| /* |
| * Copyright 2013 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 <utility> |
| |
| #include "webrtc/p2p/base/fakeportallocator.h" |
| #include "webrtc/pc/test/fakeperiodicvideocapturer.h" |
| #include "webrtc/pc/test/fakertccertificategenerator.h" |
| #include "webrtc/pc/test/mockpeerconnectionobservers.h" |
| #include "webrtc/pc/test/peerconnectiontestwrapper.h" |
| #include "webrtc/rtc_base/gunit.h" |
| |
| static const char kStreamLabelBase[] = "stream_label"; |
| static const char kVideoTrackLabelBase[] = "video_track"; |
| static const char kAudioTrackLabelBase[] = "audio_track"; |
| static const int kMaxWait = 10000; |
| static const int kTestAudioFrameCount = 3; |
| static const int kTestVideoFrameCount = 3; |
| |
| using webrtc::FakeConstraints; |
| using webrtc::FakeVideoTrackRenderer; |
| using webrtc::IceCandidateInterface; |
| using webrtc::MediaConstraintsInterface; |
| using webrtc::MediaStreamInterface; |
| using webrtc::MockSetSessionDescriptionObserver; |
| using webrtc::PeerConnectionInterface; |
| using webrtc::SessionDescriptionInterface; |
| using webrtc::VideoTrackInterface; |
| |
| void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller, |
| PeerConnectionTestWrapper* callee) { |
| caller->SignalOnIceCandidateReady.connect( |
| callee, &PeerConnectionTestWrapper::AddIceCandidate); |
| callee->SignalOnIceCandidateReady.connect( |
| caller, &PeerConnectionTestWrapper::AddIceCandidate); |
| |
| caller->SignalOnSdpReady.connect( |
| callee, &PeerConnectionTestWrapper::ReceiveOfferSdp); |
| callee->SignalOnSdpReady.connect( |
| caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp); |
| } |
| |
| PeerConnectionTestWrapper::PeerConnectionTestWrapper( |
| const std::string& name, |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread) |
| : name_(name), |
| network_thread_(network_thread), |
| worker_thread_(worker_thread) {} |
| |
| PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {} |
| |
| bool PeerConnectionTestWrapper::CreatePc( |
| const MediaConstraintsInterface* constraints, |
| const webrtc::PeerConnectionInterface::RTCConfiguration& config, |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) { |
| std::unique_ptr<cricket::PortAllocator> port_allocator( |
| new cricket::FakePortAllocator(network_thread_, nullptr)); |
| |
| fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); |
| if (fake_audio_capture_module_ == NULL) { |
| return false; |
| } |
| |
| peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( |
| network_thread_, worker_thread_, rtc::Thread::Current(), |
| fake_audio_capture_module_, audio_encoder_factory, audio_decoder_factory, |
| nullptr, nullptr); |
| if (!peer_connection_factory_) { |
| return false; |
| } |
| |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| peer_connection_ = peer_connection_factory_->CreatePeerConnection( |
| config, constraints, std::move(port_allocator), std::move(cert_generator), |
| this); |
| |
| return peer_connection_.get() != NULL; |
| } |
| |
| rtc::scoped_refptr<webrtc::DataChannelInterface> |
| PeerConnectionTestWrapper::CreateDataChannel( |
| const std::string& label, |
| const webrtc::DataChannelInit& init) { |
| return peer_connection_->CreateDataChannel(label, &init); |
| } |
| |
| void PeerConnectionTestWrapper::OnAddStream( |
| rtc::scoped_refptr<MediaStreamInterface> stream) { |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": OnAddStream"; |
| // TODO(ronghuawu): support multiple streams. |
| if (stream->GetVideoTracks().size() > 0) { |
| renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0])); |
| } |
| } |
| |
| void PeerConnectionTestWrapper::OnIceCandidate( |
| const IceCandidateInterface* candidate) { |
| std::string sdp; |
| EXPECT_TRUE(candidate->ToString(&sdp)); |
| // Give the user a chance to modify sdp for testing. |
| SignalOnIceCandidateCreated(&sdp); |
| SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(), |
| sdp); |
| } |
| |
| void PeerConnectionTestWrapper::OnDataChannel( |
| rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) { |
| SignalOnDataChannel(data_channel); |
| } |
| |
| void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) { |
| // This callback should take the ownership of |desc|. |
| std::unique_ptr<SessionDescriptionInterface> owned_desc(desc); |
| std::string sdp; |
| EXPECT_TRUE(desc->ToString(&sdp)); |
| |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": " << desc->type() << " sdp created: " << sdp; |
| |
| // Give the user a chance to modify sdp for testing. |
| SignalOnSdpCreated(&sdp); |
| |
| SetLocalDescription(desc->type(), sdp); |
| |
| SignalOnSdpReady(sdp); |
| } |
| |
| void PeerConnectionTestWrapper::CreateOffer( |
| const MediaConstraintsInterface* constraints) { |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": CreateOffer."; |
| peer_connection_->CreateOffer(this, constraints); |
| } |
| |
| void PeerConnectionTestWrapper::CreateAnswer( |
| const MediaConstraintsInterface* constraints) { |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": CreateAnswer."; |
| peer_connection_->CreateAnswer(this, constraints); |
| } |
| |
| void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) { |
| SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp); |
| CreateAnswer(NULL); |
| } |
| |
| void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) { |
| SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp); |
| } |
| |
| void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type, |
| const std::string& sdp) { |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": SetLocalDescription " << type << " " << sdp; |
| |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> |
| observer(new rtc::RefCountedObject< |
| MockSetSessionDescriptionObserver>()); |
| peer_connection_->SetLocalDescription( |
| observer, webrtc::CreateSessionDescription(type, sdp, NULL)); |
| } |
| |
| void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type, |
| const std::string& sdp) { |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": SetRemoteDescription " << type << " " << sdp; |
| |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> |
| observer(new rtc::RefCountedObject< |
| MockSetSessionDescriptionObserver>()); |
| peer_connection_->SetRemoteDescription( |
| observer, webrtc::CreateSessionDescription(type, sdp, NULL)); |
| } |
| |
| void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid, |
| int sdp_mline_index, |
| const std::string& candidate) { |
| std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate( |
| webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL)); |
| EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get())); |
| } |
| |
| void PeerConnectionTestWrapper::WaitForCallEstablished() { |
| WaitForConnection(); |
| WaitForAudio(); |
| WaitForVideo(); |
| } |
| |
| void PeerConnectionTestWrapper::WaitForConnection() { |
| EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait); |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": Connected."; |
| } |
| |
| bool PeerConnectionTestWrapper::CheckForConnection() { |
| return (peer_connection_->ice_connection_state() == |
| PeerConnectionInterface::kIceConnectionConnected) || |
| (peer_connection_->ice_connection_state() == |
| PeerConnectionInterface::kIceConnectionCompleted); |
| } |
| |
| void PeerConnectionTestWrapper::WaitForAudio() { |
| EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait); |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": Got enough audio frames."; |
| } |
| |
| bool PeerConnectionTestWrapper::CheckForAudio() { |
| return (fake_audio_capture_module_->frames_received() >= |
| kTestAudioFrameCount); |
| } |
| |
| void PeerConnectionTestWrapper::WaitForVideo() { |
| EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait); |
| LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ |
| << ": Got enough video frames."; |
| } |
| |
| bool PeerConnectionTestWrapper::CheckForVideo() { |
| if (!renderer_) { |
| return false; |
| } |
| return (renderer_->num_rendered_frames() >= kTestVideoFrameCount); |
| } |
| |
| void PeerConnectionTestWrapper::GetAndAddUserMedia( |
| bool audio, const webrtc::FakeConstraints& audio_constraints, |
| bool video, const webrtc::FakeConstraints& video_constraints) { |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| GetUserMedia(audio, audio_constraints, video, video_constraints); |
| EXPECT_TRUE(peer_connection_->AddStream(stream)); |
| } |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> |
| PeerConnectionTestWrapper::GetUserMedia( |
| bool audio, const webrtc::FakeConstraints& audio_constraints, |
| bool video, const webrtc::FakeConstraints& video_constraints) { |
| std::string label = kStreamLabelBase + |
| rtc::ToString<int>( |
| static_cast<int>(peer_connection_->local_streams()->count())); |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| peer_connection_factory_->CreateLocalMediaStream(label); |
| |
| if (audio) { |
| FakeConstraints constraints = audio_constraints; |
| // Disable highpass filter so that we can get all the test audio frames. |
| constraints.AddMandatory( |
| MediaConstraintsInterface::kHighpassFilter, false); |
| rtc::scoped_refptr<webrtc::AudioSourceInterface> source = |
| peer_connection_factory_->CreateAudioSource(&constraints); |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( |
| peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase, |
| source)); |
| stream->AddTrack(audio_track); |
| } |
| |
| if (video) { |
| // Set max frame rate to 10fps to reduce the risk of the tests to be flaky. |
| FakeConstraints constraints = video_constraints; |
| constraints.SetMandatoryMaxFrameRate(10); |
| |
| rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source = |
| peer_connection_factory_->CreateVideoSource( |
| std::unique_ptr<cricket::VideoCapturer>( |
| new webrtc::FakePeriodicVideoCapturer()), |
| &constraints); |
| std::string videotrack_label = label + kVideoTrackLabelBase; |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track( |
| peer_connection_factory_->CreateVideoTrack(videotrack_label, source)); |
| |
| stream->AddTrack(video_track); |
| } |
| return stream; |
| } |