a52722fac4
Reason for revert: About to fix problem and reland. Original issue's description: > Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) > > Reason for revert: > Breaks Chromium FYI bots. > > The problem is in the BUILD.gn file. > > Sample failure: > https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/17829 > > Sample logs: > use_goma = true > """ to /b/c/b/Linux_Builder/src/out/Release/args.gn. > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //third_party/webrtc/call/BUILD.gn:46:5: Can't load input file. > "//webrtc/base:rtc_base_approved", > ^-------------------------------- > > Original issue's description: > > Create RtcpDemuxer. Capabilities: > > 1. Demux RTCP messages according to the sender-SSRC. > > 2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP). > > 3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks"). > > > > BUG=webrtc:7135 > > > > Review-Url: https://codereview.webrtc.org/2943693003 > > Cr-Commit-Position: refs/heads/master@{#18763} > > Committed:cb83bdf01f> > BUG=webrtc:7135 > > Review-Url: https://codereview.webrtc.org/2957763002 > Cr-Commit-Position: refs/heads/master@{#18764} > Committed:0e7e7869e7BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2960623002 Cr-Commit-Position: refs/heads/master@{#18768}
120 lines
4.0 KiB
C++
120 lines
4.0 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <cstdio>
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#include "webrtc/call/rtp_rtcp_demuxer_helper.h"
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#include "webrtc/base/arraysize.h"
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/buffer.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "webrtc/test/gtest.h"
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namespace webrtc {
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namespace {
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constexpr uint32_t kSsrc = 8374;
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} // namespace
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ByePacket) {
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webrtc::rtcp::Bye rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest,
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ParseRtcpPacketSenderSsrc_ExtendedReportsPacket) {
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webrtc::rtcp::ExtendedReports rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_PsfbPacket) {
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webrtc::rtcp::Pli rtcp_packet; // Psfb is abstract; use a subclass.
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ReceiverReportPacket) {
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webrtc::rtcp::ReceiverReport rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_RtpfbPacket) {
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// Rtpfb is abstract; use a subclass.
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webrtc::rtcp::RapidResyncRequest rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_SenderReportPacket) {
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webrtc::rtcp::SenderReport rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_EQ(ssrc, kSsrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_MalformedRtcpPacket) {
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uint8_t garbage[100];
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memset(&garbage[0], 0, arraysize(garbage));
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(garbage);
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EXPECT_FALSE(ssrc);
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}
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TEST(RtpRtcpDemuxerHelperTest,
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ParseRtcpPacketSenderSsrc_RtcpMessageWithoutSenderSsrc) {
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webrtc::rtcp::ExtendedJitterReport rtcp_packet; // Has no sender SSRC.
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rtc::Buffer raw_packet = rtcp_packet.Build();
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
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EXPECT_FALSE(ssrc);
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}
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TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_TruncatedRtcpMessage) {
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webrtc::rtcp::Bye rtcp_packet;
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rtcp_packet.SetSenderSsrc(kSsrc);
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rtc::Buffer raw_packet = rtcp_packet.Build();
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constexpr size_t rtcp_length_bytes = 8;
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ASSERT_EQ(rtcp_length_bytes, raw_packet.size());
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rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(
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rtc::ArrayView<const uint8_t>(raw_packet.data(), rtcp_length_bytes - 1));
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EXPECT_FALSE(ssrc);
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}
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} // namespace webrtc
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