Only use paced packets for estimating bitrate probes.

webrtc/video_engine/vie_receiver.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 262 matching lines @@
          << ", arrival time: " << arrival_time_ms;
       if (header.extension.hasTransmissionTimeOffset)
         ss << ", toffset: " << header.extension.transmissionTimeOffset;
       if (header.extension.hasAbsoluteSendTime)
         ss << ", abs send time: " << header.extension.absoluteSendTime;
       LOG(LS_INFO) << ss.str();
       last_packet_log_ms_ = now_ms;
     }
   }
 
-  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms,
-                                            payload_length, header);
+  remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
+                                            header, true);
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
   rtp_payload_registry_->SetIncomingPayloadType(header);
   int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order)
       ? 0
       : -1;
   // Update receive statistics after ReceivePacket.
   // Receive statistics will be reset if the payload type changes (make sure
   // that the first packet is included in the stats).
@@ skipping 164 matching lines @@
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc