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 "webrtc/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 = 10000;
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 fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
79 rtc::Thread::Current());
80 if (fake_audio_capture_module_ == NULL) {
84 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
85 rtc::Thread::Current(), rtc::Thread::Current(),
86 fake_audio_capture_module_, NULL, NULL);
87 if (!peer_connection_factory_) {
91 // CreatePeerConnection with IceServers.
92 webrtc::PeerConnectionInterface::IceServers ice_servers;
93 webrtc::PeerConnectionInterface::IceServer ice_server;
94 ice_server.uri = "stun:stun.l.google.com:19302";
95 ice_servers.push_back(ice_server);
96 FakeIdentityService* dtls_service =
97 rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
98 new FakeIdentityService() : NULL;
99 peer_connection_ = peer_connection_factory_->CreatePeerConnection(
100 ice_servers, constraints, allocator_factory_.get(), dtls_service, this);
102 return peer_connection_.get() != NULL;
105 rtc::scoped_refptr<webrtc::DataChannelInterface>
106 PeerConnectionTestWrapper::CreateDataChannel(
107 const std::string& label,
108 const webrtc::DataChannelInit& init) {
109 return peer_connection_->CreateDataChannel(label, &init);
112 void PeerConnectionTestWrapper::OnAddStream(MediaStreamInterface* stream) {
113 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
115 // TODO(ronghuawu): support multiple streams.
116 if (stream->GetVideoTracks().size() > 0) {
117 renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
121 void PeerConnectionTestWrapper::OnIceCandidate(
122 const IceCandidateInterface* candidate) {
124 EXPECT_TRUE(candidate->ToString(&sdp));
125 // Give the user a chance to modify sdp for testing.
126 SignalOnIceCandidateCreated(&sdp);
127 SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
131 void PeerConnectionTestWrapper::OnDataChannel(
132 webrtc::DataChannelInterface* data_channel) {
133 SignalOnDataChannel(data_channel);
136 void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
137 // This callback should take the ownership of |desc|.
138 rtc::scoped_ptr<SessionDescriptionInterface> owned_desc(desc);
140 EXPECT_TRUE(desc->ToString(&sdp));
142 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
143 << ": " << desc->type() << " sdp created: " << sdp;
145 // Give the user a chance to modify sdp for testing.
146 SignalOnSdpCreated(&sdp);
148 SetLocalDescription(desc->type(), sdp);
150 SignalOnSdpReady(sdp);
153 void PeerConnectionTestWrapper::CreateOffer(
154 const MediaConstraintsInterface* constraints) {
155 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
157 peer_connection_->CreateOffer(this, constraints);
160 void PeerConnectionTestWrapper::CreateAnswer(
161 const MediaConstraintsInterface* constraints) {
162 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
163 << ": CreateAnswer.";
164 peer_connection_->CreateAnswer(this, constraints);
167 void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
168 SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp);
172 void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
173 SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp);
176 void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type,
177 const std::string& sdp) {
178 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
179 << ": SetLocalDescription " << type << " " << sdp;
181 rtc::scoped_refptr<MockSetSessionDescriptionObserver>
182 observer(new rtc::RefCountedObject<
183 MockSetSessionDescriptionObserver>());
184 peer_connection_->SetLocalDescription(
185 observer, webrtc::CreateSessionDescription(type, sdp, NULL));
188 void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
189 const std::string& sdp) {
190 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
191 << ": SetRemoteDescription " << type << " " << sdp;
193 rtc::scoped_refptr<MockSetSessionDescriptionObserver>
194 observer(new rtc::RefCountedObject<
195 MockSetSessionDescriptionObserver>());
196 peer_connection_->SetRemoteDescription(
197 observer, webrtc::CreateSessionDescription(type, sdp, NULL));
200 void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
202 const std::string& candidate) {
203 rtc::scoped_ptr<webrtc::IceCandidateInterface> owned_candidate(
204 webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
205 EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
208 void PeerConnectionTestWrapper::WaitForCallEstablished() {
214 void PeerConnectionTestWrapper::WaitForConnection() {
215 EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
216 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
220 bool PeerConnectionTestWrapper::CheckForConnection() {
221 return (peer_connection_->ice_connection_state() ==
222 PeerConnectionInterface::kIceConnectionConnected) ||
223 (peer_connection_->ice_connection_state() ==
224 PeerConnectionInterface::kIceConnectionCompleted);
227 void PeerConnectionTestWrapper::WaitForAudio() {
228 EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
229 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
230 << ": Got enough audio frames.";
233 bool PeerConnectionTestWrapper::CheckForAudio() {
234 return (fake_audio_capture_module_->frames_received() >=
235 kTestAudioFrameCount);
238 void PeerConnectionTestWrapper::WaitForVideo() {
239 EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
240 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
241 << ": Got enough video frames.";
244 bool PeerConnectionTestWrapper::CheckForVideo() {
248 return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
251 void PeerConnectionTestWrapper::GetAndAddUserMedia(
252 bool audio, const webrtc::FakeConstraints& audio_constraints,
253 bool video, const webrtc::FakeConstraints& video_constraints) {
254 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
255 GetUserMedia(audio, audio_constraints, video, video_constraints);
256 EXPECT_TRUE(peer_connection_->AddStream(stream, NULL));
259 rtc::scoped_refptr<webrtc::MediaStreamInterface>
260 PeerConnectionTestWrapper::GetUserMedia(
261 bool audio, const webrtc::FakeConstraints& audio_constraints,
262 bool video, const webrtc::FakeConstraints& video_constraints) {
263 std::string label = kStreamLabelBase +
265 static_cast<int>(peer_connection_->local_streams()->count()));
266 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
267 peer_connection_factory_->CreateLocalMediaStream(label);
270 FakeConstraints constraints = audio_constraints;
271 // Disable highpass filter so that we can get all the test audio frames.
272 constraints.AddMandatory(
273 MediaConstraintsInterface::kHighpassFilter, false);
274 rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
275 peer_connection_factory_->CreateAudioSource(&constraints);
276 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
277 peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
279 stream->AddTrack(audio_track);
283 // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
284 FakeConstraints constraints = video_constraints;
285 constraints.SetMandatoryMaxFrameRate(10);
287 rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
288 peer_connection_factory_->CreateVideoSource(
289 new webrtc::FakePeriodicVideoCapturer(), &constraints);
290 std::string videotrack_label = label + kVideoTrackLabelBase;
291 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
292 peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
294 stream->AddTrack(video_track);