Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8085 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 46fc1c9..a08e123 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -25,18 +25,6 @@
enum { kStartPhaseMs = 2000 };
enum { kBweConverganceTimeMs = 20000 };
-struct UmaRampUpMetric {
- std::string 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]);
-
// Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
// The formula in RFC 3448, Section 3.1, is used.
uint32_t CalcTfrcBps(int64_t rtt, uint8_t loss) {
@@ -62,7 +50,6 @@
}
}
-
SendSideBandwidthEstimation::SendSideBandwidthEstimation()
: accumulate_lost_packets_Q8_(0),
accumulate_expected_packets_(0),
@@ -77,8 +64,7 @@
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) {
+ uma_update_state_(kNoUpdate) {
}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@@ -153,20 +139,11 @@
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) {
- if (!rampup_uma_stats_updated_[i] &&
- bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
- RTC_HISTOGRAM_COUNTS_100000(kUmaRampupMetrics[i].metric_name,
- 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;
+ bitrate_at_2_seconds_kbps_ = (bitrate_ + 500) / 1000;
RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
RTC_HISTOGRAM_COUNTS(
@@ -179,8 +156,9 @@
} 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);
+ int bitrate_diff_kbps = std::max(
+ bitrate_at_2_seconds_kbps_ - static_cast<int>((bitrate_ + 500) / 1000),
+ 0);
RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
}
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 427909b..20ce5ee 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -81,7 +81,6 @@
int initially_lost_packets_;
int bitrate_at_2_seconds_kbps_;
UmaState uma_update_state_;
- std::vector<bool> rampup_uma_stats_updated_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
