Index: remoting/host/cast_extension_session.cc |
diff --git a/remoting/host/cast_extension_session.cc b/remoting/host/cast_extension_session.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..f711b466be299db9e1442b2c7b9c4e1534f1452e |
--- /dev/null |
+++ b/remoting/host/cast_extension_session.cc |
@@ -0,0 +1,675 @@ |
+// Copyright 2014 The Chromium Authors. All rights reserved. |
+// Use of this source code is governed by a BSD-style license that can be |
+// found in the LICENSE file. |
+ |
+#include "remoting/host/cast_extension_session.h" |
+ |
+#include "base/bind.h" |
+#include "base/json/json_reader.h" |
+#include "base/json/json_writer.h" |
+#include "base/logging.h" |
+#include "base/synchronization/waitable_event.h" |
+#include "net/url_request/url_request_context_getter.h" |
+#include "remoting/host/cast_video_capturer_adapter.h" |
+#include "remoting/host/chromium_port_allocator_factory.h" |
+#include "remoting/host/client_session.h" |
+#include "remoting/proto/control.pb.h" |
+#include "remoting/protocol/client_stub.h" |
+#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
+#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" |
+#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" |
+ |
+namespace remoting { |
+ |
+// Used as the type attribute of all Cast protocol::ExtensionMessages. |
+const char kExtensionMessageType[] = "cast_message"; |
+ |
+// Top-level keys used in all extension messages between host and client. |
+// Must keep synced with webapp. |
+const char kTopLevelData[] = "chromoting_data"; |
+const char kTopLevelSubject[] = "subject"; |
+ |
+// Keys used to describe the subject of a cast extension message. WebRTC-related |
+// message subjects are prepended with "webrtc_". |
+// Must keep synced with webapp. |
+const char kSubjectReady[] = "ready"; |
+const char kSubjectTest[] = "test"; |
+const char kSubjectNewCandidate[] = "webrtc_candidate"; |
+const char kSubjectOffer[] = "webrtc_offer"; |
+const char kSubjectAnswer[] = "webrtc_answer"; |
+ |
+// WebRTC headers used inside messages with subject = "webrtc_*". |
+const char kWebRtcCandidate[] = "candidate"; |
+const char kWebRtcSessionDescType[] = "type"; |
+const char kWebRtcSessionDescSDP[] = "sdp"; |
+const char kWebRtcSDPMid[] = "sdpMid"; |
+const char kWebRtcSDPMLineIndex[] = "sdpMLineIndex"; |
+ |
+// Media labels used over the PeerConnection. |
+const char kVideoLabel[] = "cast_video_label"; |
+const char kStreamLabel[] = "stream_label"; |
+ |
+// Default STUN server used to construct |
+// webrtc::PeerConnectionInterface::RTCConfiguration for the PeerConnection. |
+const char kDefaultStunURI[] = "stun:stun.l.google.com:19302"; |
+ |
+const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread"; |
+ |
+// Interval between each call to PollPeerConnectionStats(). |
+const int kStatsLogIntervalSec = 10; |
+ |
+// Minimum frame rate for video streaming over the PeerConnection in frames per |
+// second, added as a media constraint when constructing the video source for |
+// the Peer Connection. |
+const int kMinFramesPerSecond = 5; |
+ |
+// A webrtc::SetSessionDescriptionObserver implementation used to receive the |
+// results of setting local and remote descriptions of the PeerConnection. |
+class CastSetSessionDescriptionObserver |
+ : public webrtc::SetSessionDescriptionObserver { |
+ public: |
+ static CastSetSessionDescriptionObserver* Create() { |
+ return new rtc::RefCountedObject<CastSetSessionDescriptionObserver>(); |
+ } |
+ virtual void OnSuccess() OVERRIDE { |
+ VLOG(1) << "Setting session description succeeded."; |
+ } |
+ virtual void OnFailure(const std::string& error) OVERRIDE { |
+ LOG(ERROR) << "Setting session description failed: " << error; |
+ } |
+ |
+ protected: |
+ CastSetSessionDescriptionObserver() {} |
+ virtual ~CastSetSessionDescriptionObserver() {} |
+ |
+ DISALLOW_COPY_AND_ASSIGN(CastSetSessionDescriptionObserver); |
+}; |
+ |
+// A webrtc::CreateSessionDescriptionObserver implementation used to receive the |
+// results of creating descriptions for this end of the PeerConnection. |
+class CastCreateSessionDescriptionObserver |
+ : public webrtc::CreateSessionDescriptionObserver { |
+ public: |
+ static CastCreateSessionDescriptionObserver* Create( |
+ CastExtensionSession* session) { |
+ return new rtc::RefCountedObject<CastCreateSessionDescriptionObserver>( |
+ session); |
+ } |
+ virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc) OVERRIDE { |
+ if (cast_extension_session_ == NULL) { |
+ LOG(ERROR) |
+ << "No CastExtensionSession. Creating session description succeeded."; |
+ return; |
+ } |
+ cast_extension_session_->OnCreateSessionDescription(desc); |
+ } |
+ virtual void OnFailure(const std::string& error) OVERRIDE { |
+ if (cast_extension_session_ == NULL) { |
+ LOG(ERROR) |
+ << "No CastExtensionSession. Creating session description failed."; |
+ return; |
+ } |
+ cast_extension_session_->OnCreateSessionDescriptionFailure(error); |
+ } |
+ void SetCastExtensionSession(CastExtensionSession* cast_extension_session) { |
+ cast_extension_session_ = cast_extension_session; |
+ } |
+ |
+ protected: |
+ explicit CastCreateSessionDescriptionObserver(CastExtensionSession* session) |
+ : cast_extension_session_(session) {} |
+ virtual ~CastCreateSessionDescriptionObserver() {} |
+ |
+ private: |
+ CastExtensionSession* cast_extension_session_; |
+ |
+ DISALLOW_COPY_AND_ASSIGN(CastCreateSessionDescriptionObserver); |
+}; |
+ |
+// A webrtc::StatsObserver implementation used to receive statistics about the |
+// current PeerConnection. |
+class CastStatsObserver : public webrtc::StatsObserver { |
+ public: |
+ static CastStatsObserver* Create() { |
+ return new rtc::RefCountedObject<CastStatsObserver>(); |
+ } |
+ |
+ virtual void OnComplete( |
+ const std::vector<webrtc::StatsReport>& reports) OVERRIDE { |
+ typedef webrtc::StatsReport::Values::iterator ValuesIterator; |
+ |
+ VLOG(1) << "Received " << reports.size() << " new StatsReports."; |
+ |
+ int index; |
+ std::vector<webrtc::StatsReport>::const_iterator it; |
+ for (it = reports.begin(), index = 0; it != reports.end(); ++it, ++index) { |
+ webrtc::StatsReport::Values v = it->values; |
+ VLOG(1) << "Report " << index << ":"; |
+ for (ValuesIterator vIt = v.begin(); vIt != v.end(); ++vIt) { |
+ VLOG(1) << "Stat: " << vIt->name << "=" << vIt->value << "."; |
+ } |
+ } |
+ } |
+ |
+ protected: |
+ CastStatsObserver() {} |
+ virtual ~CastStatsObserver() {} |
+ |
+ DISALLOW_COPY_AND_ASSIGN(CastStatsObserver); |
+}; |
+ |
+// TODO(aiguha): Fix PeerConnnection-related tear down crash caused by premature |
+// destruction of cricket::CaptureManager (which occurs on releasing |
+// |peer_conn_factory_|). See crbug.com/403840. |
+CastExtensionSession::~CastExtensionSession() { |
+ DCHECK(caller_task_runner_->BelongsToCurrentThread()); |
+ |
+ // Explicitly clear |create_session_desc_observer_|'s pointer to |this|, |
+ // since the CastExtensionSession is destructing. Otherwise, |
+ // |create_session_desc_observer_| would be left with a dangling pointer. |
+ create_session_desc_observer_->SetCastExtensionSession(NULL); |
+ |
+ CleanupPeerConnection(); |
+} |
+ |
+// static |
+scoped_ptr<CastExtensionSession> CastExtensionSession::Create( |
+ scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner, |
+ scoped_refptr<net::URLRequestContextGetter> url_request_context_getter, |
+ const protocol::NetworkSettings& network_settings, |
+ ClientSessionControl* client_session_control, |
+ protocol::ClientStub* client_stub) { |
+ scoped_ptr<CastExtensionSession> cast_extension_session( |
+ new CastExtensionSession(caller_task_runner, |
+ url_request_context_getter, |
+ network_settings, |
+ client_session_control, |
+ client_stub)); |
+ if (!cast_extension_session->WrapTasksAndSave()) { |
+ return scoped_ptr<CastExtensionSession>(); |
+ } |
+ if (!cast_extension_session->InitializePeerConnection()) { |
+ return scoped_ptr<CastExtensionSession>(); |
+ } |
+ return cast_extension_session.Pass(); |
+} |
+ |
+void CastExtensionSession::OnCreateSessionDescription( |
+ webrtc::SessionDescriptionInterface* desc) { |
+ if (!caller_task_runner_->BelongsToCurrentThread()) { |
+ caller_task_runner_->PostTask( |
+ FROM_HERE, |
+ base::Bind(&CastExtensionSession::OnCreateSessionDescription, |
+ base::Unretained(this), |
+ desc)); |
+ return; |
+ } |
+ |
+ peer_connection_->SetLocalDescription( |
+ CastSetSessionDescriptionObserver::Create(), desc); |
+ |
+ scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue()); |
+ json->SetString(kWebRtcSessionDescType, desc->type()); |
+ std::string subject = |
+ (desc->type() == "offer") ? kSubjectOffer : kSubjectAnswer; |
+ std::string desc_str; |
+ desc->ToString(&desc_str); |
+ json->SetString(kWebRtcSessionDescSDP, desc_str); |
+ std::string json_str; |
+ if (!base::JSONWriter::Write(json.get(), &json_str)) { |
+ LOG(ERROR) << "Failed to serialize sdp message."; |
+ return; |
+ } |
+ |
+ SendMessageToClient(subject.c_str(), json_str); |
+} |
+ |
+void CastExtensionSession::OnCreateSessionDescriptionFailure( |
+ const std::string& error) { |
+ VLOG(1) << "Creating Session Description failed: " << error; |
+} |
+ |
+// TODO(aiguha): Support the case(s) where we've grabbed the capturer already, |
+// but another extension reset the video pipeline. We should remove the |
+// stream from the peer connection here, and then attempt to re-setup the |
+// peer connection in the OnRenegotiationNeeded() callback. |
+// See crbug.com/403843. |
+scoped_ptr<webrtc::DesktopCapturer> CastExtensionSession::OnCreateVideoCapturer( |
+ scoped_ptr<webrtc::DesktopCapturer> capturer) { |
+ if (has_grabbed_capturer_) { |
+ LOG(ERROR) << "The video pipeline was reset unexpectedly."; |
+ has_grabbed_capturer_ = false; |
+ peer_connection_->RemoveStream(stream_.release()); |
+ return capturer.Pass(); |
+ } |
+ |
+ if (received_offer_) { |
+ has_grabbed_capturer_ = true; |
+ if (SetupVideoStream(capturer.Pass())) { |
+ peer_connection_->CreateAnswer(create_session_desc_observer_, NULL); |
+ } else { |
+ has_grabbed_capturer_ = false; |
+ // Ignore the received offer, since we failed to setup a video stream. |
+ received_offer_ = false; |
+ } |
+ return scoped_ptr<webrtc::DesktopCapturer>(); |
+ } |
+ |
+ return capturer.Pass(); |
+} |
+ |
+bool CastExtensionSession::ModifiesVideoPipeline() const { |
+ return true; |
+} |
+ |
+// Returns true if the |message| is a Cast ExtensionMessage, even if |
+// it was badly formed or a resulting action failed. This is done so that |
+// the host does not continue to attempt to pass |message| to other |
+// HostExtensionSessions. |
+bool CastExtensionSession::OnExtensionMessage( |
+ ClientSessionControl* client_session_control, |
+ protocol::ClientStub* client_stub, |
+ const protocol::ExtensionMessage& message) { |
+ if (message.type() != kExtensionMessageType) { |
+ return false; |
+ } |
+ |
+ scoped_ptr<base::Value> value(base::JSONReader::Read(message.data())); |
+ base::DictionaryValue* client_message; |
+ if (!(value && value->GetAsDictionary(&client_message))) { |
+ LOG(ERROR) << "Could not read cast extension message."; |
+ return true; |
+ } |
+ |
+ std::string subject; |
+ if (!client_message->GetString(kTopLevelSubject, &subject)) { |
+ LOG(ERROR) << "Invalid Cast Extension Message (missing subject header)."; |
+ return true; |
+ } |
+ |
+ if (subject == kSubjectOffer && !received_offer_) { |
+ // Reset the video pipeline so we can grab the screen capturer and setup |
+ // a video stream. |
+ if (ParseAndSetRemoteDescription(client_message)) { |
+ received_offer_ = true; |
+ LOG(INFO) << "About to ResetVideoPipeline."; |
+ client_session_control_->ResetVideoPipeline(); |
+ |
+ } |
+ } else if (subject == kSubjectAnswer) { |
+ ParseAndSetRemoteDescription(client_message); |
+ } else if (subject == kSubjectNewCandidate) { |
+ ParseAndAddICECandidate(client_message); |
+ } else { |
+ VLOG(1) << "Unexpected CastExtension Message: " << message.data(); |
+ } |
+ return true; |
+} |
+ |
+// Private methods ------------------------------------------------------------ |
+ |
+CastExtensionSession::CastExtensionSession( |
+ scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner, |
+ scoped_refptr<net::URLRequestContextGetter> url_request_context_getter, |
+ const protocol::NetworkSettings& network_settings, |
+ ClientSessionControl* client_session_control, |
+ protocol::ClientStub* client_stub) |
+ : caller_task_runner_(caller_task_runner), |
+ url_request_context_getter_(url_request_context_getter), |
+ network_settings_(network_settings), |
+ client_session_control_(client_session_control), |
+ client_stub_(client_stub), |
+ stats_observer_(CastStatsObserver::Create()), |
+ received_offer_(false), |
+ has_grabbed_capturer_(false), |
+ signaling_thread_wrapper_(NULL), |
+ worker_thread_wrapper_(NULL), |
+ worker_thread_(kWorkerThreadName) { |
+ DCHECK(caller_task_runner_->BelongsToCurrentThread()); |
+ DCHECK(url_request_context_getter_); |
+ DCHECK(client_session_control_); |
+ DCHECK(client_stub_); |
+ |
+ // The worker thread is created with base::MessageLoop::TYPE_IO because |
+ // the PeerConnection performs some port allocation operations on this thread |
+ // that require it. See crbug.com/404013. |
+ base::Thread::Options options(base::MessageLoop::TYPE_IO, 0); |
+ worker_thread_.StartWithOptions(options); |
+ worker_task_runner_ = worker_thread_.task_runner(); |
+} |
+ |
+bool CastExtensionSession::ParseAndSetRemoteDescription( |
+ base::DictionaryValue* message) { |
+ DCHECK(peer_connection_.get() != NULL); |
+ |
+ base::DictionaryValue* message_data; |
+ if (!message->GetDictionary(kTopLevelData, &message_data)) { |
+ LOG(ERROR) << "Invalid Cast Extension Message (missing data)."; |
+ return false; |
+ } |
+ |
+ std::string webrtc_type; |
+ if (!message_data->GetString(kWebRtcSessionDescType, &webrtc_type)) { |
+ LOG(ERROR) |
+ << "Invalid Cast Extension Message (missing webrtc type header)."; |
+ return false; |
+ } |
+ |
+ std::string sdp; |
+ if (!message_data->GetString(kWebRtcSessionDescSDP, &sdp)) { |
+ LOG(ERROR) << "Invalid Cast Extension Message (missing webrtc sdp header)."; |
+ return false; |
+ } |
+ |
+ webrtc::SdpParseError error; |
+ webrtc::SessionDescriptionInterface* session_description( |
+ webrtc::CreateSessionDescription(webrtc_type, sdp, &error)); |
+ |
+ if (!session_description) { |
+ LOG(ERROR) << "Invalid Cast Extension Message (could not parse sdp)."; |
+ VLOG(1) << "SdpParseError was: " << error.description; |
+ return false; |
+ } |
+ |
+ peer_connection_->SetRemoteDescription( |
+ CastSetSessionDescriptionObserver::Create(), session_description); |
+ return true; |
+} |
+ |
+bool CastExtensionSession::ParseAndAddICECandidate( |
+ base::DictionaryValue* message) { |
+ DCHECK(peer_connection_.get() != NULL); |
+ |
+ base::DictionaryValue* message_data; |
+ if (!message->GetDictionary(kTopLevelData, &message_data)) { |
+ LOG(ERROR) << "Invalid Cast Extension Message (missing data)."; |
+ return false; |
+ } |
+ |
+ std::string candidate_str; |
+ std::string sdp_mid; |
+ int sdp_mlineindex = 0; |
+ if (!message_data->GetString(kWebRtcSDPMid, &sdp_mid) || |
+ !message_data->GetInteger(kWebRtcSDPMLineIndex, &sdp_mlineindex) || |
+ !message_data->GetString(kWebRtcCandidate, &candidate_str)) { |
+ LOG(ERROR) << "Invalid Cast Extension Message (could not parse)."; |
+ return false; |
+ } |
+ |
+ rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate( |
+ webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate_str)); |
+ if (!candidate.get()) { |
+ LOG(ERROR) |
+ << "Invalid Cast Extension Message (could not create candidate)."; |
+ return false; |
+ } |
+ |
+ if (!peer_connection_->AddIceCandidate(candidate.get())) { |
+ LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection."; |
+ return false; |
+ } |
+ |
+ VLOG(1) << "Received and Added ICE Candidate: " << candidate_str; |
+ |
+ return true; |
+} |
+ |
+bool CastExtensionSession::SendMessageToClient(const std::string& subject, |
+ const std::string& data) { |
+ DCHECK(caller_task_runner_->BelongsToCurrentThread()); |
+ |
+ if (client_stub_ == NULL) { |
+ LOG(ERROR) << "No Client Stub. Cannot send message to client."; |
+ return false; |
+ } |
+ |
+ base::DictionaryValue message_dict; |
+ message_dict.SetString(kTopLevelSubject, subject); |
+ message_dict.SetString(kTopLevelData, data); |
+ std::string message_json; |
+ |
+ if (!base::JSONWriter::Write(&message_dict, &message_json)) { |
+ LOG(ERROR) << "Failed to serialize JSON message."; |
+ return false; |
+ } |
+ |
+ protocol::ExtensionMessage message; |
+ message.set_type(kExtensionMessageType); |
+ message.set_data(message_json); |
+ client_stub_->DeliverHostMessage(message); |
+ return true; |
+} |
+ |
+void CastExtensionSession::EnsureTaskAndSetSend(rtc::Thread** ptr, |
+ base::WaitableEvent* event) { |
+ jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); |
+ jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); |
+ *ptr = jingle_glue::JingleThreadWrapper::current(); |
+ |
+ if (event != NULL) { |
+ event->Signal(); |
+ } |
+} |
+ |
+bool CastExtensionSession::WrapTasksAndSave() { |
+ DCHECK(caller_task_runner_->BelongsToCurrentThread()); |
+ |
+ EnsureTaskAndSetSend(&signaling_thread_wrapper_); |
+ if (signaling_thread_wrapper_ == NULL) |
+ return false; |
+ |
+ base::WaitableEvent wrap_worker_thread_event(true, false); |
+ worker_task_runner_->PostTask( |
+ FROM_HERE, |
+ base::Bind(&CastExtensionSession::EnsureTaskAndSetSend, |
+ base::Unretained(this), |
+ &worker_thread_wrapper_, |
+ &wrap_worker_thread_event)); |
+ wrap_worker_thread_event.Wait(); |
+ |
+ return (worker_thread_wrapper_ != NULL); |
+} |
+ |
+bool CastExtensionSession::InitializePeerConnection() { |
+ DCHECK(caller_task_runner_->BelongsToCurrentThread()); |
+ DCHECK(!peer_conn_factory_); |
+ DCHECK(!peer_connection_); |
+ DCHECK(worker_thread_wrapper_ != NULL); |
+ DCHECK(signaling_thread_wrapper_ != NULL); |
+ |
+ peer_conn_factory_ = webrtc::CreatePeerConnectionFactory( |
+ worker_thread_wrapper_, signaling_thread_wrapper_, NULL, NULL, NULL); |
+ |
+ if (!peer_conn_factory_.get()) { |
+ CleanupPeerConnection(); |
+ return false; |
+ } |
+ |
+ VLOG(1) << "Created PeerConnectionFactory successfully."; |
+ |
+ webrtc::PeerConnectionInterface::IceServers servers; |
+ webrtc::PeerConnectionInterface::IceServer server; |
+ server.uri = kDefaultStunURI; |
+ servers.push_back(server); |
+ webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; |
+ rtc_config.servers = servers; |
+ |
+ // DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the |
+ // peer connection uses SDES. Disabling SDES as well will cause the peer |
+ // connection to fail to connect. |
+ // Note: For protection and unprotection of SRTP packets, the libjingle |
+ // ENABLE_EXTERNAL_AUTH flag must not be set. |
+ webrtc::FakeConstraints constraints; |
+ constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
+ webrtc::MediaConstraintsInterface::kValueTrue); |
+ |
+ rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> |
+ port_allocator_factory = ChromiumPortAllocatorFactory::Create( |
+ network_settings_, url_request_context_getter_); |
+ |
+ peer_connection_ = peer_conn_factory_->CreatePeerConnection( |
+ rtc_config, &constraints, port_allocator_factory, NULL, this); |
+ |
+ if (!peer_connection_.get()) { |
+ CleanupPeerConnection(); |
+ return false; |
+ } |
+ |
+ VLOG(1) << "Created PeerConnection successfully."; |
+ |
+ create_session_desc_observer_ = |
+ CastCreateSessionDescriptionObserver::Create(this); |
+ |
+ // Send a test message to the client. Then, notify the client to start |
+ // webrtc offer/answer negotiation. |
+ if (!SendMessageToClient(kSubjectTest, "Hello, client.") || |
+ !SendMessageToClient(kSubjectReady, "Host ready to receive offers.")) { |
+ LOG(ERROR) << "Failed to send messages to client."; |
+ return false; |
+ } |
+ |
+ return true; |
+} |
+ |
+bool CastExtensionSession::SetupVideoStream( |
+ scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) { |
+ DCHECK(caller_task_runner_->BelongsToCurrentThread()); |
+ DCHECK(desktop_capturer); |
+ |
+ if (stream_) { |
+ VLOG(1) << "Already added MediaStream. Aborting Setup."; |
+ return false; |
+ } |
+ |
+ scoped_ptr<CastVideoCapturerAdapter> cast_video_capturer_adapter( |
+ new CastVideoCapturerAdapter(desktop_capturer.Pass())); |
+ |
+ // Set video stream constraints. |
+ webrtc::FakeConstraints video_constraints; |
+ video_constraints.AddMandatory( |
+ webrtc::MediaConstraintsInterface::kMinFrameRate, kMinFramesPerSecond); |
+ |
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track = |
+ peer_conn_factory_->CreateVideoTrack( |
+ kVideoLabel, |
+ peer_conn_factory_->CreateVideoSource( |
+ cast_video_capturer_adapter.release(), &video_constraints)); |
+ |
+ stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel); |
+ |
+ if (!stream_->AddTrack(video_track) || |
+ !peer_connection_->AddStream(stream_, NULL)) { |
+ return false; |
+ } |
+ |
+ VLOG(1) << "Setup video stream successfully."; |
+ |
+ return true; |
+} |
+ |
+void CastExtensionSession::PollPeerConnectionStats() { |
+ if (!connection_active()) { |
+ VLOG(1) << "Cannot poll stats while PeerConnection is inactive."; |
+ } |
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> video_track = |
+ stream_->FindVideoTrack(kVideoLabel); |
+ peer_connection_->GetStats( |
+ stats_observer_, |
+ video_track.release(), |
+ webrtc::PeerConnectionInterface::kStatsOutputLevelStandard); |
+} |
+ |
+void CastExtensionSession::CleanupPeerConnection() { |
+ peer_connection_->Close(); |
+ peer_connection_ = NULL; |
+ stream_ = NULL; |
+ peer_conn_factory_ = NULL; |
+ worker_thread_.Stop(); |
+} |
+ |
+bool CastExtensionSession::connection_active() const { |
+ return peer_connection_.get() != NULL; |
+} |
+ |
+// webrtc::PeerConnectionObserver implementation ------------------------------- |
+ |
+void CastExtensionSession::OnError() { |
+ VLOG(1) << "PeerConnectionObserver: an error occurred."; |
+} |
+ |
+void CastExtensionSession::OnSignalingChange( |
+ webrtc::PeerConnectionInterface::SignalingState new_state) { |
+ VLOG(1) << "PeerConnectionObserver: SignalingState changed to:" << new_state; |
+} |
+ |
+void CastExtensionSession::OnStateChange( |
+ webrtc::PeerConnectionObserver::StateType state_changed) { |
+ VLOG(1) << "PeerConnectionObserver: StateType changed to: " << state_changed; |
+} |
+ |
+void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) { |
+ VLOG(1) << "PeerConnectionObserver: stream added: " << stream->label(); |
+} |
+ |
+void CastExtensionSession::OnRemoveStream( |
+ webrtc::MediaStreamInterface* stream) { |
+ VLOG(1) << "PeerConnectionObserver: stream removed: " << stream->label(); |
+} |
+ |
+void CastExtensionSession::OnDataChannel( |
+ webrtc::DataChannelInterface* data_channel) { |
+ VLOG(1) << "PeerConnectionObserver: data channel: " << data_channel->label(); |
+} |
+ |
+void CastExtensionSession::OnRenegotiationNeeded() { |
+ VLOG(1) << "PeerConnectionObserver: renegotiation needed."; |
+} |
+ |
+void CastExtensionSession::OnIceConnectionChange( |
+ webrtc::PeerConnectionInterface::IceConnectionState new_state) { |
+ VLOG(1) << "PeerConnectionObserver: IceConnectionState changed to: " |
+ << new_state; |
+ |
+ // TODO(aiguha): Maybe start timer only if enabled by command-line flag or |
+ // at a particular verbosity level. |
+ if (!stats_polling_timer_.IsRunning() && |
+ new_state == webrtc::PeerConnectionInterface::kIceConnectionConnected) { |
+ stats_polling_timer_.Start( |
+ FROM_HERE, |
+ base::TimeDelta::FromSeconds(kStatsLogIntervalSec), |
+ this, |
+ &CastExtensionSession::PollPeerConnectionStats); |
+ } |
+} |
+ |
+void CastExtensionSession::OnIceGatheringChange( |
+ webrtc::PeerConnectionInterface::IceGatheringState new_state) { |
+ VLOG(1) << "PeerConnectionObserver: IceGatheringState changed to: " |
+ << new_state; |
+} |
+ |
+void CastExtensionSession::OnIceComplete() { |
+ VLOG(1) << "PeerConnectionObserver: all ICE candidates found."; |
+} |
+ |
+void CastExtensionSession::OnIceCandidate( |
+ const webrtc::IceCandidateInterface* candidate) { |
+ std::string candidate_str; |
+ if (!candidate->ToString(&candidate_str)) { |
+ LOG(ERROR) << "PeerConnectionObserver: failed to serialize candidate."; |
+ return; |
+ } |
+ scoped_ptr<base::DictionaryValue> json(new base::DictionaryValue()); |
+ json->SetString(kWebRtcSDPMid, candidate->sdp_mid()); |
+ json->SetInteger(kWebRtcSDPMLineIndex, candidate->sdp_mline_index()); |
+ json->SetString(kWebRtcCandidate, candidate_str); |
+ std::string json_str; |
+ if (!base::JSONWriter::Write(json.get(), &json_str)) { |
+ LOG(ERROR) << "Failed to serialize candidate message."; |
+ return; |
+ } |
+ SendMessageToClient(kSubjectNewCandidate, json_str); |
+} |
+ |
+} // namespace remoting |
+ |