rtcp::Rrtr block moved into own file and got Parse function

webrtc/modules/rtp_rtcp/source/rtcp_sender.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 746 matching lines @@
   if (last_xr_rr_.size() >= RTCP_NUMBER_OF_SR)
     last_xr_rr_.erase(last_xr_rr_.begin());
   last_xr_rr_.insert(std::pair<uint32_t, int64_t>(
       RTCPUtility::MidNtp(ctx.ntp_sec_, ctx.ntp_frac_),
       Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_)));
 
   rtcp::Xr* xr = new rtcp::Xr();
   xr->From(ssrc_);
 
   rtcp::Rrtr rrtr;
-  rrtr.WithNtpSec(ctx.ntp_sec_);
-  rrtr.WithNtpFrac(ctx.ntp_frac_);
+  rrtr.WithNtp(NtpTime(ctx.ntp_sec_, ctx.ntp_frac_));
 
   xr->WithRrtr(&rrtr);
 
   // TODO(sprang): Merge XR report sending to contain all of RRTR, DLRR, VOIP?
 
   return rtc::scoped_ptr<rtcp::Xr>(xr);
 }
 
 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildDlrr(
     const RtcpContext& ctx) {
@@ skipping 317 matching lines @@
     Transport* const transport_;
     bool send_failure_;
   } sender(transport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc