New PeerConnection handler in Chrome to support latest PeerConnection draft

content/renderer/media/media_stream_dependency_factory.cc

Index: content/renderer/media/media_stream_dependency_factory.cc
diff --git a/content/renderer/media/media_stream_dependency_factory.cc b/content/renderer/media/media_stream_dependency_factory.cc
index 4f4565646f9aa8a155ceeee710bf8af92158da70..8eafbaed4d7bf6a700aad5e98b8f4baf0a209043 100644
--- a/content/renderer/media/media_stream_dependency_factory.cc
+++ b/content/renderer/media/media_stream_dependency_factory.cc
@@ -10,8 +10,9 @@
 #include "base/utf_string_conversions.h"
 #include "content/renderer/media/media_stream_extra_data.h"
 #include "content/renderer/media/media_stream_source_extra_data.h"
-#include "content/renderer/media/rtc_video_capturer.h"
 #include "content/renderer/media/peer_connection_handler_jsep.h"
+#include "content/renderer/media/rtc_peer_connection_handler.h"
+#include "content/renderer/media/rtc_video_capturer.h"
 #include "content/renderer/media/video_capture_impl_manager.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
@@ -96,13 +97,26 @@ MediaStreamDependencyFactory::CreatePeerConnectionHandlerJsep(
   // webKitPeerConnection00.
   UpdateWebRTCMethodCount(WEBKIT_PEER_CONNECTION);
 
-  if (!EnsurePeerConnectionFactory()) {
+  if (!EnsurePeerConnectionFactory())
     return NULL;
-  }
 
   return new PeerConnectionHandlerJsep(client, this);
 }
 
+WebKit::WebRTCPeerConnectionHandler*
+MediaStreamDependencyFactory::CreateRTCPeerConnectionHandler(
+    WebKit::WebRTCPeerConnectionHandlerClient* client) {
+  // Save histogram data so we can see how much PeerConnetion is used.
+  // The histogram counts the number of calls to the JS API
+  // webKitRTCPeerConnection.
+  UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
+
+  if (!EnsurePeerConnectionFactory())
+    return NULL;
+
+  return new RTCPeerConnectionHandler(client, this);
+}
+
 bool MediaStreamDependencyFactory::CreateNativeLocalMediaStream(
     WebKit::WebMediaStreamDescriptor* description) {
   // Creating the peer connection factory can fail if for example the audio
@@ -203,6 +217,14 @@ MediaStreamDependencyFactory::CreatePeerConnection(
   return pc_factory_->CreatePeerConnection(config, observer);
 }
 
+talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
+MediaStreamDependencyFactory::CreatePeerConnection(
+    const webrtc::JsepInterface::IceServers& ice_servers,
+    const webrtc::MediaConstraintsInterface* constraints,
+    webrtc::PeerConnectionObserver* observer) {
+  return pc_factory_->CreatePeerConnection(ice_servers, constraints, observer);
+}
+
 talk_base::scoped_refptr<webrtc::LocalMediaStreamInterface>
 MediaStreamDependencyFactory::CreateLocalMediaStream(
     const std::string& label) {
@@ -233,6 +255,12 @@ MediaStreamDependencyFactory::CreateSessionDescription(const std::string& sdp) {
   return webrtc::CreateSessionDescription(sdp);
 }
 
+webrtc::SessionDescriptionInterface*
+MediaStreamDependencyFactory::CreateSessionDescription(const std::string& type,
+                                                       const std::string& sdp) {
+  return webrtc::CreateSessionDescription(type, sdp);
+}
+
 webrtc::IceCandidateInterface* MediaStreamDependencyFactory::CreateIceCandidate(
     const std::string& sdp_mid,
     int sdp_mline_index,