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}
140 lines
4.6 KiB
C++
140 lines
4.6 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 "webrtc/call/rtp_demuxer.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/call/rsid_resolution_observer.h"
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#include "webrtc/call/rtp_packet_sink_interface.h"
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#include "webrtc/call/rtp_rtcp_demuxer_helper.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
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namespace webrtc {
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namespace {
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constexpr size_t kMaxProcessedSsrcs = 1000; // Prevent memory overuse.
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} // namespace
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RtpDemuxer::RtpDemuxer() = default;
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RtpDemuxer::~RtpDemuxer() {
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RTC_DCHECK(sinks_.empty());
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RTC_DCHECK(rsid_sinks_.empty());
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}
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void RtpDemuxer::AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) {
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RTC_DCHECK(sink);
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RecordSsrcToSinkAssociation(ssrc, sink);
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}
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void RtpDemuxer::AddSink(const std::string& rsid,
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RtpPacketSinkInterface* sink) {
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RTC_DCHECK(StreamId::IsLegalName(rsid));
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RTC_DCHECK(sink);
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RTC_DCHECK(!MultimapAssociationExists(rsid_sinks_, rsid, sink));
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rsid_sinks_.emplace(rsid, sink);
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// This RSID might now map to an SSRC which we saw earlier.
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processed_ssrcs_.clear();
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}
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bool RtpDemuxer::RemoveSink(const RtpPacketSinkInterface* sink) {
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RTC_DCHECK(sink);
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return (RemoveFromMultimapByValue(&sinks_, sink) +
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RemoveFromMultimapByValue(&rsid_sinks_, sink)) > 0;
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}
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void RtpDemuxer::RecordSsrcToSinkAssociation(uint32_t ssrc,
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RtpPacketSinkInterface* sink) {
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RTC_DCHECK(sink);
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// The association might already have been set by a different
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// configuration source.
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if (!MultimapAssociationExists(sinks_, ssrc, sink)) {
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sinks_.emplace(ssrc, sink);
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}
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}
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bool RtpDemuxer::OnRtpPacket(const RtpPacketReceived& packet) {
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ResolveAssociations(packet);
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auto it_range = sinks_.equal_range(packet.Ssrc());
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for (auto it = it_range.first; it != it_range.second; ++it) {
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it->second->OnRtpPacket(packet);
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}
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return it_range.first != it_range.second;
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}
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void RtpDemuxer::RegisterRsidResolutionObserver(
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RsidResolutionObserver* observer) {
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RTC_DCHECK(observer);
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RTC_DCHECK(!ContainerHasKey(rsid_resolution_observers_, observer));
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rsid_resolution_observers_.push_back(observer);
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processed_ssrcs_.clear(); // New observer requires new notifications.
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}
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void RtpDemuxer::DeregisterRsidResolutionObserver(
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const RsidResolutionObserver* observer) {
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RTC_DCHECK(observer);
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auto it = std::find(rsid_resolution_observers_.begin(),
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rsid_resolution_observers_.end(), observer);
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RTC_DCHECK(it != rsid_resolution_observers_.end());
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rsid_resolution_observers_.erase(it);
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}
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void RtpDemuxer::ResolveAssociations(const RtpPacketReceived& packet) {
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// Avoid expensive string comparisons for RSID by looking the sinks up only
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// by SSRC whenever possible.
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if (processed_ssrcs_.find(packet.Ssrc()) != processed_ssrcs_.cend()) {
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return;
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}
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ResolveRsidToSsrcAssociations(packet);
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if (processed_ssrcs_.size() < kMaxProcessedSsrcs) { // Prevent memory overuse
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processed_ssrcs_.insert(packet.Ssrc()); // Avoid re-examining in-depth.
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} else if (!logged_max_processed_ssrcs_exceeded_) {
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LOG(LS_WARNING) << "More than " << kMaxProcessedSsrcs
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<< " different SSRCs seen.";
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logged_max_processed_ssrcs_exceeded_ = true;
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}
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}
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void RtpDemuxer::ResolveRsidToSsrcAssociations(
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const RtpPacketReceived& packet) {
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std::string rsid;
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if (packet.GetExtension<RtpStreamId>(&rsid)) {
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// All streams associated with this RSID need to be marked as associated
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// with this SSRC (if they aren't already).
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auto it_range = rsid_sinks_.equal_range(rsid);
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for (auto it = it_range.first; it != it_range.second; ++it) {
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RecordSsrcToSinkAssociation(packet.Ssrc(), it->second);
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}
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NotifyObserversOfRsidResolution(rsid, packet.Ssrc());
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// To prevent memory-overuse attacks, forget this RSID. Future packets
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// with this RSID, but a different SSRC, will not spawn new associations.
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rsid_sinks_.erase(it_range.first, it_range.second);
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}
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}
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void RtpDemuxer::NotifyObserversOfRsidResolution(const std::string& rsid,
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uint32_t ssrc) {
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for (auto* observer : rsid_resolution_observers_) {
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observer->OnRsidResolved(rsid, ssrc);
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}
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}
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} // namespace webrtc
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