[rtp_rtcp] Lint errors cleaned from rtp_utility

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include <stdio.h>
@@ skipping 98 matching lines @@
   }
 
   virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
 
   DeliveryStatus DeliverPacket(MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const PacketTime& packet_time) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
-    parser.Parse(header);
+    parser.Parse(&header);
     {
       rtc::CritScope lock(&crit_);
       recv_times_[header.timestamp - rtp_timestamp_delta_] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     }
 
     return receiver_->DeliverPacket(media_type, packet, length, packet_time);
   }
 
   // EncodingTimeObserver.
@@ skipping 15 matching lines @@
     }
 
     input_->IncomingCapturedFrame(video_frame);
   }
 
   bool SendRtp(const uint8_t* packet,
                size_t length,
                const PacketOptions& options) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
-    parser.Parse(header);
+    parser.Parse(&header);
 
     int64_t current_time =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     bool result = transport_->SendRtp(packet, length, options);
     {
       rtc::CritScope lock(&crit_);
       if (rtp_timestamp_delta_ == 0) {
         rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
         first_send_frame_.Reset();
       }
@@ skipping 898 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc