Upstream version 7.36.149.0
[platform/framework/web/crosswalk.git] / src / third_party / libjingle / source / talk / app / webrtc / test / peerconnectiontestwrapper.cc
1 /*
2  * libjingle
3  * Copyright 2013, Google Inc.
4  *
5  * Redistribution and use in source and binary forms, with or without
6  * modification, are permitted provided that the following conditions are met:
7  *
8  *  1. Redistributions of source code must retain the above copyright notice,
9  *     this list of conditions and the following disclaimer.
10  *  2. Redistributions in binary form must reproduce the above copyright notice,
11  *     this list of conditions and the following disclaimer in the documentation
12  *     and/or other materials provided with the distribution.
13  *  3. The name of the author may not be used to endorse or promote products
14  *     derived from this software without specific prior written permission.
15  *
16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26  */
27
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"
35
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;
42
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;
52
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);
59
60   caller->SignalOnSdpReady.connect(
61       callee, &PeerConnectionTestWrapper::ReceiveOfferSdp);
62   callee->SignalOnSdpReady.connect(
63       caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
64 }
65
66 PeerConnectionTestWrapper::PeerConnectionTestWrapper(const std::string& name)
67     : name_(name) {}
68
69 PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
70
71 bool PeerConnectionTestWrapper::CreatePc(
72   const MediaConstraintsInterface* constraints) {
73   allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
74   if (!allocator_factory_) {
75     return false;
76   }
77
78   audio_thread_.Start();
79   fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
80       &audio_thread_);
81   if (fake_audio_capture_module_ == NULL) {
82     return false;
83   }
84
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_) {
89     return false;
90   }
91
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);
102
103   return peer_connection_.get() != NULL;
104 }
105
106 void PeerConnectionTestWrapper::OnAddStream(MediaStreamInterface* stream) {
107   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
108                << ": OnAddStream";
109   // TODO(ronghuawu): support multiple streams.
110   if (stream->GetVideoTracks().size() > 0) {
111     renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
112   }
113 }
114
115 void PeerConnectionTestWrapper::OnIceCandidate(
116     const IceCandidateInterface* candidate) {
117   std::string sdp;
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(),
122                             sdp);
123 }
124
125 void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
126   // This callback should take the ownership of |desc|.
127   talk_base::scoped_ptr<SessionDescriptionInterface> owned_desc(desc);
128   std::string sdp;
129   EXPECT_TRUE(desc->ToString(&sdp));
130
131   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
132                << ": " << desc->type() << " sdp created: " << sdp;
133
134   // Give the user a chance to modify sdp for testing.
135   SignalOnSdpCreated(&sdp);
136
137   SetLocalDescription(desc->type(), sdp);
138
139   SignalOnSdpReady(sdp);
140 }
141
142 void PeerConnectionTestWrapper::CreateOffer(
143     const MediaConstraintsInterface* constraints) {
144   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
145                << ": CreateOffer.";
146   peer_connection_->CreateOffer(this, constraints);
147 }
148
149 void PeerConnectionTestWrapper::CreateAnswer(
150     const MediaConstraintsInterface* constraints) {
151   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
152                << ": CreateAnswer.";
153   peer_connection_->CreateAnswer(this, constraints);
154 }
155
156 void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
157   SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp);
158   CreateAnswer(NULL);
159 }
160
161 void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
162   SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp);
163 }
164
165 void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type,
166                                                     const std::string& sdp) {
167   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
168                << ": SetLocalDescription " << type << " " << sdp;
169
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));
175 }
176
177 void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
178                                                      const std::string& sdp) {
179   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
180                << ": SetRemoteDescription " << type << " " << sdp;
181
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));
187 }
188
189 void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
190                                                 int sdp_mline_index,
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()));
195 }
196
197 void PeerConnectionTestWrapper::WaitForCallEstablished() {
198   WaitForConnection();
199   WaitForAudio();
200   WaitForVideo();
201 }
202
203 void PeerConnectionTestWrapper::WaitForConnection() {
204   EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
205   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
206                << ": Connected.";
207 }
208
209 bool PeerConnectionTestWrapper::CheckForConnection() {
210   return (peer_connection_->ice_connection_state() ==
211           PeerConnectionInterface::kIceConnectionConnected) ||
212          (peer_connection_->ice_connection_state() ==
213           PeerConnectionInterface::kIceConnectionCompleted);
214 }
215
216 void PeerConnectionTestWrapper::WaitForAudio() {
217   EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
218   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
219                << ": Got enough audio frames.";
220 }
221
222 bool PeerConnectionTestWrapper::CheckForAudio() {
223   return (fake_audio_capture_module_->frames_received() >=
224           kTestAudioFrameCount);
225 }
226
227 void PeerConnectionTestWrapper::WaitForVideo() {
228   EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
229   LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
230                << ": Got enough video frames.";
231 }
232
233 bool PeerConnectionTestWrapper::CheckForVideo() {
234   if (!renderer_) {
235     return false;
236   }
237   return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
238 }
239
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));
246 }
247
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);
257
258   if (audio) {
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,
267                                                    source));
268     stream->AddTrack(audio_track);
269   }
270
271   if (video) {
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);
275
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));
282
283     stream->AddTrack(video_track);
284   }
285   return stream;
286 }