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..0416e68c386f824dc78746f701b202da63645250
--- /dev/null
+++ b/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
@@ -0,0 +1,102 @@
+/*
+ *  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) {

[danilchap, 2017/06/19 13:31:56]

CorrectOutput are probably redundant words.
ParseRtcpPacketSenderSsrcForBye
looks as descriptive and like a full sentence.

[eladalon, 2017/06/19 14:35:17]

I'll go with ParseRtcpPacketSenderSsrc_ByePacket, etc.

+  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_TRUE(ssrc);

[danilchap, 2017/06/19 13:31:56]

if you'll keep it, use ASSERT_TRUE
may be (to be a bit more explicit) ASSERT_TRUE(ssrc.has_value());

[eladalon, 2017/06/19 14:35:17]

I prefer without the .has_true(), as that's what's usually used in the part of
the codebase I'm currently most familiar with. (I'm not sure what's common
elsewhere.)
As for the assertion - right you are; it would crash otherwise.

[danilchap, 2017/06/19 16:24:54]

.has_value was added only recently, that's one of the reasons why it is rare in
the codebase.
But if you prefer implicit conversion to bool over .has_value() - I wouldn't
object.

[eladalon, 2017/06/19 19:13:48]

Acknowledged.

+  EXPECT_EQ(*ssrc, kSsrc);

[danilchap, 2017/06/19 13:31:56]

you may replace these two EXPECTs with single one:
EXPECT_EQ(ssrc, kSsrc);

[eladalon, 2017/06/19 14:35:17]

1. Still need to assert before.
2. I'm used to relying on rtc::Optional's implicit conversion to a bool, but not
to anything else; I prefer to keep the *.

[danilchap, 2017/06/19 16:24:54]

1. No
2. std::optional<T> allowed to be compared to T without conversions (with empty
optional never equal to a value).
rtc::Optional has explicit implementation of that feature for equality.
If you still prefer to keep * (or use .value() accessor) - I wouldn't mind. Just
want you to be aware about the alternative.

[eladalon, 2017/06/19 19:13:48]

I see now that it prints "empty optional" when comparing an empty optional with
a specific value, so long as the optional is compared directly, and not using
operator*. Cool; I'll use that.

+}
+
+TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_CorrectOutputForPsfb) {
+  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_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest,
+     ParseRtcpPacketSenderSsrc_CorrectOutputForReceiverReport) {
+  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_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 raw_packet = rtcp_packet.Build();
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
+  EXPECT_TRUE(ssrc);
+  EXPECT_EQ(*ssrc, kSsrc);
+}
+
+TEST(RtpRtcpDemuxerHelperTest,
+     ParseRtcpPacketSenderSsrc_CorrectOutputForSenderReport) {
+  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_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 raw_packet = rtcp_packet.Build();
+
+  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
+  EXPECT_FALSE(ssrc);
+}
+
+}  // namespace webrtc