This reverts commit c5f71087589b18bb4df1b78f2c452c4083edf2d9. Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995? Sample logs: STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575 STDERR: # last system error: 0 STDERR: # Check failed: (signaling_thread())->IsCurrent() STDERR: # Received signal 6 STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace() STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace() STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7f81c8d72db1 gsignal STDERR: #5 0x7f81c8d5c537 abort STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL() STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent() STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived() STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket() STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket() STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket() STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket() STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived() STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived() STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept() STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage() STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage() STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage() STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept() STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage() STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages() STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal() STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState() STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady() STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce() STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask() STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run() STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() STDERR: #40 0x7f81d395ae55 base::RunLoop::Run() STDERR: #41 0x7f81d39c87f2 base::Thread::Run() Original change's description: > Reland "Replace sigslot usages with robocaller library." > > This is a reland of 40261c3663fe316cfe40262c59cee993165ccf63 > > Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError > added a new member with a different name and used it in webrtc code. > After this change do two more follow up CLs to completely remove the old code > from google3. > > Original change's description: > > Replace sigslot usages with robocaller library. > > > > - Replace all the top level signals from jsep_transport_controller. > > - There are still sigslot usages in this file so keep the inheritance > > and that is the reason for not having a binary size gain in this CL. > > > > Bug: webrtc:11943 > > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540 > > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#32321} > > Bug: webrtc:11943 > Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946 > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32359} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11943 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32372}
239 lines
9.6 KiB
C++
239 lines
9.6 KiB
C++
/*
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* Copyright (c) 2019 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 "test/peer_scenario/scenario_connection.h"
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#include "absl/memory/memory.h"
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#include "media/base/rtp_utils.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "p2p/client/basic_port_allocator.h"
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#include "pc/jsep_transport_controller.h"
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#include "pc/rtp_transport_internal.h"
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#include "pc/session_description.h"
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namespace webrtc {
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class ScenarioIceConnectionImpl : public ScenarioIceConnection,
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public sigslot::has_slots<>,
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private JsepTransportController::Observer,
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private RtpPacketSinkInterface {
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public:
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ScenarioIceConnectionImpl(test::NetworkEmulationManagerImpl* net,
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IceConnectionObserver* observer);
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~ScenarioIceConnectionImpl() override;
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void SendRtpPacket(rtc::ArrayView<const uint8_t> packet_view) override;
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void SendRtcpPacket(rtc::ArrayView<const uint8_t> packet_view) override;
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void SetRemoteSdp(SdpType type, const std::string& remote_sdp) override;
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void SetLocalSdp(SdpType type, const std::string& local_sdp) override;
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EmulatedEndpoint* endpoint() override { return endpoint_; }
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const cricket::TransportDescription& transport_description() const override {
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return transport_description_;
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}
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private:
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JsepTransportController::Config CreateJsepConfig();
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bool OnTransportChanged(
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const std::string& mid,
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RtpTransportInternal* rtp_transport,
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rtc::scoped_refptr<DtlsTransport> dtls_transport,
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DataChannelTransportInterface* data_channel_transport) override;
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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void OnCandidates(const std::string& mid,
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const std::vector<cricket::Candidate>& candidates);
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IceConnectionObserver* const observer_;
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EmulatedEndpoint* const endpoint_;
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EmulatedNetworkManagerInterface* const manager_;
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rtc::Thread* const signaling_thread_;
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rtc::Thread* const network_thread_;
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rtc::scoped_refptr<rtc::RTCCertificate> const certificate_
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RTC_GUARDED_BY(network_thread_);
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cricket::TransportDescription const transport_description_
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RTC_GUARDED_BY(signaling_thread_);
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std::unique_ptr<cricket::BasicPortAllocator> port_allocator_
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RTC_GUARDED_BY(network_thread_);
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std::unique_ptr<JsepTransportController> jsep_controller_;
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RtpTransportInternal* rtp_transport_ RTC_GUARDED_BY(network_thread_) =
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nullptr;
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std::unique_ptr<SessionDescriptionInterface> remote_description_
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RTC_GUARDED_BY(signaling_thread_);
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std::unique_ptr<SessionDescriptionInterface> local_description_
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RTC_GUARDED_BY(signaling_thread_);
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};
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std::unique_ptr<ScenarioIceConnection> ScenarioIceConnection::Create(
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webrtc::test::NetworkEmulationManagerImpl* net,
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IceConnectionObserver* observer) {
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return std::make_unique<ScenarioIceConnectionImpl>(net, observer);
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}
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ScenarioIceConnectionImpl::ScenarioIceConnectionImpl(
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test::NetworkEmulationManagerImpl* net,
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IceConnectionObserver* observer)
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: observer_(observer),
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endpoint_(net->CreateEndpoint(EmulatedEndpointConfig())),
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manager_(net->CreateEmulatedNetworkManagerInterface({endpoint_})),
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signaling_thread_(rtc::Thread::Current()),
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network_thread_(manager_->network_thread()),
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certificate_(rtc::RTCCertificate::Create(
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rtc::SSLIdentity::Create("", ::rtc::KT_DEFAULT))),
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transport_description_(
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/*transport_options*/ {},
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rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
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rtc::CreateRandomString(cricket::ICE_PWD_LENGTH),
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cricket::IceMode::ICEMODE_FULL,
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cricket::ConnectionRole::CONNECTIONROLE_PASSIVE,
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rtc::SSLFingerprint::CreateFromCertificate(*certificate_.get())
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.get()),
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port_allocator_(
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new cricket::BasicPortAllocator(manager_->network_manager())),
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jsep_controller_(
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new JsepTransportController(signaling_thread_,
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network_thread_,
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port_allocator_.get(),
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/*async_resolver_factory*/ nullptr,
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CreateJsepConfig())) {
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network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
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RTC_DCHECK_RUN_ON(network_thread_);
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uint32_t flags = cricket::PORTALLOCATOR_DISABLE_TCP;
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port_allocator_->set_flags(port_allocator_->flags() | flags);
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port_allocator_->Initialize();
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RTC_CHECK(port_allocator_->SetConfiguration(/*stun_servers*/ {},
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/*turn_servers*/ {}, 0,
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webrtc::NO_PRUNE));
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jsep_controller_->SetLocalCertificate(certificate_);
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});
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}
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ScenarioIceConnectionImpl::~ScenarioIceConnectionImpl() {
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network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
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RTC_DCHECK_RUN_ON(network_thread_);
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jsep_controller_.reset();
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port_allocator_.reset();
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rtp_transport_ = nullptr;
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});
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}
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JsepTransportController::Config ScenarioIceConnectionImpl::CreateJsepConfig() {
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JsepTransportController::Config config;
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config.transport_observer = this;
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config.bundle_policy =
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PeerConnectionInterface::BundlePolicy::kBundlePolicyMaxBundle;
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config.rtcp_handler = [this](const rtc::CopyOnWriteBuffer& packet,
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int64_t packet_time_us) {
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RTC_DCHECK_RUN_ON(network_thread_);
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observer_->OnPacketReceived(packet);
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};
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return config;
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}
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void ScenarioIceConnectionImpl::SendRtpPacket(
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rtc::ArrayView<const uint8_t> packet_view) {
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rtc::CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
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::cricket::kMaxRtpPacketLen);
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network_thread_->PostTask(
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RTC_FROM_HERE, [this, packet = std::move(packet)]() mutable {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtp_transport_ != nullptr)
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rtp_transport_->SendRtpPacket(&packet, rtc::PacketOptions(),
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cricket::PF_SRTP_BYPASS);
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});
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}
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void ScenarioIceConnectionImpl::SendRtcpPacket(
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rtc::ArrayView<const uint8_t> packet_view) {
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rtc::CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
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::cricket::kMaxRtpPacketLen);
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network_thread_->PostTask(
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RTC_FROM_HERE, [this, packet = std::move(packet)]() mutable {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtp_transport_ != nullptr)
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rtp_transport_->SendRtcpPacket(&packet, rtc::PacketOptions(),
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cricket::PF_SRTP_BYPASS);
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});
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}
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void ScenarioIceConnectionImpl::SetRemoteSdp(SdpType type,
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const std::string& remote_sdp) {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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remote_description_ = webrtc::CreateSessionDescription(type, remote_sdp);
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jsep_controller_->SignalIceCandidatesGathered.connect(
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this, &ScenarioIceConnectionImpl::OnCandidates);
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auto res = jsep_controller_->SetRemoteDescription(
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remote_description_->GetType(), remote_description_->description());
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RTC_CHECK(res.ok()) << res.message();
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RtpDemuxerCriteria criteria;
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for (const auto& content : remote_description_->description()->contents()) {
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if (content.media_description()->as_audio()) {
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for (const auto& codec :
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content.media_description()->as_audio()->codecs()) {
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criteria.payload_types.insert(codec.id);
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}
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}
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if (content.media_description()->as_video()) {
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for (const auto& codec :
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content.media_description()->as_video()->codecs()) {
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criteria.payload_types.insert(codec.id);
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}
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}
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}
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network_thread_->PostTask(RTC_FROM_HERE, [this, criteria]() {
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RTC_DCHECK_RUN_ON(network_thread_);
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RTC_DCHECK(rtp_transport_);
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rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
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});
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}
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void ScenarioIceConnectionImpl::SetLocalSdp(SdpType type,
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const std::string& local_sdp) {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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local_description_ = webrtc::CreateSessionDescription(type, local_sdp);
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auto res = jsep_controller_->SetLocalDescription(
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local_description_->GetType(), local_description_->description());
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RTC_CHECK(res.ok()) << res.message();
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jsep_controller_->MaybeStartGathering();
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}
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bool ScenarioIceConnectionImpl::OnTransportChanged(
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const std::string& mid,
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RtpTransportInternal* rtp_transport,
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rtc::scoped_refptr<DtlsTransport> dtls_transport,
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DataChannelTransportInterface* data_channel_transport) {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtp_transport == nullptr) {
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rtp_transport_->UnregisterRtpDemuxerSink(this);
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} else {
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RTC_DCHECK(rtp_transport_ == nullptr || rtp_transport_ == rtp_transport);
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if (rtp_transport_ != rtp_transport) {
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rtp_transport_ = rtp_transport;
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}
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RtpDemuxerCriteria criteria;
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criteria.mid = mid;
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rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
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}
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return true;
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}
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void ScenarioIceConnectionImpl::OnRtpPacket(const RtpPacketReceived& packet) {
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RTC_DCHECK_RUN_ON(network_thread_);
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observer_->OnPacketReceived(packet.Buffer());
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}
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void ScenarioIceConnectionImpl::OnCandidates(
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const std::string& mid,
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const std::vector<cricket::Candidate>& candidates) {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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observer_->OnIceCandidates(mid, candidates);
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}
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} // namespace webrtc
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