[GCM] Investigatory CHECKs for crash in parsing stream
[chromium-blink-merge.git] / remoting / host / cast_extension_session.cc
blobbff87513b2437052a64fdba7b2d9cadf6212a2a8
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.
5 #include "remoting/host/cast_extension_session.h"
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"
22 namespace remoting {
24 // Used as the type attribute of all Cast protocol::ExtensionMessages.
25 const char kExtensionMessageType[] = "cast_message";
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";
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";
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";
48 // Media labels used over the PeerConnection.
49 const char kVideoLabel[] = "cast_video_label";
50 const char kStreamLabel[] = "stream_label";
52 // Default STUN server used to construct
53 // webrtc::PeerConnectionInterface::RTCConfiguration for the PeerConnection.
54 const char kDefaultStunURI[] = "stun:stun.l.google.com:19302";
56 const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread";
58 // Interval between each call to PollPeerConnectionStats().
59 const int kStatsLogIntervalSec = 10;
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;
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>();
74 virtual void OnSuccess() OVERRIDE {
75 VLOG(1) << "Setting session description succeeded.";
77 virtual void OnFailure(const std::string& error) OVERRIDE {
78 LOG(ERROR) << "Setting session description failed: " << error;
81 protected:
82 CastSetSessionDescriptionObserver() {}
83 virtual ~CastSetSessionDescriptionObserver() {}
85 DISALLOW_COPY_AND_ASSIGN(CastSetSessionDescriptionObserver);
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);
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;
104 cast_extension_session_->OnCreateSessionDescription(desc);
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;
112 cast_extension_session_->OnCreateSessionDescriptionFailure(error);
114 void SetCastExtensionSession(CastExtensionSession* cast_extension_session) {
115 cast_extension_session_ = cast_extension_session;
118 protected:
119 explicit CastCreateSessionDescriptionObserver(CastExtensionSession* session)
120 : cast_extension_session_(session) {}
121 virtual ~CastCreateSessionDescriptionObserver() {}
123 private:
124 CastExtensionSession* cast_extension_session_;
126 DISALLOW_COPY_AND_ASSIGN(CastCreateSessionDescriptionObserver);
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>();
137 virtual void OnComplete(
138 const std::vector<webrtc::StatsReport>& reports) OVERRIDE {
139 typedef webrtc::StatsReport::Values::iterator ValuesIterator;
141 VLOG(1) << "Received " << reports.size() << " new StatsReports.";
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 << ".";
154 protected:
155 CastStatsObserver() {}
156 virtual ~CastStatsObserver() {}
158 DISALLOW_COPY_AND_ASSIGN(CastStatsObserver);
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());
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);
172 CleanupPeerConnection();
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>();
191 if (!cast_extension_session->InitializePeerConnection()) {
192 return scoped_ptr<CastExtensionSession>();
194 return cast_extension_session.Pass();
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;
208 peer_connection_->SetLocalDescription(
209 CastSetSessionDescriptionObserver::Create(), desc);
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;
224 SendMessageToClient(subject.c_str(), json_str);
227 void CastExtensionSession::OnCreateSessionDescriptionFailure(
228 const std::string& error) {
229 VLOG(1) << "Creating Session Description failed: " << error;
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;
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;
255 return;
259 bool CastExtensionSession::ModifiesVideoPipeline() const {
260 return true;
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;
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;
282 std::string subject;
283 if (!client_message->GetString(kTopLevelSubject, &subject)) {
284 LOG(ERROR) << "Invalid Cast Extension Message (missing subject header).";
285 return true;
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();
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();
304 return true;
307 // Private methods ------------------------------------------------------------
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_);
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();
339 bool CastExtensionSession::ParseAndSetRemoteDescription(
340 base::DictionaryValue* message) {
341 DCHECK(peer_connection_.get() != NULL);
343 base::DictionaryValue* message_data;
344 if (!message->GetDictionary(kTopLevelData, &message_data)) {
345 LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
346 return false;
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;
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;
362 webrtc::SdpParseError error;
363 webrtc::SessionDescriptionInterface* session_description(
364 webrtc::CreateSessionDescription(webrtc_type, sdp, &error));
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;
372 peer_connection_->SetRemoteDescription(
373 CastSetSessionDescriptionObserver::Create(), session_description);
374 return true;
377 bool CastExtensionSession::ParseAndAddICECandidate(
378 base::DictionaryValue* message) {
379 DCHECK(peer_connection_.get() != NULL);
381 base::DictionaryValue* message_data;
382 if (!message->GetDictionary(kTopLevelData, &message_data)) {
383 LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
384 return false;
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;
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;
405 if (!peer_connection_->AddIceCandidate(candidate.get())) {
406 LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection.";
407 return false;
410 VLOG(1) << "Received and Added ICE Candidate: " << candidate_str;
412 return true;
415 bool CastExtensionSession::SendMessageToClient(const std::string& subject,
416 const std::string& data) {
417 DCHECK(caller_task_runner_->BelongsToCurrentThread());
419 if (client_stub_ == NULL) {
420 LOG(ERROR) << "No Client Stub. Cannot send message to client.";
421 return false;
424 base::DictionaryValue message_dict;
425 message_dict.SetString(kTopLevelSubject, subject);
426 message_dict.SetString(kTopLevelData, data);
427 std::string message_json;
429 if (!base::JSONWriter::Write(&message_dict, &message_json)) {
430 LOG(ERROR) << "Failed to serialize JSON message.";
431 return false;
434 protocol::ExtensionMessage message;
435 message.set_type(kExtensionMessageType);
436 message.set_data(message_json);
437 client_stub_->DeliverHostMessage(message);
438 return true;
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();
447 if (event != NULL) {
448 event->Signal();
452 bool CastExtensionSession::WrapTasksAndSave() {
453 DCHECK(caller_task_runner_->BelongsToCurrentThread());
455 EnsureTaskAndSetSend(&signaling_thread_wrapper_);
456 if (signaling_thread_wrapper_ == NULL)
457 return false;
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();
468 return (worker_thread_wrapper_ != NULL);
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);
478 peer_conn_factory_ = webrtc::CreatePeerConnectionFactory(
479 worker_thread_wrapper_, signaling_thread_wrapper_, NULL, NULL, NULL);
481 if (!peer_conn_factory_.get()) {
482 CleanupPeerConnection();
483 return false;
486 VLOG(1) << "Created PeerConnectionFactory successfully.";
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;
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);
504 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
505 port_allocator_factory = ChromiumPortAllocatorFactory::Create(
506 network_settings_, url_request_context_getter_);
508 peer_connection_ = peer_conn_factory_->CreatePeerConnection(
509 rtc_config, &constraints, port_allocator_factory, NULL, this);
511 if (!peer_connection_.get()) {
512 CleanupPeerConnection();
513 return false;
516 VLOG(1) << "Created PeerConnection successfully.";
518 create_session_desc_observer_ =
519 CastCreateSessionDescriptionObserver::Create(this);
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;
529 return true;
532 bool CastExtensionSession::SetupVideoStream(
533 scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) {
534 DCHECK(caller_task_runner_->BelongsToCurrentThread());
535 DCHECK(desktop_capturer);
537 if (stream_) {
538 VLOG(1) << "Already added MediaStream. Aborting Setup.";
539 return false;
542 scoped_ptr<CastVideoCapturerAdapter> cast_video_capturer_adapter(
543 new CastVideoCapturerAdapter(desktop_capturer.Pass()));
545 // Set video stream constraints.
546 webrtc::FakeConstraints video_constraints;
547 video_constraints.AddMandatory(
548 webrtc::MediaConstraintsInterface::kMinFrameRate, kMinFramesPerSecond);
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));
556 stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel);
558 if (!stream_->AddTrack(video_track) ||
559 !peer_connection_->AddStream(stream_, NULL)) {
560 return false;
563 VLOG(1) << "Setup video stream successfully.";
565 return true;
568 void CastExtensionSession::PollPeerConnectionStats() {
569 if (!connection_active()) {
570 VLOG(1) << "Cannot poll stats while PeerConnection is inactive.";
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);
580 void CastExtensionSession::CleanupPeerConnection() {
581 peer_connection_->Close();
582 peer_connection_ = NULL;
583 stream_ = NULL;
584 peer_conn_factory_ = NULL;
585 worker_thread_.Stop();
588 bool CastExtensionSession::connection_active() const {
589 return peer_connection_.get() != NULL;
592 // webrtc::PeerConnectionObserver implementation -------------------------------
594 void CastExtensionSession::OnError() {
595 VLOG(1) << "PeerConnectionObserver: an error occurred.";
598 void CastExtensionSession::OnSignalingChange(
599 webrtc::PeerConnectionInterface::SignalingState new_state) {
600 VLOG(1) << "PeerConnectionObserver: SignalingState changed to:" << new_state;
603 void CastExtensionSession::OnStateChange(
604 webrtc::PeerConnectionObserver::StateType state_changed) {
605 VLOG(1) << "PeerConnectionObserver: StateType changed to: " << state_changed;
608 void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) {
609 VLOG(1) << "PeerConnectionObserver: stream added: " << stream->label();
612 void CastExtensionSession::OnRemoveStream(
613 webrtc::MediaStreamInterface* stream) {
614 VLOG(1) << "PeerConnectionObserver: stream removed: " << stream->label();
617 void CastExtensionSession::OnDataChannel(
618 webrtc::DataChannelInterface* data_channel) {
619 VLOG(1) << "PeerConnectionObserver: data channel: " << data_channel->label();
622 void CastExtensionSession::OnRenegotiationNeeded() {
623 VLOG(1) << "PeerConnectionObserver: renegotiation needed.";
626 void CastExtensionSession::OnIceConnectionChange(
627 webrtc::PeerConnectionInterface::IceConnectionState new_state) {
628 VLOG(1) << "PeerConnectionObserver: IceConnectionState changed to: "
629 << new_state;
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);
643 void CastExtensionSession::OnIceGatheringChange(
644 webrtc::PeerConnectionInterface::IceGatheringState new_state) {
645 VLOG(1) << "PeerConnectionObserver: IceGatheringState changed to: "
646 << new_state;
649 void CastExtensionSession::OnIceComplete() {
650 VLOG(1) << "PeerConnectionObserver: all ICE candidates found.";
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;
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;
669 SendMessageToClient(kSubjectNewCandidate, json_str);
672 } // namespace remoting