diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/pc/peer_connection_interface_unittest.cc | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_interface_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/peer_connection_interface_unittest.cc | 235 |
1 files changed, 103 insertions, 132 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc b/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc index 1f5ab2f449..5ee9881b84 100644 --- a/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc +++ b/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc @@ -22,9 +22,9 @@ #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/call/call_factory_interface.h" #include "api/create_peerconnection_factory.h" #include "api/data_channel_interface.h" +#include "api/enable_media_with_defaults.h" #include "api/jsep.h" #include "api/media_stream_interface.h" #include "api/media_types.h" @@ -53,7 +53,6 @@ #include "media/base/media_engine.h" #include "media/base/stream_params.h" #include "media/engine/webrtc_media_engine.h" -#include "media/engine/webrtc_media_engine_defaults.h" #include "media/sctp/sctp_transport_internal.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_processing/include/audio_processing.h" @@ -475,8 +474,7 @@ bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) { // Get the ufrags out of an SDP blob. Useful for testing ICE restart // behavior. -std::vector<std::string> GetUfrags( - const webrtc::SessionDescriptionInterface* desc) { +std::vector<std::string> GetUfrags(const SessionDescriptionInterface* desc) { std::vector<std::string> ufrags; for (const cricket::TransportInfo& info : desc->description()->transport_infos()) { @@ -545,21 +543,19 @@ rtc::scoped_refptr<StreamCollection> CreateStreamCollection( StreamCollection::Create()); for (int i = 0; i < number_of_streams; ++i) { - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream( - webrtc::MediaStream::Create(kStreams[i])); + rtc::scoped_refptr<MediaStreamInterface> stream( + MediaStream::Create(kStreams[i])); for (int j = 0; j < tracks_per_stream; ++j) { // Add a local audio track. - rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( - webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], - nullptr)); + rtc::scoped_refptr<AudioTrackInterface> audio_track( + AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], nullptr)); stream->AddTrack(audio_track); // Add a local video track. - rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( - webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j], - webrtc::FakeVideoTrackSource::Create(), - rtc::Thread::Current())); + rtc::scoped_refptr<VideoTrackInterface> video_track(VideoTrack::Create( + kVideoTracks[i * tracks_per_stream + j], + FakeVideoTrackSource::Create(), rtc::Thread::Current())); stream->AddTrack(video_track); } @@ -579,10 +575,10 @@ bool CompareStreamCollections(StreamCollectionInterface* s1, if (s1->at(i)->id() != s2->at(i)->id()) { return false; } - webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks(); - webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks(); - webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks(); - webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks(); + AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks(); + AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks(); + VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks(); + VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks(); if (audio_tracks1.size() != audio_tracks2.size()) { return false; @@ -631,7 +627,7 @@ class MockTrackObserver : public ObserverInterface { // constraints are propagated into the PeerConnection's MediaConfig. These // settings are intended for MediaChannel constructors, but that is not // exercised by these unittest. -class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory { +class PeerConnectionFactoryForTest : public PeerConnectionFactory { public: static rtc::scoped_refptr<PeerConnectionFactoryForTest> CreatePeerConnectionFactoryForTest() { @@ -641,16 +637,10 @@ class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory { dependencies.signaling_thread = rtc::Thread::Current(); dependencies.task_queue_factory = CreateDefaultTaskQueueFactory(); dependencies.trials = std::make_unique<FieldTrialBasedConfig>(); - cricket::MediaEngineDependencies media_deps; - media_deps.task_queue_factory = dependencies.task_queue_factory.get(); // Use fake audio device module since we're only testing the interface // level, and using a real one could make tests flaky when run in parallel. - media_deps.adm = FakeAudioCaptureModule::Create(); - SetMediaEngineDefaults(&media_deps); - media_deps.trials = dependencies.trials.get(); - dependencies.media_engine = - cricket::CreateMediaEngine(std::move(media_deps)); - dependencies.call_factory = webrtc::CreateCallFactory(); + dependencies.adm = FakeAudioCaptureModule::Create(); + EnableMediaWithDefaults(dependencies); dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>( dependencies.task_queue_factory.get()); @@ -672,7 +662,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { main_(vss_.get()), sdp_semantics_(sdp_semantics) { #ifdef WEBRTC_ANDROID - webrtc::InitializeAndroidObjects(); + InitializeAndroidObjects(); #endif } @@ -680,22 +670,16 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { // Use fake audio capture module since we're only testing the interface // level, and using a real one could make tests flaky when run in parallel. fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); - pc_factory_ = webrtc::CreatePeerConnectionFactory( + pc_factory_ = CreatePeerConnectionFactory( rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), - rtc::scoped_refptr<webrtc::AudioDeviceModule>( - fake_audio_capture_module_), - webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), - std::make_unique<webrtc::VideoEncoderFactoryTemplate< - webrtc::LibvpxVp8EncoderTemplateAdapter, - webrtc::LibvpxVp9EncoderTemplateAdapter, - webrtc::OpenH264EncoderTemplateAdapter, - webrtc::LibaomAv1EncoderTemplateAdapter>>(), - std::make_unique<webrtc::VideoDecoderFactoryTemplate< - webrtc::LibvpxVp8DecoderTemplateAdapter, - webrtc::LibvpxVp9DecoderTemplateAdapter, - webrtc::OpenH264DecoderTemplateAdapter, - webrtc::Dav1dDecoderTemplateAdapter>>(), + rtc::scoped_refptr<AudioDeviceModule>(fake_audio_capture_module_), + CreateBuiltinAudioEncoderFactory(), 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(pc_factory_); } @@ -953,8 +937,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { // Call the standards-compliant GetStats function. bool DoGetRTCStats() { - auto callback = - rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>(); + auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>(); pc_->GetStats(callback.get()); EXPECT_TRUE_WAIT(callback->called(), kTimeout); return callback->called(); @@ -994,14 +977,14 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(offer->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); } void CreateAndSetRemoteOffer(const std::string& sdp) { std::unique_ptr<SessionDescriptionInterface> remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); } @@ -1020,7 +1003,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(answer->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> new_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer))); EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); } @@ -1032,7 +1015,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(answer->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> pr_answer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer))); EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_); } @@ -1057,7 +1040,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE(offer->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> new_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_); @@ -1067,7 +1050,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { void CreateAnswerAsRemoteDescription(const std::string& sdp) { std::unique_ptr<SessionDescriptionInterface> answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); ASSERT_TRUE(answer); EXPECT_TRUE(DoSetRemoteDescription(std::move(answer))); EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); @@ -1075,12 +1058,12 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) { std::unique_ptr<SessionDescriptionInterface> pr_answer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); ASSERT_TRUE(pr_answer); EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_); std::unique_ptr<SessionDescriptionInterface> answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); ASSERT_TRUE(answer); EXPECT_TRUE(DoSetRemoteDescription(std::move(answer))); EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_); @@ -1124,8 +1107,8 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string mediastream_id = kStreams[0]; - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream( - webrtc::MediaStream::Create(mediastream_id)); + rtc::scoped_refptr<MediaStreamInterface> stream( + MediaStream::Create(mediastream_id)); reference_collection_->AddStream(stream); if (number_of_audio_tracks > 0) { @@ -1149,22 +1132,20 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { } return std::unique_ptr<SessionDescriptionInterface>( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1)); + CreateSessionDescription(SdpType::kOffer, sdp_ms1)); } void AddAudioTrack(const std::string& track_id, MediaStreamInterface* stream) { - rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( - webrtc::AudioTrack::Create(track_id, nullptr)); + rtc::scoped_refptr<AudioTrackInterface> audio_track( + AudioTrack::Create(track_id, nullptr)); ASSERT_TRUE(stream->AddTrack(audio_track)); } void AddVideoTrack(const std::string& track_id, MediaStreamInterface* stream) { - rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( - webrtc::VideoTrack::Create(track_id, - webrtc::FakeVideoTrackSource::Create(), - rtc::Thread::Current())); + rtc::scoped_refptr<VideoTrackInterface> video_track(VideoTrack::Create( + track_id, FakeVideoTrackSource::Create(), rtc::Thread::Current())); ASSERT_TRUE(stream->AddTrack(video_track)); } @@ -1224,7 +1205,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE((*desc)->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_); } @@ -1237,7 +1218,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { std::string sdp; EXPECT_TRUE((*desc)->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> new_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer))); EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_); @@ -1246,8 +1227,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { bool HasCNCodecs(const cricket::ContentInfo* content) { RTC_DCHECK(content); RTC_DCHECK(content->media_description()); - for (const cricket::AudioCodec& codec : - content->media_description()->as_audio()->codecs()) { + for (const cricket::Codec& codec : content->media_description()->codecs()) { if (codec.name == "CN") { return true; } @@ -1273,13 +1253,13 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test { rtc::SocketServer* socket_server() const { return vss_.get(); } - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; std::unique_ptr<rtc::VirtualSocketServer> vss_; rtc::AutoSocketServerThread main_; rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; cricket::FakePortAllocator* port_allocator_ = nullptr; FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr; - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_; + rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_; rtc::scoped_refptr<PeerConnectionInterface> pc_; MockPeerConnectionObserver observer_; rtc::scoped_refptr<StreamCollection> reference_collection_; @@ -1399,22 +1379,19 @@ TEST_P(PeerConnectionInterfaceTest, config.prune_turn_ports = true; // Create the PC factory and PC with the above config. - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory( - webrtc::CreatePeerConnectionFactory( + rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory( + CreatePeerConnectionFactory( rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(), fake_audio_capture_module_, - webrtc::CreateBuiltinAudioEncoderFactory(), - webrtc::CreateBuiltinAudioDecoderFactory(), - std::make_unique<webrtc::VideoEncoderFactoryTemplate< - webrtc::LibvpxVp8EncoderTemplateAdapter, - webrtc::LibvpxVp9EncoderTemplateAdapter, - webrtc::OpenH264EncoderTemplateAdapter, - webrtc::LibaomAv1EncoderTemplateAdapter>>(), - std::make_unique<webrtc::VideoDecoderFactoryTemplate< - webrtc::LibvpxVp8DecoderTemplateAdapter, - webrtc::LibvpxVp9DecoderTemplateAdapter, - webrtc::OpenH264DecoderTemplateAdapter, - webrtc::Dav1dDecoderTemplateAdapter>>(), + CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory(), + std::make_unique<VideoEncoderFactoryTemplate< + LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter, + OpenH264EncoderTemplateAdapter, + LibaomAv1EncoderTemplateAdapter>>(), + std::make_unique<VideoDecoderFactoryTemplate< + LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter, + OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(), nullptr /* audio_mixer */, nullptr /* audio_processing */)); PeerConnectionDependencies pc_dependencies(&observer_); pc_dependencies.allocator = std::move(port_allocator); @@ -1431,7 +1408,7 @@ TEST_P(PeerConnectionInterfaceTest, EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP); EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS); - EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY, + EXPECT_EQ(PRUNE_BASED_ON_PRIORITY, raw_port_allocator->turn_port_prune_policy()); } @@ -1453,8 +1430,7 @@ TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) { TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) { PeerConnectionInterface::RTCConfiguration starting_config; starting_config.sdp_semantics = sdp_semantics_; - starting_config.bundle_policy = - webrtc::PeerConnection::kBundlePolicyMaxBundle; + starting_config.bundle_policy = PeerConnection::kBundlePolicyMaxBundle; CreatePeerConnection(starting_config); PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration(); @@ -1985,7 +1961,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) { RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - webrtc::DataChannelInit config; + DataChannelInit config; auto channel = pc_->CreateDataChannelOrError("1", &config); EXPECT_TRUE(channel.ok()); EXPECT_TRUE(channel.value()->reliable()); @@ -2017,7 +1993,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWhenClosed) { RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); pc_->Close(); - webrtc::DataChannelInit config; + DataChannelInit config; auto ret = pc_->CreateDataChannelOrError("1", &config); ASSERT_FALSE(ret.ok()); EXPECT_EQ(ret.error().type(), RTCErrorType::INVALID_STATE); @@ -2029,7 +2005,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) { RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - webrtc::DataChannelInit config; + DataChannelInit config; config.maxRetransmitTime = -1; config.maxRetransmits = -1; auto channel = pc_->CreateDataChannelOrError("1", &config); @@ -2044,7 +2020,7 @@ TEST_P(PeerConnectionInterfaceTest, CreatePeerConnection(rtc_config); std::string label = "test"; - webrtc::DataChannelInit config; + DataChannelInit config; config.maxRetransmits = 0; config.maxRetransmitTime = 0; @@ -2059,7 +2035,7 @@ TEST_P(PeerConnectionInterfaceTest, RTCConfiguration rtc_config; CreatePeerConnection(rtc_config); - webrtc::DataChannelInit config; + DataChannelInit config; config.id = 1; config.negotiated = true; @@ -2113,7 +2089,7 @@ TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer) std::string sdp; EXPECT_TRUE(pc_->local_description()->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); ASSERT_TRUE(answer); cricket::ContentInfo* data_info = cricket::GetFirstDataContent(answer->description()); @@ -2132,8 +2108,7 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) { AddAudioTrack("audio_label"); AddVideoTrack("video_label"); std::unique_ptr<SessionDescriptionInterface> desc( - webrtc::CreateSessionDescription(SdpType::kOffer, - webrtc::kFireFoxSdpOffer, nullptr)); + CreateSessionDescription(SdpType::kOffer, kFireFoxSdpOffer, nullptr)); EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false)); CreateAnswerAsLocalDescription(); ASSERT_TRUE(pc_->local_description() != nullptr); @@ -2170,8 +2145,7 @@ TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) { EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(), kTimeout); std::unique_ptr<SessionDescriptionInterface> desc( - webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, - nullptr)); + CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, nullptr)); EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false)); } @@ -2184,18 +2158,17 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) { CreateOfferAsLocalDescription(); const char* answer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED - ? webrtc::kAudioSdpPlanB - : webrtc::kAudioSdpUnifiedPlan); + ? kAudioSdpPlanB + : kAudioSdpUnifiedPlan); std::unique_ptr<SessionDescriptionInterface> answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr)); + CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr)); EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false)); - const char* reoffer_sdp = - (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED - ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB - : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan); + const char* reoffer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED + ? kAudioSdpWithUnsupportedCodecsPlanB + : kAudioSdpWithUnsupportedCodecsUnifiedPlan); std::unique_ptr<SessionDescriptionInterface> updated_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr)); + CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr)); EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false)); CreateAnswerAsLocalDescription(); } @@ -2282,12 +2255,11 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) { config.prune_turn_ports = false; CreatePeerConnection(config); config = pc_->GetConfiguration(); - EXPECT_EQ(webrtc::NO_PRUNE, port_allocator_->turn_port_prune_policy()); + EXPECT_EQ(NO_PRUNE, port_allocator_->turn_port_prune_policy()); config.prune_turn_ports = true; EXPECT_TRUE(pc_->SetConfiguration(config).ok()); - EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY, - port_allocator_->turn_port_prune_policy()); + EXPECT_EQ(PRUNE_BASED_ON_PRIORITY, port_allocator_->turn_port_prune_policy()); } // Test that the ice check interval can be changed. This does not verify that @@ -2556,12 +2528,12 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) { std::string sdp; ASSERT_TRUE(pc_->remote_description()->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer))); ASSERT_TRUE(pc_->local_description()->ToString(&sdp)); std::unique_ptr<SessionDescriptionInterface> local_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer))); } @@ -2621,10 +2593,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB, reference_collection_.get())); rtc::scoped_refptr<AudioTrackInterface> audio_track2 = observer_.remote_streams()->at(0)->GetAudioTracks()[1]; - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track2->state()); rtc::scoped_refptr<VideoTrackInterface> video_track2 = observer_.remote_streams()->at(0)->GetVideoTracks()[1]; - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track2->state()); // Remove the extra audio and video tracks. std::unique_ptr<SessionDescriptionInterface> desc_ms2 = @@ -2638,10 +2610,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB, EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(), reference_collection_.get())); // Track state may be updated asynchronously. - EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded, - audio_track2->state(), kTimeout); - EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded, - video_track2->state(), kTimeout); + EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, audio_track2->state(), + kTimeout); + EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, video_track2->state(), + kTimeout); } // This tests that remote tracks are ended if a local session description is set @@ -2659,7 +2631,7 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) { rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio = audio_receiver->track(); - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state()); rtc::scoped_refptr<MediaStreamTrackInterface> remote_video = video_receiver->track(); EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state()); @@ -2703,8 +2675,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) { remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]); std::unique_ptr<SessionDescriptionInterface> local_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, - GetSdpStringWithStream1(), nullptr)); + CreateSessionDescription(SdpType::kAnswer, GetSdpStringWithStream1(), + nullptr)); cricket::ContentInfo* video_info = local_answer->description()->GetContentByName("video"); video_info->rejected = true; @@ -2993,9 +2965,9 @@ TEST_P(PeerConnectionInterfaceTest, ASSERT_TRUE(DoCreateOffer(&offer, nullptr)); // Grab a copy of the offer before it gets passed into the PC. std::unique_ptr<SessionDescriptionInterface> modified_offer = - webrtc::CreateSessionDescription( - webrtc::SdpType::kOffer, offer->session_id(), - offer->session_version(), offer->description()->Clone()); + CreateSessionDescription(SdpType::kOffer, offer->session_id(), + offer->session_version(), + offer->description()->Clone()); EXPECT_TRUE(DoSetLocalDescription(std::move(offer))); auto senders = pc_->GetSenders(); @@ -3051,8 +3023,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0])); // Add a new MediaStream but with the same tracks as in the first stream. - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1( - webrtc::MediaStream::Create(kStreams[1])); + rtc::scoped_refptr<MediaStreamInterface> stream_1( + MediaStream::Create(kStreams[1])); stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]); stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]); pc_->AddStream(stream_1.get()); @@ -3173,9 +3145,9 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) { EXPECT_TRUE(pc_->SetConfiguration(config).ok()); // Do ICE restart for the first m= section, initiated by remote peer. - std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, - GetSdpStringWithStream1(), nullptr)); + std::unique_ptr<SessionDescriptionInterface> remote_offer( + CreateSessionDescription(SdpType::kOffer, GetSdpStringWithStream1(), + nullptr)); ASSERT_TRUE(remote_offer); remote_offer->description()->transport_infos()[0].description.ice_ufrag = "modified"; @@ -3221,7 +3193,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set remote pranswer. std::unique_ptr<SessionDescriptionInterface> remote_pranswer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get(); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer))); EXPECT_EQ(local_offer_ptr, pc_->pending_local_description()); @@ -3231,7 +3203,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set remote answer. std::unique_ptr<SessionDescriptionInterface> remote_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); SessionDescriptionInterface* remote_answer_ptr = remote_answer.get(); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer))); EXPECT_EQ(nullptr, pc_->pending_local_description()); @@ -3241,7 +3213,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set remote offer. std::unique_ptr<SessionDescriptionInterface> remote_offer( - webrtc::CreateSessionDescription(SdpType::kOffer, sdp)); + CreateSessionDescription(SdpType::kOffer, sdp)); SessionDescriptionInterface* remote_offer_ptr = remote_offer.get(); EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer))); EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description()); @@ -3251,7 +3223,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set local pranswer. std::unique_ptr<SessionDescriptionInterface> local_pranswer( - webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp)); + CreateSessionDescription(SdpType::kPrAnswer, sdp)); SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get(); EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer))); EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description()); @@ -3261,7 +3233,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) { // Set local answer. std::unique_ptr<SessionDescriptionInterface> local_answer( - webrtc::CreateSessionDescription(SdpType::kAnswer, sdp)); + CreateSessionDescription(SdpType::kAnswer, sdp)); SessionDescriptionInterface* local_answer_ptr = local_answer.get(); EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer))); EXPECT_EQ(nullptr, pc_->pending_remote_description()); @@ -3280,9 +3252,8 @@ TEST_P(PeerConnectionInterfaceTest, // The RtcEventLog will be reset when the PeerConnection is closed. pc_->Close(); - EXPECT_FALSE( - pc_->StartRtcEventLog(std::make_unique<webrtc::RtcEventLogOutputNull>(), - webrtc::RtcEventLog::kImmediateOutput)); + EXPECT_FALSE(pc_->StartRtcEventLog(std::make_unique<RtcEventLogOutputNull>(), + RtcEventLog::kImmediateOutput)); pc_->StopRtcEventLog(); } |