Always call RemoteBitrateEstimator::IncomingPacket from Call.

webrtc/video/rtp_stream_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 320 matching lines @@
     if (!receiving_) {
       return false;
     }
   }
 
   RTPHeader header;
   if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length,
                                  &header)) {
     return false;
   }
-  size_t payload_length = rtp_packet_length - header.headerLength;
   int64_t arrival_time_ms;
   int64_t now_ms = clock_->TimeInMilliseconds();
   if (packet_time.timestamp != -1)
     arrival_time_ms = (packet_time.timestamp + 500) / 1000;
   else
     arrival_time_ms = now_ms;
 
   {
     // Periodically log the RTP header of incoming packets.
     rtc::CritScope lock(&receive_cs_);
     if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
       std::stringstream ss;
       ss << "Packet received on SSRC: " << header.ssrc << " with payload type: "
          << static_cast<int>(header.payloadType) << ", timestamp: "
          << header.timestamp << ", sequence number: " << header.sequenceNumber
          << ", 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);
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
   rtp_payload_registry_.SetIncomingPayloadType(header);
   bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
   // 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).
   rtp_receive_statistics_->IncomingPacket(
       header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
@@ skipping 297 matching lines @@
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc