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

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The previous CL was reverted for two reasons:
- Added a static initializer because std::string.
- Landed before the corresponding chromium CL, which has now been landed.

BUG=crbug:425925
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8094 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2015-01-19 15:44:47 +00:00
parent f9d3555ec3
commit 474e36e623
2 changed files with 27 additions and 5 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

31
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
View File

@ -25,6 +25,18 @@ enum { kAvgPacketSizeBytes = 1000 };
enum { kStartPhaseMs = 2000 };
enum { kBweConverganceTimeMs = 20000 };
struct UmaRampUpMetric {
const char* 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) {
@ -64,7 +76,8 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
first_report_time_ms_(-1),
initially_lost_packets_(0),
bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate) {
uma_update_state_(kNoUpdate),
rampup_uma_stats_updated_(kNumUmaRampupMetrics, false) {
}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@ -139,11 +152,20 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
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_ + 500) / 1000;
bitrate_at_2_seconds_kbps_ = bitrate_kbps;
RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
RTC_HISTOGRAM_COUNTS(
@ -156,9 +178,8 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
} 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_ - static_cast<int>((bitrate_ + 500) / 1000),
0);
int bitrate_diff_kbps =
std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
}

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

@ -81,6 +81,7 @@ class SendSideBandwidthEstimation {
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_