Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h

Index: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 15894f939543d265309d109693ece2778db8f17c..0c8c769b500423f2043c8f77850589907b5ce2a9 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -26,7 +26,7 @@ class RtcEventLog;
 
 class SendSideBandwidthEstimation {
  public:
-  SendSideBandwidthEstimation();
+  explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
   virtual ~SendSideBandwidthEstimation();
 
   void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
@@ -50,8 +50,6 @@ class SendSideBandwidthEstimation {
   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
   int GetMinBitrate() const;
 
-  void SetEventLog(RtcEventLog* event_log);
-
  private:
   enum UmaState { kNoUpdate, kFirstDone, kDone };
 
@@ -93,6 +91,7 @@ class SendSideBandwidthEstimation {
   UmaState uma_update_state_;
   std::vector<bool> rampup_uma_stats_updated_;
   RtcEventLog* event_log_;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendSideBandwidthEstimation);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_