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..d1a56426c28bf344d0ce73008a5e8bd389377d73
--- /dev/null
+++ b/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
@@ -0,0 +1,116 @@
+/*
+ *  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, ParseSenderSsrc_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());
+
+  uint32_t ssrc;
+  EXPECT_TRUE(ParseSenderSsrc(array, &ssrc));

[danilchap, 2017/06/16 16:46:19]

you should be able to pass buffer directly, without converting it to an array

[eladalon, 2017/06/19 11:28:47]

Done.

+  EXPECT_EQ(ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseSenderSsrc_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());
+
+  uint32_t ssrc;
+  EXPECT_TRUE(ParseSenderSsrc(array, &ssrc));
+  EXPECT_EQ(ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseSenderSsrc_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());
+
+  uint32_t ssrc;
+  EXPECT_TRUE(ParseSenderSsrc(array, &ssrc));
+  EXPECT_EQ(ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseSenderSsrc_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());
+
+  uint32_t ssrc;
+  EXPECT_TRUE(ParseSenderSsrc(array, &ssrc));
+  EXPECT_EQ(ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseSenderSsrc_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());
+
+  uint32_t ssrc;
+  EXPECT_TRUE(ParseSenderSsrc(array, &ssrc));
+  EXPECT_EQ(ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseSenderSsrc_BadRtcpPacket) {

[danilchap, 2017/06/16 16:46:19]

may be test for other kind of bad packets too:
e.g. valid header but only 2 bytes of rtcp payload.

[eladalon, 2017/06/19 11:28:47]

IMHO, that would be the sink's problem; the demuxer should have as little
knowledge about RTCP, and what constitutes a valid RTCP packet, as possible.
Also, once we move to passing parsed packets instead of raw, this validation
will already have been done before this module is reached.
Wdyt?

+  uint8_t garbage[100];
+  memset(&garbage[0], 0, arraysize(garbage));
+
+  uint32_t ssrc;
+  EXPECT_FALSE(ParseSenderSsrc(garbage, &ssrc));
+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseSenderSsrc_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());
+
+  uint32_t ssrc;
+  EXPECT_FALSE(ParseSenderSsrc(array, &ssrc));
+}
+
+}  // namespace webrtc