Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.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 26 matching lines @@
 
 const UmaRampUpMetric kUmaRampupMetrics[] = {
     {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
     {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
     {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
 const size_t kNumUmaRampupMetrics =
     sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);
 
 }  // namespace
 
-SendSideBandwidthEstimation::SendSideBandwidthEstimation()
+SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
     : lost_packets_since_last_loss_update_Q8_(0),
       expected_packets_since_last_loss_update_(0),
       bitrate_(0),
       min_bitrate_configured_(kDefaultMinBitrateBps),
       max_bitrate_configured_(kDefaultMaxBitrateBps),
       last_low_bitrate_log_ms_(-1),
       has_decreased_since_last_fraction_loss_(false),
       time_last_receiver_block_ms_(-1),
       last_fraction_loss_(0),
       last_round_trip_time_ms_(0),
       bwe_incoming_(0),
       delay_based_bitrate_bps_(0),
       time_last_decrease_ms_(0),
       first_report_time_ms_(-1),
       initially_lost_packets_(0),
       bitrate_at_2_seconds_kbps_(0),
       uma_update_state_(kNoUpdate),
       rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
-      event_log_(nullptr) {}
+      event_log_(event_log) {}
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
 void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
   RTC_DCHECK_GT(bitrate, 0);
   bitrate_ = bitrate;
 
   // Clear last sent bitrate history so the new value can be used directly
   // and not capped.
   min_bitrate_history_.clear();
@@ skipping 135 matching lines @@
       //   If instead one would do: bitrate_ *= 1.08^(delta time), it would
       //   take over one second since the lower packet loss to achieve 108kbps.
       bitrate_ = static_cast<uint32_t>(
           min_bitrate_history_.front().second * 1.08 + 0.5);
 
       // Add 1 kbps extra, just to make sure that we do not get stuck
       // (gives a little extra increase at low rates, negligible at higher
       // rates).
       bitrate_ += 1000;
 
       if (event_log_) {

[stefan-webrtc, 2016/03/11 10:11:53]

I guess we can remove these ifs now that we always set the event log? For tests
we can pass in a mock.

[ivoc, 2016/03/16 17:00:32]

I guess that makes sense, I can add a DCHECK in the constructor

         event_log_->LogBwePacketLossEvent(
             bitrate_, last_fraction_loss_,
             expected_packets_since_last_loss_update_);
       }
     } else if (last_fraction_loss_ <= 26) {
       // Loss between 2% - 10%: Do nothing.
     } else {
       // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs +
       // rtt.
       if (!has_decreased_since_last_fraction_loss_ &&
@@ skipping 60 matching lines @@
         now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) {
       LOG(LS_WARNING) << "Estimated available bandwidth " << bitrate / 1000
                       << " kbps is below configured min bitrate "
                       << min_bitrate_configured_ / 1000 << " kbps.";
       last_low_bitrate_log_ms_ = now_ms;
     }
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
-
-void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
-  event_log_ = event_log;
-}
-
 }  // namespace webrtc