Enabling rtcp-rsize negotiation and fixing some issues with it.

webrtc/modules/rtp_rtcp/source/rtcp_receiver.h

 /*
  *  Copyright (c) 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 20 matching lines @@
 public:
  RTCPReceiver(Clock* clock,
               bool receiver_only,
               RtcpPacketTypeCounterObserver* packet_type_counter_observer,
               RtcpBandwidthObserver* rtcp_bandwidth_observer,
               RtcpIntraFrameObserver* rtcp_intra_frame_observer,
               TransportFeedbackObserver* transport_feedback_observer,
               ModuleRtpRtcpImpl* owner);
     virtual ~RTCPReceiver();
 
-    RtcpMode Status() const;
-    void SetRTCPStatus(RtcpMode method);
-
     int64_t LastReceived();
     int64_t LastReceivedReceiverReport() const;
 
     void SetSsrcs(uint32_t main_ssrc,
                   const std::set<uint32_t>& registered_ssrcs);
     void SetRelaySSRC(uint32_t ssrc);
     void SetRemoteSSRC(uint32_t ssrc);
     uint32_t RemoteSSRC() const;
 
     uint32_t RelaySSRC() const;
@@ skipping 263 matching lines @@
   RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
   RtcpPacketTypeCounter packet_type_counter_;
 
   RTCPUtility::NackStats nack_stats_;
 
   size_t num_skipped_packets_;
   int64_t last_skipped_packets_warning_;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_