| /* |
| * Copyright (c) 2019 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 "test/pc/e2e/peer_connection_quality_test.h" |
| |
| #include <algorithm> |
| #include <memory> |
| #include <set> |
| #include <utility> |
| |
| #include "absl/flags/flag.h" |
| #include "absl/strings/string_view.h" |
| #include "api/jsep.h" |
| #include "api/media_stream_interface.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/rtc_event_log_output_file.h" |
| #include "api/scoped_refptr.h" |
| #include "api/test/metrics/metric.h" |
| #include "api/test/pclf/media_configuration.h" |
| #include "api/test/pclf/peer_configurer.h" |
| #include "api/test/time_controller.h" |
| #include "api/test/video_quality_analyzer_interface.h" |
| #include "pc/sdp_utils.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "system_wrappers/include/cpu_info.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "test/field_trial.h" |
| #include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h" |
| #include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h" |
| #include "test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h" |
| #include "test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h" |
| #include "test/pc/e2e/cross_media_metrics_reporter.h" |
| #include "test/pc/e2e/metric_metadata_keys.h" |
| #include "test/pc/e2e/peer_params_preprocessor.h" |
| #include "test/pc/e2e/stats_poller.h" |
| #include "test/pc/e2e/test_peer_factory.h" |
| #include "test/test_flags.h" |
| #include "test/testsupport/file_utils.h" |
| |
| namespace webrtc { |
| namespace webrtc_pc_e2e { |
| namespace { |
| |
| using ::webrtc::test::ImprovementDirection; |
| using ::webrtc::test::Unit; |
| |
| constexpr TimeDelta kDefaultTimeout = TimeDelta::Seconds(10); |
| constexpr char kSignalThreadName[] = "signaling_thread"; |
| // 1 signaling, 2 network, 2 worker and 2 extra for codecs etc. |
| constexpr int kPeerConnectionUsedThreads = 7; |
| // Framework has extra thread for network layer and extra thread for peer |
| // connection stats polling. |
| constexpr int kFrameworkUsedThreads = 2; |
| constexpr int kMaxVideoAnalyzerThreads = 8; |
| |
| constexpr TimeDelta kStatsUpdateInterval = TimeDelta::Seconds(1); |
| |
| constexpr TimeDelta kAliveMessageLogInterval = TimeDelta::Seconds(30); |
| |
| constexpr TimeDelta kQuickTestModeRunDuration = TimeDelta::Millis(100); |
| |
| // Field trials to enable Flex FEC advertising and receiving. |
| constexpr char kFlexFecEnabledFieldTrials[] = |
| "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/"; |
| constexpr char kUseStandardsBytesStats[] = |
| "WebRTC-UseStandardBytesStats/Enabled/"; |
| |
| class FixturePeerConnectionObserver : public MockPeerConnectionObserver { |
| public: |
| // `on_track_callback` will be called when any new track will be added to peer |
| // connection. |
| // `on_connected_callback` will be called when peer connection will come to |
| // either connected or completed state. Client should notice that in the case |
| // of reconnect this callback can be called again, so it should be tolerant |
| // to such behavior. |
| FixturePeerConnectionObserver( |
| std::function<void(rtc::scoped_refptr<RtpTransceiverInterface>)> |
| on_track_callback, |
| std::function<void()> on_connected_callback) |
| : on_track_callback_(std::move(on_track_callback)), |
| on_connected_callback_(std::move(on_connected_callback)) {} |
| |
| void OnTrack( |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override { |
| MockPeerConnectionObserver::OnTrack(transceiver); |
| on_track_callback_(transceiver); |
| } |
| |
| void OnIceConnectionChange( |
| PeerConnectionInterface::IceConnectionState new_state) override { |
| MockPeerConnectionObserver::OnIceConnectionChange(new_state); |
| if (ice_connected_) { |
| on_connected_callback_(); |
| } |
| } |
| |
| private: |
| std::function<void(rtc::scoped_refptr<RtpTransceiverInterface>)> |
| on_track_callback_; |
| std::function<void()> on_connected_callback_; |
| }; |
| |
| void ValidateP2PSimulcastParams( |
| const std::vector<std::unique_ptr<PeerConfigurer>>& peers) { |
| for (size_t i = 0; i < peers.size(); ++i) { |
| Params* params = peers[i]->params(); |
| ConfigurableParams* configurable_params = peers[i]->configurable_params(); |
| for (const VideoConfig& video_config : configurable_params->video_configs) { |
| if (video_config.simulcast_config) { |
| // When we simulate SFU we support only one video codec. |
| RTC_CHECK_EQ(params->video_codecs.size(), 1) |
| << "Only 1 video codec is supported when simulcast is enabled in " |
| << "at least 1 video config"; |
| } |
| } |
| } |
| } |
| |
| } // namespace |
| |
| PeerConnectionE2EQualityTest::PeerConnectionE2EQualityTest( |
| std::string test_case_name, |
| TimeController& time_controller, |
| std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer, |
| std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer) |
| : PeerConnectionE2EQualityTest(std::move(test_case_name), |
| time_controller, |
| std::move(audio_quality_analyzer), |
| std::move(video_quality_analyzer), |
| /*metrics_logger_=*/nullptr) {} |
| |
| PeerConnectionE2EQualityTest::PeerConnectionE2EQualityTest( |
| std::string test_case_name, |
| TimeController& time_controller, |
| std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer, |
| std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer, |
| test::MetricsLogger* metrics_logger) |
| : time_controller_(time_controller), |
| task_queue_factory_(time_controller_.CreateTaskQueueFactory()), |
| test_case_name_(std::move(test_case_name)), |
| executor_(std::make_unique<TestActivitiesExecutor>( |
| time_controller_.GetClock())), |
| metrics_logger_(metrics_logger) { |
| // Create default video quality analyzer. We will always create an analyzer, |
| // even if there are no video streams, because it will be installed into video |
| // encoder/decoder factories. |
| if (video_quality_analyzer == nullptr) { |
| video_quality_analyzer = std::make_unique<DefaultVideoQualityAnalyzer>( |
| time_controller_.GetClock(), metrics_logger_); |
| } |
| if (field_trial::IsEnabled("WebRTC-VideoFrameTrackingIdAdvertised")) { |
| encoded_image_data_propagator_ = |
| std::make_unique<VideoFrameTrackingIdInjector>(); |
| } else { |
| encoded_image_data_propagator_ = |
| std::make_unique<SingleProcessEncodedImageDataInjector>(); |
| } |
| video_quality_analyzer_injection_helper_ = |
| std::make_unique<VideoQualityAnalyzerInjectionHelper>( |
| time_controller_.GetClock(), std::move(video_quality_analyzer), |
| encoded_image_data_propagator_.get(), |
| encoded_image_data_propagator_.get()); |
| |
| if (audio_quality_analyzer == nullptr) { |
| audio_quality_analyzer = |
| std::make_unique<DefaultAudioQualityAnalyzer>(metrics_logger_); |
| } |
| audio_quality_analyzer_.swap(audio_quality_analyzer); |
| } |
| |
| void PeerConnectionE2EQualityTest::ExecuteAt( |
| TimeDelta target_time_since_start, |
| std::function<void(TimeDelta)> func) { |
| executor_->ScheduleActivity(target_time_since_start, absl::nullopt, func); |
| } |
| |
| void PeerConnectionE2EQualityTest::ExecuteEvery( |
| TimeDelta initial_delay_since_start, |
| TimeDelta interval, |
| std::function<void(TimeDelta)> func) { |
| executor_->ScheduleActivity(initial_delay_since_start, interval, func); |
| } |
| |
| void PeerConnectionE2EQualityTest::AddQualityMetricsReporter( |
| std::unique_ptr<QualityMetricsReporter> quality_metrics_reporter) { |
| quality_metrics_reporters_.push_back(std::move(quality_metrics_reporter)); |
| } |
| |
| PeerConnectionE2EQualityTest::PeerHandle* PeerConnectionE2EQualityTest::AddPeer( |
| std::unique_ptr<PeerConfigurer> configurer) { |
| peer_configurations_.push_back(std::move(configurer)); |
| peer_handles_.push_back(PeerHandleImpl()); |
| return &peer_handles_.back(); |
| } |
| |
| void PeerConnectionE2EQualityTest::Run(RunParams run_params) { |
| webrtc::webrtc_pc_e2e::PeerParamsPreprocessor params_preprocessor; |
| for (auto& peer_configuration : peer_configurations_) { |
| params_preprocessor.SetDefaultValuesForMissingParams(*peer_configuration); |
| params_preprocessor.ValidateParams(*peer_configuration); |
| } |
| ValidateP2PSimulcastParams(peer_configurations_); |
| RTC_CHECK_EQ(peer_configurations_.size(), 2) |
| << "Only peer to peer calls are allowed, please add 2 peers"; |
| |
| std::unique_ptr<PeerConfigurer> alice_configurer = |
| std::move(peer_configurations_[0]); |
| std::unique_ptr<PeerConfigurer> bob_configurer = |
| std::move(peer_configurations_[1]); |
| peer_configurations_.clear(); |
| |
| for (size_t i = 0; |
| i < bob_configurer->configurable_params()->video_configs.size(); ++i) { |
| // We support simulcast only from caller. |
| RTC_CHECK(!bob_configurer->configurable_params() |
| ->video_configs[i] |
| .simulcast_config) |
| << "Only simulcast stream from first peer is supported"; |
| } |
| |
| test::ScopedFieldTrials field_trials(GetFieldTrials(run_params)); |
| |
| // Print test summary |
| RTC_LOG(LS_INFO) |
| << "Media quality test: " << *alice_configurer->params()->name |
| << " will make a call to " << *bob_configurer->params()->name |
| << " with media video=" |
| << !alice_configurer->configurable_params()->video_configs.empty() |
| << "; audio=" << alice_configurer->params()->audio_config.has_value() |
| << ". " << *bob_configurer->params()->name |
| << " will respond with media video=" |
| << !bob_configurer->configurable_params()->video_configs.empty() |
| << "; audio=" << bob_configurer->params()->audio_config.has_value(); |
| |
| const std::unique_ptr<rtc::Thread> signaling_thread = |
| time_controller_.CreateThread(kSignalThreadName); |
| media_helper_ = std::make_unique<MediaHelper>( |
| video_quality_analyzer_injection_helper_.get(), task_queue_factory_.get(), |
| time_controller_.GetClock()); |
| |
| // Create a `task_queue_`. |
| task_queue_ = std::make_unique<webrtc::TaskQueueForTest>( |
| time_controller_.GetTaskQueueFactory()->CreateTaskQueue( |
| "pc_e2e_quality_test", webrtc::TaskQueueFactory::Priority::NORMAL)); |
| |
| // Create call participants: Alice and Bob. |
| // Audio streams are intercepted in AudioDeviceModule, so if it is required to |
| // catch output of Alice's stream, Alice's output_dump_file_name should be |
| // passed to Bob's TestPeer setup as audio output file name. |
| absl::optional<RemotePeerAudioConfig> alice_remote_audio_config = |
| RemotePeerAudioConfig::Create(bob_configurer->params()->audio_config); |
| absl::optional<RemotePeerAudioConfig> bob_remote_audio_config = |
| RemotePeerAudioConfig::Create(alice_configurer->params()->audio_config); |
| // Copy Alice and Bob video configs, subscriptions and names to correctly pass |
| // them into lambdas. |
| VideoSubscription alice_subscription = |
| alice_configurer->configurable_params()->video_subscription; |
| std::vector<VideoConfig> alice_video_configs = |
| alice_configurer->configurable_params()->video_configs; |
| std::string alice_name = alice_configurer->params()->name.value(); |
| VideoSubscription bob_subscription = |
| alice_configurer->configurable_params()->video_subscription; |
| std::vector<VideoConfig> bob_video_configs = |
| bob_configurer->configurable_params()->video_configs; |
| std::string bob_name = bob_configurer->params()->name.value(); |
| |
| TestPeerFactory test_peer_factory( |
| signaling_thread.get(), time_controller_, |
| video_quality_analyzer_injection_helper_.get(), task_queue_.get()); |
| alice_ = test_peer_factory.CreateTestPeer( |
| std::move(alice_configurer), |
| std::make_unique<FixturePeerConnectionObserver>( |
| [this, alice_name, alice_subscription, bob_video_configs]( |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver) { |
| OnTrackCallback(alice_name, alice_subscription, transceiver, |
| bob_video_configs); |
| }, |
| [this]() { StartVideo(alice_video_sources_); }), |
| alice_remote_audio_config, run_params.echo_emulation_config); |
| bob_ = test_peer_factory.CreateTestPeer( |
| std::move(bob_configurer), |
| std::make_unique<FixturePeerConnectionObserver>( |
| [this, bob_name, bob_subscription, alice_video_configs]( |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver) { |
| OnTrackCallback(bob_name, bob_subscription, transceiver, |
| alice_video_configs); |
| }, |
| [this]() { StartVideo(bob_video_sources_); }), |
| bob_remote_audio_config, run_params.echo_emulation_config); |
| |
| int num_cores = CpuInfo::DetectNumberOfCores(); |
| RTC_DCHECK_GE(num_cores, 1); |
| |
| int video_analyzer_threads = |
| num_cores - kPeerConnectionUsedThreads - kFrameworkUsedThreads; |
| if (video_analyzer_threads <= 0) { |
| video_analyzer_threads = 1; |
| } |
| video_analyzer_threads = |
| std::min(video_analyzer_threads, kMaxVideoAnalyzerThreads); |
| RTC_LOG(LS_INFO) << "video_analyzer_threads=" << video_analyzer_threads; |
| quality_metrics_reporters_.push_back( |
| std::make_unique<VideoQualityMetricsReporter>(time_controller_.GetClock(), |
| metrics_logger_)); |
| quality_metrics_reporters_.push_back( |
| std::make_unique<CrossMediaMetricsReporter>(metrics_logger_)); |
| |
| video_quality_analyzer_injection_helper_->Start( |
| test_case_name_, |
| std::vector<std::string>{alice_->params().name.value(), |
| bob_->params().name.value()}, |
| video_analyzer_threads); |
| audio_quality_analyzer_->Start(test_case_name_, &analyzer_helper_); |
| for (auto& reporter : quality_metrics_reporters_) { |
| reporter->Start(test_case_name_, &analyzer_helper_); |
| } |
| |
| // Start RTCEventLog recording if requested. |
| if (alice_->params().rtc_event_log_path) { |
| auto alice_rtc_event_log = std::make_unique<webrtc::RtcEventLogOutputFile>( |
| alice_->params().rtc_event_log_path.value()); |
| alice_->pc()->StartRtcEventLog(std::move(alice_rtc_event_log), |
| webrtc::RtcEventLog::kImmediateOutput); |
| } |
| if (bob_->params().rtc_event_log_path) { |
| auto bob_rtc_event_log = std::make_unique<webrtc::RtcEventLogOutputFile>( |
| bob_->params().rtc_event_log_path.value()); |
| bob_->pc()->StartRtcEventLog(std::move(bob_rtc_event_log), |
| webrtc::RtcEventLog::kImmediateOutput); |
| } |
| |
| // Setup alive logging. It is done to prevent test infra to think that test is |
| // dead. |
| RepeatingTaskHandle::DelayedStart(task_queue_->Get(), |
| kAliveMessageLogInterval, []() { |
| std::printf("Test is still running...\n"); |
| return kAliveMessageLogInterval; |
| }); |
| |
| RTC_LOG(LS_INFO) << "Configuration is done. Now " << *alice_->params().name |
| << " is calling to " << *bob_->params().name << "..."; |
| |
| // Setup stats poller. |
| std::vector<StatsObserverInterface*> observers = { |
| audio_quality_analyzer_.get(), |
| video_quality_analyzer_injection_helper_.get()}; |
| for (auto& reporter : quality_metrics_reporters_) { |
| observers.push_back(reporter.get()); |
| } |
| StatsPoller stats_poller(observers, |
| std::map<std::string, StatsProvider*>{ |
| {*alice_->params().name, alice_.get()}, |
| {*bob_->params().name, bob_.get()}}); |
| executor_->ScheduleActivity(TimeDelta::Zero(), kStatsUpdateInterval, |
| [&stats_poller](TimeDelta) { |
| stats_poller.PollStatsAndNotifyObservers(); |
| }); |
| |
| // Setup call. |
| SendTask(signaling_thread.get(), |
| [this, &run_params] { SetupCallOnSignalingThread(run_params); }); |
| std::unique_ptr<SignalingInterceptor> signaling_interceptor = |
| CreateSignalingInterceptor(run_params); |
| // Connect peers. |
| SendTask(signaling_thread.get(), [this, &signaling_interceptor] { |
| ExchangeOfferAnswer(signaling_interceptor.get()); |
| }); |
| WaitUntilIceCandidatesGathered(signaling_thread.get()); |
| |
| SendTask(signaling_thread.get(), [this, &signaling_interceptor] { |
| ExchangeIceCandidates(signaling_interceptor.get()); |
| }); |
| WaitUntilPeersAreConnected(signaling_thread.get()); |
| |
| executor_->Start(task_queue_.get()); |
| Timestamp start_time = Now(); |
| |
| bool is_quick_test_enabled = absl::GetFlag(FLAGS_webrtc_quick_perf_test); |
| if (is_quick_test_enabled) { |
| time_controller_.AdvanceTime(kQuickTestModeRunDuration); |
| } else { |
| time_controller_.AdvanceTime(run_params.run_duration); |
| } |
| |
| RTC_LOG(LS_INFO) << "Test is done, initiating disconnect sequence."; |
| |
| // Stop all client started tasks to prevent their access to any call related |
| // objects after these objects will be destroyed during call tear down. |
| executor_->Stop(); |
| // There is no guarantee, that last stats collection will happen at the end |
| // of the call, so we force it after executor, which is among others is doing |
| // stats collection, was stopped. |
| task_queue_->SendTask([&stats_poller]() { |
| // Get final end-of-call stats. |
| stats_poller.PollStatsAndNotifyObservers(); |
| }); |
| // We need to detach AEC dumping from peers, because dump uses `task_queue_` |
| // inside. |
| alice_->DetachAecDump(); |
| bob_->DetachAecDump(); |
| // Tear down the call. |
| SendTask(signaling_thread.get(), [this] { TearDownCallOnSignalingThread(); }); |
| |
| Timestamp end_time = Now(); |
| RTC_LOG(LS_INFO) << "All peers are disconnected."; |
| { |
| MutexLock lock(&lock_); |
| real_test_duration_ = end_time - start_time; |
| } |
| |
| ReportGeneralTestResults(); |
| audio_quality_analyzer_->Stop(); |
| video_quality_analyzer_injection_helper_->Stop(); |
| for (auto& reporter : quality_metrics_reporters_) { |
| reporter->StopAndReportResults(); |
| } |
| |
| // Reset `task_queue_` after test to cleanup. |
| task_queue_.reset(); |
| |
| alice_ = nullptr; |
| bob_ = nullptr; |
| // Ensuring that TestVideoCapturerVideoTrackSource are destroyed on the right |
| // thread. |
| RTC_CHECK(alice_video_sources_.empty()); |
| RTC_CHECK(bob_video_sources_.empty()); |
| } |
| |
| std::string PeerConnectionE2EQualityTest::GetFieldTrials( |
| const RunParams& run_params) { |
| std::vector<absl::string_view> default_field_trials = { |
| kUseStandardsBytesStats}; |
| if (run_params.enable_flex_fec_support) { |
| default_field_trials.push_back(kFlexFecEnabledFieldTrials); |
| } |
| rtc::StringBuilder sb; |
| sb << field_trial::GetFieldTrialString(); |
| for (const absl::string_view& field_trial : default_field_trials) { |
| sb << field_trial; |
| } |
| return sb.Release(); |
| } |
| |
| void PeerConnectionE2EQualityTest::OnTrackCallback( |
| absl::string_view peer_name, |
| VideoSubscription peer_subscription, |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver, |
| std::vector<VideoConfig> remote_video_configs) { |
| const rtc::scoped_refptr<MediaStreamTrackInterface>& track = |
| transceiver->receiver()->track(); |
| RTC_CHECK_EQ(transceiver->receiver()->stream_ids().size(), 2) |
| << "Expected 2 stream ids: 1st - sync group, 2nd - unique stream label"; |
| std::string sync_group = transceiver->receiver()->stream_ids()[0]; |
| std::string stream_label = transceiver->receiver()->stream_ids()[1]; |
| analyzer_helper_.AddTrackToStreamMapping(track->id(), peer_name, stream_label, |
| sync_group); |
| if (track->kind() != MediaStreamTrackInterface::kVideoKind) { |
| return; |
| } |
| |
| // It is safe to cast here, because it is checked above that |
| // track->kind() is kVideoKind. |
| auto* video_track = static_cast<VideoTrackInterface*>(track.get()); |
| std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> video_sink = |
| video_quality_analyzer_injection_helper_->CreateVideoSink( |
| peer_name, peer_subscription, /*report_infra_stats=*/false); |
| video_track->AddOrUpdateSink(video_sink.get(), rtc::VideoSinkWants()); |
| output_video_sinks_.push_back(std::move(video_sink)); |
| } |
| |
| void PeerConnectionE2EQualityTest::SetupCallOnSignalingThread( |
| const RunParams& run_params) { |
| // We need receive-only transceivers for Bob's media stream, so there will |
| // be media section in SDP for that streams in Alice's offer, because it is |
| // forbidden to add new media sections in answer in Unified Plan. |
| RtpTransceiverInit receive_only_transceiver_init; |
| receive_only_transceiver_init.direction = RtpTransceiverDirection::kRecvOnly; |
| int alice_transceivers_counter = 0; |
| if (bob_->params().audio_config) { |
| // Setup receive audio transceiver if Bob has audio to send. If we'll need |
| // multiple audio streams, then we need transceiver for each Bob's audio |
| // stream. |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result = |
| alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_AUDIO, |
| receive_only_transceiver_init); |
| RTC_CHECK(result.ok()); |
| alice_transceivers_counter++; |
| } |
| |
| size_t alice_video_transceivers_non_simulcast_counter = 0; |
| for (auto& video_config : alice_->configurable_params().video_configs) { |
| RtpTransceiverInit transceiver_params; |
| if (video_config.simulcast_config) { |
| transceiver_params.direction = RtpTransceiverDirection::kSendOnly; |
| // Because simulcast enabled `alice_->params().video_codecs` has only 1 |
| // element. |
| if (alice_->params().video_codecs[0].name == cricket::kVp8CodecName) { |
| // For Vp8 simulcast we need to add as many RtpEncodingParameters to the |
| // track as many simulcast streams requested. If they specified in |
| // `video_config.simulcast_config` it should be copied from there. |
| for (int i = 0; |
| i < video_config.simulcast_config->simulcast_streams_count; ++i) { |
| RtpEncodingParameters enc_params; |
| if (!video_config.encoding_params.empty()) { |
| enc_params = video_config.encoding_params[i]; |
| } |
| // We need to be sure, that all rids will be unique with all mids. |
| enc_params.rid = std::to_string(alice_transceivers_counter) + "000" + |
| std::to_string(i); |
| transceiver_params.send_encodings.push_back(enc_params); |
| } |
| } |
| } else { |
| transceiver_params.direction = RtpTransceiverDirection::kSendRecv; |
| RtpEncodingParameters enc_params; |
| if (video_config.encoding_params.size() == 1) { |
| enc_params = video_config.encoding_params[0]; |
| } |
| transceiver_params.send_encodings.push_back(enc_params); |
| |
| alice_video_transceivers_non_simulcast_counter++; |
| } |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result = |
| alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO, |
| transceiver_params); |
| RTC_CHECK(result.ok()); |
| |
| alice_transceivers_counter++; |
| } |
| |
| // Add receive only transceivers in case Bob has more video_configs than |
| // Alice. |
| for (size_t i = alice_video_transceivers_non_simulcast_counter; |
| i < bob_->configurable_params().video_configs.size(); ++i) { |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result = |
| alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO, |
| receive_only_transceiver_init); |
| RTC_CHECK(result.ok()); |
| alice_transceivers_counter++; |
| } |
| |
| // Then add media for Alice and Bob |
| media_helper_->MaybeAddAudio(alice_.get()); |
| alice_video_sources_ = media_helper_->MaybeAddVideo(alice_.get()); |
| media_helper_->MaybeAddAudio(bob_.get()); |
| bob_video_sources_ = media_helper_->MaybeAddVideo(bob_.get()); |
| |
| SetPeerCodecPreferences(alice_.get()); |
| SetPeerCodecPreferences(bob_.get()); |
| } |
| |
| void PeerConnectionE2EQualityTest::TearDownCallOnSignalingThread() { |
| TearDownCall(); |
| } |
| |
| void PeerConnectionE2EQualityTest::SetPeerCodecPreferences(TestPeer* peer) { |
| std::vector<RtpCodecCapability> with_rtx_video_capabilities = |
| FilterVideoCodecCapabilities( |
| peer->params().video_codecs, true, peer->params().use_ulp_fec, |
| peer->params().use_flex_fec, |
| peer->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO) |
| .codecs); |
| std::vector<RtpCodecCapability> without_rtx_video_capabilities = |
| FilterVideoCodecCapabilities( |
| peer->params().video_codecs, false, peer->params().use_ulp_fec, |
| peer->params().use_flex_fec, |
| peer->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO) |
| .codecs); |
| |
| // Set codecs for transceivers |
| for (auto transceiver : peer->pc()->GetTransceivers()) { |
| if (transceiver->media_type() == cricket::MediaType::MEDIA_TYPE_VIDEO) { |
| if (transceiver->sender()->init_send_encodings().size() > 1) { |
| // If transceiver's sender has more then 1 send encodings, it means it |
| // has multiple simulcast streams, so we need disable RTX on it. |
| RTCError result = |
| transceiver->SetCodecPreferences(without_rtx_video_capabilities); |
| RTC_CHECK(result.ok()); |
| } else { |
| RTCError result = |
| transceiver->SetCodecPreferences(with_rtx_video_capabilities); |
| RTC_CHECK(result.ok()); |
| } |
| } |
| } |
| } |
| |
| std::unique_ptr<SignalingInterceptor> |
| PeerConnectionE2EQualityTest::CreateSignalingInterceptor( |
| const RunParams& run_params) { |
| std::map<std::string, int> stream_label_to_simulcast_streams_count; |
| // We add only Alice here, because simulcast/svc is supported only from the |
| // first peer. |
| for (auto& video_config : alice_->configurable_params().video_configs) { |
| if (video_config.simulcast_config) { |
| stream_label_to_simulcast_streams_count.insert( |
| {*video_config.stream_label, |
| video_config.simulcast_config->simulcast_streams_count}); |
| } |
| } |
| PatchingParams patching_params(run_params.use_conference_mode, |
| stream_label_to_simulcast_streams_count); |
| return std::make_unique<SignalingInterceptor>(patching_params); |
| } |
| |
| void PeerConnectionE2EQualityTest::WaitUntilIceCandidatesGathered( |
| rtc::Thread* signaling_thread) { |
| ASSERT_TRUE(time_controller_.Wait( |
| [&]() { |
| bool result; |
| SendTask(signaling_thread, [&]() { |
| result = alice_->IsIceGatheringDone() && bob_->IsIceGatheringDone(); |
| }); |
| return result; |
| }, |
| 2 * kDefaultTimeout)); |
| } |
| |
| void PeerConnectionE2EQualityTest::WaitUntilPeersAreConnected( |
| rtc::Thread* signaling_thread) { |
| // This means that ICE and DTLS are connected. |
| alice_connected_ = time_controller_.Wait( |
| [&]() { |
| bool result; |
| SendTask(signaling_thread, [&] { result = alice_->IsIceConnected(); }); |
| return result; |
| }, |
| kDefaultTimeout); |
| bob_connected_ = time_controller_.Wait( |
| [&]() { |
| bool result; |
| SendTask(signaling_thread, [&] { result = bob_->IsIceConnected(); }); |
| return result; |
| }, |
| kDefaultTimeout); |
| } |
| |
| void PeerConnectionE2EQualityTest::ExchangeOfferAnswer( |
| SignalingInterceptor* signaling_interceptor) { |
| std::string log_output; |
| |
| auto offer = alice_->CreateOffer(); |
| RTC_CHECK(offer); |
| offer->ToString(&log_output); |
| RTC_LOG(LS_INFO) << "Original offer: " << log_output; |
| LocalAndRemoteSdp patch_result = signaling_interceptor->PatchOffer( |
| std::move(offer), alice_->params().video_codecs[0]); |
| patch_result.local_sdp->ToString(&log_output); |
| RTC_LOG(LS_INFO) << "Offer to set as local description: " << log_output; |
| patch_result.remote_sdp->ToString(&log_output); |
| RTC_LOG(LS_INFO) << "Offer to set as remote description: " << log_output; |
| |
| bool set_local_offer = |
| alice_->SetLocalDescription(std::move(patch_result.local_sdp)); |
| RTC_CHECK(set_local_offer); |
| bool set_remote_offer = |
| bob_->SetRemoteDescription(std::move(patch_result.remote_sdp)); |
| RTC_CHECK(set_remote_offer); |
| auto answer = bob_->CreateAnswer(); |
| RTC_CHECK(answer); |
| answer->ToString(&log_output); |
| RTC_LOG(LS_INFO) << "Original answer: " << log_output; |
| patch_result = signaling_interceptor->PatchAnswer( |
| std::move(answer), bob_->params().video_codecs[0]); |
| patch_result.local_sdp->ToString(&log_output); |
| RTC_LOG(LS_INFO) << "Answer to set as local description: " << log_output; |
| patch_result.remote_sdp->ToString(&log_output); |
| RTC_LOG(LS_INFO) << "Answer to set as remote description: " << log_output; |
| |
| bool set_local_answer = |
| bob_->SetLocalDescription(std::move(patch_result.local_sdp)); |
| RTC_CHECK(set_local_answer); |
| bool set_remote_answer = |
| alice_->SetRemoteDescription(std::move(patch_result.remote_sdp)); |
| RTC_CHECK(set_remote_answer); |
| } |
| |
| void PeerConnectionE2EQualityTest::ExchangeIceCandidates( |
| SignalingInterceptor* signaling_interceptor) { |
| // Connect an ICE candidate pairs. |
| std::vector<std::unique_ptr<IceCandidateInterface>> alice_candidates = |
| signaling_interceptor->PatchOffererIceCandidates( |
| alice_->observer()->GetAllCandidates()); |
| for (auto& candidate : alice_candidates) { |
| std::string candidate_str; |
| RTC_CHECK(candidate->ToString(&candidate_str)); |
| RTC_LOG(LS_INFO) << *alice_->params().name |
| << " ICE candidate(mid= " << candidate->sdp_mid() |
| << "): " << candidate_str; |
| } |
| ASSERT_TRUE(bob_->AddIceCandidates(std::move(alice_candidates))); |
| std::vector<std::unique_ptr<IceCandidateInterface>> bob_candidates = |
| signaling_interceptor->PatchAnswererIceCandidates( |
| bob_->observer()->GetAllCandidates()); |
| for (auto& candidate : bob_candidates) { |
| std::string candidate_str; |
| RTC_CHECK(candidate->ToString(&candidate_str)); |
| RTC_LOG(LS_INFO) << *bob_->params().name |
| << " ICE candidate(mid= " << candidate->sdp_mid() |
| << "): " << candidate_str; |
| } |
| ASSERT_TRUE(alice_->AddIceCandidates(std::move(bob_candidates))); |
| } |
| |
| void PeerConnectionE2EQualityTest::StartVideo( |
| const std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>& |
| sources) { |
| for (auto& source : sources) { |
| if (source->state() != MediaSourceInterface::SourceState::kLive) { |
| source->Start(); |
| } |
| } |
| } |
| |
| void PeerConnectionE2EQualityTest::TearDownCall() { |
| for (const auto& video_source : alice_video_sources_) { |
| video_source->Stop(); |
| } |
| for (const auto& video_source : bob_video_sources_) { |
| video_source->Stop(); |
| } |
| |
| alice_video_sources_.clear(); |
| bob_video_sources_.clear(); |
| |
| alice_->Close(); |
| bob_->Close(); |
| |
| media_helper_ = nullptr; |
| } |
| |
| void PeerConnectionE2EQualityTest::ReportGeneralTestResults() { |
| // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey. |
| metrics_logger_->LogSingleValueMetric( |
| *alice_->params().name + "_connected", test_case_name_, alice_connected_, |
| Unit::kUnitless, ImprovementDirection::kBiggerIsBetter, |
| {{MetricMetadataKey::kPeerMetadataKey, *alice_->params().name}, |
| {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}}); |
| // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey. |
| metrics_logger_->LogSingleValueMetric( |
| *bob_->params().name + "_connected", test_case_name_, bob_connected_, |
| Unit::kUnitless, ImprovementDirection::kBiggerIsBetter, |
| {{MetricMetadataKey::kPeerMetadataKey, *bob_->params().name}, |
| {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}}); |
| } |
| |
| Timestamp PeerConnectionE2EQualityTest::Now() const { |
| return time_controller_.GetClock()->CurrentTime(); |
| } |
| |
| } // namespace webrtc_pc_e2e |
| } // namespace webrtc |
