Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).

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 137 matching lines @@
   UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
 }
 
 void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
                                                  int64_t rtt,
                                                  int lost_packets) {
   int bitrate_kbps = static_cast<int>((bitrate_ + 500) / 1000);
   for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
     if (!rampup_uma_stats_updated_[i] &&
         bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
-      RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
-                                   now_ms - first_report_time_ms_);
+      RTC_LOGGED_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
+                                          now_ms - first_report_time_ms_);
       rampup_uma_stats_updated_[i] = true;
     }
   }
   if (IsInStartPhase(now_ms)) {
     initially_lost_packets_ += lost_packets;
   } else if (uma_update_state_ == kNoUpdate) {
     uma_update_state_ = kFirstDone;
     bitrate_at_2_seconds_kbps_ = bitrate_kbps;
-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
-                         initially_lost_packets_, 0, 100, 50);
-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0,
-                         2000, 50);
-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
-                         bitrate_at_2_seconds_kbps_, 0, 2000, 50);
+    RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
+                                initially_lost_packets_, 0, 100, 50);
+    RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt),
+                                0, 2000, 50);
+    RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
+                                bitrate_at_2_seconds_kbps_, 0, 2000, 50);
   } else if (uma_update_state_ == kFirstDone &&
              now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
     uma_update_state_ = kDone;
     int bitrate_diff_kbps =
         std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
-                         0, 2000, 50);
+    RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff",
+                                bitrate_diff_kbps, 0, 2000, 50);
   }
 }
 
 void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
   // We trust the REMB and/or delay-based estimate during the first 2 seconds if
   // we haven't had any packet loss reported, to allow startup bitrate probing.
   if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms)) {
     uint32_t prev_bitrate = bitrate_;
     if (bwe_incoming_ > bitrate_)
       bitrate_ = CapBitrateToThresholds(now_ms, bwe_incoming_);
@@ skipping 108 matching lines @@
     bitrate = min_bitrate_configured_;
   }
   return bitrate;
 }
 
 void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
   event_log_ = event_log;
 }
 
 }  // namespace webrtc