Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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 112 matching lines @@
 
 enum RtxMode {
   kRtxOff                 = 0x0,
   kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
   kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
                                   // instead of padding.
 };
 
 const size_t kRtxHeaderSize = 2;
 
-struct RTCPSenderInfo {
-    uint32_t NTPseconds;
-    uint32_t NTPfraction;
-    uint32_t RTPtimeStamp;
-    uint32_t sendPacketCount;
-    uint32_t sendOctetCount;
-};
-
 struct RTCPReportBlock {
   RTCPReportBlock()
       : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
         extendedHighSeqNum(0), jitter(0), lastSR(0),
         delaySinceLastSR(0) {}
 
   RTCPReportBlock(uint32_t remote_ssrc,
                   uint32_t source_ssrc,
                   uint8_t fraction_lost,
                   uint32_t cumulative_lost,
@@ skipping 306 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_