1 // Copyright 2014 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 "remoting/host/cast_extension_session.h" 6 7 #include "base/bind.h" 8 #include "base/json/json_reader.h" 9 #include "base/json/json_writer.h" 10 #include "base/logging.h" 11 #include "base/synchronization/waitable_event.h" 12 #include "net/url_request/url_request_context_getter.h" 13 #include "remoting/host/cast_video_capturer_adapter.h" 14 #include "remoting/host/chromium_port_allocator_factory.h" 15 #include "remoting/host/client_session.h" 16 #include "remoting/proto/control.pb.h" 17 #include "remoting/protocol/client_stub.h" 18 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 19 #include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" 20 #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" 21 22 namespace remoting { 23 24 // Used as the type attribute of all Cast protocol::ExtensionMessages. 25 const char kExtensionMessageType[] = "cast_message"; 26 27 // Top-level keys used in all extension messages between host and client. 28 // Must keep synced with webapp. 29 const char kTopLevelData[] = "chromoting_data"; 30 const char kTopLevelSubject[] = "subject"; 31 32 // Keys used to describe the subject of a cast extension message. WebRTC-related 33 // message subjects are prepended with "webrtc_". 34 // Must keep synced with webapp. 35 const char kSubjectReady[] = "ready"; 36 const char kSubjectTest[] = "test"; 37 const char kSubjectNewCandidate[] = "webrtc_candidate"; 38 const char kSubjectOffer[] = "webrtc_offer"; 39 const char kSubjectAnswer[] = "webrtc_answer"; 40 41 // WebRTC headers used inside messages with subject = "webrtc_*". 42 const char kWebRtcCandidate[] = "candidate"; 43 const char kWebRtcSessionDescType[] = "type"; 44 const char kWebRtcSessionDescSDP[] = "sdp"; 45 const char kWebRtcSDPMid[] = "sdpMid"; 46 const char kWebRtcSDPMLineIndex[] = "sdpMLineIndex"; 47 48 // Media labels used over the PeerConnection. 49 const char kVideoLabel[] = "cast_video_label"; 50 const char kStreamLabel[] = "stream_label"; 51 52 // Default STUN server used to construct 53 // webrtc::PeerConnectionInterface::RTCConfiguration for the PeerConnection. 54 const char kDefaultStunURI[] = "stun:stun.l.google.com:19302"; 55 56 const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread"; 57 58 // Interval between each call to PollPeerConnectionStats(). 59 const int kStatsLogIntervalSec = 10; 60 61 // Minimum frame rate for video streaming over the PeerConnection in frames per 62 // second, added as a media constraint when constructing the video source for 63 // the Peer Connection. 64 const int kMinFramesPerSecond = 5; 65 66 // A webrtc::SetSessionDescriptionObserver implementation used to receive the 67 // results of setting local and remote descriptions of the PeerConnection. 68 class CastSetSessionDescriptionObserver 69 : public webrtc::SetSessionDescriptionObserver { 70 public: 71 static CastSetSessionDescriptionObserver* Create() { 72 return new rtc::RefCountedObject<CastSetSessionDescriptionObserver>(); 73 } 74 virtual void OnSuccess() OVERRIDE { 75 VLOG(1) << "Setting session description succeeded."; 76 } 77 virtual void OnFailure(const std::string& error) OVERRIDE { 78 LOG(ERROR) << "Setting session description failed: " << error; 79 } 80 81 protected: 82 CastSetSessionDescriptionObserver() {} 83 virtual ~CastSetSessionDescriptionObserver() {} 84 85 DISALLOW_COPY_AND_ASSIGN(CastSetSessionDescriptionObserver); 86 }; 87 88 // A webrtc::CreateSessionDescriptionObserver implementation used to receive the 89 // results of creating descriptions for this end of the PeerConnection. 90 class CastCreateSessionDescriptionObserver 91 : public webrtc::CreateSessionDescriptionObserver { 92 public: 93 static CastCreateSessionDescriptionObserver* Create( 94 CastExtensionSession* session) { 95 return new rtc::RefCountedObject<CastCreateSessionDescriptionObserver>( 96 session); 97 } 98 virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc) OVERRIDE { 99 if (cast_extension_session_ == NULL) { 100 LOG(ERROR) 101 << "No CastExtensionSession. Creating session description succeeded."; 102 return; 103 } 104 cast_extension_session_->OnCreateSessionDescription(desc); 105 } 106 virtual void OnFailure(const std::string& error) OVERRIDE { 107 if (cast_extension_session_ == NULL) { 108 LOG(ERROR) 109 << "No CastExtensionSession. Creating session description failed."; 110 return; 111 } 112 cast_extension_session_->OnCreateSessionDescriptionFailure(error); 113 } 114 void SetCastExtensionSession(CastExtensionSession* cast_extension_session) { 115 cast_extension_session_ = cast_extension_session; 116 } 117 118 protected: 119 explicit CastCreateSessionDescriptionObserver(CastExtensionSession* session) 120 : cast_extension_session_(session) {} 121 virtual ~CastCreateSessionDescriptionObserver() {} 122 123 private: 124 CastExtensionSession* cast_extension_session_; 125 126 DISALLOW_COPY_AND_ASSIGN(CastCreateSessionDescriptionObserver); 127 }; 128 129 // A webrtc::StatsObserver implementation used to receive statistics about the 130 // current PeerConnection. 131 class CastStatsObserver : public webrtc::StatsObserver { 132 public: 133 static CastStatsObserver* Create() { 134 return new rtc::RefCountedObject<CastStatsObserver>(); 135 } 136 137 virtual void OnComplete( 138 const std::vector<webrtc::StatsReport>& reports) OVERRIDE { 139 typedef webrtc::StatsReport::Values::iterator ValuesIterator; 140 141 VLOG(1) << "Received " << reports.size() << " new StatsReports."; 142 143 int index; 144 std::vector<webrtc::StatsReport>::const_iterator it; 145 for (it = reports.begin(), index = 0; it != reports.end(); ++it, ++index) { 146 webrtc::StatsReport::Values v = it->values; 147 VLOG(1) << "Report " << index << ":"; 148 for (ValuesIterator vIt = v.begin(); vIt != v.end(); ++vIt) { 149 VLOG(1) << "Stat: " << vIt->name << "=" << vIt->value << "."; 150 } 151 } 152 } 153 154 protected: 155 CastStatsObserver() {} 156 virtual ~CastStatsObserver() {} 157 158 DISALLOW_COPY_AND_ASSIGN(CastStatsObserver); 159 }; 160 161 // TODO(aiguha): Fix PeerConnnection-related tear down crash caused by premature 162 // destruction of cricket::CaptureManager (which occurs on releasing 163 // |peer_conn_factory_|). See crbug.com/403840. 164 CastExtensionSession::~CastExtensionSession() { 165 DCHECK(caller_task_runner_->BelongsToCurrentThread()); 166 167 // Explicitly clear |create_session_desc_observer_|'s pointer to |this|, 168 // since the CastExtensionSession is destructing. Otherwise, 169 // |create_session_desc_observer_| would be left with a dangling pointer. 170 create_session_desc_observer_->SetCastExtensionSession(NULL); 171 172 CleanupPeerConnection(); 173 } 174 175 // static 176 scoped_ptr<CastExtensionSession> CastExtensionSession::Create( 177 scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner, 178 scoped_refptr<net::URLRequestContextGetter> url_request_context_getter, 179 const protocol::NetworkSettings& network_settings, 180 ClientSessionControl* client_session_control, 181 protocol::ClientStub* client_stub) { 182 scoped_ptr<CastExtensionSession> cast_extension_session( 183 new CastExtensionSession(caller_task_runner, 184 url_request_context_getter, 185 network_settings, 186 client_session_control, 187 client_stub)); 188 if (!cast_extension_session->WrapTasksAndSave()) { 189 return scoped_ptr<CastExtensionSession>(); 190 } 191 if (!cast_extension_session->InitializePeerConnection()) { 192 return scoped_ptr<CastExtensionSession>(); 193 } 194 return cast_extension_session.Pass(); 195 } 196 197 void CastExtensionSession::OnCreateSessionDescription( 198 webrtc::SessionDescriptionInterface* desc) { 199 if (!caller_task_runner_->BelongsToCurrentThread()) { 200 caller_task_runner_->PostTask( 201 FROM_HERE, 202 base::Bind(&CastExtensionSession::OnCreateSessionDescription, 203 base::Unretained(this), 204 desc)); 205 return; 206 } 207 208 peer_connection_->SetLocalDescription( 209 CastSetSessionDescriptionObserver::Create(), desc); 210 211 scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue()); 212 json->SetString(kWebRtcSessionDescType, desc->type()); 213 std::string subject = 214 (desc->type() == "offer") ? kSubjectOffer : kSubjectAnswer; 215 std::string desc_str; 216 desc->ToString(&desc_str); 217 json->SetString(kWebRtcSessionDescSDP, desc_str); 218 std::string json_str; 219 if (!base::JSONWriter::Write(json.get(), &json_str)) { 220 LOG(ERROR) << "Failed to serialize sdp message."; 221 return; 222 } 223 224 SendMessageToClient(subject.c_str(), json_str); 225 } 226 227 void CastExtensionSession::OnCreateSessionDescriptionFailure( 228 const std::string& error) { 229 VLOG(1) << "Creating Session Description failed: " << error; 230 } 231 232 // TODO(aiguha): Support the case(s) where we've grabbed the capturer already, 233 // but another extension reset the video pipeline. We should remove the 234 // stream from the peer connection here, and then attempt to re-setup the 235 // peer connection in the OnRenegotiationNeeded() callback. 236 // See crbug.com/403843. 237 void CastExtensionSession::OnCreateVideoCapturer( 238 scoped_ptr<webrtc::DesktopCapturer>* capturer) { 239 if (has_grabbed_capturer_) { 240 LOG(ERROR) << "The video pipeline was reset unexpectedly."; 241 has_grabbed_capturer_ = false; 242 peer_connection_->RemoveStream(stream_.release()); 243 return; 244 } 245 246 if (received_offer_) { 247 has_grabbed_capturer_ = true; 248 if (SetupVideoStream(capturer->Pass())) { 249 peer_connection_->CreateAnswer(create_session_desc_observer_, NULL); 250 } else { 251 has_grabbed_capturer_ = false; 252 // Ignore the received offer, since we failed to setup a video stream. 253 received_offer_ = false; 254 } 255 return; 256 } 257 } 258 259 bool CastExtensionSession::ModifiesVideoPipeline() const { 260 return true; 261 } 262 263 // Returns true if the |message| is a Cast ExtensionMessage, even if 264 // it was badly formed or a resulting action failed. This is done so that 265 // the host does not continue to attempt to pass |message| to other 266 // HostExtensionSessions. 267 bool CastExtensionSession::OnExtensionMessage( 268 ClientSessionControl* client_session_control, 269 protocol::ClientStub* client_stub, 270 const protocol::ExtensionMessage& message) { 271 if (message.type() != kExtensionMessageType) { 272 return false; 273 } 274 275 scoped_ptr<base::Value> value(base::JSONReader::Read(message.data())); 276 base::DictionaryValue* client_message; 277 if (!(value && value->GetAsDictionary(&client_message))) { 278 LOG(ERROR) << "Could not read cast extension message."; 279 return true; 280 } 281 282 std::string subject; 283 if (!client_message->GetString(kTopLevelSubject, &subject)) { 284 LOG(ERROR) << "Invalid Cast Extension Message (missing subject header)."; 285 return true; 286 } 287 288 if (subject == kSubjectOffer && !received_offer_) { 289 // Reset the video pipeline so we can grab the screen capturer and setup 290 // a video stream. 291 if (ParseAndSetRemoteDescription(client_message)) { 292 received_offer_ = true; 293 LOG(INFO) << "About to ResetVideoPipeline."; 294 client_session_control_->ResetVideoPipeline(); 295 296 } 297 } else if (subject == kSubjectAnswer) { 298 ParseAndSetRemoteDescription(client_message); 299 } else if (subject == kSubjectNewCandidate) { 300 ParseAndAddICECandidate(client_message); 301 } else { 302 VLOG(1) << "Unexpected CastExtension Message: " << message.data(); 303 } 304 return true; 305 } 306 307 // Private methods ------------------------------------------------------------ 308 309 CastExtensionSession::CastExtensionSession( 310 scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner, 311 scoped_refptr<net::URLRequestContextGetter> url_request_context_getter, 312 const protocol::NetworkSettings& network_settings, 313 ClientSessionControl* client_session_control, 314 protocol::ClientStub* client_stub) 315 : caller_task_runner_(caller_task_runner), 316 url_request_context_getter_(url_request_context_getter), 317 network_settings_(network_settings), 318 client_session_control_(client_session_control), 319 client_stub_(client_stub), 320 stats_observer_(CastStatsObserver::Create()), 321 received_offer_(false), 322 has_grabbed_capturer_(false), 323 signaling_thread_wrapper_(NULL), 324 worker_thread_wrapper_(NULL), 325 worker_thread_(kWorkerThreadName) { 326 DCHECK(caller_task_runner_->BelongsToCurrentThread()); 327 DCHECK(url_request_context_getter_.get()); 328 DCHECK(client_session_control_); 329 DCHECK(client_stub_); 330 331 // The worker thread is created with base::MessageLoop::TYPE_IO because 332 // the PeerConnection performs some port allocation operations on this thread 333 // that require it. See crbug.com/404013. 334 base::Thread::Options options(base::MessageLoop::TYPE_IO, 0); 335 worker_thread_.StartWithOptions(options); 336 worker_task_runner_ = worker_thread_.task_runner(); 337 } 338 339 bool CastExtensionSession::ParseAndSetRemoteDescription( 340 base::DictionaryValue* message) { 341 DCHECK(peer_connection_.get() != NULL); 342 343 base::DictionaryValue* message_data; 344 if (!message->GetDictionary(kTopLevelData, &message_data)) { 345 LOG(ERROR) << "Invalid Cast Extension Message (missing data)."; 346 return false; 347 } 348 349 std::string webrtc_type; 350 if (!message_data->GetString(kWebRtcSessionDescType, &webrtc_type)) { 351 LOG(ERROR) 352 << "Invalid Cast Extension Message (missing webrtc type header)."; 353 return false; 354 } 355 356 std::string sdp; 357 if (!message_data->GetString(kWebRtcSessionDescSDP, &sdp)) { 358 LOG(ERROR) << "Invalid Cast Extension Message (missing webrtc sdp header)."; 359 return false; 360 } 361 362 webrtc::SdpParseError error; 363 webrtc::SessionDescriptionInterface* session_description( 364 webrtc::CreateSessionDescription(webrtc_type, sdp, &error)); 365 366 if (!session_description) { 367 LOG(ERROR) << "Invalid Cast Extension Message (could not parse sdp)."; 368 VLOG(1) << "SdpParseError was: " << error.description; 369 return false; 370 } 371 372 peer_connection_->SetRemoteDescription( 373 CastSetSessionDescriptionObserver::Create(), session_description); 374 return true; 375 } 376 377 bool CastExtensionSession::ParseAndAddICECandidate( 378 base::DictionaryValue* message) { 379 DCHECK(peer_connection_.get() != NULL); 380 381 base::DictionaryValue* message_data; 382 if (!message->GetDictionary(kTopLevelData, &message_data)) { 383 LOG(ERROR) << "Invalid Cast Extension Message (missing data)."; 384 return false; 385 } 386 387 std::string candidate_str; 388 std::string sdp_mid; 389 int sdp_mlineindex = 0; 390 if (!message_data->GetString(kWebRtcSDPMid, &sdp_mid) || 391 !message_data->GetInteger(kWebRtcSDPMLineIndex, &sdp_mlineindex) || 392 !message_data->GetString(kWebRtcCandidate, &candidate_str)) { 393 LOG(ERROR) << "Invalid Cast Extension Message (could not parse)."; 394 return false; 395 } 396 397 rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate( 398 webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate_str)); 399 if (!candidate.get()) { 400 LOG(ERROR) 401 << "Invalid Cast Extension Message (could not create candidate)."; 402 return false; 403 } 404 405 if (!peer_connection_->AddIceCandidate(candidate.get())) { 406 LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection."; 407 return false; 408 } 409 410 VLOG(1) << "Received and Added ICE Candidate: " << candidate_str; 411 412 return true; 413 } 414 415 bool CastExtensionSession::SendMessageToClient(const std::string& subject, 416 const std::string& data) { 417 DCHECK(caller_task_runner_->BelongsToCurrentThread()); 418 419 if (client_stub_ == NULL) { 420 LOG(ERROR) << "No Client Stub. Cannot send message to client."; 421 return false; 422 } 423 424 base::DictionaryValue message_dict; 425 message_dict.SetString(kTopLevelSubject, subject); 426 message_dict.SetString(kTopLevelData, data); 427 std::string message_json; 428 429 if (!base::JSONWriter::Write(&message_dict, &message_json)) { 430 LOG(ERROR) << "Failed to serialize JSON message."; 431 return false; 432 } 433 434 protocol::ExtensionMessage message; 435 message.set_type(kExtensionMessageType); 436 message.set_data(message_json); 437 client_stub_->DeliverHostMessage(message); 438 return true; 439 } 440 441 void CastExtensionSession::EnsureTaskAndSetSend(rtc::Thread** ptr, 442 base::WaitableEvent* event) { 443 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); 444 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); 445 *ptr = jingle_glue::JingleThreadWrapper::current(); 446 447 if (event != NULL) { 448 event->Signal(); 449 } 450 } 451 452 bool CastExtensionSession::WrapTasksAndSave() { 453 DCHECK(caller_task_runner_->BelongsToCurrentThread()); 454 455 EnsureTaskAndSetSend(&signaling_thread_wrapper_); 456 if (signaling_thread_wrapper_ == NULL) 457 return false; 458 459 base::WaitableEvent wrap_worker_thread_event(true, false); 460 worker_task_runner_->PostTask( 461 FROM_HERE, 462 base::Bind(&CastExtensionSession::EnsureTaskAndSetSend, 463 base::Unretained(this), 464 &worker_thread_wrapper_, 465 &wrap_worker_thread_event)); 466 wrap_worker_thread_event.Wait(); 467 468 return (worker_thread_wrapper_ != NULL); 469 } 470 471 bool CastExtensionSession::InitializePeerConnection() { 472 DCHECK(caller_task_runner_->BelongsToCurrentThread()); 473 DCHECK(!peer_conn_factory_); 474 DCHECK(!peer_connection_); 475 DCHECK(worker_thread_wrapper_ != NULL); 476 DCHECK(signaling_thread_wrapper_ != NULL); 477 478 peer_conn_factory_ = webrtc::CreatePeerConnectionFactory( 479 worker_thread_wrapper_, signaling_thread_wrapper_, NULL, NULL, NULL); 480 481 if (!peer_conn_factory_.get()) { 482 CleanupPeerConnection(); 483 return false; 484 } 485 486 VLOG(1) << "Created PeerConnectionFactory successfully."; 487 488 webrtc::PeerConnectionInterface::IceServers servers; 489 webrtc::PeerConnectionInterface::IceServer server; 490 server.uri = kDefaultStunURI; 491 servers.push_back(server); 492 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; 493 rtc_config.servers = servers; 494 495 // DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the 496 // peer connection uses SDES. Disabling SDES as well will cause the peer 497 // connection to fail to connect. 498 // Note: For protection and unprotection of SRTP packets, the libjingle 499 // ENABLE_EXTERNAL_AUTH flag must not be set. 500 webrtc::FakeConstraints constraints; 501 constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, 502 webrtc::MediaConstraintsInterface::kValueTrue); 503 504 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> 505 port_allocator_factory = ChromiumPortAllocatorFactory::Create( 506 network_settings_, url_request_context_getter_); 507 508 peer_connection_ = peer_conn_factory_->CreatePeerConnection( 509 rtc_config, &constraints, port_allocator_factory, NULL, this); 510 511 if (!peer_connection_.get()) { 512 CleanupPeerConnection(); 513 return false; 514 } 515 516 VLOG(1) << "Created PeerConnection successfully."; 517 518 create_session_desc_observer_ = 519 CastCreateSessionDescriptionObserver::Create(this); 520 521 // Send a test message to the client. Then, notify the client to start 522 // webrtc offer/answer negotiation. 523 if (!SendMessageToClient(kSubjectTest, "Hello, client.") || 524 !SendMessageToClient(kSubjectReady, "Host ready to receive offers.")) { 525 LOG(ERROR) << "Failed to send messages to client."; 526 return false; 527 } 528 529 return true; 530 } 531 532 bool CastExtensionSession::SetupVideoStream( 533 scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) { 534 DCHECK(caller_task_runner_->BelongsToCurrentThread()); 535 DCHECK(desktop_capturer); 536 537 if (stream_) { 538 VLOG(1) << "Already added MediaStream. Aborting Setup."; 539 return false; 540 } 541 542 scoped_ptr<CastVideoCapturerAdapter> cast_video_capturer_adapter( 543 new CastVideoCapturerAdapter(desktop_capturer.Pass())); 544 545 // Set video stream constraints. 546 webrtc::FakeConstraints video_constraints; 547 video_constraints.AddMandatory( 548 webrtc::MediaConstraintsInterface::kMinFrameRate, kMinFramesPerSecond); 549 550 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track = 551 peer_conn_factory_->CreateVideoTrack( 552 kVideoLabel, 553 peer_conn_factory_->CreateVideoSource( 554 cast_video_capturer_adapter.release(), &video_constraints)); 555 556 stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel); 557 558 if (!stream_->AddTrack(video_track) || 559 !peer_connection_->AddStream(stream_, NULL)) { 560 return false; 561 } 562 563 VLOG(1) << "Setup video stream successfully."; 564 565 return true; 566 } 567 568 void CastExtensionSession::PollPeerConnectionStats() { 569 if (!connection_active()) { 570 VLOG(1) << "Cannot poll stats while PeerConnection is inactive."; 571 } 572 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> video_track = 573 stream_->FindVideoTrack(kVideoLabel); 574 peer_connection_->GetStats( 575 stats_observer_, 576 video_track.release(), 577 webrtc::PeerConnectionInterface::kStatsOutputLevelStandard); 578 } 579 580 void CastExtensionSession::CleanupPeerConnection() { 581 peer_connection_->Close(); 582 peer_connection_ = NULL; 583 stream_ = NULL; 584 peer_conn_factory_ = NULL; 585 worker_thread_.Stop(); 586 } 587 588 bool CastExtensionSession::connection_active() const { 589 return peer_connection_.get() != NULL; 590 } 591 592 // webrtc::PeerConnectionObserver implementation ------------------------------- 593 594 void CastExtensionSession::OnError() { 595 VLOG(1) << "PeerConnectionObserver: an error occurred."; 596 } 597 598 void CastExtensionSession::OnSignalingChange( 599 webrtc::PeerConnectionInterface::SignalingState new_state) { 600 VLOG(1) << "PeerConnectionObserver: SignalingState changed to:" << new_state; 601 } 602 603 void CastExtensionSession::OnStateChange( 604 webrtc::PeerConnectionObserver::StateType state_changed) { 605 VLOG(1) << "PeerConnectionObserver: StateType changed to: " << state_changed; 606 } 607 608 void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) { 609 VLOG(1) << "PeerConnectionObserver: stream added: " << stream->label(); 610 } 611 612 void CastExtensionSession::OnRemoveStream( 613 webrtc::MediaStreamInterface* stream) { 614 VLOG(1) << "PeerConnectionObserver: stream removed: " << stream->label(); 615 } 616 617 void CastExtensionSession::OnDataChannel( 618 webrtc::DataChannelInterface* data_channel) { 619 VLOG(1) << "PeerConnectionObserver: data channel: " << data_channel->label(); 620 } 621 622 void CastExtensionSession::OnRenegotiationNeeded() { 623 VLOG(1) << "PeerConnectionObserver: renegotiation needed."; 624 } 625 626 void CastExtensionSession::OnIceConnectionChange( 627 webrtc::PeerConnectionInterface::IceConnectionState new_state) { 628 VLOG(1) << "PeerConnectionObserver: IceConnectionState changed to: " 629 << new_state; 630 631 // TODO(aiguha): Maybe start timer only if enabled by command-line flag or 632 // at a particular verbosity level. 633 if (!stats_polling_timer_.IsRunning() && 634 new_state == webrtc::PeerConnectionInterface::kIceConnectionConnected) { 635 stats_polling_timer_.Start( 636 FROM_HERE, 637 base::TimeDelta::FromSeconds(kStatsLogIntervalSec), 638 this, 639 &CastExtensionSession::PollPeerConnectionStats); 640 } 641 } 642 643 void CastExtensionSession::OnIceGatheringChange( 644 webrtc::PeerConnectionInterface::IceGatheringState new_state) { 645 VLOG(1) << "PeerConnectionObserver: IceGatheringState changed to: " 646 << new_state; 647 } 648 649 void CastExtensionSession::OnIceComplete() { 650 VLOG(1) << "PeerConnectionObserver: all ICE candidates found."; 651 } 652 653 void CastExtensionSession::OnIceCandidate( 654 const webrtc::IceCandidateInterface* candidate) { 655 std::string candidate_str; 656 if (!candidate->ToString(&candidate_str)) { 657 LOG(ERROR) << "PeerConnectionObserver: failed to serialize candidate."; 658 return; 659 } 660 scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue()); 661 json->SetString(kWebRtcSDPMid, candidate->sdp_mid()); 662 json->SetInteger(kWebRtcSDPMLineIndex, candidate->sdp_mline_index()); 663 json->SetString(kWebRtcCandidate, candidate_str); 664 std::string json_str; 665 if (!base::JSONWriter::Write(json.get(), &json_str)) { 666 LOG(ERROR) << "Failed to serialize candidate message."; 667 return; 668 } 669 SendMessageToClient(kSubjectNewCandidate, json_str); 670 } 671 672 } // namespace remoting 673