New PeerConnection handler in Chrome to support latest PeerConnection draft

content/renderer/media/media_stream_dependency_factory.cc

 // Copyright (c) 2012 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 "content/renderer/media/media_stream_dependency_factory.h"
 
 #include <vector>
 
 #include "base/synchronization/waitable_event.h"
 #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/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/peer_connection_handler_jsep.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"
 #include "content/renderer/p2p/ipc_network_manager.h"
 #include "content/renderer/p2p/ipc_socket_factory.h"
 #include "content/renderer/p2p/port_allocator.h"
 #include "jingle/glue/thread_wrapper.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamComponent.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamDescriptor.h"
@@ skipping 64 matching lines @@
 }
 
 WebKit::WebPeerConnection00Handler*
 MediaStreamDependencyFactory::CreatePeerConnectionHandlerJsep(
     WebKit::WebPeerConnection00HandlerClient* client) {
   // Save histogram data so we can see how much PeerConnetion is used.
   // The histogram counts the number of calls to the JS API
   // 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
   // (input or output) or video device cannot be opened. Handling such cases
   // better is a higher level design discussion which involves the media
   // manager, webrtc and libjingle. We cannot create any native
   // track objects however, so we'll just have to skip that. Furthermore,
   // creating a peer connection later on will fail if we don't have a factory.
   if (!EnsurePeerConnectionFactory())
     return false;
@@ skipping 80 matching lines @@
   return pc_factory_.get() != NULL;
 }
 
 talk_base::scoped_refptr<webrtc::PeerConnectionInterface>
 MediaStreamDependencyFactory::CreatePeerConnection(
     const std::string& config,
     webrtc::PeerConnectionObserver* observer) {
   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) {
   return pc_factory_->CreateLocalMediaStream(label);
 }
 
 talk_base::scoped_refptr<webrtc::LocalVideoTrackInterface>
 MediaStreamDependencyFactory::CreateLocalVideoTrack(
     const std::string& label,
     int video_session_id) {
@@ skipping 10 matching lines @@
     const std::string& label,
     webrtc::AudioDeviceModule* audio_device) {
   return pc_factory_->CreateLocalAudioTrack(label, audio_device);
 }
 
 webrtc::SessionDescriptionInterface*
 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,
     const std::string& sdp) {
   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
 }
 
 void MediaStreamDependencyFactory::SetAudioDeviceSessionId(int session_id) {
   audio_device_->SetSessionId(session_id);
 }
@@ skipping 87 matching lines @@
           base::Unretained(this)));
       // Stopping the thread will wait until all tasks have been
       // processed before returning. We wait for the above task to finish before
       // letting the the function continue to avoid any potential race issues.
       chrome_worker_thread_.Stop();
     } else {
       NOTREACHED() << "Worker thread not running.";
     }
   }
 }