src/content/renderer/media/rtc_peer_connection_handler_unittest.cc

   1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include <string>
   6 #include <vector>
   7
   8 #include "base/memory/scoped_ptr.h"
   9 #include "base/message_loop/message_loop.h"
  10 #include "base/strings/utf_string_conversions.h"
  11 #include "base/values.h"
  12 #include "content/child/child_process.h"
  13 #include "content/renderer/media/media_stream.h"
  14 #include "content/renderer/media/media_stream_audio_source.h"
  15 #include "content/renderer/media/media_stream_source.h"
  16 #include "content/renderer/media/media_stream_video_track.h"
  17 #include "content/renderer/media/mock_media_stream_video_source.h"
  18 #include "content/renderer/media/mock_peer_connection_impl.h"
  19 #include "content/renderer/media/mock_web_rtc_peer_connection_handler_client.h"
  20 #include "content/renderer/media/peer_connection_tracker.h"
  21 #include "content/renderer/media/rtc_media_constraints.h"
  22 #include "content/renderer/media/rtc_peer_connection_handler.h"
  23 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
  24 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
  25 #include "content/renderer/media/webrtc_audio_capturer.h"
  26 #include "testing/gmock/include/gmock/gmock.h"
  27 #include "testing/gtest/include/gtest/gtest.h"
  28 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
  29 #include "third_party/WebKit/public/platform/WebMediaStream.h"
  30 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
  31 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  32 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
  33 #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h"
  34 #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h"
  35 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
  36 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
  37 #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandlerClient.h"
  38 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
  39 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
  40 #include "third_party/WebKit/public/platform/WebRTCStatsRequest.h"
  41 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
  42 #include "third_party/WebKit/public/platform/WebURL.h"
  43 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
  44
  45 static const char kDummySdp[] = "dummy sdp";
  46 static const char kDummySdpType[] = "dummy type";
  47
  48 using blink::WebRTCPeerConnectionHandlerClient;
  49 using testing::NiceMock;
  50 using testing::_;
  51 using testing::Ref;
  52
  53 namespace content {
  54
  55 class MockRTCStatsResponse : public LocalRTCStatsResponse {
  56  public:
  57   MockRTCStatsResponse()
  58       : report_count_(0),
  59         statistic_count_(0) {
  60   }
  61
  62   virtual size_t addReport(blink::WebString type,
  63                            blink::WebString id,
  64                            double timestamp) OVERRIDE {
  65     ++report_count_;
  66     return report_count_;
  67   }
  68
  69   virtual void addStatistic(size_t report,
  70                             blink::WebString name, blink::WebString value)
  71       OVERRIDE {
  72     ++statistic_count_;
  73   }
  74   int report_count() const { return report_count_; }
  75
  76  private:
  77   int report_count_;
  78   int statistic_count_;
  79 };
  80
  81 // Mocked wrapper for blink::WebRTCStatsRequest
  82 class MockRTCStatsRequest : public LocalRTCStatsRequest {
  83  public:
  84   MockRTCStatsRequest()
  85       : has_selector_(false),
  86         request_succeeded_called_(false) {}
  87
  88   virtual bool hasSelector() const OVERRIDE {
  89     return has_selector_;
  90   }
  91   virtual blink::WebMediaStreamTrack component() const OVERRIDE {
  92     return component_;
  93   }
  94   virtual scoped_refptr<LocalRTCStatsResponse> createResponse() OVERRIDE {
  95     DCHECK(!response_.get());
  96     response_ = new rtc::RefCountedObject<MockRTCStatsResponse>();
  97     return response_;
  98   }
  99
 100   virtual void requestSucceeded(const LocalRTCStatsResponse* response)
 101       OVERRIDE {
 102     EXPECT_EQ(response, response_.get());
 103     request_succeeded_called_ = true;
 104   }
 105
 106   // Function for setting whether or not a selector is available.
 107   void setSelector(const blink::WebMediaStreamTrack& component) {
 108     has_selector_ = true;
 109     component_ = component;
 110   }
 111
 112   // Function for inspecting the result of a stats request.
 113   MockRTCStatsResponse* result() {
 114     if (request_succeeded_called_) {
 115       return response_.get();
 116     } else {
 117       return NULL;
 118     }
 119   }
 120
 121  private:
 122   bool has_selector_;
 123   blink::WebMediaStreamTrack component_;
 124   scoped_refptr<MockRTCStatsResponse> response_;
 125   bool request_succeeded_called_;
 126 };
 127
 128 class MockPeerConnectionTracker : public PeerConnectionTracker {
 129  public:
 130   MOCK_METHOD1(UnregisterPeerConnection,
 131                void(RTCPeerConnectionHandler* pc_handler));
 132   // TODO(jiayl): add coverage for the following methods
 133   MOCK_METHOD2(TrackCreateOffer,
 134                void(RTCPeerConnectionHandler* pc_handler,
 135                     const RTCMediaConstraints& constraints));
 136   MOCK_METHOD2(TrackCreateAnswer,
 137                void(RTCPeerConnectionHandler* pc_handler,
 138                     const RTCMediaConstraints& constraints));
 139   MOCK_METHOD3(TrackSetSessionDescription,
 140                void(RTCPeerConnectionHandler* pc_handler,
 141                     const blink::WebRTCSessionDescription& desc,
 142                     Source source));
 143   MOCK_METHOD3(
 144       TrackUpdateIce,
 145       void(RTCPeerConnectionHandler* pc_handler,
 146            const webrtc::PeerConnectionInterface::RTCConfiguration& config,
 147            const RTCMediaConstraints& options));
 148   MOCK_METHOD3(TrackAddIceCandidate,
 149                void(RTCPeerConnectionHandler* pc_handler,
 150                     const blink::WebRTCICECandidate& candidate,
 151                     Source source));
 152   MOCK_METHOD3(TrackAddStream,
 153                void(RTCPeerConnectionHandler* pc_handler,
 154                     const blink::WebMediaStream& stream,
 155                     Source source));
 156   MOCK_METHOD3(TrackRemoveStream,
 157                void(RTCPeerConnectionHandler* pc_handler,
 158                     const blink::WebMediaStream& stream,
 159                     Source source));
 160   MOCK_METHOD1(TrackOnIceComplete,
 161                void(RTCPeerConnectionHandler* pc_handler));
 162   MOCK_METHOD3(TrackCreateDataChannel,
 163                void(RTCPeerConnectionHandler* pc_handler,
 164                     const webrtc::DataChannelInterface* data_channel,
 165                     Source source));
 166   MOCK_METHOD1(TrackStop, void(RTCPeerConnectionHandler* pc_handler));
 167   MOCK_METHOD2(TrackSignalingStateChange,
 168                void(RTCPeerConnectionHandler* pc_handler,
 169                     WebRTCPeerConnectionHandlerClient::SignalingState state));
 170   MOCK_METHOD2(
 171       TrackIceConnectionStateChange,
 172       void(RTCPeerConnectionHandler* pc_handler,
 173            WebRTCPeerConnectionHandlerClient::ICEConnectionState state));
 174   MOCK_METHOD2(
 175       TrackIceGatheringStateChange,
 176       void(RTCPeerConnectionHandler* pc_handler,
 177            WebRTCPeerConnectionHandlerClient::ICEGatheringState state));
 178   MOCK_METHOD1(TrackOnRenegotiationNeeded,
 179                void(RTCPeerConnectionHandler* pc_handler));
 180   MOCK_METHOD2(TrackCreateDTMFSender,
 181                void(RTCPeerConnectionHandler* pc_handler,
 182                      const blink::WebMediaStreamTrack& track));
 183 };
 184
 185 class RTCPeerConnectionHandlerUnderTest : public RTCPeerConnectionHandler {
 186  public:
 187   RTCPeerConnectionHandlerUnderTest(
 188       WebRTCPeerConnectionHandlerClient* client,
 189       PeerConnectionDependencyFactory* dependency_factory)
 190       : RTCPeerConnectionHandler(client, dependency_factory) {
 191   }
 192
 193   MockPeerConnectionImpl* native_peer_connection() {
 194     return static_cast<MockPeerConnectionImpl*>(
 195         RTCPeerConnectionHandler::native_peer_connection());
 196   }
 197 };
 198
 199 class RTCPeerConnectionHandlerTest : public ::testing::Test {
 200  public:
 201   RTCPeerConnectionHandlerTest() : mock_peer_connection_(NULL) {
 202     child_process_.reset(new ChildProcess());
 203   }
 204
 205   virtual void SetUp() {
 206     mock_client_.reset(new NiceMock<MockWebRTCPeerConnectionHandlerClient>());
 207     mock_dependency_factory_.reset(new MockPeerConnectionDependencyFactory());
 208     pc_handler_.reset(
 209         new RTCPeerConnectionHandlerUnderTest(mock_client_.get(),
 210                                               mock_dependency_factory_.get()));
 211     mock_tracker_.reset(new NiceMock<MockPeerConnectionTracker>());
 212     blink::WebRTCConfiguration config;
 213     blink::WebMediaConstraints constraints;
 214     EXPECT_TRUE(pc_handler_->InitializeForTest(config, constraints,
 215                                                mock_tracker_.get()));
 216
 217     mock_peer_connection_ = pc_handler_->native_peer_connection();
 218     ASSERT_TRUE(mock_peer_connection_);
 219   }
 220
 221   // Creates a WebKit local MediaStream.
 222   blink::WebMediaStream CreateLocalMediaStream(
 223       const std::string& stream_label) {
 224     std::string video_track_label("video-label");
 225     std::string audio_track_label("audio-label");
 226
 227     blink::WebMediaStreamSource audio_source;
 228     audio_source.initialize(blink::WebString::fromUTF8(audio_track_label),
 229                             blink::WebMediaStreamSource::TypeAudio,
 230                             blink::WebString::fromUTF8("audio_track"));
 231     audio_source.setExtraData(new MediaStreamAudioSource());
 232     blink::WebMediaStreamSource video_source;
 233     video_source.initialize(blink::WebString::fromUTF8(video_track_label),
 234                             blink::WebMediaStreamSource::TypeVideo,
 235                             blink::WebString::fromUTF8("video_track"));
 236     MockMediaStreamVideoSource* native_video_source =
 237         new MockMediaStreamVideoSource(false);
 238     video_source.setExtraData(native_video_source);
 239
 240     blink::WebVector<blink::WebMediaStreamTrack> audio_tracks(
 241         static_cast<size_t>(1));
 242     audio_tracks[0].initialize(audio_source.id(), audio_source);
 243     audio_tracks[0].setExtraData(
 244         new MediaStreamTrack(
 245             WebRtcLocalAudioTrackAdapter::Create(audio_track_label,
 246                                                  NULL),
 247                                                  true));
 248     blink::WebVector<blink::WebMediaStreamTrack> video_tracks(
 249         static_cast<size_t>(1));
 250     blink::WebMediaConstraints constraints;
 251     constraints.initialize();
 252     video_tracks[0] = MediaStreamVideoTrack::CreateVideoTrack(
 253         native_video_source, constraints,
 254         MediaStreamVideoSource::ConstraintsCallback(), true);
 255
 256     blink::WebMediaStream local_stream;
 257     local_stream.initialize(base::UTF8ToUTF16(stream_label), audio_tracks,
 258                             video_tracks);
 259     local_stream.setExtraData(
 260         new MediaStream(local_stream));
 261     return local_stream;
 262   }
 263
 264   // Creates a remote MediaStream and adds it to the mocked native
 265   // peer connection.
 266   scoped_refptr<webrtc::MediaStreamInterface>
 267   AddRemoteMockMediaStream(const std::string& stream_label,
 268                            const std::string& video_track_label,
 269                            const std::string& audio_track_label) {
 270     scoped_refptr<webrtc::MediaStreamInterface> stream(
 271         mock_dependency_factory_->CreateLocalMediaStream(stream_label));
 272     if (!video_track_label.empty()) {
 273       webrtc::VideoSourceInterface* source = NULL;
 274       scoped_refptr<webrtc::VideoTrackInterface> video_track(
 275           mock_dependency_factory_->CreateLocalVideoTrack(
 276               video_track_label, source));
 277       stream->AddTrack(video_track.get());
 278     }
 279     if (!audio_track_label.empty()) {
 280       scoped_refptr<WebRtcAudioCapturer> capturer;
 281       scoped_refptr<webrtc::AudioTrackInterface> audio_track(
 282           WebRtcLocalAudioTrackAdapter::Create(audio_track_label, NULL));
 283       stream->AddTrack(audio_track.get());
 284     }
 285     mock_peer_connection_->AddRemoteStream(stream.get());
 286     return stream;
 287   }
 288
 289   base::MessageLoop message_loop_;
 290   scoped_ptr<ChildProcess> child_process_;
 291   scoped_ptr<MockWebRTCPeerConnectionHandlerClient> mock_client_;
 292   scoped_ptr<MockPeerConnectionDependencyFactory> mock_dependency_factory_;
 293   scoped_ptr<NiceMock<MockPeerConnectionTracker> > mock_tracker_;
 294   scoped_ptr<RTCPeerConnectionHandlerUnderTest> pc_handler_;
 295
 296   // Weak reference to the mocked native peer connection implementation.
 297   MockPeerConnectionImpl* mock_peer_connection_;
 298 };
 299
 300 TEST_F(RTCPeerConnectionHandlerTest, Destruct) {
 301   EXPECT_CALL(*mock_tracker_.get(), UnregisterPeerConnection(pc_handler_.get()))
 302       .Times(1);
 303   pc_handler_.reset(NULL);
 304 }
 305
 306 TEST_F(RTCPeerConnectionHandlerTest, CreateOffer) {
 307   blink::WebRTCSessionDescriptionRequest request;
 308   blink::WebMediaConstraints options;
 309   EXPECT_CALL(*mock_tracker_.get(), TrackCreateOffer(pc_handler_.get(), _));
 310
 311   // TODO(perkj): Can blink::WebRTCSessionDescriptionRequest be changed so
 312   // the |reqest| requestSucceeded can be tested? Currently the |request| object
 313   // can not be initialized from a unit test.
 314   EXPECT_FALSE(mock_peer_connection_->created_session_description() != NULL);
 315   pc_handler_->createOffer(request, options);
 316   EXPECT_TRUE(mock_peer_connection_->created_session_description() != NULL);
 317 }
 318
 319 TEST_F(RTCPeerConnectionHandlerTest, CreateAnswer) {
 320   blink::WebRTCSessionDescriptionRequest request;
 321   blink::WebMediaConstraints options;
 322   EXPECT_CALL(*mock_tracker_.get(), TrackCreateAnswer(pc_handler_.get(), _));
 323   // TODO(perkj): Can blink::WebRTCSessionDescriptionRequest be changed so
 324   // the |reqest| requestSucceeded can be tested? Currently the |request| object
 325   // can not be initialized from a unit test.
 326   EXPECT_FALSE(mock_peer_connection_->created_session_description() != NULL);
 327   pc_handler_->createAnswer(request, options);
 328   EXPECT_TRUE(mock_peer_connection_->created_session_description() != NULL);
 329 }
 330
 331 TEST_F(RTCPeerConnectionHandlerTest, setLocalDescription) {
 332   blink::WebRTCVoidRequest request;
 333   blink::WebRTCSessionDescription description;
 334   description.initialize(kDummySdpType, kDummySdp);
 335   // PeerConnectionTracker::TrackSetSessionDescription is expected to be called
 336   // before |mock_peer_connection| is called.
 337   testing::InSequence sequence;
 338   EXPECT_CALL(*mock_tracker_.get(),
 339               TrackSetSessionDescription(pc_handler_.get(), Ref(description),
 340                                          PeerConnectionTracker::SOURCE_LOCAL));
 341   EXPECT_CALL(*mock_peer_connection_, SetLocalDescription(_, _));
 342
 343   pc_handler_->setLocalDescription(request, description);
 344   EXPECT_EQ(description.type(), pc_handler_->localDescription().type());
 345   EXPECT_EQ(description.sdp(), pc_handler_->localDescription().sdp());
 346
 347   std::string sdp_string;
 348   ASSERT_TRUE(mock_peer_connection_->local_description() != NULL);
 349   EXPECT_EQ(kDummySdpType, mock_peer_connection_->local_description()->type());
 350   mock_peer_connection_->local_description()->ToString(&sdp_string);
 351   EXPECT_EQ(kDummySdp, sdp_string);
 352 }
 353
 354 TEST_F(RTCPeerConnectionHandlerTest, setRemoteDescription) {
 355   blink::WebRTCVoidRequest request;
 356   blink::WebRTCSessionDescription description;
 357   description.initialize(kDummySdpType, kDummySdp);
 358
 359   // PeerConnectionTracker::TrackSetSessionDescription is expected to be called
 360   // before |mock_peer_connection| is called.
 361   testing::InSequence sequence;
 362   EXPECT_CALL(*mock_tracker_.get(),
 363               TrackSetSessionDescription(pc_handler_.get(), Ref(description),
 364                                          PeerConnectionTracker::SOURCE_REMOTE));
 365   EXPECT_CALL(*mock_peer_connection_, SetRemoteDescription(_, _));
 366
 367   pc_handler_->setRemoteDescription(request, description);
 368   EXPECT_EQ(description.type(), pc_handler_->remoteDescription().type());
 369   EXPECT_EQ(description.sdp(), pc_handler_->remoteDescription().sdp());
 370
 371   std::string sdp_string;
 372   ASSERT_TRUE(mock_peer_connection_->remote_description() != NULL);
 373   EXPECT_EQ(kDummySdpType, mock_peer_connection_->remote_description()->type());
 374   mock_peer_connection_->remote_description()->ToString(&sdp_string);
 375   EXPECT_EQ(kDummySdp, sdp_string);
 376 }
 377
 378 TEST_F(RTCPeerConnectionHandlerTest, updateICE) {
 379   blink::WebRTCConfiguration config;
 380   blink::WebMediaConstraints constraints;
 381
 382   EXPECT_CALL(*mock_tracker_.get(), TrackUpdateIce(pc_handler_.get(), _, _));
 383   // TODO(perkj): Test that the parameters in |config| can be translated when a
 384   // WebRTCConfiguration can be constructed. It's WebKit class and can't be
 385   // initialized from a test.
 386   EXPECT_TRUE(pc_handler_->updateICE(config, constraints));
 387 }
 388
 389 TEST_F(RTCPeerConnectionHandlerTest, addICECandidate) {
 390   blink::WebRTCICECandidate candidate;
 391   candidate.initialize(kDummySdp, "mid", 1);
 392
 393   EXPECT_CALL(*mock_tracker_.get(),
 394               TrackAddIceCandidate(pc_handler_.get(),
 395                                    testing::Ref(candidate),
 396                                    PeerConnectionTracker::SOURCE_REMOTE));
 397   EXPECT_TRUE(pc_handler_->addICECandidate(candidate));
 398   EXPECT_EQ(kDummySdp, mock_peer_connection_->ice_sdp());
 399   EXPECT_EQ(1, mock_peer_connection_->sdp_mline_index());
 400   EXPECT_EQ("mid", mock_peer_connection_->sdp_mid());
 401 }
 402
 403 TEST_F(RTCPeerConnectionHandlerTest, addAndRemoveStream) {
 404   std::string stream_label = "local_stream";
 405   blink::WebMediaStream local_stream(
 406       CreateLocalMediaStream(stream_label));
 407   blink::WebMediaConstraints constraints;
 408
 409   EXPECT_CALL(*mock_tracker_.get(),
 410               TrackAddStream(pc_handler_.get(),
 411                              testing::Ref(local_stream),
 412                              PeerConnectionTracker::SOURCE_LOCAL));
 413   EXPECT_CALL(*mock_tracker_.get(),
 414               TrackRemoveStream(pc_handler_.get(),
 415                                 testing::Ref(local_stream),
 416                                 PeerConnectionTracker::SOURCE_LOCAL));
 417   EXPECT_TRUE(pc_handler_->addStream(local_stream, constraints));
 418   EXPECT_EQ(stream_label, mock_peer_connection_->stream_label());
 419   EXPECT_EQ(1u,
 420       mock_peer_connection_->local_streams()->at(0)->GetAudioTracks().size());
 421   EXPECT_EQ(1u,
 422       mock_peer_connection_->local_streams()->at(0)->GetVideoTracks().size());
 423
 424   EXPECT_FALSE(pc_handler_->addStream(local_stream, constraints));
 425
 426   pc_handler_->removeStream(local_stream);
 427   EXPECT_EQ(0u, mock_peer_connection_->local_streams()->count());
 428 }
 429
 430 TEST_F(RTCPeerConnectionHandlerTest, addStreamWithStoppedAudioAndVideoTrack) {
 431   std::string stream_label = "local_stream";
 432   blink::WebMediaStream local_stream(
 433       CreateLocalMediaStream(stream_label));
 434   blink::WebMediaConstraints constraints;
 435
 436   blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
 437   local_stream.audioTracks(audio_tracks);
 438   MediaStreamAudioSource* native_audio_source =
 439       static_cast<MediaStreamAudioSource*>(
 440           audio_tracks[0].source().extraData());
 441   native_audio_source->StopSource();
 442
 443   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
 444   local_stream.videoTracks(video_tracks);
 445   MediaStreamVideoSource* native_video_source =
 446       static_cast<MediaStreamVideoSource*>(
 447           video_tracks[0].source().extraData());
 448   native_video_source->StopSource();
 449
 450   EXPECT_TRUE(pc_handler_->addStream(local_stream, constraints));
 451   EXPECT_EQ(stream_label, mock_peer_connection_->stream_label());
 452   EXPECT_EQ(
 453       1u,
 454       mock_peer_connection_->local_streams()->at(0)->GetAudioTracks().size());
 455   EXPECT_EQ(
 456       1u,
 457       mock_peer_connection_->local_streams()->at(0)->GetVideoTracks().size());
 458 }
 459
 460 TEST_F(RTCPeerConnectionHandlerTest, GetStatsNoSelector) {
 461   scoped_refptr<MockRTCStatsRequest> request(
 462       new rtc::RefCountedObject<MockRTCStatsRequest>());
 463   pc_handler_->getStats(request.get());
 464   // Note that callback gets executed synchronously by mock.
 465   ASSERT_TRUE(request->result());
 466   EXPECT_LT(1, request->result()->report_count());
 467 }
 468
 469 TEST_F(RTCPeerConnectionHandlerTest, GetStatsAfterClose) {
 470   scoped_refptr<MockRTCStatsRequest> request(
 471       new rtc::RefCountedObject<MockRTCStatsRequest>());
 472   pc_handler_->stop();
 473   pc_handler_->getStats(request.get());
 474   // Note that callback gets executed synchronously by mock.
 475   ASSERT_TRUE(request->result());
 476   EXPECT_LT(1, request->result()->report_count());
 477 }
 478
 479 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithLocalSelector) {
 480   blink::WebMediaStream local_stream(
 481       CreateLocalMediaStream("local_stream"));
 482   blink::WebMediaConstraints constraints;
 483   pc_handler_->addStream(local_stream, constraints);
 484   blink::WebVector<blink::WebMediaStreamTrack> tracks;
 485   local_stream.audioTracks(tracks);
 486   ASSERT_LE(1ul, tracks.size());
 487
 488   scoped_refptr<MockRTCStatsRequest> request(
 489       new rtc::RefCountedObject<MockRTCStatsRequest>());
 490   request->setSelector(tracks[0]);
 491   pc_handler_->getStats(request.get());
 492   EXPECT_EQ(1, request->result()->report_count());
 493 }
 494
 495 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithRemoteSelector) {
 496   scoped_refptr<webrtc::MediaStreamInterface> stream(
 497       AddRemoteMockMediaStream("remote_stream", "video", "audio"));
 498   pc_handler_->OnAddStream(stream.get());
 499   const blink::WebMediaStream& remote_stream = mock_client_->remote_stream();
 500
 501   blink::WebVector<blink::WebMediaStreamTrack> tracks;
 502   remote_stream.audioTracks(tracks);
 503   ASSERT_LE(1ul, tracks.size());
 504
 505   scoped_refptr<MockRTCStatsRequest> request(
 506       new rtc::RefCountedObject<MockRTCStatsRequest>());
 507   request->setSelector(tracks[0]);
 508   pc_handler_->getStats(request.get());
 509   EXPECT_EQ(1, request->result()->report_count());
 510 }
 511
 512 TEST_F(RTCPeerConnectionHandlerTest, GetStatsWithBadSelector) {
 513   // The setup is the same as GetStatsWithLocalSelector, but the stream is not
 514   // added to the PeerConnection.
 515   blink::WebMediaStream local_stream(
 516       CreateLocalMediaStream("local_stream_2"));
 517   blink::WebMediaConstraints constraints;
 518   blink::WebVector<blink::WebMediaStreamTrack> tracks;
 519
 520   local_stream.audioTracks(tracks);
 521   blink::WebMediaStreamTrack component = tracks[0];
 522   mock_peer_connection_->SetGetStatsResult(false);
 523
 524   scoped_refptr<MockRTCStatsRequest> request(
 525       new rtc::RefCountedObject<MockRTCStatsRequest>());
 526   request->setSelector(component);
 527   pc_handler_->getStats(request.get());
 528   EXPECT_EQ(0, request->result()->report_count());
 529 }
 530
 531 TEST_F(RTCPeerConnectionHandlerTest, OnSignalingChange) {
 532   testing::InSequence sequence;
 533
 534   webrtc::PeerConnectionInterface::SignalingState new_state =
 535       webrtc::PeerConnectionInterface::kHaveRemoteOffer;
 536   EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
 537       pc_handler_.get(),
 538       WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemoteOffer));
 539   EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
 540       WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemoteOffer));
 541   pc_handler_->OnSignalingChange(new_state);
 542
 543   new_state = webrtc::PeerConnectionInterface::kHaveLocalPrAnswer;
 544   EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
 545       pc_handler_.get(),
 546       WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalPrAnswer));
 547   EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
 548       WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalPrAnswer));
 549   pc_handler_->OnSignalingChange(new_state);
 550
 551   new_state = webrtc::PeerConnectionInterface::kHaveLocalOffer;
 552   EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
 553       pc_handler_.get(),
 554       WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalOffer));
 555   EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
 556       WebRTCPeerConnectionHandlerClient::SignalingStateHaveLocalOffer));
 557   pc_handler_->OnSignalingChange(new_state);
 558
 559   new_state = webrtc::PeerConnectionInterface::kHaveRemotePrAnswer;
 560   EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
 561       pc_handler_.get(),
 562       WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemotePrAnswer));
 563   EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
 564       WebRTCPeerConnectionHandlerClient::SignalingStateHaveRemotePrAnswer));
 565   pc_handler_->OnSignalingChange(new_state);
 566
 567   new_state = webrtc::PeerConnectionInterface::kClosed;
 568   EXPECT_CALL(*mock_tracker_.get(), TrackSignalingStateChange(
 569       pc_handler_.get(),
 570       WebRTCPeerConnectionHandlerClient::SignalingStateClosed));
 571   EXPECT_CALL(*mock_client_.get(), didChangeSignalingState(
 572       WebRTCPeerConnectionHandlerClient::SignalingStateClosed));
 573   pc_handler_->OnSignalingChange(new_state);
 574 }
 575
 576 TEST_F(RTCPeerConnectionHandlerTest, OnIceConnectionChange) {
 577   testing::InSequence sequence;
 578
 579   webrtc::PeerConnectionInterface::IceConnectionState new_state =
 580       webrtc::PeerConnectionInterface::kIceConnectionNew;
 581   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 582       pc_handler_.get(),
 583       WebRTCPeerConnectionHandlerClient::ICEConnectionStateStarting));
 584   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 585       WebRTCPeerConnectionHandlerClient::ICEConnectionStateStarting));
 586   pc_handler_->OnIceConnectionChange(new_state);
 587
 588   new_state = webrtc::PeerConnectionInterface::kIceConnectionChecking;
 589   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 590       pc_handler_.get(),
 591       WebRTCPeerConnectionHandlerClient::ICEConnectionStateChecking));
 592   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 593       WebRTCPeerConnectionHandlerClient::ICEConnectionStateChecking));
 594   pc_handler_->OnIceConnectionChange(new_state);
 595
 596   new_state = webrtc::PeerConnectionInterface::kIceConnectionConnected;
 597   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 598       pc_handler_.get(),
 599       WebRTCPeerConnectionHandlerClient::ICEConnectionStateConnected));
 600   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 601       WebRTCPeerConnectionHandlerClient::ICEConnectionStateConnected));
 602   pc_handler_->OnIceConnectionChange(new_state);
 603
 604   new_state = webrtc::PeerConnectionInterface::kIceConnectionCompleted;
 605   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 606       pc_handler_.get(),
 607       WebRTCPeerConnectionHandlerClient::ICEConnectionStateCompleted));
 608   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 609       WebRTCPeerConnectionHandlerClient::ICEConnectionStateCompleted));
 610   pc_handler_->OnIceConnectionChange(new_state);
 611
 612   new_state = webrtc::PeerConnectionInterface::kIceConnectionFailed;
 613   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 614       pc_handler_.get(),
 615       WebRTCPeerConnectionHandlerClient::ICEConnectionStateFailed));
 616   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 617       WebRTCPeerConnectionHandlerClient::ICEConnectionStateFailed));
 618   pc_handler_->OnIceConnectionChange(new_state);
 619
 620   new_state = webrtc::PeerConnectionInterface::kIceConnectionDisconnected;
 621   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 622       pc_handler_.get(),
 623       WebRTCPeerConnectionHandlerClient::ICEConnectionStateDisconnected));
 624   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 625       WebRTCPeerConnectionHandlerClient::ICEConnectionStateDisconnected));
 626   pc_handler_->OnIceConnectionChange(new_state);
 627
 628   new_state = webrtc::PeerConnectionInterface::kIceConnectionClosed;
 629   EXPECT_CALL(*mock_tracker_.get(), TrackIceConnectionStateChange(
 630       pc_handler_.get(),
 631       WebRTCPeerConnectionHandlerClient::ICEConnectionStateClosed));
 632   EXPECT_CALL(*mock_client_.get(), didChangeICEConnectionState(
 633       WebRTCPeerConnectionHandlerClient::ICEConnectionStateClosed));
 634   pc_handler_->OnIceConnectionChange(new_state);
 635 }
 636
 637 TEST_F(RTCPeerConnectionHandlerTest, OnIceGatheringChange) {
 638   testing::InSequence sequence;
 639   EXPECT_CALL(*mock_tracker_.get(), TrackIceGatheringStateChange(
 640       pc_handler_.get(),
 641       WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew));
 642   EXPECT_CALL(*mock_client_.get(), didChangeICEGatheringState(
 643       WebRTCPeerConnectionHandlerClient::ICEGatheringStateNew));
 644   EXPECT_CALL(*mock_tracker_.get(), TrackIceGatheringStateChange(
 645       pc_handler_.get(),
 646       WebRTCPeerConnectionHandlerClient::ICEGatheringStateGathering));
 647   EXPECT_CALL(*mock_client_.get(), didChangeICEGatheringState(
 648       WebRTCPeerConnectionHandlerClient::ICEGatheringStateGathering));
 649   EXPECT_CALL(*mock_tracker_.get(), TrackIceGatheringStateChange(
 650       pc_handler_.get(),
 651       WebRTCPeerConnectionHandlerClient::ICEGatheringStateComplete));
 652   EXPECT_CALL(*mock_client_.get(), didChangeICEGatheringState(
 653       WebRTCPeerConnectionHandlerClient::ICEGatheringStateComplete));
 654
 655   webrtc::PeerConnectionInterface::IceGatheringState new_state =
 656         webrtc::PeerConnectionInterface::kIceGatheringNew;
 657   pc_handler_->OnIceGatheringChange(new_state);
 658
 659   new_state = webrtc::PeerConnectionInterface::kIceGatheringGathering;
 660   pc_handler_->OnIceGatheringChange(new_state);
 661
 662   new_state = webrtc::PeerConnectionInterface::kIceGatheringComplete;
 663   pc_handler_->OnIceGatheringChange(new_state);
 664
 665   // Check NULL candidate after ice gathering is completed.
 666   EXPECT_EQ("", mock_client_->candidate_mid());
 667   EXPECT_EQ(-1, mock_client_->candidate_mlineindex());
 668   EXPECT_EQ("", mock_client_->candidate_sdp());
 669 }
 670
 671 TEST_F(RTCPeerConnectionHandlerTest, OnAddAndOnRemoveStream) {
 672   std::string remote_stream_label("remote_stream");
 673   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
 674       AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
 675
 676   testing::InSequence sequence;
 677   EXPECT_CALL(*mock_tracker_.get(), TrackAddStream(
 678       pc_handler_.get(),
 679       testing::Property(&blink::WebMediaStream::id,
 680                         base::UTF8ToUTF16(remote_stream_label)),
 681       PeerConnectionTracker::SOURCE_REMOTE));
 682   EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
 683       testing::Property(&blink::WebMediaStream::id,
 684                         base::UTF8ToUTF16(remote_stream_label))));
 685
 686   EXPECT_CALL(*mock_tracker_.get(), TrackRemoveStream(
 687       pc_handler_.get(),
 688       testing::Property(&blink::WebMediaStream::id,
 689                         base::UTF8ToUTF16(remote_stream_label)),
 690       PeerConnectionTracker::SOURCE_REMOTE));
 691   EXPECT_CALL(*mock_client_.get(), didRemoveRemoteStream(
 692       testing::Property(&blink::WebMediaStream::id,
 693                         base::UTF8ToUTF16(remote_stream_label))));
 694
 695   pc_handler_->OnAddStream(remote_stream.get());
 696   pc_handler_->OnRemoveStream(remote_stream.get());
 697 }
 698
 699 // This test that WebKit is notified about remote track state changes.
 700 TEST_F(RTCPeerConnectionHandlerTest, RemoteTrackState) {
 701   std::string remote_stream_label("remote_stream");
 702   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
 703       AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
 704
 705   testing::InSequence sequence;
 706   EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
 707       testing::Property(&blink::WebMediaStream::id,
 708                         base::UTF8ToUTF16(remote_stream_label))));
 709   pc_handler_->OnAddStream(remote_stream.get());
 710   const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
 711
 712   blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
 713   webkit_stream.audioTracks(audio_tracks);
 714   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateLive,
 715             audio_tracks[0].source().readyState());
 716
 717   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
 718     webkit_stream.videoTracks(video_tracks);
 719   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateLive,
 720             video_tracks[0].source().readyState());
 721
 722   remote_stream->GetAudioTracks()[0]->set_state(
 723       webrtc::MediaStreamTrackInterface::kEnded);
 724   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateEnded,
 725             audio_tracks[0].source().readyState());
 726
 727   remote_stream->GetVideoTracks()[0]->set_state(
 728       webrtc::MediaStreamTrackInterface::kEnded);
 729   EXPECT_EQ(blink::WebMediaStreamSource::ReadyStateEnded,
 730             video_tracks[0].source().readyState());
 731 }
 732
 733 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddAudioTrackFromRemoteStream) {
 734   std::string remote_stream_label("remote_stream");
 735   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
 736       AddRemoteMockMediaStream(remote_stream_label, "video", "audio"));
 737
 738   EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
 739       testing::Property(&blink::WebMediaStream::id,
 740                         base::UTF8ToUTF16(remote_stream_label))));
 741   pc_handler_->OnAddStream(remote_stream.get());
 742   const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
 743
 744   {
 745     // Test in a small scope so that  |audio_tracks| don't hold on to destroyed
 746     // source later.
 747     blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
 748     webkit_stream.audioTracks(audio_tracks);
 749     EXPECT_EQ(1u, audio_tracks.size());
 750   }
 751
 752   // Remove the Webrtc audio track from the Webrtc MediaStream.
 753   scoped_refptr<webrtc::AudioTrackInterface> webrtc_track =
 754       remote_stream->GetAudioTracks()[0].get();
 755   remote_stream->RemoveTrack(webrtc_track.get());
 756
 757   {
 758     blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks1;
 759     webkit_stream.audioTracks(modified_audio_tracks1);
 760     EXPECT_EQ(0u, modified_audio_tracks1.size());
 761   }
 762
 763   // Add the WebRtc audio track again.
 764   remote_stream->AddTrack(webrtc_track.get());
 765   blink::WebVector<blink::WebMediaStreamTrack> modified_audio_tracks2;
 766   webkit_stream.audioTracks(modified_audio_tracks2);
 767   EXPECT_EQ(1u, modified_audio_tracks2.size());
 768 }
 769
 770 TEST_F(RTCPeerConnectionHandlerTest, RemoveAndAddVideoTrackFromRemoteStream) {
 771   std::string remote_stream_label("remote_stream");
 772   scoped_refptr<webrtc::MediaStreamInterface> remote_stream(
 773       AddRemoteMockMediaStream(remote_stream_label, "video", "video"));
 774
 775   EXPECT_CALL(*mock_client_.get(), didAddRemoteStream(
 776       testing::Property(&blink::WebMediaStream::id,
 777                         base::UTF8ToUTF16(remote_stream_label))));
 778   pc_handler_->OnAddStream(remote_stream.get());
 779   const blink::WebMediaStream& webkit_stream = mock_client_->remote_stream();
 780
 781   {
 782     // Test in a small scope so that  |video_tracks| don't hold on to destroyed
 783     // source later.
 784     blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
 785     webkit_stream.videoTracks(video_tracks);
 786     EXPECT_EQ(1u, video_tracks.size());
 787   }
 788
 789   // Remove the Webrtc video track from the Webrtc MediaStream.
 790   scoped_refptr<webrtc::VideoTrackInterface> webrtc_track =
 791       remote_stream->GetVideoTracks()[0].get();
 792   remote_stream->RemoveTrack(webrtc_track.get());
 793   {
 794     blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks1;
 795     webkit_stream.videoTracks(modified_video_tracks1);
 796     EXPECT_EQ(0u, modified_video_tracks1.size());
 797   }
 798
 799   // Add the WebRtc video track again.
 800   remote_stream->AddTrack(webrtc_track.get());
 801   blink::WebVector<blink::WebMediaStreamTrack> modified_video_tracks2;
 802   webkit_stream.videoTracks(modified_video_tracks2);
 803   EXPECT_EQ(1u, modified_video_tracks2.size());
 804 }
 805
 806 TEST_F(RTCPeerConnectionHandlerTest, OnIceCandidate) {
 807   testing::InSequence sequence;
 808   EXPECT_CALL(*mock_tracker_.get(),
 809               TrackAddIceCandidate(pc_handler_.get(), _,
 810                                    PeerConnectionTracker::SOURCE_LOCAL));
 811   EXPECT_CALL(*mock_client_.get(), didGenerateICECandidate(_));
 812
 813   scoped_ptr<webrtc::IceCandidateInterface> native_candidate(
 814       mock_dependency_factory_->CreateIceCandidate("mid", 1, kDummySdp));
 815   pc_handler_->OnIceCandidate(native_candidate.get());
 816   EXPECT_EQ("mid", mock_client_->candidate_mid());
 817   EXPECT_EQ(1, mock_client_->candidate_mlineindex());
 818   EXPECT_EQ(kDummySdp, mock_client_->candidate_sdp());
 819 }
 820
 821 TEST_F(RTCPeerConnectionHandlerTest, OnRenegotiationNeeded) {
 822   testing::InSequence sequence;
 823   EXPECT_CALL(*mock_tracker_.get(),
 824               TrackOnRenegotiationNeeded(pc_handler_.get()));
 825   EXPECT_CALL(*mock_client_.get(), negotiationNeeded());
 826   pc_handler_->OnRenegotiationNeeded();
 827 }
 828
 829 TEST_F(RTCPeerConnectionHandlerTest, CreateDataChannel) {
 830   blink::WebString label = "d1";
 831   EXPECT_CALL(*mock_tracker_.get(),
 832               TrackCreateDataChannel(pc_handler_.get(),
 833                                      testing::NotNull(),
 834                                      PeerConnectionTracker::SOURCE_LOCAL));
 835   scoped_ptr<blink::WebRTCDataChannelHandler> channel(
 836       pc_handler_->createDataChannel("d1", blink::WebRTCDataChannelInit()));
 837   EXPECT_TRUE(channel.get() != NULL);
 838   EXPECT_EQ(label, channel->label());
 839 }
 840
 841 TEST_F(RTCPeerConnectionHandlerTest, CreateDtmfSender) {
 842   std::string stream_label = "local_stream";
 843   blink::WebMediaStream local_stream(CreateLocalMediaStream(stream_label));
 844   blink::WebMediaConstraints constraints;
 845   pc_handler_->addStream(local_stream, constraints);
 846
 847   blink::WebVector<blink::WebMediaStreamTrack> tracks;
 848   local_stream.videoTracks(tracks);
 849
 850   ASSERT_LE(1ul, tracks.size());
 851   EXPECT_FALSE(pc_handler_->createDTMFSender(tracks[0]));
 852
 853   local_stream.audioTracks(tracks);
 854   ASSERT_LE(1ul, tracks.size());
 855
 856   EXPECT_CALL(*mock_tracker_.get(),
 857               TrackCreateDTMFSender(pc_handler_.get(),
 858                                     testing::Ref(tracks[0])));
 859
 860   scoped_ptr<blink::WebRTCDTMFSenderHandler> sender(
 861       pc_handler_->createDTMFSender(tracks[0]));
 862   EXPECT_TRUE(sender.get());
 863 }
 864
 865 }  // namespace content