Remove deprecated functions in rtp_rtcp module

webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc

 /*
  *  Copyright (c) 2013 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 212 matching lines @@
 
 bool RTPPayloadRegistry::IsRtx(const RTPHeader& header) const {
   rtc::CritScope cs(&crit_sect_);
   return IsRtxInternal(header);
 }
 
 bool RTPPayloadRegistry::IsRtxInternal(const RTPHeader& header) const {
   return rtx_ && ssrc_rtx_ == header.ssrc;
 }
 
-bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t** restored_packet,
-                                               const uint8_t* packet,
-                                               size_t* packet_length,
-                                               uint32_t original_ssrc,
-                                               const RTPHeader& header) const {
-  return RestoreOriginalPacket(*restored_packet, packet, packet_length,
-                               original_ssrc, header);
-}
-
 bool RTPPayloadRegistry::RestoreOriginalPacket(uint8_t* restored_packet,
                                                const uint8_t* packet,
                                                size_t* packet_length,
                                                uint32_t original_ssrc,
                                                const RTPHeader& header) const {
   if (kRtxHeaderSize + header.headerLength + header.paddingLength >
       *packet_length) {
     return false;
   }
   const uint8_t* rtx_header = packet + header.headerLength;
@@ skipping 221 matching lines @@
 RTPPayloadStrategy* RTPPayloadStrategy::CreateStrategy(
     const bool handling_audio) {
   if (handling_audio) {
     return new RTPPayloadAudioStrategy();
   } else {
     return new RTPPayloadVideoStrategy();
   }
 }
 
 }  // namespace webrtc