| 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);
|
| }
|
| }
|
|
|
|
|