[rtp_rtcp] Lint errors cleaned from rtp_utility

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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 332 matching lines @@
     }
   }
   {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     _lastPayloadType = payloadType;
   }
   // Update audio level extension, if included.
   size_t packetSize = payloadSize + rtpHeaderLength;
   RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
                                (frameType == kAudioFrameSpeech),
                                audio_level_dbov);
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
                          _rtpSender->Timestamp(), "seqnum",
                          _rtpSender->SequenceNumber());
   return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
                                    TickTime::MillisecondTimestamp(),
                                    kAllowRetransmission,
                                    RtpPacketSender::kHighPriority);
@@ skipping 91 matching lines @@
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
         dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
 
   return retVal;
 }
 }  // namespace webrtc