| /* |
| * 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 <cstdint> |
| #include <memory> |
| |
| #include "absl/memory/memory.h" |
| #include "api/test/create_network_emulation_manager.h" |
| #include "api/test/create_peerconnection_quality_test_fixture.h" |
| #include "api/test/network_emulation_manager.h" |
| #include "api/test/peerconnection_quality_test_fixture.h" |
| #include "call/simulated_network.h" |
| #include "test/gtest.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/testsupport/file_utils.h" |
| |
| namespace webrtc { |
| namespace webrtc_pc_e2e { |
| |
| // IOS debug builds can be quite slow, disabling to avoid issues with timeouts. |
| #if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG) |
| #define MAYBE_RunWithEmulatedNetwork DISABLED_RunWithEmulatedNetwork |
| #else |
| #define MAYBE_RunWithEmulatedNetwork RunWithEmulatedNetwork |
| #endif |
| TEST(PeerConnectionE2EQualityTestSmokeTest, MAYBE_RunWithEmulatedNetwork) { |
| using PeerConfigurer = PeerConnectionE2EQualityTestFixture::PeerConfigurer; |
| using RunParams = PeerConnectionE2EQualityTestFixture::RunParams; |
| using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig; |
| using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig; |
| |
| // Setup emulated network |
| std::unique_ptr<NetworkEmulationManager> network_emulation_manager = |
| CreateNetworkEmulationManager(); |
| |
| auto alice_network_behavior = |
| absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()); |
| SimulatedNetwork* alice_network_behavior_ptr = alice_network_behavior.get(); |
| EmulatedNetworkNode* alice_node = |
| network_emulation_manager->CreateEmulatedNode( |
| std::move(alice_network_behavior)); |
| EmulatedNetworkNode* bob_node = network_emulation_manager->CreateEmulatedNode( |
| absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())); |
| EmulatedEndpoint* alice_endpoint = |
| network_emulation_manager->CreateEndpoint(EmulatedEndpointConfig()); |
| EmulatedEndpoint* bob_endpoint = |
| network_emulation_manager->CreateEndpoint(EmulatedEndpointConfig()); |
| network_emulation_manager->CreateRoute(alice_endpoint, {alice_node}, |
| bob_endpoint); |
| network_emulation_manager->CreateRoute(bob_endpoint, {bob_node}, |
| alice_endpoint); |
| |
| // Create analyzers. |
| std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer = |
| absl::make_unique<DefaultVideoQualityAnalyzer>(); |
| // This is only done for the sake of smoke testing. In general there should |
| // be no need to explicitly pull data from analyzers after the run. |
| auto* video_analyzer_ptr = |
| static_cast<DefaultVideoQualityAnalyzer*>(video_quality_analyzer.get()); |
| |
| std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer = |
| absl::make_unique<DefaultAudioQualityAnalyzer>(); |
| |
| auto fixture = CreatePeerConnectionE2EQualityTestFixture( |
| "smoke_test", std::move(audio_quality_analyzer), |
| std::move(video_quality_analyzer)); |
| fixture->ExecuteAt(TimeDelta::seconds(2), |
| [alice_network_behavior_ptr](TimeDelta) { |
| BuiltInNetworkBehaviorConfig config; |
| config.loss_percent = 5; |
| alice_network_behavior_ptr->SetConfig(config); |
| }); |
| |
| // Setup components. We need to provide rtc::NetworkManager compatible with |
| // emulated network layer. |
| EmulatedNetworkManagerInterface* alice_network = |
| network_emulation_manager->CreateEmulatedNetworkManagerInterface( |
| {alice_endpoint}); |
| fixture->AddPeer(alice_network->network_thread(), |
| alice_network->network_manager(), [](PeerConfigurer* alice) { |
| VideoConfig video_config(640, 360, 30); |
| video_config.stream_label = "alice-video"; |
| alice->AddVideoConfig(std::move(video_config)); |
| AudioConfig audio_config; |
| audio_config.stream_label = "alice-audio"; |
| audio_config.mode = AudioConfig::Mode::kFile; |
| audio_config.input_file_name = test::ResourcePath( |
| "pc_quality_smoke_test_alice_source", "wav"); |
| alice->SetAudioConfig(std::move(audio_config)); |
| }); |
| |
| EmulatedNetworkManagerInterface* bob_network = |
| network_emulation_manager->CreateEmulatedNetworkManagerInterface( |
| {bob_endpoint}); |
| fixture->AddPeer(bob_network->network_thread(), |
| bob_network->network_manager(), [](PeerConfigurer* bob) { |
| VideoConfig video_config(640, 360, 30); |
| video_config.stream_label = "bob-video"; |
| bob->AddVideoConfig(std::move(video_config)); |
| AudioConfig audio_config; |
| audio_config.stream_label = "bob-audio"; |
| audio_config.mode = AudioConfig::Mode::kFile; |
| audio_config.input_file_name = test::ResourcePath( |
| "pc_quality_smoke_test_bob_source", "wav"); |
| bob->SetAudioConfig(std::move(audio_config)); |
| }); |
| |
| RunParams run_params(TimeDelta::seconds(7)); |
| run_params.video_encoder_bitrate_multiplier = 1.1; |
| fixture->Run(run_params); |
| |
| for (auto stream_label : video_analyzer_ptr->GetKnownVideoStreams()) { |
| FrameCounters stream_conters = |
| video_analyzer_ptr->GetPerStreamCounters().at(stream_label); |
| // 150 = 30fps * 5s. On some devices pipeline can be too slow, so it can |
| // happen, that frames will stuck in the middle, so we actually can't force |
| // real constraints here, so lets just check, that at least 1 frame passed |
| // whole pipeline. |
| EXPECT_GE(stream_conters.captured, 150); |
| EXPECT_GE(stream_conters.pre_encoded, 1); |
| EXPECT_GE(stream_conters.encoded, 1); |
| EXPECT_GE(stream_conters.received, 1); |
| EXPECT_GE(stream_conters.decoded, 1); |
| EXPECT_GE(stream_conters.rendered, 1); |
| } |
| } |
| |
| } // namespace webrtc_pc_e2e |
| } // namespace webrtc |
