Remove static locals from histograms.

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

Index: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index ad8a6f9e77140d7a7af703c08df4d746be51c569..0e6cddf8e4711326ca2017b7fea4559b667e2061 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -34,23 +34,22 @@ const int64_t kRtcEventLogPeriodMs = 5000;
 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()),
@@ -70,7 +69,9 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
       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(
@@ -175,11 +176,10 @@ 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;
     }
   }
@@ -188,19 +188,15 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
   } 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);
   }
 }