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Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."

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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
This commit is contained in:
stefan@webrtc.org 2015-01-16 13:52:52 +00:00
parent 4ba1e44ff0
commit a1aea10af2
2 changed files with 5 additions and 28 deletions
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32
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
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@ -25,18 +25,6 @@ enum { kAvgPacketSizeBytes = 1000 };
enum { kStartPhaseMs = 2000 }; enum { kStartPhaseMs = 2000 };
enum { kBweConverganceTimeMs = 20000 }; 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. // Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
// The formula in RFC 3448, Section 3.1, is used. // The formula in RFC 3448, Section 3.1, is used.
uint32_t CalcTfrcBps(int64_t rtt, uint8_t loss) { uint32_t CalcTfrcBps(int64_t rtt, uint8_t loss) {
@ -62,7 +50,6 @@ uint32_t CalcTfrcBps(int64_t rtt, uint8_t loss) {
} }
} }
SendSideBandwidthEstimation::SendSideBandwidthEstimation() SendSideBandwidthEstimation::SendSideBandwidthEstimation()
: accumulate_lost_packets_Q8_(0), : accumulate_lost_packets_Q8_(0),
accumulate_expected_packets_(0), accumulate_expected_packets_(0),
@ -77,8 +64,7 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
first_report_time_ms_(-1), first_report_time_ms_(-1),
initially_lost_packets_(0), initially_lost_packets_(0),
bitrate_at_2_seconds_kbps_(0), bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate), uma_update_state_(kNoUpdate) {
rampup_uma_stats_updated_(kNumUmaRampupMetrics, false) {
} }
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {} SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@ -153,20 +139,11 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms, void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
int64_t rtt, int64_t rtt,
int lost_packets) { 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)) { if (IsInStartPhase(now_ms)) {
initially_lost_packets_ += lost_packets; initially_lost_packets_ += lost_packets;
} else if (uma_update_state_ == kNoUpdate) { } else if (uma_update_state_ == kNoUpdate) {
uma_update_state_ = kFirstDone; uma_update_state_ = kFirstDone;
bitrate_at_2_seconds_kbps_ = bitrate_kbps; bitrate_at_2_seconds_kbps_ = (bitrate_ + 500) / 1000;
RTC_HISTOGRAM_COUNTS( RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50); "WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
RTC_HISTOGRAM_COUNTS( RTC_HISTOGRAM_COUNTS(
@ -179,8 +156,9 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
} else if (uma_update_state_ == kFirstDone && } else if (uma_update_state_ == kFirstDone &&
now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) { now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
uma_update_state_ = kDone; uma_update_state_ = kDone;
int bitrate_diff_kbps = int bitrate_diff_kbps = std::max(
std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0); bitrate_at_2_seconds_kbps_ - static_cast<int>((bitrate_ + 500) / 1000),
0);
RTC_HISTOGRAM_COUNTS( RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50); "WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
} }

1
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
View File

@ -81,7 +81,6 @@ class SendSideBandwidthEstimation {
int initially_lost_packets_; int initially_lost_packets_;
int bitrate_at_2_seconds_kbps_; int bitrate_at_2_seconds_kbps_;
UmaState uma_update_state_; UmaState uma_update_state_;
std::vector<bool> rampup_uma_stats_updated_;
}; };
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_ #endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_