Implement current/pending session description methods.

webrtc/api/peerconnection.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1370 matching lines @@
 }
 
 const SessionDescriptionInterface* PeerConnection::local_description() const {
   return session_->local_description();
 }
 
 const SessionDescriptionInterface* PeerConnection::remote_description() const {
   return session_->remote_description();
 }
 
+const SessionDescriptionInterface* PeerConnection::current_local_description()
+    const {
+  return session_->current_local_description();
+}
+
+const SessionDescriptionInterface* PeerConnection::current_remote_description()
+    const {
+  return session_->current_remote_description();
+}
+
+const SessionDescriptionInterface* PeerConnection::pending_local_description()
+    const {
+  return session_->pending_local_description();
+}
+
+const SessionDescriptionInterface* PeerConnection::pending_remote_description()
+    const {
+  return session_->pending_remote_description();
+}
+
 void PeerConnection::Close() {
   TRACE_EVENT0("webrtc", "PeerConnection::Close");
   // Update stats here so that we have the most recent stats for tracks and
   // streams before the channels are closed.
   stats_->UpdateStats(kStatsOutputLevelStandard);
 
   session_->Close();
 }
 
 void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
@@ skipping 1009 matching lines @@
 
 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
                                         int64_t max_size_bytes) {
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   event_log_->StopLogging();
 }
 }  // namespace webrtc