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 68912907e6f6faa0a081597eb53c5fd8da444a8e..269bcdfdded7511b6219aa51db40dbcf603f5b36 100644 |
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc |
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc |
@@ -164,19 +164,19 @@ 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_SPARSE("WebRTC.BWE.InitiallyLostPackets", |
- initially_lost_packets_, 0, 100, 50); |
- RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), |
- 0, 2000, 50); |
- RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialBandwidthEstimate", |
- bitrate_at_2_seconds_kbps_, 0, 2000, 50); |
+ 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); |
} 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_SPARSE("WebRTC.BWE.InitialVsConvergedDiff", |
- bitrate_diff_kbps, 0, 2000, 50); |
+ RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, |
+ 0, 2000, 50); |
} |
} |