfb) {
+  // Rtpfb is abstract; use a subclass.
+  webrtc::rtcp::RapidResyncRequest rtcp_packet;
+  rtcp_packet.SetSenderSsrc(kSsrc);
+
+  rtc::Buffer buffer = rtcp_packet.Build();
+  rtc::ArrayView<const uint8_t> array =
+      rtc::ArrayView<const uint8_t>(buffer.data(), buffer.size());
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(array);
+  EXPECT_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest,
+     ParseRtcpPacketSenderSsrc_CorrectOutputForSenderReport) {
+  webrtc::rtcp::SenderReport rtcp_packet;
+  rtcp_packet.SetSenderSsrc(kSsrc);
+
+  rtc::Buffer buffer = rtcp_packet.Build();
+  rtc::ArrayView<const uint8_t> array =
+      rtc::ArrayView<const uint8_t>(buffer.data(), buffer.size());
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(array);
+  EXPECT_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_BadRtcpPacket) {
+  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 buffer = rtcp_packet.Build();
+  rtc::ArrayView<const uint8_t> array =
+      rtc::ArrayView<const uint8_t>(buffer.data(), buffer.size());
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(array);
+  EXPECT_FALSE(ssrc);
+}
+
+}  // namespace webrtc