Reland "Wire up send-side bandwidth estimation."

webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.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_receiver_help.h"
 
 #include <assert.h>  // assert
 #include <string.h>  // memset
 
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 using namespace RTCPHelp;
 
 RTCPPacketInformation::RTCPPacketInformation()
     : rtcpPacketTypeFlags(0),
       remoteSSRC(0),
       nackSequenceNumbers(),
       applicationSubType(0),
       applicationName(0),
       applicationData(),
       applicationLength(0),
       rtt(0),
       interArrivalJitter(0),
       sliPictureId(0),
       rpsiPictureId(0),
       receiverEstimatedMaxBitrate(0),
       ntp_secs(0),
       ntp_frac(0),
       rtp_timestamp(0),
       xr_originator_ssrc(0),
       xr_dlrr_item(false),
-      VoIPMetric(NULL) {
-}
+      VoIPMetric(nullptr) {}
 
 RTCPPacketInformation::~RTCPPacketInformation()
 {
     delete [] applicationData;
     delete VoIPMetric;
 }
 
 void
 RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
 {
@@ skipping 133 matching lines @@
                          TmmbrSet.PacketOH(sourceIdx),
                          TmmbrSet.Ssrc(sourceIdx));
   return 0;
 }
 
 void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
     const uint32_t minimumSize) {
   TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
 }
 }  // namespace webrtc