CastExtension Impl for Chromoting Host

remoting/host/cast_extension_session.cc

+// 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/callback.h"
+#include "base/json/json_reader.h"
+#include "base/json/json_writer.h"
+#include "base/logging.h"
+#include "base/message_loop/message_loop.h"
+#include "base/synchronization/waitable_event.h"
+#include "net/url_request/url_request_context_getter.h"
+#include "remoting/host/chromium_port_allocator_factory.h"
+// #include "remoting/host/cast_video_capturer.h"
+#include "remoting/host/client_session.h"
+#include "remoting/proto/control.pb.h"
+#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
+#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h"
+#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h"
+#include "third_party/webrtc/modules/desktop_capture/mouse_cursor_shape.h"
+
+namespace remoting {
+
+// Constant keys used in JSON messages to the host.
+// Must keep synced with webapp.
+const char kMessageData[] = "data";  // Use chromoting_data for CC v2.
+const char kMessageSubject[] = "subject";
+const char kMessageType[] = "cast_message";
+
+const char kSubjectNewCandidate[] = "webrtc_candidate";
+const char kSubjectReady[] = "ready";
+const char kSubjectSDP[] = "webrtc_sdp";
+const char kSubjectTest[] = "test";
+
+const char kWebRtcPrefix[] = "webrtc_";
+
+// WebRTC Headers inside cast extension messages.
+const char kWebRtcCandidate[] = "candidate";
+const char kWebRtcSessionDescType[] = "type";
+const char kWebRtcSessionDescSDP[] = "sdp";
+const char kWebRtcSDPMid[] = "sdpMid";
+const char kWebRtcSDPMLineIndex[] = "sdpMLineIndex";
+
+// Constants used by PeerConnection.
+const char kVideoLabel[] = "cast_video_label";
+const char kStreamLabel[] = "stream_label";
+const char kDefaultStunURI[] = "stun:stun.l.google.com:19302";
+
+//------------------------------------------------------------------------------
+
+// webrtc::SetSessionDescriptionObserver implementation.
+class CastSetSessionDescriptionObserver
+    : public webrtc::SetSessionDescriptionObserver {
+ public:
+  static CastSetSessionDescriptionObserver* Create() {
+    return new talk_base::RefCountedObject<CastSetSessionDescriptionObserver>();
+  }
+  virtual void OnSuccess() OVERRIDE {
+    VLOG(1) << "SetSessionDescriptionObserver success.";
+  }
+  virtual void OnFailure(const std::string& error) OVERRIDE {
+    LOG(ERROR) << "CastSetSessionDescriptionObserver" << __FUNCTION__ << " "
+               << error;
+  }
+
+ protected:
+  CastSetSessionDescriptionObserver() {}
+  virtual ~CastSetSessionDescriptionObserver() {}
+};
+
+// webrtc::CreateSessionDescriptionObserver implementation.
+class CastCreateSessionDescriptionObserver
+    : public webrtc::CreateSessionDescriptionObserver {
+ public:
+  static CastCreateSessionDescriptionObserver* Create(
+      CastExtensionSession* session) {
+    return new talk_base::RefCountedObject<
+        CastCreateSessionDescriptionObserver>(session);
+  }
+  virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc) OVERRIDE {
+    if (session_ == NULL) {
+      LOG(ERROR) << "No Session, cannot create session description.";
+      return;
+    }
+    session_->OnSuccess(desc);
+  }
+  virtual void OnFailure(const std::string& error) OVERRIDE {
+    if (session_ == NULL) {
+      LOG(ERROR) << "No Session, cannot create session description.";
+      return;
+    }
+    session_->OnFailure(error);
+  }
+
+ protected:
+  explicit CastCreateSessionDescriptionObserver(CastExtensionSession* session)
+      : session_(session) {}
+  virtual ~CastCreateSessionDescriptionObserver() {}
+
+ private:
+  CastExtensionSession* session_;
+};
+
+// webrtc::StatsObserver implementation.
+class CastStatsObserver : public webrtc::StatsObserver {
+ public:
+  static CastStatsObserver* Create() {
+    return new talk_base::RefCountedObject<CastStatsObserver>();
+  }
+
+  virtual void OnComplete(
+      const std::vector<webrtc::StatsReport>& reports) OVERRIDE {
+    if (reports.empty()) {
+      VLOG(1) << "Received 0 StatsReports.";
+    }
+    VLOG(1) << "Received " << reports.size() << " new StatsReports.";
+    std::vector<webrtc::StatsReport>::const_iterator it;
+    for (it = reports.begin(); it != reports.end(); ++it) {
+      LogStatsReport(*it);
+    }
+  }
+
+ protected:
+  CastStatsObserver() {}
+  virtual ~CastStatsObserver() {}
+
+  void LogStatsReport(const webrtc::StatsReport& report) {
+    typedef webrtc::StatsReport StatsReport;
+    typedef webrtc::StatsReport::Values::iterator ValuesIterator;
+    webrtc::StatsReport::Values v = report.values;
+    for (ValuesIterator it = v.begin(); it != v.end(); ++it) {
+      VLOG(1) << "Param: " << it->name << "; Value: " << it->value << ".";
+    }
+  }
+};
+
+//------------------------------------------------------------------------------
+
+// static
+CastExtensionSession* CastExtensionSession::Create(
+    scoped_refptr<base::SingleThreadTaskRunner> network_task_runner,
+    scoped_refptr<base::SingleThreadTaskRunner> video_capture_task_runner,
+    scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
+    const protocol::NetworkSettings& network_settings,
+    ClientSession* client_session) {
+  scoped_ptr<CastExtensionSession> cast_extension_session(
+      new CastExtensionSession(network_task_runner,
+                               video_capture_task_runner,
+                               url_request_context_getter,
+                               network_settings,
+                               client_session));
+  bool success = cast_extension_session->WrapTasksAndSave();
+  success = cast_extension_session->InitializePeerConnection();
+  return (success ? cast_extension_session.release() : NULL);
+}
+
+CastExtensionSession::CastExtensionSession(
+    scoped_refptr<base::SingleThreadTaskRunner> network_task_runner,
+    scoped_refptr<base::SingleThreadTaskRunner> video_capture_task_runner,
+    scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
+    const protocol::NetworkSettings& network_settings,
+    ClientSession* client_session)
+    : network_task_runner_(network_task_runner),
+      capture_task_runner_(video_capture_task_runner),
+      network_thread_wrapper_(NULL),
+      capture_thread_wrapper_(NULL),
+      url_request_context_getter_(url_request_context_getter),
+      network_settings_(network_settings),
+      client_session_(client_session),
+      stats_observer_(CastStatsObserver::Create()) {
+  DCHECK(network_task_runner_.get() != NULL);
+  DCHECK(capture_task_runner_.get() != NULL);
+  DCHECK(url_request_context_getter_.get() != NULL);
+  DCHECK(client_session_);
+}
+
+CastExtensionSession::~CastExtensionSession() {
+  DeletePeerConnection();
+}
+
+std::string append_path(const char* first, const char* second) {
+  std::string path(first);
+  path.append(".");
+  path.append(second);
+  return path;
+}
+
+// Returns true if the ExtensionMessage was of type |kMessageType|, 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(
+    ClientSession* client_session,
+    const protocol::ExtensionMessage& message) {
+  if (!message.has_type() || message.type().compare(kMessageType) != 0) {
+    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(kMessageSubject, &subject)) {
+    LOG(ERROR) << "Invalid Cast Extension Message (missing subject header).";
+    return true;
+  }
+
+  if (subject == kSubjectSDP) {
+    std::string webrtc_type;
+    std::string sdp;
+    if (!client_message->GetString(
+            append_path(kMessageData, kWebRtcSessionDescType), &webrtc_type)) {
+      LOG(ERROR)
+          << "Invalid Cast Extension Message (missing webrtc type header).";
+      return true;
+    }
+    if (!client_message->GetString(
+            append_path(kMessageData, kWebRtcSessionDescSDP), &sdp)) {
+      LOG(ERROR)
+          << "Invalid Cast Extension Message (missing webrtc sdp header).";
+      return true;
+    }
+    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.line << "; "
+              << error.description + ". (Ignore if empty).";
+      return true;
+    }
+
+    // Save the type because session_description is going to be given off to
+    // PeerConnection.
+    std::string sdp_type = session_description->type();
+    VLOG(1) << "Setting Remote Description.";
+    peer_connection_->SetRemoteDescription(
+        CastSetSessionDescriptionObserver::Create(), session_description);
+    if (sdp_type == webrtc::SessionDescriptionInterface::kOffer) {
+      // Setup MediaStream in PeerConnection because client has made an offer.
+      // TODO(aiguha): Notify client of failure.
+      if (!InitializeAndAddMediaStream()) {
+        LOG(ERROR) << "InitializeAndAddMediaStream failed.";
+        return true;
+      }
+
+      VLOG(1) << "Received Offer. Creating Answer.";
+      peer_connection_->CreateAnswer(
+          CastCreateSessionDescriptionObserver::Create(this), NULL);
+    }
+    return true;
+  } else if (subject == kSubjectNewCandidate) {
+    std::string candidate_str;
+    std::string sdp_mid;
+    int sdp_mlineindex = 0;
+    if (!client_message->GetString(append_path(kMessageData, kWebRtcSDPMid),
+                                   &sdp_mid) ||
+        !client_message->GetInteger(
+            append_path(kMessageData, kWebRtcSDPMLineIndex), &sdp_mlineindex) ||
+        !client_message->GetString(append_path(kMessageData, kWebRtcCandidate),
+                                   &candidate_str)) {
+      LOG(ERROR) << "Invalid Cast Extension Message (could not parse).";
+      return true;
+    }
+    talk_base::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 true;
+    }
+    if (!peer_connection_->AddIceCandidate(candidate.get())) {
+      LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection.";
+      return true;
+    }
+    VLOG(1) << "Received ICE Candidate: " << candidate_str;
+  } else {
+    VLOG(1) << "Unexpected CastExtension Message: " << message.data();
+  }
+  return true;
+}
+
+// Private Methods -------------------------------------------------------------
+
+bool CastExtensionSession::SendMessageToClient(const char* subject,
+                                               const std::string& data) {
+  DCHECK(network_task_runner_->BelongsToCurrentThread());
+  if (client_session_ == NULL) {
+    LOG(ERROR) << "No Client Session. Cannot send message to client.";
+    return false;
+  }
+
+  base::DictionaryValue message_dict;
+  message_dict.SetString(kMessageSubject, subject);
+  message_dict.SetString(kMessageData, data);
+  std::string message_json;
+
+  if (!base::JSONWriter::Write(&message_dict, &message_json)) {
+    LOG(ERROR) << "Failed to create message json.";
+    return false;
+  }
+
+  protocol::ExtensionMessage message;
+  message.set_type(kMessageType);
+  message.set_data(message_json);
+  client_session_->connection()->client_stub()->DeliverHostMessage(message);
+  return true;
+}
+
+void CastExtensionSession::SendCursorShape(
+    scoped_ptr<protocol::CursorShapeInfo> cursor_shape) {
+  DCHECK(network_task_runner_->BelongsToCurrentThread());
+  if (client_session_ == NULL)
+    return;
+
+  protocol::ClientStub* client_stub =
+      client_session_->connection()->client_stub();
+
+  if (client_stub == NULL)
+    return;
+
+  client_stub->SetCursorShape(*cursor_shape);
+}
+
+void CastExtensionSession::EnsureTaskAndSetSend(talk_base::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(network_task_runner_->BelongsToCurrentThread());
+  EnsureTaskAndSetSend(&network_thread_wrapper_);
+
+  if (network_thread_wrapper_ == NULL)
+    return false;
+
+  base::WaitableEvent wrap_capture_thread_event(true, false);
+  capture_task_runner_->PostTask(
+      FROM_HERE,
+      base::Bind(&CastExtensionSession::EnsureTaskAndSetSend,
+                 base::Unretained(this),
+                 &capture_thread_wrapper_,
+                 &wrap_capture_thread_event));
+  wrap_capture_thread_event.Wait();
+
+  return (capture_thread_wrapper_ != NULL);
+}
+
+bool CastExtensionSession::InitializePeerConnection() {
+  DCHECK(network_task_runner_->BelongsToCurrentThread());
+  DCHECK(peer_conn_factory_.get() == NULL);
+  DCHECK(peer_connection_.get() == NULL);
+  DCHECK(capture_thread_wrapper_ != NULL);
+  DCHECK(network_thread_wrapper_ != NULL);
+
+  // TODO(aiguha): Confirm that worker and signalling threads are being
+  // assigned appropriately. For the below configuration to work,
+  // |capture_task_runner_| was changed to have a TYPE_IO MessageLoop.
+  peer_conn_factory_ = webrtc::CreatePeerConnectionFactory(
+      capture_thread_wrapper_, network_thread_wrapper_, NULL, NULL, NULL);
+
+  if (!peer_conn_factory_.get()) {
+    LOG(ERROR) << "Failed to initialize PeerConnectionFactory";
+    DeletePeerConnection();
+    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;
+  webrtc::FakeConstraints constraints;
+
+  // 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.
+  constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
+                           webrtc::MediaConstraintsInterface::kValueTrue);
+
+  peer_connection_ = peer_conn_factory_->CreatePeerConnection(
+      rtc_config,
+      &constraints,
+      ChromiumPortAllocatorFactory::Create(network_settings_,
+                                       url_request_context_getter_),
+      NULL,
+      this);
+
+  if (!peer_connection_.get()) {
+    LOG(ERROR) << "Failed to initialize PeerConnection.";
+    DeletePeerConnection();
+    return false;
+  }
+
+  VLOG(1) << "Created PeerConnection successfully.";
+
+  if (!SendMessageToClient(kSubjectTest, "Hello, client.")) {
+    LOG(ERROR) << "Failed to send test message to client.";
+    return false;
+  }
+  // Sending this message triggers the client to start the peer connection
+  // offer/answer negotiation.
+  if (!SendMessageToClient(kSubjectReady, "Host ready to receive offers.")) {
+    LOG(ERROR) << "Failed to send ready message to client.";
+    return false;
+  }
+
+  return true;
+}
+
+bool CastExtensionSession::InitializeAndAddMediaStream() {
+  DCHECK(network_task_runner_->BelongsToCurrentThread());
+
+  if (stream_) {
+    VLOG(1) << "Already added MediaStream.";
+    return false;  // Already added.
+  }
+  scoped_ptr<webrtc::ScreenCapturer> screen_capturer =
+      client_session_->RequestScreenCapturer();
+  screen_capturer->SetMouseShapeObserver(this);
+  // scoped_ptr<CastVideoCapturer> cast_video_capturer(
+  //     new CastVideoCapturer(capture_task_runner_, screen_capturer.Pass()));

[aiguha, 2014/07/29 04:23:45]

This is the only bit of code that relies on the other CL linked in the
description. I've tried to separate this as best as possible, but this was
unavoidable :( I'm confident that we can get the CastVideoCapturer CL in soon,
after which I can merge those changes in here.

+  std::string minFrameRate = "3";
+  webrtc::FakeConstraints video_constraints;
+  video_constraints.AddMandatory(
+      webrtc::MediaConstraintsInterface::kMinFrameRate, minFrameRate);
+
+  talk_base::scoped_refptr<webrtc::VideoTrackInterface> video_track =
+      peer_conn_factory_->CreateVideoTrack(
+          kVideoLabel,
+          peer_conn_factory_->CreateVideoSource(NULL,
+                                                &video_constraints));
+
+  VLOG(1) << "Created VideoTrack successfully.";
+  stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel);
+
+  if (!stream_->AddTrack(video_track)) {
+    LOG(ERROR) << "Failed to add VideoTrack to MediaStream.";
+    return false;
+  } else {
+    VLOG(1) << "Added VideoTrack to MediaStream successfully.";
+  }
+
+  if (!peer_connection_->AddStream(stream_, NULL)) {
+    VLOG(1) << "Failed to add MediaStream to PeerConnection.";
+    return false;
+  } else {
+    VLOG(1) << "Added Stream to PeerConnection successfully.";
+  }
+  return true;
+}
+
+void CastExtensionSession::PollPeerConnectionStats() {
+  if (!connection_active()) {
+    VLOG(1) << "Cannot poll stats while PeerConnection is inactive.";
+  }
+  talk_base::scoped_refptr<webrtc::MediaStreamTrackInterface> video_track =
+      stream_->FindVideoTrack(kVideoLabel);
+  peer_connection_->GetStats(
+      stats_observer_,
+      video_track.release(),
+      webrtc::PeerConnectionInterface::kStatsOutputLevelStandard);
+}
+
+void CastExtensionSession::DeletePeerConnection() {
+  peer_connection_->Close();
+  // peer_connection_ = NULL;
+  stream_ = NULL;
+  peer_conn_factory_ = NULL;
+}
+
+bool CastExtensionSession::connection_active() const {
+  return peer_connection_.get() != NULL;
+}
+
+// MouseShapeObserver implementation -------------------------------------------
+
+// TODO(aiguha): To reduce duplication, it would perhaps be
+// better to create a single MouseShapeObserver outside of VideoScheduler
+// that can be attached to the ScreenCapturer on its creation, in
+// ClientSession.
+void CastExtensionSession::OnCursorShapeChanged(
+    webrtc::MouseCursorShape* cursor_shape) {
+  DCHECK(capture_task_runner_->BelongsToCurrentThread());
+  VLOG(1) << __FUNCTION__ << " called.";
+  scoped_ptr<webrtc::MouseCursorShape> owned_cursor(cursor_shape);
+
+  scoped_ptr<protocol::CursorShapeInfo> cursor_proto(
+      new protocol::CursorShapeInfo());
+  cursor_proto->set_width(cursor_shape->size.width());
+  cursor_proto->set_height(cursor_shape->size.height());
+  cursor_proto->set_hotspot_x(cursor_shape->hotspot.x());
+  cursor_proto->set_hotspot_y(cursor_shape->hotspot.y());
+  cursor_proto->set_data(cursor_shape->data);
+
+  network_task_runner_->PostTask(
+      FROM_HERE,
+      base::Bind(&CastExtensionSession::SendCursorShape,
+                 base::Unretained(this),
+                 base::Passed(&cursor_proto)));
+}
+
+// CreateSessionDescriptionObserver related methods ----------------------------
+
+void CastExtensionSession::OnSuccess(
+    webrtc::SessionDescriptionInterface* desc) {
+  if (!network_task_runner_->BelongsToCurrentThread()) {
+    network_task_runner_->PostTask(
+        FROM_HERE,
+        base::Bind(
+            &CastExtensionSession::OnSuccess, 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 = kWebRtcPrefix + desc->type();
+  std::string desc_str;
+  desc->ToString(&desc_str);
+  json->SetString(kWebRtcSessionDescSDP, desc_str);
+  std::string json_str;
+  base::JSONWriter::Write(json.get(), &json_str);
+  SendMessageToClient(subject.c_str(), json_str);
+}
+
+void CastExtensionSession::OnFailure(const std::string& error) {
+  VLOG(1) << __FUNCTION__ << " called with: " << error;
+}
+
+// PeerConnectionObserver implementation ---------------------------------------
+
+void CastExtensionSession::OnError() {
+  VLOG(1) << __FUNCTION__;
+}
+void CastExtensionSession::OnSignalingChange(
+    webrtc::PeerConnectionInterface::SignalingState new_state) {
+  VLOG(1) << "Function CastExtensionSession::OnSignalingChange called with "
+          << new_state;
+}
+void CastExtensionSession::OnStateChange(
+    webrtc::PeerConnectionObserver::StateType state_changed) {
+  VLOG(1) << __FUNCTION__ << " called with input " << state_changed;
+}
+void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) {
+  VLOG(1) << __FUNCTION__ << " " << stream->label();
+}
+void CastExtensionSession::OnRemoveStream(
+    webrtc::MediaStreamInterface* stream) {
+  VLOG(1) << __FUNCTION__ << " " << stream->label();
+}
+void CastExtensionSession::OnDataChannel(
+    webrtc::DataChannelInterface* data_channel) {
+  VLOG(1) << __FUNCTION__ << " called with " << data_channel->label();
+}
+void CastExtensionSession::OnRenegotiationNeeded() {
+  VLOG(1) << __FUNCTION__ << " called.";
+}
+void CastExtensionSession::OnIceConnectionChange(
+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
+  VLOG(1) << __FUNCTION__ << " called with new IceConnectionState "
+          << new_state;
+}
+void CastExtensionSession::OnIceGatheringChange(
+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
+  VLOG(1) << __FUNCTION__ << " called with new IceGatheringState " << new_state;
+}
+
+void CastExtensionSession::OnIceComplete() {
+  VLOG(1) << __FUNCTION__ << " called.";
+  // TODO(aiguha): Maybe start timer only if enabled by command-line flag or
+  // at a particular verbosity level.
+  stats_polling_timer_.Start(FROM_HERE,
+                             base::TimeDelta::FromSeconds(10),
+                             this,
+                             &CastExtensionSession::PollPeerConnectionStats);
+}
+
+void CastExtensionSession::OnIceCandidate(
+    const webrtc::IceCandidateInterface* candidate) {
+  std::string candidate_str;
+  if (!candidate->ToString(&candidate_str)) {
+    LOG(ERROR) << __FUNCTION__ << " called, but could not serialize candidate.";
+    return;
+  }
+  VLOG(1) << __FUNCTION__ << " called with " << candidate_str;
+  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;
+  base::JSONWriter::Write(json.get(), &json_str);
+  SendMessageToClient(kSubjectNewCandidate, json_str);
+}
+
+}  // namespace remoting
+