Log to RtcEventLog when loss based estimate is changed.

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h

 /*
  *  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.
  *
  *  FEC and NACK added bitrate is handled outside class
@@ skipping 42 matching lines @@
   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
   int GetMinBitrate() const;
 
  private:
   enum UmaState { kNoUpdate, kFirstDone, kDone };
 
   bool IsInStartPhase(int64_t now_ms) const;
 
   void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
 
-  // Returns the input bitrate capped to the thresholds defined by the max,
-  // min and incoming bandwidth.
-  uint32_t CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate);
-
   // Updates history of min bitrates.
   // After this method returns min_bitrate_history_.front().second contains the
   // min bitrate used during last kBweIncreaseIntervalMs.
   void UpdateMinHistory(int64_t now_ms);
 
+  // Cap |bitrate_bps| to [min_bitrate_configured_, max_bitrate_configured_] and
+  // set |current_bitrate_bps_| to the capped value and updates the event log.
+  void CapBitrateToThresholds(int64_t now_ms, uint32_t bitrate_bps);
+
   std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
 
   // incoming filters
   int lost_packets_since_last_loss_update_Q8_;
   int expected_packets_since_last_loss_update_;
 
-  uint32_t bitrate_;
+  uint32_t current_bitrate_bps_;
   uint32_t min_bitrate_configured_;
   uint32_t max_bitrate_configured_;
   int64_t last_low_bitrate_log_ms_;
 
   bool has_decreased_since_last_fraction_loss_;
   int64_t last_feedback_ms_;
   int64_t last_packet_report_ms_;
   int64_t last_timeout_ms_;
   uint8_t last_fraction_loss_;
   uint8_t last_logged_fraction_loss_;
   int64_t last_round_trip_time_ms_;
 
   uint32_t bwe_incoming_;
   uint32_t delay_based_bitrate_bps_;
   int64_t time_last_decrease_ms_;
   int64_t first_report_time_ms_;
   int initially_lost_packets_;
   int bitrate_at_2_seconds_kbps_;
   UmaState uma_update_state_;
   std::vector<bool> rampup_uma_stats_updated_;
   RtcEventLog* event_log_;
   int64_t last_rtc_event_log_ms_;
   bool in_timeout_experiment_;
   float low_loss_threshold_;
   float high_loss_threshold_;
   uint32_t bitrate_threshold_bps_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_