| 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
|
| +
|
|
|