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Side by Side Diff: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

Issue 1564923008: Integrate helper macros for calling histograms with different names (real-time vs screenshare and r… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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148 UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8); 148 UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
149 } 149 }
150 150
151 void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms, 151 void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
152 int64_t rtt, 152 int64_t rtt,
153 int lost_packets) { 153 int lost_packets) {
154 int bitrate_kbps = static_cast<int>((bitrate_ + 500) / 1000); 154 int bitrate_kbps = static_cast<int>((bitrate_ + 500) / 1000);
155 for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) { 155 for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
156 if (!rampup_uma_stats_updated_[i] && 156 if (!rampup_uma_stats_updated_[i] &&
157 bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) { 157 bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
158 RTC_HISTOGRAM_COUNTS_SPARSE_100000(kUmaRampupMetrics[i].metric_name, 158 RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
159 now_ms - first_report_time_ms_); 159 now_ms - first_report_time_ms_);
160 rampup_uma_stats_updated_[i] = true; 160 rampup_uma_stats_updated_[i] = true;
161 } 161 }
162 } 162 }
163 if (IsInStartPhase(now_ms)) { 163 if (IsInStartPhase(now_ms)) {
164 initially_lost_packets_ += lost_packets; 164 initially_lost_packets_ += lost_packets;
165 } else if (uma_update_state_ == kNoUpdate) { 165 } else if (uma_update_state_ == kNoUpdate) {
166 uma_update_state_ = kFirstDone; 166 uma_update_state_ = kFirstDone;
167 bitrate_at_2_seconds_kbps_ = bitrate_kbps; 167 bitrate_at_2_seconds_kbps_ = bitrate_kbps;
168 RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets", 168 RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
169 initially_lost_packets_, 0, 100, 50); 169 initially_lost_packets_, 0, 100, 50);
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299 bitrate = min_bitrate_configured_; 299 bitrate = min_bitrate_configured_;
300 } 300 }
301 return bitrate; 301 return bitrate;
302 } 302 }
303 303
304 void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) { 304 void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
305 event_log_ = event_log; 305 event_log_ = event_log;
306 } 306 }
307 307
308 } // namespace webrtc 308 } // namespace webrtc
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