Add WebRTC support in ChromotingClient

Now ChromotingClient creates WebrtcConnectionToHost or
IceConnectionToHost depending on protocol_config.

BUG=547158

Review URL: https://codereview.chromium.org/1649513002

Cr-Commit-Position: refs/heads/master@{#372204}

remoting/client/chromoting_client.cc

@@ -15,11 +15,11 @@
 #include "remoting/protocol/connection_to_host.h"
 #include "remoting/protocol/host_stub.h"
 #include "remoting/protocol/ice_connection_to_host.h"
-#include "remoting/protocol/ice_transport.h"
 #include "remoting/protocol/jingle_session_manager.h"
 #include "remoting/protocol/session_config.h"
 #include "remoting/protocol/transport_context.h"
 #include "remoting/protocol/video_renderer.h"
+#include "remoting/protocol/webrtc_connection_to_host.h"
 
 namespace remoting {
 
@@ -27,9 +27,7 @@ ChromotingClient::ChromotingClient(ClientContext* client_context,
                                    ClientUserInterface* user_interface,
                                    protocol::VideoRenderer* video_renderer,
                                    scoped_ptr<AudioPlayer> audio_player)
-    : user_interface_(user_interface),
-      video_renderer_(video_renderer),
-      connection_(new protocol::IceConnectionToHost()) {
+    : user_interface_(user_interface), video_renderer_(video_renderer) {
   DCHECK(client_context->main_task_runner()->BelongsToCurrentThread());
   if (audio_player) {
     audio_decode_scheduler_.reset(new AudioDecodeScheduler(
@@ -65,17 +63,30 @@ void ChromotingClient::Start(
   host_jid_ = host_jid;
   local_capabilities_ = capabilities;
 
+  if (!protocol_config_)
+    protocol_config_ = protocol::CandidateSessionConfig::CreateDefault();
+  if (!audio_decode_scheduler_)
+    protocol_config_->DisableAudioChannel();
+
+  if (!connection_) {
+    if (protocol_config_->webrtc_supported()) {
+      DCHECK(!protocol_config_->ice_supported());
+#if defined(OS_NACL)
+      LOG(FATAL) << "WebRTC is not supported in webapp.";
+#else   // defined(OS_NACL)
+      connection_.reset(new protocol::WebrtcConnectionToHost());
+#endif  // !defined(OS_NACL)
+    } else {
+      DCHECK(protocol_config_->ice_supported());
+      connection_.reset(new protocol::IceConnectionToHost());
+    }
+  }
   connection_->set_client_stub(this);
   connection_->set_clipboard_stub(this);
   connection_->set_video_renderer(video_renderer_);
   connection_->set_audio_stub(audio_decode_scheduler_.get());
 
   session_manager_.reset(new protocol::JingleSessionManager(signal_strategy));
-
-  if (!protocol_config_)
-    protocol_config_ = protocol::CandidateSessionConfig::CreateDefault();
-  if (!audio_decode_scheduler_)
-    protocol_config_->DisableAudioChannel();
   session_manager_->set_protocol_config(std::move(protocol_config_));
 
   authenticator_ = std::move(authenticator);

remoting/protocol/jingle_messages.cc

@@ -25,6 +25,8 @@ const char kJingleNamespace[] = "urn:xmpp:jingle:1";
 // Namespace for transport messages when using standard ICE.
 const char kIceTransportNamespace[] = "google:remoting:ice";
 
+const char kWebrtcTransportNamespace[] = "google:remoting:webrtc";
+
 const char kEmptyNamespace[] = "";
 const char kXmlNamespace[] = "http://www.w3.org/XML/1998/namespace";
 
@@ -332,11 +334,15 @@ scoped_ptr<buzz::XmlElement> JingleMessage::ToXml() const {
                          ContentDescription::kChromotingContentName);
     content_tag->AddAttr(QName(kEmptyNamespace, "creator"), "initiator");
 
-    if (description.get())
+    if (description)
       content_tag->AddElement(description->ToXml());
 
-    if (transport_info)
+    if (transport_info) {
       content_tag->AddElement(new XmlElement(*transport_info));
+    } else if (description && description->config()->webrtc_supported()) {
+      content_tag->AddElement(
+          new XmlElement(QName(kWebrtcTransportNamespace, "transport")));
+    }
   }
 
   return root;