Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop f…

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
  */
 
 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
 #define WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
 
 #include <deque>
 #include <utility>
 #include <vector>
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 namespace webrtc {
 
 class RtcEventLog;
 
 class SendSideBandwidthEstimation {
  public:
-  SendSideBandwidthEstimation();
+  SendSideBandwidthEstimation() = delete;
+  explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
   virtual ~SendSideBandwidthEstimation();
 
   void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
 
   // Call periodically to update estimate.
   void UpdateEstimate(int64_t now_ms);
 
   // Call when we receive a RTCP message with TMMBR or REMB.
   void UpdateReceiverEstimate(int64_t now_ms, uint32_t bandwidth);
 
   // Call when a new delay-based estimate is available.
   void UpdateDelayBasedEstimate(int64_t now_ms, uint32_t bitrate_bps);
 
   // Call when we receive a RTCP message with a ReceiveBlock.
   void UpdateReceiverBlock(uint8_t fraction_loss,
                            int64_t rtt,
                            int number_of_packets,
                            int64_t now_ms);
 
   void SetBitrates(int send_bitrate,
                    int min_bitrate,
                    int max_bitrate);
   void SetSendBitrate(int bitrate);
   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
   int GetMinBitrate() const;
 
-  void SetEventLog(RtcEventLog* event_log);
-
  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);
@@ skipping 24 matching lines @@
   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_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_