| /* |
| * 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 "api/media_stream_interface.h" |
| #include "api/test/create_network_emulation_manager.h" |
| #include "api/test/create_peer_connection_quality_test_frame_generator.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 "system_wrappers/include/field_trial.h" |
| #include "test/field_trial.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/pc/e2e/network_quality_metrics_reporter.h" |
| #include "test/testsupport/file_utils.h" |
| |
| namespace webrtc { |
| namespace webrtc_pc_e2e { |
| namespace { |
| |
| class PeerConnectionE2EQualityTestSmokeTest : public ::testing::Test { |
| public: |
| using PeerConfigurer = PeerConnectionE2EQualityTestFixture::PeerConfigurer; |
| using RunParams = PeerConnectionE2EQualityTestFixture::RunParams; |
| using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig; |
| using VideoCodecConfig = |
| PeerConnectionE2EQualityTestFixture::VideoCodecConfig; |
| using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig; |
| using ScreenShareConfig = |
| PeerConnectionE2EQualityTestFixture::ScreenShareConfig; |
| using ScrollingParams = PeerConnectionE2EQualityTestFixture::ScrollingParams; |
| using VideoSimulcastConfig = |
| PeerConnectionE2EQualityTestFixture::VideoSimulcastConfig; |
| using EchoEmulationConfig = |
| PeerConnectionE2EQualityTestFixture::EchoEmulationConfig; |
| |
| void RunTest(const std::string& test_case_name, |
| const RunParams& run_params, |
| rtc::FunctionView<void(PeerConfigurer*)> alice_configurer, |
| rtc::FunctionView<void(PeerConfigurer*)> bob_configurer) { |
| // Setup emulated network |
| std::unique_ptr<NetworkEmulationManager> network_emulation_manager = |
| CreateNetworkEmulationManager(); |
| |
| auto alice_network_behavior = |
| std::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( |
| std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())); |
| auto* 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 = |
| std::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()); |
| |
| auto fixture = CreatePeerConnectionE2EQualityTestFixture( |
| test_case_name, /*audio_quality_analyzer=*/nullptr, |
| 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}); |
| EmulatedNetworkManagerInterface* bob_network = |
| network_emulation_manager->CreateEmulatedNetworkManagerInterface( |
| {bob_endpoint}); |
| |
| fixture->AddPeer(alice_network->network_thread(), |
| alice_network->network_manager(), alice_configurer); |
| fixture->AddPeer(bob_network->network_thread(), |
| bob_network->network_manager(), bob_configurer); |
| fixture->AddQualityMetricsReporter( |
| std::make_unique<NetworkQualityMetricsReporter>(alice_network, |
| bob_network)); |
| |
| fixture->Run(run_params); |
| |
| EXPECT_GE(fixture->GetRealTestDuration(), run_params.run_duration); |
| for (auto stream_label : video_analyzer_ptr->GetKnownVideoStreams()) { |
| FrameCounters stream_conters = |
| video_analyzer_ptr->GetPerStreamCounters().at(stream_label); |
| // On some devices the pipeline can be too slow, so we actually can't |
| // force real constraints here. Lets just check, that at least 1 |
| // frame passed whole pipeline. |
| int64_t expected_min_fps = run_params.run_duration.seconds() * 30; |
| EXPECT_GE(stream_conters.captured, expected_min_fps); |
| 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 |
| |
| // 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_Smoke DISABLED_Smoke |
| #else |
| #define MAYBE_Smoke Smoke |
| #endif |
| TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Smoke) { |
| RunParams run_params(TimeDelta::Seconds(2)); |
| run_params.video_codecs = { |
| VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})}; |
| run_params.use_flex_fec = true; |
| run_params.use_ulp_fec = true; |
| run_params.video_encoder_bitrate_multiplier = 1.1; |
| test::ScopedFieldTrials field_trials( |
| std::string(field_trial::GetFieldTrialString()) + |
| "WebRTC-UseStandardBytesStats/Enabled/"); |
| RunTest( |
| "smoke", run_params, |
| [](PeerConfigurer* alice) { |
| VideoConfig video(640, 360, 30); |
| video.stream_label = "alice-video"; |
| video.sync_group = "alice-media"; |
| alice->AddVideoConfig(std::move(video)); |
| |
| AudioConfig audio; |
| audio.stream_label = "alice-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_alice_source", "wav"); |
| audio.sampling_frequency_in_hz = 48000; |
| audio.sync_group = "alice-media"; |
| alice->SetAudioConfig(std::move(audio)); |
| }, |
| [](PeerConfigurer* charlie) { |
| charlie->SetName("charlie"); |
| VideoConfig video(640, 360, 30); |
| video.stream_label = "charlie-video"; |
| video.temporal_layers_count = 2; |
| charlie->AddVideoConfig(std::move(video)); |
| |
| VideoConfig screenshare(640, 360, 30); |
| screenshare.stream_label = "charlie-screenshare"; |
| screenshare.content_hint = VideoTrackInterface::ContentHint::kText; |
| ScreenShareConfig screen_share_config = |
| ScreenShareConfig(TimeDelta::Seconds(2)); |
| screen_share_config.scrolling_params = ScrollingParams( |
| TimeDelta::Millis(1800), kDefaultSlidesWidth, kDefaultSlidesHeight); |
| auto screen_share_frame_generator = |
| CreateScreenShareFrameGenerator(screenshare, screen_share_config); |
| charlie->AddVideoConfig(std::move(screenshare), |
| std::move(screen_share_frame_generator)); |
| |
| AudioConfig audio; |
| audio.stream_label = "charlie-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_bob_source", "wav"); |
| charlie->SetAudioConfig(std::move(audio)); |
| }); |
| } |
| |
| // 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_Echo DISABLED_Echo |
| #else |
| #define MAYBE_Echo Echo |
| #endif |
| TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Echo) { |
| RunParams run_params(TimeDelta::Seconds(2)); |
| run_params.echo_emulation_config = EchoEmulationConfig(); |
| RunTest( |
| "smoke", run_params, |
| [](PeerConfigurer* alice) { |
| AudioConfig audio; |
| audio.stream_label = "alice-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_alice_source", "wav"); |
| audio.sampling_frequency_in_hz = 48000; |
| alice->SetAudioConfig(std::move(audio)); |
| }, |
| [](PeerConfigurer* bob) { |
| AudioConfig audio; |
| audio.stream_label = "bob-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_bob_source", "wav"); |
| bob->SetAudioConfig(std::move(audio)); |
| }); |
| } |
| |
| // 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_Simulcast DISABLED_Simulcast |
| #else |
| #define MAYBE_Simulcast Simulcast |
| #endif |
| TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Simulcast) { |
| RunParams run_params(TimeDelta::Seconds(2)); |
| run_params.video_codecs = {VideoCodecConfig(cricket::kVp8CodecName)}; |
| RunTest( |
| "simulcast", run_params, |
| [](PeerConfigurer* alice) { |
| VideoConfig simulcast(1280, 720, 30); |
| simulcast.stream_label = "alice-simulcast"; |
| simulcast.simulcast_config = VideoSimulcastConfig(3, 0); |
| alice->AddVideoConfig(std::move(simulcast)); |
| |
| AudioConfig audio; |
| audio.stream_label = "alice-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_alice_source", "wav"); |
| alice->SetAudioConfig(std::move(audio)); |
| }, |
| [](PeerConfigurer* bob) { |
| VideoConfig video(640, 360, 30); |
| video.stream_label = "bob-video"; |
| bob->AddVideoConfig(std::move(video)); |
| |
| AudioConfig audio; |
| audio.stream_label = "bob-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_bob_source", "wav"); |
| bob->SetAudioConfig(std::move(audio)); |
| }); |
| } |
| |
| // 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_Svc DISABLED_Svc |
| #else |
| #define MAYBE_Svc Svc |
| #endif |
| TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Svc) { |
| RunParams run_params(TimeDelta::Seconds(2)); |
| run_params.video_codecs = {VideoCodecConfig(cricket::kVp9CodecName)}; |
| RunTest( |
| "simulcast", run_params, |
| [](PeerConfigurer* alice) { |
| VideoConfig simulcast(1280, 720, 30); |
| simulcast.stream_label = "alice-svc"; |
| // Because we have network with packets loss we can analyze only the |
| // highest spatial layer in SVC mode. |
| simulcast.simulcast_config = VideoSimulcastConfig(3, 2); |
| alice->AddVideoConfig(std::move(simulcast)); |
| |
| AudioConfig audio; |
| audio.stream_label = "alice-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_alice_source", "wav"); |
| alice->SetAudioConfig(std::move(audio)); |
| }, |
| [](PeerConfigurer* bob) { |
| VideoConfig video(640, 360, 30); |
| video.stream_label = "bob-video"; |
| bob->AddVideoConfig(std::move(video)); |
| |
| AudioConfig audio; |
| audio.stream_label = "bob-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_bob_source", "wav"); |
| bob->SetAudioConfig(std::move(audio)); |
| }); |
| } |
| |
| // 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_HighBitrate DISABLED_HighBitrate |
| #else |
| #define MAYBE_HighBitrate HighBitrate |
| #endif |
| TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_HighBitrate) { |
| RunParams run_params(TimeDelta::Seconds(2)); |
| run_params.video_codecs = { |
| VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})}; |
| |
| RunTest( |
| "smoke", run_params, |
| [](PeerConfigurer* alice) { |
| PeerConnectionInterface::BitrateParameters bitrate_params; |
| bitrate_params.current_bitrate_bps = 3'000'000; |
| bitrate_params.max_bitrate_bps = 3'000'000; |
| alice->SetBitrateParameters(bitrate_params); |
| VideoConfig video(800, 600, 30); |
| video.stream_label = "alice-video"; |
| video.min_encode_bitrate_bps = 500'000; |
| video.max_encode_bitrate_bps = 3'000'000; |
| alice->AddVideoConfig(std::move(video)); |
| |
| AudioConfig audio; |
| audio.stream_label = "alice-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_alice_source", "wav"); |
| audio.sampling_frequency_in_hz = 48000; |
| alice->SetAudioConfig(std::move(audio)); |
| }, |
| [](PeerConfigurer* bob) { |
| PeerConnectionInterface::BitrateParameters bitrate_params; |
| bitrate_params.current_bitrate_bps = 3'000'000; |
| bitrate_params.max_bitrate_bps = 3'000'000; |
| bob->SetBitrateParameters(bitrate_params); |
| VideoConfig video(800, 600, 30); |
| video.stream_label = "bob-video"; |
| video.min_encode_bitrate_bps = 500'000; |
| video.max_encode_bitrate_bps = 3'000'000; |
| bob->AddVideoConfig(std::move(video)); |
| |
| AudioConfig audio; |
| audio.stream_label = "bob-audio"; |
| audio.mode = AudioConfig::Mode::kFile; |
| audio.input_file_name = |
| test::ResourcePath("pc_quality_smoke_test_bob_source", "wav"); |
| bob->SetAudioConfig(std::move(audio)); |
| }); |
| } |
| |
| } // namespace webrtc_pc_e2e |
| } // namespace webrtc |
