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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/peer_connection_factory_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_factory_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/peer_connection_factory_unittest.cc | 648 |
1 files changed, 648 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc b/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc new file mode 100644 index 0000000000..6aa7f49079 --- /dev/null +++ b/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc @@ -0,0 +1,648 @@ +/* + * 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 <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/test/mock_packet_socket_factory.h" +#include "api/video_codecs/builtin_video_decoder_factory.h" +#include "api/video_codecs/builtin_video_encoder_factory.h" +#include "media/base/fake_frame_source.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" + +using webrtc::DataChannelInterface; +using webrtc::FakeVideoTrackRenderer; +using webrtc::MediaStreamInterface; +using webrtc::PeerConnectionFactoryInterface; +using webrtc::PeerConnectionInterface; +using webrtc::PeerConnectionObserver; +using webrtc::VideoTrackInterface; +using webrtc::VideoTrackSourceInterface; + +using ::testing::_; +using ::testing::AtLeast; +using ::testing::InvokeWithoutArgs; +using ::testing::NiceMock; +using ::testing::Return; +using ::testing::UnorderedElementsAre; + +namespace { + +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)); +}; + +} // namespace + +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(), + webrtc::CreateBuiltinVideoEncoderFactory(), + webrtc::CreateBuiltinVideoDecoderFactory(), 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_; +}; + +// 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(), + webrtc::CreateBuiltinVideoEncoderFactory(), + webrtc::CreateBuiltinVideoDecoderFactory(), 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, 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, 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("testlabel", source.get())); + 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); +} |