[rtp_rtcp] rtcp::Xr moved into own file and renamed to ExtendedReports on the way

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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 
 #include <assert.h>  // assert
 #include <string.h>  // memcpy
 
 #include <algorithm>  // min
 #include <limits>     // max
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
@@ skipping 695 matching lines @@
 }
 
 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildReceiverReferenceTime(
     const RtcpContext& ctx) {
   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();
+  rtcp::ExtendedReports* xr = new rtcp::ExtendedReports();
   xr->From(ssrc_);
 
   rtcp::Rrtr rrtr;
   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);
+  return rtc::scoped_ptr<rtcp::RtcpPacket>(xr);
 }
 
 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildDlrr(
     const RtcpContext& ctx) {
-  rtcp::Xr* xr = new rtcp::Xr();
+  rtcp::ExtendedReports* xr = new rtcp::ExtendedReports();
   xr->From(ssrc_);
 
   rtcp::Dlrr dlrr;
   const RtcpReceiveTimeInfo& info = ctx.feedback_state_.last_xr_rr;
   dlrr.WithDlrrItem(info.sourceSSRC, info.lastRR, info.delaySinceLastRR);
 
   xr->WithDlrr(&dlrr);
 
-  return rtc::scoped_ptr<rtcp::Xr>(xr);
+  return rtc::scoped_ptr<rtcp::RtcpPacket>(xr);
 }
 
 // TODO(sprang): Add a unit test for this, or remove if the code isn't used.
 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildVoIPMetric(
     const RtcpContext& context) {
-  rtcp::Xr* xr = new rtcp::Xr();
+  rtcp::ExtendedReports* xr = new rtcp::ExtendedReports();
   xr->From(ssrc_);
 
   rtcp::VoipMetric voip;
   voip.To(remote_ssrc_);
   voip.WithVoipMetric(xr_voip_metric_);
 
   xr->WithVoipMetric(&voip);
 
-  return rtc::scoped_ptr<rtcp::Xr>(xr);
+  return rtc::scoped_ptr<rtcp::RtcpPacket>(xr);
 }
 
 int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
                              RTCPPacketType packetType,
                              int32_t nack_size,
                              const uint16_t* nack_list,
                              bool repeat,
                              uint64_t pictureID) {
   return SendCompoundRTCP(
       feedback_state, std::set<RTCPPacketType>(&packetType, &packetType + 1),
@@ skipping 260 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