Only log BWE updates if the actual estimate changed or if we have non-zero loss reports.

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 11 matching lines @@
 namespace webrtc {
 namespace {
 const int64_t kBweIncreaseIntervalMs = 1000;
 const int64_t kBweDecreaseIntervalMs = 300;
 const int64_t kStartPhaseMs = 2000;
 const int64_t kBweConverganceTimeMs = 20000;
 const int kLimitNumPackets = 20;
 const int kDefaultMinBitrateBps = 10000;
 const int kDefaultMaxBitrateBps = 1000000000;
 const int64_t kLowBitrateLogPeriodMs = 10000;
+const int64_t kRtcEventLogPeriodMs = 5000;
 
 struct UmaRampUpMetric {
   const char* metric_name;
   int bitrate_kbps;
 };
 
 const UmaRampUpMetric kUmaRampupMetrics[] = {
     {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
     {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
     {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
@@ skipping 14 matching lines @@
       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_(event_log) {
+      event_log_(event_log),
+      last_rtc_event_log_ms_(-1) {
   RTC_DCHECK(event_log);
 }
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
 void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
                                               int min_bitrate,
                                               int max_bitrate) {
   if (send_bitrate > 0)
     SetSendBitrate(send_bitrate);
@@ skipping 144 matching lines @@
       //   whenever a receiver report is received with lower packet loss.
       //   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;
-
-      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_ &&
           (now_ms - time_last_decrease_ms_) >=
               (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {
         time_last_decrease_ms_ = now_ms;
 
         // Reduce rate:
         //   newRate = rate * (1 - 0.5*lossRate);
         //   where packetLoss = 256*lossRate;
         bitrate_ = static_cast<uint32_t>(
             (bitrate_ * static_cast<double>(512 - last_fraction_loss_)) /
             512.0);
         has_decreased_since_last_fraction_loss_ = true;
       }
-      event_log_->LogBwePacketLossEvent(
-          bitrate_, last_fraction_loss_,
-          expected_packets_since_last_loss_update_);
     }
   }
-  bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
+  uint32_t capped_bitrate = CapBitrateToThresholds(now_ms, bitrate_);
+  if (capped_bitrate != bitrate_ || last_fraction_loss_ > 0 ||

[philipel, 2016/09/02 11:26:07]

Maybe add the same type of logic to fraction loss, i.e. only log in the case of
a change in packet loss (line 151).

[stefan-webrtc, 2016/09/02 13:13:17]

Done.

+      last_rtc_event_log_ms_ == -1 ||
+      now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
+    event_log_->LogBwePacketLossEvent(capped_bitrate, last_fraction_loss_,
+                                      expected_packets_since_last_loss_update_);
+    last_rtc_event_log_ms_ = now_ms;
+  }
+  bitrate_ = capped_bitrate;
 }
 
 bool SendSideBandwidthEstimation::IsInStartPhase(int64_t now_ms) const {
   return first_report_time_ms_ == -1 ||
          now_ms - first_report_time_ms_ < kStartPhaseMs;
 }
 
 void SendSideBandwidthEstimation::UpdateMinHistory(int64_t now_ms) {
   // Remove old data points from history.
   // Since history precision is in ms, add one so it is able to increase
@@ skipping 31 matching lines @@
       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;
 }
 }  // namespace webrtc