Adds peer scenario connection interface.
This allows implementing custom clients for test in peer connection scenario tests. For example server side behavior. Bug: webrtc:10839 Change-Id: I5627b7a4d967d401f31d2e9a8f861d0849eb0184 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151907 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29125}
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Sebastian Jansson
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parent
0cd61b6e28
commit
d181ee798d
@ -16,6 +16,8 @@ if (rtc_include_tests) {
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"peer_scenario.h",
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"peer_scenario_client.cc",
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"peer_scenario_client.h",
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"scenario_connection.cc",
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"scenario_connection.h",
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"sdp_callbacks.cc",
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"sdp_callbacks.h",
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"signaling_route.cc",
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@ -33,9 +35,12 @@ if (rtc_include_tests) {
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"../../api/video_codecs:builtin_video_decoder_factory",
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"../../api/video_codecs:builtin_video_encoder_factory",
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"../../media:rtc_audio_video",
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"../../media:rtc_media_base",
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"../../modules/audio_device:audio_device_impl",
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"../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../p2p:rtc_p2p",
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"../../pc:pc_test_utils",
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"../../pc:rtc_pc_base",
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"..//network:emulated_network",
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"../scenario",
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"//third_party/abseil-cpp/absl/memory:memory",
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248
test/peer_scenario/scenario_connection.cc
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248
test/peer_scenario/scenario_connection.cc
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@ -0,0 +1,248 @@
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/*
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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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MediaTransportInterface* media_transport,
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DataChannelTransportInterface* data_channel_transport) override;
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void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet_ptr,
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int64_t packet_time_us);
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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 absl::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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absl::WrapUnique(rtc::SSLIdentity::Generate("", ::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, false));
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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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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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// TODO(srte): Move |packet| into lambda when we have c++14.
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network_thread_->PostTask(RTC_FROM_HERE, [this, packet]() mutable {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtp_transport_ == nullptr)
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return;
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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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// TODO(srte): Move |packet| into lambda when we have c++14.
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network_thread_->PostTask(RTC_FROM_HERE, [this, packet]() mutable {
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RTC_DCHECK_RUN_ON(network_thread_);
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if (rtp_transport_ == nullptr)
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return;
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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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MediaTransportInterface* media_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_->SignalRtcpPacketReceived.disconnect(this);
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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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rtp_transport_->SignalRtcpPacketReceived.connect(
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this, &ScenarioIceConnectionImpl::OnRtcpPacketReceived);
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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::OnRtcpPacketReceived(
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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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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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63
test/peer_scenario/scenario_connection.h
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63
test/peer_scenario/scenario_connection.h
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@ -0,0 +1,63 @@
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/*
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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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#ifndef TEST_PEER_SCENARIO_SCENARIO_CONNECTION_H_
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#define TEST_PEER_SCENARIO_SCENARIO_CONNECTION_H_
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#include <functional>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/candidate.h"
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#include "api/jsep.h"
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#include "p2p/base/transport_description.h"
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#include "test/network/network_emulation_manager.h"
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namespace webrtc {
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// ScenarioIceConnection provides the transport level functionality of a
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// PeerConnection for use in peer connection scenario tests. This allows
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// implementing custom server side behavior in tests.
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class ScenarioIceConnection {
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public:
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class IceConnectionObserver {
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public:
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// Called on network thread.
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virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet) = 0;
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// Called on signaling thread.
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virtual void OnIceCandidates(
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const std::string& mid,
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const std::vector<cricket::Candidate>& candidates) = 0;
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protected:
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~IceConnectionObserver() = default;
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};
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static std::unique_ptr<ScenarioIceConnection> Create(
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test::NetworkEmulationManagerImpl* net,
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IceConnectionObserver* observer);
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virtual ~ScenarioIceConnection() = default;
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// Posts tasks to send packets to network thread.
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virtual void SendRtpPacket(rtc::ArrayView<const uint8_t> packet_view) = 0;
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virtual void SendRtcpPacket(rtc::ArrayView<const uint8_t> packet_view) = 0;
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// Used for ICE configuration, called on signaling thread.
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virtual void SetRemoteSdp(SdpType type, const std::string& remote_sdp) = 0;
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virtual void SetLocalSdp(SdpType type, const std::string& local_sdp) = 0;
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virtual EmulatedEndpoint* endpoint() = 0;
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virtual const cricket::TransportDescription& transport_description()
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const = 0;
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};
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} // namespace webrtc
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#endif // TEST_PEER_SCENARIO_SCENARIO_CONNECTION_H_
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