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 833 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;
   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?
 
   PacketBuiltCallback callback(ctx);
   if (!callback.BuildPacket(xr))
     return BuildResult::kTruncated;
 
   return BuildResult::kSuccess;
@@ skipping 356 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