3 * Copyright 2013, Google Inc.
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6 * modification, are permitted provided that the following conditions are met:
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28 #include "talk/app/webrtc/fakeportallocatorfactory.h"
29 #include "talk/app/webrtc/test/fakedtlsidentityservice.h"
30 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
31 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
32 #include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
33 #include "talk/app/webrtc/videosourceinterface.h"
34 #include "talk/base/gunit.h"
36 static const char kStreamLabelBase[] = "stream_label";
37 static const char kVideoTrackLabelBase[] = "video_track";
38 static const char kAudioTrackLabelBase[] = "audio_track";
39 static const int kMaxWait = 5000;
40 static const int kTestAudioFrameCount = 3;
41 static const int kTestVideoFrameCount = 3;
43 using webrtc::FakeConstraints;
44 using webrtc::FakeVideoTrackRenderer;
45 using webrtc::IceCandidateInterface;
46 using webrtc::MediaConstraintsInterface;
47 using webrtc::MediaStreamInterface;
48 using webrtc::MockSetSessionDescriptionObserver;
49 using webrtc::PeerConnectionInterface;
50 using webrtc::SessionDescriptionInterface;
51 using webrtc::VideoTrackInterface;
53 void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
54 PeerConnectionTestWrapper* callee) {
55 caller->SignalOnIceCandidateReady.connect(
56 callee, &PeerConnectionTestWrapper::AddIceCandidate);
57 callee->SignalOnIceCandidateReady.connect(
58 caller, &PeerConnectionTestWrapper::AddIceCandidate);
60 caller->SignalOnSdpReady.connect(
61 callee, &PeerConnectionTestWrapper::ReceiveOfferSdp);
62 callee->SignalOnSdpReady.connect(
63 caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
66 PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name)
69 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
71 bool PeerConnectionTestWrapper::CreatePc(
72 const MediaConstraintsInterface* constraints) {
73 allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
74 if (!allocator_factory_) {
78 audio_thread_.Start();
79 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
81 if (fake_audio_capture_module_ == NULL) {
85 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
86 talk_base::Thread::Current(), talk_base::Thread::Current(),
87 fake_audio_capture_module_, NULL, NULL);
88 if (!peer_connection_factory_) {
92 // CreatePeerConnection with IceServers.
93 webrtc::PeerConnectionInterface::IceServers ice_servers;
94 webrtc::PeerConnectionInterface::IceServer ice_server;
95 ice_server.uri = "stun:stun.l.google.com:19302";
96 ice_servers.push_back(ice_server);
97 FakeIdentityService* dtls_service =
98 talk_base::SSLStreamAdapter::HaveDtlsSrtp() ?
99 new FakeIdentityService() : NULL;
100 peer_connection_ = peer_connection_factory_->CreatePeerConnection(
101 ice_servers, constraints, allocator_factory_.get(), dtls_service, this);
103 return peer_connection_.get() != NULL;
106 void PeerConnectionTestWrapper::OnAddStream(MediaStreamInterface* stream) {
107 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
109 // TODO(ronghuawu): support multiple streams.
110 if (stream->GetVideoTracks().size() > 0) {
111 renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
115 void PeerConnectionTestWrapper::OnIceCandidate(
116 const IceCandidateInterface* candidate) {
118 EXPECT_TRUE(candidate->ToString(&sdp));
119 // Give the user a chance to modify sdp for testing.
120 SignalOnIceCandidateCreated(&sdp);
121 SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
125 void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
126 // This callback should take the ownership of |desc|.
127 talk_base::scoped_ptr<SessionDescriptionInterface> owned_desc(desc);
129 EXPECT_TRUE(desc->ToString(&sdp));
131 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
132 << ": " << desc->type() << " sdp created: " << sdp;
134 // Give the user a chance to modify sdp for testing.
135 SignalOnSdpCreated(&sdp);
137 SetLocalDescription(desc->type(), sdp);
139 SignalOnSdpReady(sdp);
142 void PeerConnectionTestWrapper::CreateOffer(
143 const MediaConstraintsInterface* constraints) {
144 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
146 peer_connection_->CreateOffer(this, constraints);
149 void PeerConnectionTestWrapper::CreateAnswer(
150 const MediaConstraintsInterface* constraints) {
151 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
152 << ": CreateAnswer.";
153 peer_connection_->CreateAnswer(this, constraints);
156 void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
157 SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp);
161 void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
162 SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp);
165 void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type,
166 const std::string& sdp) {
167 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
168 << ": SetLocalDescription " << type << " " << sdp;
170 talk_base::scoped_refptr<MockSetSessionDescriptionObserver>
171 observer(new talk_base::RefCountedObject<
172 MockSetSessionDescriptionObserver>());
173 peer_connection_->SetLocalDescription(
174 observer, webrtc::CreateSessionDescription(type, sdp, NULL));
177 void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
178 const std::string& sdp) {
179 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
180 << ": SetRemoteDescription " << type << " " << sdp;
182 talk_base::scoped_refptr<MockSetSessionDescriptionObserver>
183 observer(new talk_base::RefCountedObject<
184 MockSetSessionDescriptionObserver>());
185 peer_connection_->SetRemoteDescription(
186 observer, webrtc::CreateSessionDescription(type, sdp, NULL));
189 void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
191 const std::string& candidate) {
192 talk_base::scoped_ptr<webrtc::IceCandidateInterface> owned_candidate(
193 webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
194 EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
197 void PeerConnectionTestWrapper::WaitForCallEstablished() {
203 void PeerConnectionTestWrapper::WaitForConnection() {
204 EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
205 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
209 bool PeerConnectionTestWrapper::CheckForConnection() {
210 return (peer_connection_->ice_connection_state() ==
211 PeerConnectionInterface::kIceConnectionConnected) ||
212 (peer_connection_->ice_connection_state() ==
213 PeerConnectionInterface::kIceConnectionCompleted);
216 void PeerConnectionTestWrapper::WaitForAudio() {
217 EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
218 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
219 << ": Got enough audio frames.";
222 bool PeerConnectionTestWrapper::CheckForAudio() {
223 return (fake_audio_capture_module_->frames_received() >=
224 kTestAudioFrameCount);
227 void PeerConnectionTestWrapper::WaitForVideo() {
228 EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
229 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
230 << ": Got enough video frames.";
233 bool PeerConnectionTestWrapper::CheckForVideo() {
237 return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
240 void PeerConnectionTestWrapper::GetAndAddUserMedia(
241 bool audio, const webrtc::FakeConstraints& audio_constraints,
242 bool video, const webrtc::FakeConstraints& video_constraints) {
243 talk_base::scoped_refptr<webrtc::MediaStreamInterface> stream =
244 GetUserMedia(audio, audio_constraints, video, video_constraints);
245 EXPECT_TRUE(peer_connection_->AddStream(stream, NULL));
248 talk_base::scoped_refptr<webrtc::MediaStreamInterface>
249 PeerConnectionTestWrapper::GetUserMedia(
250 bool audio, const webrtc::FakeConstraints& audio_constraints,
251 bool video, const webrtc::FakeConstraints& video_constraints) {
252 std::string label = kStreamLabelBase +
253 talk_base::ToString<int>(
254 static_cast<int>(peer_connection_->local_streams()->count()));
255 talk_base::scoped_refptr<webrtc::MediaStreamInterface> stream =
256 peer_connection_factory_->CreateLocalMediaStream(label);
259 FakeConstraints constraints = audio_constraints;
260 // Disable highpass filter so that we can get all the test audio frames.
261 constraints.AddMandatory(
262 MediaConstraintsInterface::kHighpassFilter, false);
263 talk_base::scoped_refptr<webrtc::AudioSourceInterface> source =
264 peer_connection_factory_->CreateAudioSource(&constraints);
265 talk_base::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
266 peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
268 stream->AddTrack(audio_track);
272 // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
273 FakeConstraints constraints = video_constraints;
274 constraints.SetMandatoryMaxFrameRate(10);
276 talk_base::scoped_refptr<webrtc::VideoSourceInterface> source =
277 peer_connection_factory_->CreateVideoSource(
278 new webrtc::FakePeriodicVideoCapturer(), &constraints);
279 std::string videotrack_label = label + kVideoTrackLabelBase;
280 talk_base::scoped_refptr<webrtc::VideoTrackInterface> video_track(
281 peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
283 stream->AddTrack(video_track);