Update bitrate only when we have incoming packet.

webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc

 /*
  *  Copyright (c) 2015 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 34 matching lines @@
                                           size_t payload_size,
                                           const RTPHeader& header,
                                           bool was_paced) {
   if (!header.extension.hasTransportSequenceNumber) {
     LOG(LS_WARNING) << "RemoteEstimatorProxy: Incoming packet "
                        "is missing the transport sequence number extension!";
     return;
   }
   rtc::CritScope cs(&lock_);
   media_ssrc_ = header.ssrc;
+
   OnPacketArrival(header.extension.transportSequenceNumber, arrival_time_ms);
 }
 
-void RemoteEstimatorProxy::RemoveStream(unsigned int ssrc) {}
-
 bool RemoteEstimatorProxy::LatestEstimate(std::vector<unsigned int>* ssrcs,
                                           unsigned int* bitrate_bps) const {
   return false;
 }
 
-bool RemoteEstimatorProxy::GetStats(
-    ReceiveBandwidthEstimatorStats* output) const {
-  return false;
-}
-
-
 int64_t RemoteEstimatorProxy::TimeUntilNextProcess() {
   int64_t now = clock_->TimeInMilliseconds();
   int64_t time_until_next = 0;
   if (last_process_time_ms_ != -1 &&
       now - last_process_time_ms_ < kDefaultProcessIntervalMs) {
     time_until_next = (last_process_time_ms_ + kDefaultProcessIntervalMs - now);
   }
   return time_until_next;
 }
 
@@ skipping 30 matching lines @@
          arrival_time - it->second >= kBackWindowMs;) {
       auto delete_it = it;
       ++it;
       packet_arrival_times_.erase(delete_it);
     }
   } else if (seq < window_start_seq_) {
     window_start_seq_ = seq;
   }
 
   RTC_DCHECK(packet_arrival_times_.end() == packet_arrival_times_.find(seq));
+
   packet_arrival_times_[seq] = arrival_time;
 }
 
 bool RemoteEstimatorProxy::BuildFeedbackPacket(
     rtcp::TransportFeedback* feedback_packet) {
   rtc::CritScope cs(&lock_);
   if (window_start_seq_ == -1)
     return false;
 
   // window_start_seq_ is the first sequence number to include in the current
@@ skipping 24 matching lines @@
     // they need to be re-sent after a reordering. Removal will be handled
     // by OnPacketArrival once packets are too old.
   }
   if (it == packet_arrival_times_.end())
     window_start_seq_ = -1;
 
   return true;
 }
 
 }  // namespace webrtc