Revert 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
deleted file mode 100644
index e51002f6ad2c56592b5c327df14be55815b50c49..0000000000000000000000000000000000000000
--- a/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- *  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