webrtc::RtpPacket name freed for better RtpPacket

webrtc/modules/rtp_rtcp/source/producer_fec.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 14 matching lines @@
 // level) in order to trigger GenerateFEC(), before |params_.max_fec_frames| is
 // reached.
 enum { kMinimumMediaPackets = 4 };
 // Threshold on the received FEC protection level, above which we enforce at
 // least |kMinimumMediaPackets| packets for the FEC code. Below this
 // threshold |kMinimumMediaPackets| is set to default value of 1.
 enum { kHighProtectionThreshold = 80 };  // Corresponds to ~30 overhead, range
 // is 0 to 255, where 255 corresponds to 100% overhead (relative to number of
 // media packets).
 
-struct RtpPacket {
-  uint16_t rtpHeaderLength;
-  ForwardErrorCorrection::Packet* pkt;
-};
-
 RedPacket::RedPacket(size_t length)
     : data_(new uint8_t[length]),
       length_(length),
       header_length_(0) {
 }
 
 RedPacket::~RedPacket() {
   delete [] data_;
 }
 
@@ skipping 207 matching lines @@
 
 void ProducerFec::DeletePackets() {
   while (!media_packets_fec_.empty()) {
     delete media_packets_fec_.front();
     media_packets_fec_.pop_front();
   }
   assert(media_packets_fec_.empty());
 }
 
 }  // namespace webrtc