| /* |
| * Copyright 2018 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 "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/stats/rtcstats_objects.h" |
| #include "api/test/fakeconstraints.h" |
| #include "api/video_codecs/builtin_video_decoder_factory.h" |
| #include "api/video_codecs/builtin_video_encoder_factory.h" |
| #include "p2p/base/testturnserver.h" |
| #include "p2p/client/basicportallocator.h" |
| #include "pc/peerconnection.h" |
| #include "pc/peerconnectionwrapper.h" |
| #include "pc/test/fakeaudiocapturemodule.h" |
| #include "pc/test/fakeperiodicvideotracksource.h" |
| #include "pc/test/fakevideotrackrenderer.h" |
| #include "pc/test/framegeneratorcapturervideotracksource.h" |
| #include "rtc_base/fakenetwork.h" |
| #include "rtc_base/firewallsocketserver.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/platform_thread.h" |
| #include "rtc_base/socketaddress.h" |
| #include "rtc_base/virtualsocketserver.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/perf_test.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| static const int kDefaultTestTimeMs = 15000; |
| static const int kRampUpTimeMs = 5000; |
| static const int kPollIntervalTimeMs = 50; |
| static const int kDefaultTimeoutMs = 10000; |
| static const rtc::SocketAddress kDefaultLocalAddress("1.1.1.1", 0); |
| // The video's configured max bitrate in webrtcvideoengine.cc is 1.7 Mbps. |
| // Setting the network bandwidth to 1 Mbps allows the video's bitrate to push |
| // the network's limitations. |
| static const int kNetworkBandwidth = 1000000; |
| } // namespace |
| |
| using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| |
| // This is an end to end test to verify that BWE is functioning when setting |
| // up a one to one call at the PeerConnection level. The intention of the test |
| // is to catch potential regressions for different ICE path configurations. The |
| // test uses a VirtualSocketServer for it's underlying simulated network and |
| // fake audio and video sources. The test is based upon rampup_tests.cc, but |
| // instead is at the PeerConnection level and uses a different fake network |
| // (rampup_tests.cc uses SimulatedNetwork). In the future, this test could |
| // potentially test different network conditions and test video quality as well |
| // (video_quality_test.cc does this, but at the call level). |
| // |
| // The perf test results are printed using the perf test support. If the |
| // isolated_script_test_perf_output flag is specified in test_main.cc, then |
| // the results are written to a JSON formatted file for the Chrome perf |
| // dashboard. Since this test is a webrtc_perf_test, it will be run in the perf |
| // console every webrtc commit. |
| class PeerConnectionWrapperForRampUpTest : public PeerConnectionWrapper { |
| public: |
| using PeerConnectionWrapper::PeerConnectionWrapper; |
| |
| PeerConnectionWrapperForRampUpTest( |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory, |
| rtc::scoped_refptr<PeerConnectionInterface> pc, |
| std::unique_ptr<MockPeerConnectionObserver> observer, |
| rtc::FakeNetworkManager* fake_network_manager) |
| : PeerConnectionWrapper::PeerConnectionWrapper(pc_factory, |
| pc, |
| std::move(observer)), |
| fake_network_manager_(std::move(fake_network_manager)) {} |
| |
| bool AddIceCandidates(std::vector<const IceCandidateInterface*> candidates) { |
| bool success = true; |
| for (const auto candidate : candidates) { |
| if (!pc()->AddIceCandidate(candidate)) { |
| success = false; |
| } |
| } |
| return success; |
| } |
| |
| rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack( |
| FrameGeneratorCapturerVideoTrackSource::Config config, |
| Clock* clock) { |
| video_track_sources_.emplace_back( |
| new rtc::RefCountedObject<FrameGeneratorCapturerVideoTrackSource>( |
| config, clock)); |
| video_track_sources_.back()->Start(); |
| return rtc::scoped_refptr<VideoTrackInterface>( |
| pc_factory()->CreateVideoTrack(rtc::CreateRandomUuid(), |
| video_track_sources_.back())); |
| } |
| |
| rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack( |
| const cricket::AudioOptions options) { |
| rtc::scoped_refptr<AudioSourceInterface> source = |
| pc_factory()->CreateAudioSource(options); |
| return pc_factory()->CreateAudioTrack(rtc::CreateRandomUuid(), source); |
| } |
| |
| private: |
| // This is owned by the Test, not the Wrapper. It needs to outlive the |
| // Wrapper, because the port allocator expects its lifetime to be longer than |
| // the PeerConnection's lifetime. |
| rtc::FakeNetworkManager* fake_network_manager_; |
| std::vector<rtc::scoped_refptr<FrameGeneratorCapturerVideoTrackSource>> |
| video_track_sources_; |
| }; |
| |
| // TODO(shampson): Paramaterize the test to run for both Plan B & Unified Plan. |
| class PeerConnectionRampUpTest : public ::testing::Test { |
| public: |
| PeerConnectionRampUpTest() |
| : clock_(Clock::GetRealTimeClock()), |
| virtual_socket_server_(new rtc::VirtualSocketServer()), |
| firewall_socket_server_( |
| new rtc::FirewallSocketServer(virtual_socket_server_.get())), |
| network_thread_(new rtc::Thread(firewall_socket_server_.get())), |
| worker_thread_(rtc::Thread::Create()) { |
| network_thread_->SetName("PCNetworkThread", this); |
| worker_thread_->SetName("PCWorkerThread", this); |
| RTC_CHECK(network_thread_->Start()); |
| RTC_CHECK(worker_thread_->Start()); |
| |
| virtual_socket_server_->set_bandwidth(kNetworkBandwidth / 8); |
| pc_factory_ = CreatePeerConnectionFactory( |
| network_thread_.get(), worker_thread_.get(), rtc::Thread::Current(), |
| rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()), |
| CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(), |
| CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(), |
| nullptr /* audio_mixer */, nullptr /* audio_processing */); |
| } |
| |
| virtual ~PeerConnectionRampUpTest() { |
| network_thread()->Invoke<void>(RTC_FROM_HERE, |
| [this] { turn_servers_.clear(); }); |
| } |
| |
| bool CreatePeerConnectionWrappers(const RTCConfiguration& caller_config, |
| const RTCConfiguration& callee_config) { |
| caller_ = CreatePeerConnectionWrapper(caller_config); |
| callee_ = CreatePeerConnectionWrapper(callee_config); |
| return caller_ && callee_; |
| } |
| |
| std::unique_ptr<PeerConnectionWrapperForRampUpTest> |
| CreatePeerConnectionWrapper(const RTCConfiguration& config) { |
| auto* fake_network_manager = new rtc::FakeNetworkManager(); |
| fake_network_manager->AddInterface(kDefaultLocalAddress); |
| fake_network_managers_.emplace_back(fake_network_manager); |
| auto port_allocator = |
| rtc::MakeUnique<cricket::BasicPortAllocator>(fake_network_manager); |
| |
| port_allocator->set_step_delay(cricket::kDefaultStepDelay); |
| auto observer = rtc::MakeUnique<MockPeerConnectionObserver>(); |
| auto pc = pc_factory_->CreatePeerConnection( |
| config, std::move(port_allocator), nullptr, observer.get()); |
| if (!pc) { |
| return nullptr; |
| } |
| |
| return rtc::MakeUnique<PeerConnectionWrapperForRampUpTest>( |
| pc_factory_, pc, std::move(observer), fake_network_manager); |
| } |
| |
| void SetupOneWayCall() { |
| ASSERT_TRUE(caller_); |
| ASSERT_TRUE(callee_); |
| FrameGeneratorCapturerVideoTrackSource::Config config; |
| caller_->AddTrack(caller_->CreateLocalVideoTrack(config, clock_)); |
| // Disable highpass filter so that we can get all the test audio frames. |
| cricket::AudioOptions options; |
| options.highpass_filter = false; |
| caller_->AddTrack(caller_->CreateLocalAudioTrack(options)); |
| |
| // Do the SDP negotiation, and also exchange ice candidates. |
| ASSERT_TRUE(caller_->ExchangeOfferAnswerWith(callee_.get())); |
| ASSERT_TRUE_WAIT( |
| caller_->signaling_state() == PeerConnectionInterface::kStable, |
| kDefaultTimeoutMs); |
| ASSERT_TRUE_WAIT(caller_->IsIceGatheringDone(), kDefaultTimeoutMs); |
| ASSERT_TRUE_WAIT(callee_->IsIceGatheringDone(), kDefaultTimeoutMs); |
| |
| // Connect an ICE candidate pairs. |
| ASSERT_TRUE( |
| callee_->AddIceCandidates(caller_->observer()->GetAllCandidates())); |
| ASSERT_TRUE( |
| caller_->AddIceCandidates(callee_->observer()->GetAllCandidates())); |
| // This means that ICE and DTLS are connected. |
| ASSERT_TRUE_WAIT(callee_->IsIceConnected(), kDefaultTimeoutMs); |
| ASSERT_TRUE_WAIT(caller_->IsIceConnected(), kDefaultTimeoutMs); |
| } |
| |
| void CreateTurnServer(cricket::ProtocolType type) { |
| rtc::Thread* thread = network_thread(); |
| std::unique_ptr<cricket::TestTurnServer> turn_server = |
| network_thread_->Invoke<std::unique_ptr<cricket::TestTurnServer>>( |
| RTC_FROM_HERE, [thread, type] { |
| static const rtc::SocketAddress turn_server_internal_address{ |
| "88.88.88.0", 3478}; |
| static const rtc::SocketAddress turn_server_external_address{ |
| "88.88.88.1", 0}; |
| return rtc::MakeUnique<cricket::TestTurnServer>( |
| thread, turn_server_internal_address, |
| turn_server_external_address, type); |
| }); |
| turn_servers_.push_back(std::move(turn_server)); |
| } |
| |
| // First runs the call for kRampUpTimeMs to ramp up the bandwidth estimate. |
| // Then runs the test for the remaining test time, grabbing the bandwidth |
| // estimation stat, every kPollIntervalTimeMs. When finished, averages the |
| // bandwidth estimations and prints the bandwidth estimation result as a perf |
| // metric. |
| void RunTest(const std::string& test_string) { |
| rtc::Thread::Current()->ProcessMessages(kRampUpTimeMs); |
| int number_of_polls = |
| (kDefaultTestTimeMs - kRampUpTimeMs) / kPollIntervalTimeMs; |
| int total_bwe = 0; |
| for (int i = 0; i < number_of_polls; ++i) { |
| rtc::Thread::Current()->ProcessMessages(kPollIntervalTimeMs); |
| total_bwe += static_cast<int>(GetCallerAvailableBitrateEstimate()); |
| } |
| double average_bandwidth_estimate = total_bwe / number_of_polls; |
| std::string value_description = |
| "bwe_after_" + std::to_string(kDefaultTestTimeMs / 1000) + "_seconds"; |
| test::PrintResult("peerconnection_ramp_up_", test_string, value_description, |
| average_bandwidth_estimate, "bwe", false); |
| } |
| |
| rtc::Thread* network_thread() { return network_thread_.get(); } |
| |
| PeerConnectionWrapperForRampUpTest* caller() { return caller_.get(); } |
| |
| PeerConnectionWrapperForRampUpTest* callee() { return callee_.get(); } |
| |
| private: |
| // Gets the caller's outgoing available bitrate from the stats. Returns 0 if |
| // something went wrong. It takes the outgoing bitrate from the current |
| // selected ICE candidate pair's stats. |
| double GetCallerAvailableBitrateEstimate() { |
| auto stats = caller_->GetStats(); |
| auto transport_stats = stats->GetStatsOfType<RTCTransportStats>(); |
| if (transport_stats.size() == 0u || |
| !transport_stats[0]->selected_candidate_pair_id.is_defined()) { |
| return 0; |
| } |
| std::string selected_ice_id = |
| transport_stats[0]->selected_candidate_pair_id.ValueToString(); |
| // Use the selected ICE candidate pair ID to get the appropriate ICE stats. |
| const RTCIceCandidatePairStats ice_candidate_pair_stats = |
| stats->Get(selected_ice_id)->cast_to<const RTCIceCandidatePairStats>(); |
| if (ice_candidate_pair_stats.available_outgoing_bitrate.is_defined()) { |
| return *ice_candidate_pair_stats.available_outgoing_bitrate; |
| } |
| // We couldn't get the |available_outgoing_bitrate| for the active candidate |
| // pair. |
| return 0; |
| } |
| |
| Clock* const clock_; |
| // The turn servers should be accessed & deleted on the network thread to |
| // avoid a race with the socket read/write which occurs on the network thread. |
| std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_; |
| // |virtual_socket_server_| is used by |network_thread_| so it must be |
| // destroyed later. |
| // TODO(bugs.webrtc.org/7668): We would like to update the virtual network we |
| // use for this test. VirtualSocketServer isn't ideal because: |
| // 1) It uses the same queue & network capacity for both directions. |
| // 2) VirtualSocketServer implements how the network bandwidth affects the |
| // send delay differently than the SimulatedNetwork, used by the |
| // FakeNetworkPipe. It would be ideal if all of levels of virtual |
| // networks used in testing were consistent. |
| // We would also like to update this test to record the time to ramp up, |
| // down, and back up (similar to in rampup_tests.cc). This is problematic with |
| // the VirtualSocketServer. The first ramp down time is very noisy and the |
| // second ramp up time can take up to 300 seconds, most likely due to a built |
| // up queue. |
| std::unique_ptr<rtc::VirtualSocketServer> virtual_socket_server_; |
| std::unique_ptr<rtc::FirewallSocketServer> firewall_socket_server_; |
| std::unique_ptr<rtc::Thread> network_thread_; |
| std::unique_ptr<rtc::Thread> worker_thread_; |
| // The |pc_factory| uses |network_thread_| & |worker_thread_|, so it must be |
| // destroyed first. |
| std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_network_managers_; |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_; |
| std::unique_ptr<PeerConnectionWrapperForRampUpTest> caller_; |
| std::unique_ptr<PeerConnectionWrapperForRampUpTest> callee_; |
| }; |
| |
| TEST_F(PeerConnectionRampUpTest, TurnOverTCP) { |
| CreateTurnServer(cricket::ProtocolType::PROTO_TCP); |
| PeerConnectionInterface::IceServer ice_server; |
| ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp"); |
| ice_server.username = "test"; |
| ice_server.password = "test"; |
| PeerConnectionInterface::RTCConfiguration client_1_config; |
| client_1_config.servers.push_back(ice_server); |
| client_1_config.type = PeerConnectionInterface::kRelay; |
| PeerConnectionInterface::RTCConfiguration client_2_config; |
| client_2_config.servers.push_back(ice_server); |
| client_2_config.type = PeerConnectionInterface::kRelay; |
| ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config)); |
| |
| SetupOneWayCall(); |
| RunTest("turn_over_tcp"); |
| } |
| |
| // TODO(bugs.webrtc.org/7668): Test other ICE configurations. |
| |
| } // namespace webrtc |