summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc
diff options
context:
space:
mode:
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_interface_unittest.cc')
-rw-r--r--third_party/libwebrtc/pc/peer_connection_interface_unittest.cc235
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();
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-22 09:46:50 +0000