Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  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 594 matching lines @@
   if (stats_source) {
     loss_stats->single_packet_loss_count =
         stats_source->GetSingleLossCount();
     loss_stats->multiple_packet_loss_event_count =
         stats_source->GetMultipleLossEventCount();
     loss_stats->multiple_packet_loss_packet_count =
         stats_source->GetMultipleLossPacketCount();
   }
 }
 
-int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) {
-  return rtcp_receiver_.SenderInfoReceived(sender_info);
-}
-
 // Received RTCP report.
 int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
     std::vector<RTCPReportBlock>* receive_blocks) const {
   return rtcp_receiver_.StatisticsReceived(receive_blocks);
 }
 
 // (REMB) Receiver Estimated Max Bitrate.
 bool ModuleRtpRtcpImpl::REMB() const {
   return rtcp_sender_.REMB();
 }
@@ skipping 292 matching lines @@
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc