Revert of Refactoring of RTCP options in BaseChannel.

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 536 matching lines @@
   // Gather the usage of IPv4/IPv6 as best connection.
   void ReportBestConnectionState(const cricket::TransportStats& stats);
 
   void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
 
   void OnSentPacket_w(const rtc::SentPacket& sent_packet);
 
   const std::string GetTransportName(const std::string& content_name);
 
   void DestroyTransport(const std::string& transport_name, int component);
-  void DestroyRtcpTransport_n(const std::string& transport_name);
+  void OnDestroyRtcpTransport_n(const std::string& transport_name);
   void DestroyVideoChannel();
   void DestroyVoiceChannel();
   void DestroyDataChannel();
 
   rtc::Thread* const network_thread_;
   rtc::Thread* const worker_thread_;
   rtc::Thread* const signaling_thread_;
 
   State state_ = STATE_INIT;
   Error error_ = ERROR_NONE;
@@ skipping 78 matching lines @@
 
 #ifdef HAVE_QUIC
   std::unique_ptr<QuicDataTransport> quic_data_transport_;
 #endif  // HAVE_QUIC
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_WEBRTCSESSION_H_