| /* |
| * Copyright 2012 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 "pc/peer_connection_factory.h" |
| |
| #include <algorithm> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/create_peerconnection_factory.h" |
| #include "api/data_channel_interface.h" |
| #include "api/jsep.h" |
| #include "api/media_stream_interface.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/test/mock_packet_socket_factory.h" |
| #include "api/video_codecs/video_decoder_factory_template.h" |
| #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template.h" |
| #include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h" |
| #include "media/base/fake_frame_source.h" |
| #include "media/engine/webrtc_media_engine.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "p2p/base/fake_port_allocator.h" |
| #include "p2p/base/port.h" |
| #include "p2p/base/port_allocator.h" |
| #include "p2p/base/port_interface.h" |
| #include "pc/test/fake_audio_capture_module.h" |
| #include "pc/test/fake_video_track_source.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/internal/default_socket_server.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/socket_address.h" |
| #include "rtc_base/time_utils.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/scoped_key_value_config.h" |
| |
| #ifdef WEBRTC_ANDROID |
| #include "pc/test/android_test_initializer.h" |
| #endif |
| #include "pc/test/fake_rtc_certificate_generator.h" |
| #include "pc/test/fake_video_track_renderer.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| using ::testing::_; |
| using ::testing::AtLeast; |
| using ::testing::InvokeWithoutArgs; |
| using ::testing::NiceMock; |
| using ::testing::Return; |
| using ::testing::UnorderedElementsAre; |
| |
| static const char kStunIceServer[] = "stun:stun.l.google.com:19302"; |
| static const char kTurnIceServer[] = "turn:test.com:1234"; |
| static const char kTurnIceServerWithTransport[] = |
| "turn:hello.com?transport=tcp"; |
| static const char kSecureTurnIceServer[] = "turns:hello.com?transport=tcp"; |
| static const char kSecureTurnIceServerWithoutTransportParam[] = |
| "turns:hello.com:443"; |
| static const char kSecureTurnIceServerWithoutTransportAndPortParam[] = |
| "turns:hello.com"; |
| static const char kTurnIceServerWithNoUsernameInUri[] = "turn:test.com:1234"; |
| static const char kTurnPassword[] = "turnpassword"; |
| static const int kDefaultStunPort = 3478; |
| static const int kDefaultStunTlsPort = 5349; |
| static const char kTurnUsername[] = "test"; |
| static const char kStunIceServerWithIPv4Address[] = "stun:1.2.3.4:1234"; |
| static const char kStunIceServerWithIPv4AddressWithoutPort[] = "stun:1.2.3.4"; |
| static const char kStunIceServerWithIPv6Address[] = "stun:[2401:fa00:4::]:1234"; |
| static const char kStunIceServerWithIPv6AddressWithoutPort[] = |
| "stun:[2401:fa00:4::]"; |
| static const char kTurnIceServerWithIPv6Address[] = "turn:[2401:fa00:4::]:1234"; |
| |
| class NullPeerConnectionObserver : public PeerConnectionObserver { |
| public: |
| virtual ~NullPeerConnectionObserver() = default; |
| void OnSignalingChange( |
| PeerConnectionInterface::SignalingState new_state) override {} |
| void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override {} |
| void OnRemoveStream( |
| rtc::scoped_refptr<MediaStreamInterface> stream) override {} |
| void OnDataChannel( |
| rtc::scoped_refptr<DataChannelInterface> data_channel) override {} |
| void OnRenegotiationNeeded() override {} |
| void OnIceConnectionChange( |
| PeerConnectionInterface::IceConnectionState new_state) override {} |
| void OnIceGatheringChange( |
| PeerConnectionInterface::IceGatheringState new_state) override {} |
| void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { |
| } |
| }; |
| |
| class MockNetworkManager : public rtc::NetworkManager { |
| public: |
| MOCK_METHOD(void, StartUpdating, (), (override)); |
| MOCK_METHOD(void, StopUpdating, (), (override)); |
| MOCK_METHOD(std::vector<const rtc::Network*>, |
| GetNetworks, |
| (), |
| (const override)); |
| MOCK_METHOD(std::vector<const rtc::Network*>, |
| GetAnyAddressNetworks, |
| (), |
| (override)); |
| }; |
| |
| class PeerConnectionFactoryTest : public ::testing::Test { |
| public: |
| PeerConnectionFactoryTest() |
| : socket_server_(rtc::CreateDefaultSocketServer()), |
| main_thread_(socket_server_.get()) {} |
| |
| private: |
| void SetUp() { |
| #ifdef WEBRTC_ANDROID |
| webrtc::InitializeAndroidObjects(); |
| #endif |
| // Use fake audio device module since we're only testing the interface |
| // level, and using a real one could make tests flaky e.g. when run in |
| // parallel. |
| factory_ = webrtc::CreatePeerConnectionFactory( |
| rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), |
| rtc::scoped_refptr<webrtc::AudioDeviceModule>( |
| FakeAudioCaptureModule::Create()), |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| std::make_unique<VideoEncoderFactoryTemplate< |
| LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter, |
| OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(), |
| std::make_unique<VideoDecoderFactoryTemplate< |
| LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter, |
| OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(), |
| nullptr /* audio_mixer */, nullptr /* audio_processing */); |
| |
| ASSERT_TRUE(factory_.get() != NULL); |
| packet_socket_factory_.reset( |
| new rtc::BasicPacketSocketFactory(socket_server_.get())); |
| port_allocator_.reset(new cricket::FakePortAllocator( |
| rtc::Thread::Current(), packet_socket_factory_.get(), &field_trials_)); |
| raw_port_allocator_ = port_allocator_.get(); |
| } |
| |
| protected: |
| void VerifyStunServers(cricket::ServerAddresses stun_servers) { |
| EXPECT_EQ(stun_servers, raw_port_allocator_->stun_servers()); |
| } |
| |
| void VerifyTurnServers(std::vector<cricket::RelayServerConfig> turn_servers) { |
| EXPECT_EQ(turn_servers.size(), raw_port_allocator_->turn_servers().size()); |
| for (size_t i = 0; i < turn_servers.size(); ++i) { |
| ASSERT_EQ(1u, turn_servers[i].ports.size()); |
| EXPECT_EQ(1u, raw_port_allocator_->turn_servers()[i].ports.size()); |
| EXPECT_EQ( |
| turn_servers[i].ports[0].address.ToString(), |
| raw_port_allocator_->turn_servers()[i].ports[0].address.ToString()); |
| EXPECT_EQ(turn_servers[i].ports[0].proto, |
| raw_port_allocator_->turn_servers()[i].ports[0].proto); |
| EXPECT_EQ(turn_servers[i].credentials.username, |
| raw_port_allocator_->turn_servers()[i].credentials.username); |
| EXPECT_EQ(turn_servers[i].credentials.password, |
| raw_port_allocator_->turn_servers()[i].credentials.password); |
| } |
| } |
| |
| void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) { |
| EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO); |
| EXPECT_FALSE(codec.name.empty()); |
| EXPECT_GT(codec.clock_rate, 0); |
| EXPECT_GT(codec.num_channels, 0); |
| } |
| |
| void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec, |
| bool sender) { |
| EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO); |
| EXPECT_FALSE(codec.name.empty()); |
| EXPECT_GT(codec.clock_rate, 0); |
| if (sender) { |
| if (codec.name == "VP8" || codec.name == "H264") { |
| EXPECT_THAT(codec.scalability_modes, |
| UnorderedElementsAre(webrtc::ScalabilityMode::kL1T1, |
| webrtc::ScalabilityMode::kL1T2, |
| webrtc::ScalabilityMode::kL1T3)) |
| << "Codec: " << codec.name; |
| } else if (codec.name == "VP9" || codec.name == "AV1") { |
| EXPECT_THAT( |
| codec.scalability_modes, |
| UnorderedElementsAre( |
| // clang-format off |
| webrtc::ScalabilityMode::kL1T1, |
| webrtc::ScalabilityMode::kL1T2, |
| webrtc::ScalabilityMode::kL1T3, |
| webrtc::ScalabilityMode::kL2T1, |
| webrtc::ScalabilityMode::kL2T1h, |
| webrtc::ScalabilityMode::kL2T1_KEY, |
| webrtc::ScalabilityMode::kL2T2, |
| webrtc::ScalabilityMode::kL2T2h, |
| webrtc::ScalabilityMode::kL2T2_KEY, |
| webrtc::ScalabilityMode::kL2T2_KEY_SHIFT, |
| webrtc::ScalabilityMode::kL2T3, |
| webrtc::ScalabilityMode::kL2T3h, |
| webrtc::ScalabilityMode::kL2T3_KEY, |
| webrtc::ScalabilityMode::kL3T1, |
| webrtc::ScalabilityMode::kL3T1h, |
| webrtc::ScalabilityMode::kL3T1_KEY, |
| webrtc::ScalabilityMode::kL3T2, |
| webrtc::ScalabilityMode::kL3T2h, |
| webrtc::ScalabilityMode::kL3T2_KEY, |
| webrtc::ScalabilityMode::kL3T3, |
| webrtc::ScalabilityMode::kL3T3h, |
| webrtc::ScalabilityMode::kL3T3_KEY, |
| webrtc::ScalabilityMode::kS2T1, |
| webrtc::ScalabilityMode::kS2T1h, |
| webrtc::ScalabilityMode::kS2T2, |
| webrtc::ScalabilityMode::kS2T2h, |
| webrtc::ScalabilityMode::kS2T3, |
| webrtc::ScalabilityMode::kS2T3h, |
| webrtc::ScalabilityMode::kS3T1, |
| webrtc::ScalabilityMode::kS3T1h, |
| webrtc::ScalabilityMode::kS3T2, |
| webrtc::ScalabilityMode::kS3T2h, |
| webrtc::ScalabilityMode::kS3T3, |
| webrtc::ScalabilityMode::kS3T3h) |
| // clang-format on |
| ) |
| << "Codec: " << codec.name; |
| } else { |
| EXPECT_TRUE(codec.scalability_modes.empty()); |
| } |
| } else { |
| EXPECT_TRUE(codec.scalability_modes.empty()); |
| } |
| } |
| |
| webrtc::test::ScopedKeyValueConfig field_trials_; |
| std::unique_ptr<rtc::SocketServer> socket_server_; |
| rtc::AutoSocketServerThread main_thread_; |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_; |
| NullPeerConnectionObserver observer_; |
| std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory_; |
| std::unique_ptr<cricket::FakePortAllocator> port_allocator_; |
| // Since the PC owns the port allocator after it's been initialized, |
| // this should only be used when known to be safe. |
| cricket::FakePortAllocator* raw_port_allocator_; |
| }; |
| |
| // Since there is no public PeerConnectionFactory API to control RTX usage, need |
| // to reconstruct factory with our own ConnectionContext. |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> |
| CreatePeerConnectionFactoryWithRtxDisabled() { |
| webrtc::PeerConnectionFactoryDependencies pcf_dependencies; |
| pcf_dependencies.signaling_thread = rtc::Thread::Current(); |
| pcf_dependencies.worker_thread = rtc::Thread::Current(); |
| pcf_dependencies.network_thread = rtc::Thread::Current(); |
| pcf_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); |
| pcf_dependencies.call_factory = CreateCallFactory(); |
| pcf_dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>(); |
| |
| cricket::MediaEngineDependencies media_dependencies; |
| media_dependencies.task_queue_factory = |
| pcf_dependencies.task_queue_factory.get(); |
| media_dependencies.adm = rtc::scoped_refptr<webrtc::AudioDeviceModule>( |
| FakeAudioCaptureModule::Create()); |
| media_dependencies.audio_encoder_factory = |
| webrtc::CreateBuiltinAudioEncoderFactory(); |
| media_dependencies.audio_decoder_factory = |
| webrtc::CreateBuiltinAudioDecoderFactory(); |
| media_dependencies.video_encoder_factory = |
| std::make_unique<VideoEncoderFactoryTemplate< |
| LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter, |
| OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(); |
| media_dependencies.video_decoder_factory = |
| std::make_unique<VideoDecoderFactoryTemplate< |
| LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter, |
| OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(), |
| media_dependencies.trials = pcf_dependencies.trials.get(); |
| pcf_dependencies.media_engine = |
| cricket::CreateMediaEngine(std::move(media_dependencies)); |
| |
| rtc::scoped_refptr<webrtc::ConnectionContext> context = |
| ConnectionContext::Create(&pcf_dependencies); |
| context->set_use_rtx(false); |
| return rtc::make_ref_counted<PeerConnectionFactory>(context, |
| &pcf_dependencies); |
| } |
| |
| // Verify creation of PeerConnection using internal ADM, video factory and |
| // internal libjingle threads. |
| // TODO(henrika): disabling this test since relying on real audio can result in |
| // flaky tests and focus on details that are out of scope for you might expect |
| // for a PeerConnectionFactory unit test. |
| // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7806 for details. |
| TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) { |
| #ifdef WEBRTC_ANDROID |
| webrtc::InitializeAndroidObjects(); |
| #endif |
| |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> factory( |
| webrtc::CreatePeerConnectionFactory( |
| nullptr /* network_thread */, nullptr /* worker_thread */, |
| nullptr /* signaling_thread */, nullptr /* default_adm */, |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| nullptr /* video_encoder_factory */, |
| nullptr /* video_decoder_factory */, nullptr /* audio_mixer */, |
| nullptr /* audio_processing */)); |
| |
| NullPeerConnectionObserver observer; |
| webrtc::PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer); |
| pc_dependencies.cert_generator = std::move(cert_generator); |
| auto result = |
| factory->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| |
| EXPECT_TRUE(result.ok()); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) { |
| webrtc::RtpCapabilities audio_capabilities = |
| factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO); |
| EXPECT_FALSE(audio_capabilities.codecs.empty()); |
| for (const auto& codec : audio_capabilities.codecs) { |
| VerifyAudioCodecCapability(codec); |
| } |
| EXPECT_FALSE(audio_capabilities.header_extensions.empty()); |
| for (const auto& header_extension : audio_capabilities.header_extensions) { |
| EXPECT_FALSE(header_extension.uri.empty()); |
| } |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) { |
| webrtc::RtpCapabilities video_capabilities = |
| factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO); |
| EXPECT_FALSE(video_capabilities.codecs.empty()); |
| for (const auto& codec : video_capabilities.codecs) { |
| VerifyVideoCodecCapability(codec, true); |
| } |
| EXPECT_FALSE(video_capabilities.header_extensions.empty()); |
| for (const auto& header_extension : video_capabilities.header_extensions) { |
| EXPECT_FALSE(header_extension.uri.empty()); |
| } |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) { |
| webrtc::RtpCapabilities video_capabilities = |
| factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO); |
| const auto it = std::find_if( |
| video_capabilities.codecs.begin(), video_capabilities.codecs.end(), |
| [](const auto& c) { return c.name == cricket::kRtxCodecName; }); |
| EXPECT_TRUE(it != video_capabilities.codecs.end()); |
| } |
| |
| TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) { |
| auto factory = CreatePeerConnectionFactoryWithRtxDisabled(); |
| webrtc::RtpCapabilities video_capabilities = |
| factory->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO); |
| const auto it = std::find_if( |
| video_capabilities.codecs.begin(), video_capabilities.codecs.end(), |
| [](const auto& c) { return c.name == cricket::kRtxCodecName; }); |
| EXPECT_TRUE(it == video_capabilities.codecs.end()); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) { |
| webrtc::RtpCapabilities data_capabilities = |
| factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA); |
| EXPECT_TRUE(data_capabilities.codecs.empty()); |
| EXPECT_TRUE(data_capabilities.header_extensions.empty()); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) { |
| webrtc::RtpCapabilities audio_capabilities = |
| factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO); |
| EXPECT_FALSE(audio_capabilities.codecs.empty()); |
| for (const auto& codec : audio_capabilities.codecs) { |
| VerifyAudioCodecCapability(codec); |
| } |
| EXPECT_FALSE(audio_capabilities.header_extensions.empty()); |
| for (const auto& header_extension : audio_capabilities.header_extensions) { |
| EXPECT_FALSE(header_extension.uri.empty()); |
| } |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) { |
| webrtc::RtpCapabilities video_capabilities = |
| factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO); |
| EXPECT_FALSE(video_capabilities.codecs.empty()); |
| for (const auto& codec : video_capabilities.codecs) { |
| VerifyVideoCodecCapability(codec, false); |
| } |
| EXPECT_FALSE(video_capabilities.header_extensions.empty()); |
| for (const auto& header_extension : video_capabilities.header_extensions) { |
| EXPECT_FALSE(header_extension.uri.empty()); |
| } |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) { |
| webrtc::RtpCapabilities video_capabilities = |
| factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO); |
| const auto it = std::find_if( |
| video_capabilities.codecs.begin(), video_capabilities.codecs.end(), |
| [](const auto& c) { return c.name == cricket::kRtxCodecName; }); |
| EXPECT_TRUE(it != video_capabilities.codecs.end()); |
| } |
| |
| TEST(PeerConnectionFactoryTestInternal, |
| CheckRtpReceiverRtxDisabledCapabilities) { |
| auto factory = CreatePeerConnectionFactoryWithRtxDisabled(); |
| webrtc::RtpCapabilities video_capabilities = |
| factory->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO); |
| const auto it = std::find_if( |
| video_capabilities.codecs.begin(), video_capabilities.codecs.end(), |
| [](const auto& c) { return c.name == cricket::kRtxCodecName; }); |
| EXPECT_TRUE(it == video_capabilities.codecs.end()); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) { |
| webrtc::RtpCapabilities data_capabilities = |
| factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA); |
| EXPECT_TRUE(data_capabilities.codecs.empty()); |
| EXPECT_TRUE(data_capabilities.header_extensions.empty()); |
| } |
| |
| // This test verifies creation of PeerConnection with valid STUN and TURN |
| // configuration. Also verifies the URL's parsed correctly as expected. |
| TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = kStunIceServer; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kTurnIceServer; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kTurnIceServerWithTransport; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer_); |
| pc_dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| pc_dependencies.allocator = std::move(port_allocator_); |
| auto result = |
| factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| ASSERT_TRUE(result.ok()); |
| cricket::ServerAddresses stun_servers; |
| rtc::SocketAddress stun1("stun.l.google.com", 19302); |
| stun_servers.insert(stun1); |
| VerifyStunServers(stun_servers); |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| cricket::RelayServerConfig turn1("test.com", 1234, kTurnUsername, |
| kTurnPassword, cricket::PROTO_UDP); |
| turn_servers.push_back(turn1); |
| cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, kTurnUsername, |
| kTurnPassword, cricket::PROTO_TCP); |
| turn_servers.push_back(turn2); |
| VerifyTurnServers(turn_servers); |
| } |
| |
| // This test verifies creation of PeerConnection with valid STUN and TURN |
| // configuration. Also verifies the list of URL's parsed correctly as expected. |
| TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.urls.push_back(kStunIceServer); |
| ice_server.urls.push_back(kTurnIceServer); |
| ice_server.urls.push_back(kTurnIceServerWithTransport); |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer_); |
| pc_dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| pc_dependencies.allocator = std::move(port_allocator_); |
| auto result = |
| factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| ASSERT_TRUE(result.ok()); |
| cricket::ServerAddresses stun_servers; |
| rtc::SocketAddress stun1("stun.l.google.com", 19302); |
| stun_servers.insert(stun1); |
| VerifyStunServers(stun_servers); |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| cricket::RelayServerConfig turn1("test.com", 1234, kTurnUsername, |
| kTurnPassword, cricket::PROTO_UDP); |
| turn_servers.push_back(turn1); |
| cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, kTurnUsername, |
| kTurnPassword, cricket::PROTO_TCP); |
| turn_servers.push_back(turn2); |
| VerifyTurnServers(turn_servers); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = kStunIceServer; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kTurnIceServerWithNoUsernameInUri; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer_); |
| pc_dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| pc_dependencies.allocator = std::move(port_allocator_); |
| auto result = |
| factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| ASSERT_TRUE(result.ok()); |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| cricket::RelayServerConfig turn("test.com", 1234, kTurnUsername, |
| kTurnPassword, cricket::PROTO_UDP); |
| turn_servers.push_back(turn); |
| VerifyTurnServers(turn_servers); |
| } |
| |
| // This test verifies the PeerConnection created properly with TURN url which |
| // has transport parameter in it. |
| TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = kTurnIceServerWithTransport; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer_); |
| pc_dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| pc_dependencies.allocator = std::move(port_allocator_); |
| auto result = |
| factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| ASSERT_TRUE(result.ok()); |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| cricket::RelayServerConfig turn("hello.com", kDefaultStunPort, kTurnUsername, |
| kTurnPassword, cricket::PROTO_TCP); |
| turn_servers.push_back(turn); |
| VerifyTurnServers(turn_servers); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = kSecureTurnIceServer; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kSecureTurnIceServerWithoutTransportParam; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kSecureTurnIceServerWithoutTransportAndPortParam; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer_); |
| pc_dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| pc_dependencies.allocator = std::move(port_allocator_); |
| auto result = |
| factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| ASSERT_TRUE(result.ok()); |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| cricket::RelayServerConfig turn1("hello.com", kDefaultStunTlsPort, |
| kTurnUsername, kTurnPassword, |
| cricket::PROTO_TLS); |
| turn_servers.push_back(turn1); |
| // TURNS with transport param should be default to tcp. |
| cricket::RelayServerConfig turn2("hello.com", 443, kTurnUsername, |
| kTurnPassword, cricket::PROTO_TLS); |
| turn_servers.push_back(turn2); |
| cricket::RelayServerConfig turn3("hello.com", kDefaultStunTlsPort, |
| kTurnUsername, kTurnPassword, |
| cricket::PROTO_TLS); |
| turn_servers.push_back(turn3); |
| VerifyTurnServers(turn_servers); |
| } |
| |
| TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) { |
| PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = kStunIceServerWithIPv4Address; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kStunIceServerWithIPv4AddressWithoutPort; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kStunIceServerWithIPv6Address; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kStunIceServerWithIPv6AddressWithoutPort; |
| config.servers.push_back(ice_server); |
| ice_server.uri = kTurnIceServerWithIPv6Address; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| webrtc::PeerConnectionDependencies pc_dependencies(&observer_); |
| pc_dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| pc_dependencies.allocator = std::move(port_allocator_); |
| auto result = |
| factory_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)); |
| ASSERT_TRUE(result.ok()); |
| cricket::ServerAddresses stun_servers; |
| rtc::SocketAddress stun1("1.2.3.4", 1234); |
| stun_servers.insert(stun1); |
| rtc::SocketAddress stun2("1.2.3.4", 3478); |
| stun_servers.insert(stun2); // Default port |
| rtc::SocketAddress stun3("2401:fa00:4::", 1234); |
| stun_servers.insert(stun3); |
| rtc::SocketAddress stun4("2401:fa00:4::", 3478); |
| stun_servers.insert(stun4); // Default port |
| VerifyStunServers(stun_servers); |
| |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| cricket::RelayServerConfig turn1("2401:fa00:4::", 1234, kTurnUsername, |
| kTurnPassword, cricket::PROTO_UDP); |
| turn_servers.push_back(turn1); |
| VerifyTurnServers(turn_servers); |
| } |
| |
| // This test verifies the captured stream is rendered locally using a |
| // local video track. |
| TEST_F(PeerConnectionFactoryTest, LocalRendering) { |
| rtc::scoped_refptr<webrtc::FakeVideoTrackSource> source = |
| webrtc::FakeVideoTrackSource::Create(/*is_screencast=*/false); |
| |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| |
| ASSERT_TRUE(source.get() != NULL); |
| rtc::scoped_refptr<VideoTrackInterface> track( |
| factory_->CreateVideoTrack(source, "testlabel")); |
| ASSERT_TRUE(track.get() != NULL); |
| FakeVideoTrackRenderer local_renderer(track.get()); |
| |
| EXPECT_EQ(0, local_renderer.num_rendered_frames()); |
| source->InjectFrame(frame_source.GetFrame()); |
| EXPECT_EQ(1, local_renderer.num_rendered_frames()); |
| EXPECT_FALSE(local_renderer.black_frame()); |
| |
| track->set_enabled(false); |
| source->InjectFrame(frame_source.GetFrame()); |
| EXPECT_EQ(2, local_renderer.num_rendered_frames()); |
| EXPECT_TRUE(local_renderer.black_frame()); |
| |
| track->set_enabled(true); |
| source->InjectFrame(frame_source.GetFrame()); |
| EXPECT_EQ(3, local_renderer.num_rendered_frames()); |
| EXPECT_FALSE(local_renderer.black_frame()); |
| } |
| |
| TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) { |
| constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10); |
| auto mock_network_manager = std::make_unique<NiceMock<MockNetworkManager>>(); |
| |
| rtc::Event called; |
| EXPECT_CALL(*mock_network_manager, StartUpdating()) |
| .Times(AtLeast(1)) |
| .WillRepeatedly(InvokeWithoutArgs([&] { called.Set(); })); |
| |
| webrtc::PeerConnectionFactoryDependencies pcf_dependencies; |
| pcf_dependencies.network_manager = std::move(mock_network_manager); |
| |
| rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf = |
| CreateModularPeerConnectionFactory(std::move(pcf_dependencies)); |
| |
| PeerConnectionInterface::RTCConfiguration config; |
| config.ice_candidate_pool_size = 2; |
| NullPeerConnectionObserver observer; |
| auto pc = pcf->CreatePeerConnectionOrError( |
| config, webrtc::PeerConnectionDependencies(&observer)); |
| ASSERT_TRUE(pc.ok()); |
| |
| called.Wait(kWaitTimeout); |
| } |
| |
| TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) { |
| constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10); |
| auto mock_socket_factory = |
| std::make_unique<NiceMock<rtc::MockPacketSocketFactory>>(); |
| |
| rtc::Event called; |
| EXPECT_CALL(*mock_socket_factory, CreateUdpSocket(_, _, _)) |
| .WillOnce(InvokeWithoutArgs([&] { |
| called.Set(); |
| return nullptr; |
| })) |
| .WillRepeatedly(Return(nullptr)); |
| |
| webrtc::PeerConnectionFactoryDependencies pcf_dependencies; |
| pcf_dependencies.packet_socket_factory = std::move(mock_socket_factory); |
| |
| rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf = |
| CreateModularPeerConnectionFactory(std::move(pcf_dependencies)); |
| |
| // By default, localhost addresses are ignored, which makes tests fail if test |
| // machine is offline. |
| PeerConnectionFactoryInterface::Options options; |
| options.network_ignore_mask = 0; |
| pcf->SetOptions(options); |
| |
| PeerConnectionInterface::RTCConfiguration config; |
| config.ice_candidate_pool_size = 2; |
| NullPeerConnectionObserver observer; |
| auto pc = pcf->CreatePeerConnectionOrError( |
| config, webrtc::PeerConnectionDependencies(&observer)); |
| ASSERT_TRUE(pc.ok()); |
| |
| called.Wait(kWaitTimeout); |
| } |
| |
| } // namespace |
| } // namespace webrtc |
