[rtp_rtcp] Lint errors cleaned from rtp_utility

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 286 matching lines @@
           kRtpExtensionVideoRotation));
     } else if (cvo_mode == RTPSenderInterface::kCVOActivated) {
       // Checking whether CVO header extension is registered will require taking
       // a lock. It'll be a no-op if it's not registered.
       // TODO(guoweis): For now, all packets sent will carry the CVO such that
       // the RTP header length is consistent, although the receiver side will
       // only exam the packets with marker bit set.
       size_t packetSize = payloadSize + rtp_header_length;
       RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
       RTPHeader rtp_header;
-      rtp_parser.Parse(rtp_header);
+      rtp_parser.Parse(&rtp_header);
       _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
                                      rtpHdr->rotation);
     }
     if (fec_enabled) {
       SendVideoPacketAsRed(dataBuffer, payload_bytes_in_packet,
                            rtp_header_length, _rtpSender.SequenceNumber(),
                            captureTimeStamp, capture_time_ms, storage,
                            packetizer->GetProtectionType() == kProtectedPacket);
     } else {
       SendVideoPacket(dataBuffer, payload_bytes_in_packet, rtp_header_length,
@@ skipping 32 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   return _retransmissionSettings;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   CriticalSectionScoped cs(crit_.get());
   _retransmissionSettings = settings;
 }
 
 }  // namespace webrtc