Battery Status API: add UMA logging for Linux.
[chromium-blink-merge.git] / remoting / host / cast_extension_session.cc
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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 scoped_ptr<webrtc::DesktopCapturer> 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 capturer.Pass();
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 scoped_ptr<webrtc::DesktopCapturer>();
258 return capturer.Pass();
261 bool CastExtensionSession::ModifiesVideoPipeline() const {
262 return true;
265 // Returns true if the |message| is a Cast ExtensionMessage, even if
266 // it was badly formed or a resulting action failed. This is done so that
267 // the host does not continue to attempt to pass |message| to other
268 // HostExtensionSessions.
269 bool CastExtensionSession::OnExtensionMessage(
270 ClientSessionControl* client_session_control,
271 protocol::ClientStub* client_stub,
272 const protocol::ExtensionMessage& message) {
273 if (message.type() != kExtensionMessageType) {
274 return false;
277 scoped_ptr<base::Value> value(base::JSONReader::Read(message.data()));
278 base::DictionaryValue* client_message;
279 if (!(value && value->GetAsDictionary(&client_message))) {
280 LOG(ERROR) << "Could not read cast extension message.";
281 return true;
284 std::string subject;
285 if (!client_message->GetString(kTopLevelSubject, &subject)) {
286 LOG(ERROR) << "Invalid Cast Extension Message (missing subject header).";
287 return true;
290 if (subject == kSubjectOffer && !received_offer_) {
291 // Reset the video pipeline so we can grab the screen capturer and setup
292 // a video stream.
293 if (ParseAndSetRemoteDescription(client_message)) {
294 received_offer_ = true;
295 LOG(INFO) << "About to ResetVideoPipeline.";
296 client_session_control_->ResetVideoPipeline();
299 } else if (subject == kSubjectAnswer) {
300 ParseAndSetRemoteDescription(client_message);
301 } else if (subject == kSubjectNewCandidate) {
302 ParseAndAddICECandidate(client_message);
303 } else {
304 VLOG(1) << "Unexpected CastExtension Message: " << message.data();
306 return true;
309 // Private methods ------------------------------------------------------------
311 CastExtensionSession::CastExtensionSession(
312 scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner,
313 scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
314 const protocol::NetworkSettings& network_settings,
315 ClientSessionControl* client_session_control,
316 protocol::ClientStub* client_stub)
317 : caller_task_runner_(caller_task_runner),
318 url_request_context_getter_(url_request_context_getter),
319 network_settings_(network_settings),
320 client_session_control_(client_session_control),
321 client_stub_(client_stub),
322 stats_observer_(CastStatsObserver::Create()),
323 received_offer_(false),
324 has_grabbed_capturer_(false),
325 signaling_thread_wrapper_(NULL),
326 worker_thread_wrapper_(NULL),
327 worker_thread_(kWorkerThreadName) {
328 DCHECK(caller_task_runner_->BelongsToCurrentThread());
329 DCHECK(url_request_context_getter_.get());
330 DCHECK(client_session_control_);
331 DCHECK(client_stub_);
333 // The worker thread is created with base::MessageLoop::TYPE_IO because
334 // the PeerConnection performs some port allocation operations on this thread
335 // that require it. See crbug.com/404013.
336 base::Thread::Options options(base::MessageLoop::TYPE_IO, 0);
337 worker_thread_.StartWithOptions(options);
338 worker_task_runner_ = worker_thread_.task_runner();
341 bool CastExtensionSession::ParseAndSetRemoteDescription(
342 base::DictionaryValue* message) {
343 DCHECK(peer_connection_.get() != NULL);
345 base::DictionaryValue* message_data;
346 if (!message->GetDictionary(kTopLevelData, &message_data)) {
347 LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
348 return false;
351 std::string webrtc_type;
352 if (!message_data->GetString(kWebRtcSessionDescType, &webrtc_type)) {
353 LOG(ERROR)
354 << "Invalid Cast Extension Message (missing webrtc type header).";
355 return false;
358 std::string sdp;
359 if (!message_data->GetString(kWebRtcSessionDescSDP, &sdp)) {
360 LOG(ERROR) << "Invalid Cast Extension Message (missing webrtc sdp header).";
361 return false;
364 webrtc::SdpParseError error;
365 webrtc::SessionDescriptionInterface* session_description(
366 webrtc::CreateSessionDescription(webrtc_type, sdp, &error));
368 if (!session_description) {
369 LOG(ERROR) << "Invalid Cast Extension Message (could not parse sdp).";
370 VLOG(1) << "SdpParseError was: " << error.description;
371 return false;
374 peer_connection_->SetRemoteDescription(
375 CastSetSessionDescriptionObserver::Create(), session_description);
376 return true;
379 bool CastExtensionSession::ParseAndAddICECandidate(
380 base::DictionaryValue* message) {
381 DCHECK(peer_connection_.get() != NULL);
383 base::DictionaryValue* message_data;
384 if (!message->GetDictionary(kTopLevelData, &message_data)) {
385 LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
386 return false;
389 std::string candidate_str;
390 std::string sdp_mid;
391 int sdp_mlineindex = 0;
392 if (!message_data->GetString(kWebRtcSDPMid, &sdp_mid) ||
393 !message_data->GetInteger(kWebRtcSDPMLineIndex, &sdp_mlineindex) ||
394 !message_data->GetString(kWebRtcCandidate, &candidate_str)) {
395 LOG(ERROR) << "Invalid Cast Extension Message (could not parse).";
396 return false;
399 rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
400 webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate_str));
401 if (!candidate.get()) {
402 LOG(ERROR)
403 << "Invalid Cast Extension Message (could not create candidate).";
404 return false;
407 if (!peer_connection_->AddIceCandidate(candidate.get())) {
408 LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection.";
409 return false;
412 VLOG(1) << "Received and Added ICE Candidate: " << candidate_str;
414 return true;
417 bool CastExtensionSession::SendMessageToClient(const std::string& subject,
418 const std::string& data) {
419 DCHECK(caller_task_runner_->BelongsToCurrentThread());
421 if (client_stub_ == NULL) {
422 LOG(ERROR) << "No Client Stub. Cannot send message to client.";
423 return false;
426 base::DictionaryValue message_dict;
427 message_dict.SetString(kTopLevelSubject, subject);
428 message_dict.SetString(kTopLevelData, data);
429 std::string message_json;
431 if (!base::JSONWriter::Write(&message_dict, &message_json)) {
432 LOG(ERROR) << "Failed to serialize JSON message.";
433 return false;
436 protocol::ExtensionMessage message;
437 message.set_type(kExtensionMessageType);
438 message.set_data(message_json);
439 client_stub_->DeliverHostMessage(message);
440 return true;
443 void CastExtensionSession::EnsureTaskAndSetSend(rtc::Thread** ptr,
444 base::WaitableEvent* event) {
445 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
446 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
447 *ptr = jingle_glue::JingleThreadWrapper::current();
449 if (event != NULL) {
450 event->Signal();
454 bool CastExtensionSession::WrapTasksAndSave() {
455 DCHECK(caller_task_runner_->BelongsToCurrentThread());
457 EnsureTaskAndSetSend(&signaling_thread_wrapper_);
458 if (signaling_thread_wrapper_ == NULL)
459 return false;
461 base::WaitableEvent wrap_worker_thread_event(true, false);
462 worker_task_runner_->PostTask(
463 FROM_HERE,
464 base::Bind(&CastExtensionSession::EnsureTaskAndSetSend,
465 base::Unretained(this),
466 &worker_thread_wrapper_,
467 &wrap_worker_thread_event));
468 wrap_worker_thread_event.Wait();
470 return (worker_thread_wrapper_ != NULL);
473 bool CastExtensionSession::InitializePeerConnection() {
474 DCHECK(caller_task_runner_->BelongsToCurrentThread());
475 DCHECK(!peer_conn_factory_);
476 DCHECK(!peer_connection_);
477 DCHECK(worker_thread_wrapper_ != NULL);
478 DCHECK(signaling_thread_wrapper_ != NULL);
480 peer_conn_factory_ = webrtc::CreatePeerConnectionFactory(
481 worker_thread_wrapper_, signaling_thread_wrapper_, NULL, NULL, NULL);
483 if (!peer_conn_factory_.get()) {
484 CleanupPeerConnection();
485 return false;
488 VLOG(1) << "Created PeerConnectionFactory successfully.";
490 webrtc::PeerConnectionInterface::IceServers servers;
491 webrtc::PeerConnectionInterface::IceServer server;
492 server.uri = kDefaultStunURI;
493 servers.push_back(server);
494 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
495 rtc_config.servers = servers;
497 // DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the
498 // peer connection uses SDES. Disabling SDES as well will cause the peer
499 // connection to fail to connect.
500 // Note: For protection and unprotection of SRTP packets, the libjingle
501 // ENABLE_EXTERNAL_AUTH flag must not be set.
502 webrtc::FakeConstraints constraints;
503 constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
504 webrtc::MediaConstraintsInterface::kValueTrue);
506 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
507 port_allocator_factory = ChromiumPortAllocatorFactory::Create(
508 network_settings_, url_request_context_getter_);
510 peer_connection_ = peer_conn_factory_->CreatePeerConnection(
511 rtc_config, &constraints, port_allocator_factory, NULL, this);
513 if (!peer_connection_.get()) {
514 CleanupPeerConnection();
515 return false;
518 VLOG(1) << "Created PeerConnection successfully.";
520 create_session_desc_observer_ =
521 CastCreateSessionDescriptionObserver::Create(this);
523 // Send a test message to the client. Then, notify the client to start
524 // webrtc offer/answer negotiation.
525 if (!SendMessageToClient(kSubjectTest, "Hello, client.") ||
526 !SendMessageToClient(kSubjectReady, "Host ready to receive offers.")) {
527 LOG(ERROR) << "Failed to send messages to client.";
528 return false;
531 return true;
534 bool CastExtensionSession::SetupVideoStream(
535 scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) {
536 DCHECK(caller_task_runner_->BelongsToCurrentThread());
537 DCHECK(desktop_capturer);
539 if (stream_) {
540 VLOG(1) << "Already added MediaStream. Aborting Setup.";
541 return false;
544 scoped_ptr<CastVideoCapturerAdapter> cast_video_capturer_adapter(
545 new CastVideoCapturerAdapter(desktop_capturer.Pass()));
547 // Set video stream constraints.
548 webrtc::FakeConstraints video_constraints;
549 video_constraints.AddMandatory(
550 webrtc::MediaConstraintsInterface::kMinFrameRate, kMinFramesPerSecond);
552 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
553 peer_conn_factory_->CreateVideoTrack(
554 kVideoLabel,
555 peer_conn_factory_->CreateVideoSource(
556 cast_video_capturer_adapter.release(), &video_constraints));
558 stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel);
560 if (!stream_->AddTrack(video_track) ||
561 !peer_connection_->AddStream(stream_, NULL)) {
562 return false;
565 VLOG(1) << "Setup video stream successfully.";
567 return true;
570 void CastExtensionSession::PollPeerConnectionStats() {
571 if (!connection_active()) {
572 VLOG(1) << "Cannot poll stats while PeerConnection is inactive.";
574 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> video_track =
575 stream_->FindVideoTrack(kVideoLabel);
576 peer_connection_->GetStats(
577 stats_observer_,
578 video_track.release(),
579 webrtc::PeerConnectionInterface::kStatsOutputLevelStandard);
582 void CastExtensionSession::CleanupPeerConnection() {
583 peer_connection_->Close();
584 peer_connection_ = NULL;
585 stream_ = NULL;
586 peer_conn_factory_ = NULL;
587 worker_thread_.Stop();
590 bool CastExtensionSession::connection_active() const {
591 return peer_connection_.get() != NULL;
594 // webrtc::PeerConnectionObserver implementation -------------------------------
596 void CastExtensionSession::OnError() {
597 VLOG(1) << "PeerConnectionObserver: an error occurred.";
600 void CastExtensionSession::OnSignalingChange(
601 webrtc::PeerConnectionInterface::SignalingState new_state) {
602 VLOG(1) << "PeerConnectionObserver: SignalingState changed to:" << new_state;
605 void CastExtensionSession::OnStateChange(
606 webrtc::PeerConnectionObserver::StateType state_changed) {
607 VLOG(1) << "PeerConnectionObserver: StateType changed to: " << state_changed;
610 void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) {
611 VLOG(1) << "PeerConnectionObserver: stream added: " << stream->label();
614 void CastExtensionSession::OnRemoveStream(
615 webrtc::MediaStreamInterface* stream) {
616 VLOG(1) << "PeerConnectionObserver: stream removed: " << stream->label();
619 void CastExtensionSession::OnDataChannel(
620 webrtc::DataChannelInterface* data_channel) {
621 VLOG(1) << "PeerConnectionObserver: data channel: " << data_channel->label();
624 void CastExtensionSession::OnRenegotiationNeeded() {
625 VLOG(1) << "PeerConnectionObserver: renegotiation needed.";
628 void CastExtensionSession::OnIceConnectionChange(
629 webrtc::PeerConnectionInterface::IceConnectionState new_state) {
630 VLOG(1) << "PeerConnectionObserver: IceConnectionState changed to: "
631 << new_state;
633 // TODO(aiguha): Maybe start timer only if enabled by command-line flag or
634 // at a particular verbosity level.
635 if (!stats_polling_timer_.IsRunning() &&
636 new_state == webrtc::PeerConnectionInterface::kIceConnectionConnected) {
637 stats_polling_timer_.Start(
638 FROM_HERE,
639 base::TimeDelta::FromSeconds(kStatsLogIntervalSec),
640 this,
641 &CastExtensionSession::PollPeerConnectionStats);
645 void CastExtensionSession::OnIceGatheringChange(
646 webrtc::PeerConnectionInterface::IceGatheringState new_state) {
647 VLOG(1) << "PeerConnectionObserver: IceGatheringState changed to: "
648 << new_state;
651 void CastExtensionSession::OnIceComplete() {
652 VLOG(1) << "PeerConnectionObserver: all ICE candidates found.";
655 void CastExtensionSession::OnIceCandidate(
656 const webrtc::IceCandidateInterface* candidate) {
657 std::string candidate_str;
658 if (!candidate->ToString(&candidate_str)) {
659 LOG(ERROR) << "PeerConnectionObserver: failed to serialize candidate.";
660 return;
662 scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue());
663 json->SetString(kWebRtcSDPMid, candidate->sdp_mid());
664 json->SetInteger(kWebRtcSDPMLineIndex, candidate->sdp_mline_index());
665 json->SetString(kWebRtcCandidate, candidate_str);
666 std::string json_str;
667 if (!base::JSONWriter::Write(json.get(), &json_str)) {
668 LOG(ERROR) << "Failed to serialize candidate message.";
669 return;
671 SendMessageToClient(kSubjectNewCandidate, json_str);
674 } // namespace remoting