Wire up send-side bandwidth estimation.

webrtc/modules/rtp_rtcp/source/rtcp_utility.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_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 
 #include <assert.h>
 #include <math.h>   // ceil
 #include <string.h> // memcpy
 
 namespace webrtc {
 
 namespace RTCPUtility {
 
 NackStats::NackStats()
@@ skipping 47 matching lines @@
 {
     return _packetType;
 }
 
 const RTCPUtility::RTCPPacket&
 RTCPUtility::RTCPParserV2::Packet() const
 {
     return _packet;
 }
 
+rtcp::RtcpPacket* RTCPUtility::RTCPParserV2::ReleaseRtcpPacket() {
+  return rtcp_packet_.release();
+}
 RTCPUtility::RTCPPacketTypes
 RTCPUtility::RTCPParserV2::Begin()
 {
     _ptrRTCPData = _ptrRTCPDataBegin;
 
     return Iterate();
 }
 
 RTCPUtility::RTCPPacketTypes
 RTCPUtility::RTCPParserV2::Iterate()
@@ skipping 1118 matching lines @@
 
             return true;
         }
         case 5:
          {
             // RTCP-SR-REQ Rapid Synchronisation of RTP Flows
             // draft-perkins-avt-rapid-rtp-sync-03.txt
             // trigger a new RTCP SR
           _packetType = RTCPPacketTypes::kRtpfbSrReq;
 
             // Note: No state transition, SR REQ is empty!

[stefan-webrtc, 2015/09/17 10:33:19]

May want to add a comment similar to this below as well.

[sprang_webrtc, 2015/09/17 15:08:28]

Done.

             return true;
         }
+        case 15: {
+          _packetType = RTCPPacketTypes::kTransportFeedback;
+          rtcp_packet_ =
+              rtcp::TransportFeedback::ParseFrom(_ptrRTCPData - 12, length);
+          if (rtcp_packet_.get() == nullptr) {
+            EndCurrentBlock();
+            return false;
+          }
+
+          _ptrRTCPData += rtcp_packet_->BlockLength();
+          return true;
+        }
         default:
             break;
         }
         EndCurrentBlock();
         return false;
     }
     else if (header.PT == PT_PSFB)
     {
         // Payload specific feedback
         switch (header.IC)
@@ skipping 453 matching lines @@
 RTCPUtility::RTCPPacketIterator::Current()
 {
     if (!_ptrBlock)
     {
         return NULL;
     }
 
     return &_header;
 }
 }  // namespace webrtc