Remove static locals from histograms.

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 16 matching lines @@
 const int64_t kStartPhaseMs = 2000;
 const int64_t kBweConverganceTimeMs = 20000;
 const int kLimitNumPackets = 20;
 const int kDefaultMaxBitrateBps = 1000000000;
 const int64_t kLowBitrateLogPeriodMs = 10000;
 const int64_t kRtcEventLogPeriodMs = 5000;
 // Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
 const int64_t kFeedbackIntervalMs = 1500;
 const int64_t kFeedbackTimeoutIntervals = 3;
 const int64_t kTimeoutIntervalMs = 1000;
-
-struct UmaRampUpMetric {
-  const char* metric_name;
-  int bitrate_kbps;
-};
-
-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(RtcEventLog* event_log)
-    : lost_packets_since_last_loss_update_Q8_(0),
+    : rampup_histograms_{
+          {rtc::Histogram("WebRTC.BWE.RampUpTimeTo500kbpsInMs", 100000), 500},
+          {rtc::Histogram("WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 100000), 1000},
+          {rtc::Histogram("WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 100000), 2000},
+      },
+      initially_lost_packets_histogram_(
+          rtc::Histogram("WebRTC.BWE.InitiallyLostPackets", 0, 100)),
+      initial_rtt_histogram_(rtc::Histogram("WebRTC.BWE.InitialRtt", 0, 2000)),
+      initial_bwe_histogram_(
+          rtc::Histogram("WebRTC.BWE.InitialBandwidthEstimate", 0, 2000)),
+      initial_vs_converged_diff_histogram_(
+          rtc::Histogram("WebRTC.BWE.InitialVsConvergedDiff", 0, 2000)),
+      lost_packets_since_last_loss_update_Q8_(0),
       expected_packets_since_last_loss_update_(0),
       bitrate_(0),
       min_bitrate_configured_(congestion_controller::GetMinBitrateBps()),
       max_bitrate_configured_(kDefaultMaxBitrateBps),
       last_low_bitrate_log_ms_(-1),
       has_decreased_since_last_fraction_loss_(false),
       last_feedback_ms_(-1),
       last_packet_report_ms_(-1),
       last_timeout_ms_(-1),
       last_fraction_loss_(0),
       last_logged_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),
+      rampup_uma_stats_updated_(
+          sizeof(rampup_histograms_) / sizeof(*rampup_histograms_),
+          false),
       event_log_(event_log),
       last_rtc_event_log_ms_(-1),
       in_timeout_experiment_(webrtc::field_trial::FindFullName(
                                  "WebRTC-FeedbackTimeout") == "Enabled") {
   RTC_DCHECK(event_log);
 }
 
 SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
 
 void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
@@ skipping 84 matching lines @@
     last_packet_report_ms_ = now_ms;
     UpdateEstimate(now_ms);
   }
   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) {
+  for (size_t i = 0; i < rampup_uma_stats_updated_.size(); ++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_);
+        bitrate_kbps >= rampup_histograms_[i].bitrate_kbps) {
+      rampup_histograms_[i].histogram.AddSample(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);
+    initially_lost_packets_histogram_.AddSample(initially_lost_packets_);
+    initial_rtt_histogram_.AddSample(static_cast<int>(rtt));
+    initial_bwe_histogram_.AddSample(bitrate_at_2_seconds_kbps_);
   } 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);
+    initial_vs_converged_diff_histogram_.AddSample(bitrate_diff_kbps);
   }
 }
 
 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 122 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