| /* |
| * 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 <stddef.h> |
| |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio_codecs/audio_decoder_factory.h" |
| #include "api/audio_codecs/audio_encoder_factory.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/video_codecs/builtin_video_decoder_factory.h" |
| #include "api/video_codecs/builtin_video_encoder_factory.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/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_interface.h" |
| #include "pc/test/fake_audio_capture_module.h" |
| #include "pc/test/fake_video_track_source.h" |
| #include "rtc_base/socket_address.h" |
| #include "test/gtest.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; |
| |
| 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 { |
| } |
| }; |
| |
| } // namespace |
| |
| class PeerConnectionFactoryTest : public ::testing::Test { |
| 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); |
| port_allocator_.reset( |
| new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr)); |
| 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) { |
| EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO); |
| EXPECT_FALSE(codec.name.empty()); |
| EXPECT_GT(codec.clock_rate, 0); |
| } |
| |
| rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_; |
| NullPeerConnectionObserver observer_; |
| 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; |
| |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection( |
| config, nullptr, std::move(cert_generator), &observer)); |
| |
| EXPECT_TRUE(pc.get() != nullptr); |
| } |
| |
| 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); |
| } |
| 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); |
| } |
| 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; |
| 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); |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| factory_->CreatePeerConnection(config, std::move(port_allocator_), |
| std::move(cert_generator), &observer_)); |
| ASSERT_TRUE(pc.get() != NULL); |
| 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; |
| 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); |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| factory_->CreatePeerConnection(config, std::move(port_allocator_), |
| std::move(cert_generator), &observer_)); |
| ASSERT_TRUE(pc.get() != NULL); |
| 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; |
| 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); |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| factory_->CreatePeerConnection(config, std::move(port_allocator_), |
| std::move(cert_generator), &observer_)); |
| ASSERT_TRUE(pc.get() != NULL); |
| 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; |
| webrtc::PeerConnectionInterface::IceServer ice_server; |
| ice_server.uri = kTurnIceServerWithTransport; |
| ice_server.username = kTurnUsername; |
| ice_server.password = kTurnPassword; |
| config.servers.push_back(ice_server); |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| factory_->CreatePeerConnection(config, std::move(port_allocator_), |
| std::move(cert_generator), &observer_)); |
| ASSERT_TRUE(pc.get() != NULL); |
| 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; |
| 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); |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| factory_->CreatePeerConnection(config, std::move(port_allocator_), |
| std::move(cert_generator), &observer_)); |
| ASSERT_TRUE(pc.get() != NULL); |
| 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; |
| 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); |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| rtc::scoped_refptr<PeerConnectionInterface> pc( |
| factory_->CreatePeerConnection(config, std::move(port_allocator_), |
| std::move(cert_generator), &observer_)); |
| ASSERT_TRUE(pc.get() != NULL); |
| 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)); |
| ASSERT_TRUE(track.get() != NULL); |
| FakeVideoTrackRenderer local_renderer(track); |
| |
| 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()); |
| } |