Add PeerConnection IsClosed check.

webrtc/api/webrtcsession.h

 /*
  *  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 123 matching lines @@
   enum Error {
     ERROR_NONE = 0,       // no error
     ERROR_CONTENT = 1,    // channel errors in SetLocalContent/SetRemoteContent
     ERROR_TRANSPORT = 2,  // transport error of some kind
   };
 
   WebRtcSession(webrtc::MediaControllerInterface* media_controller,
                 rtc::Thread* network_thread,
                 rtc::Thread* worker_thread,
                 rtc::Thread* signaling_thread,
-                cricket::PortAllocator* port_allocator);
+                cricket::PortAllocator* port_allocator,
+                cricket::TransportController* transport_controller);
   virtual ~WebRtcSession();
 
   // These are const to allow them to be called from const methods.
   rtc::Thread* worker_thread() const { return worker_thread_; }
   rtc::Thread* signaling_thread() const { return signaling_thread_; }
 
   // The ID of this session.
   const std::string& id() const { return sid_; }
 
   bool Initialize(
@@ skipping 375 matching lines @@
   PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
 
   bool received_first_video_packet_ = false;
   bool received_first_audio_packet_ = false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_WEBRTCSESSION_H_