webrtc::RtpPacket name freed for better RtpPacket

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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 11 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 enum { REDForFECHeaderLength = 1 };
 
-struct RtpPacket {
-  uint16_t rtpHeaderLength;
-  ForwardErrorCorrection::Packet* pkt;
-};
-
 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender)
     : _rtpSender(*rtpSender),
       crit_(CriticalSectionWrapper::CreateCriticalSection()),
       _videoType(kRtpVideoGeneric),
       _retransmissionSettings(kRetransmitBaseLayer),
       // Generic FEC
       fec_(),
       fec_enabled_(false),
       red_payload_type_(-1),
       fec_payload_type_(-1),
@@ skipping 291 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   return _retransmissionSettings;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   CriticalSectionScoped cs(crit_.get());
   _retransmissionSettings = settings;
 }
 
 }  // namespace webrtc