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Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.

Browse Source 
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2014-11-04 19:32:10 +00:00
parent 8944c9d08b
commit db26247a9b
2 changed files with 31 additions and 7 deletions
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31
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
View File

@ -22,6 +22,7 @@ enum { kBweDecreaseIntervalMs = 300 };
enum { kLimitNumPackets = 20 };
enum { kAvgPacketSizeBytes = 1000 };
enum { kStartPhaseMs = 2000 };
enum { kBweConverganceTimeMs = 20000 };
// Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
// The formula in RFC 3448, Section 3.1, is used.
@ -61,7 +62,8 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation()
time_last_decrease_ms_(0),
first_report_time_ms_(-1),
initially_lost_packets_(0),
uma_updated_(false) {
bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate) {
}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@ -130,18 +132,35 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
if (first_report_time_ms_ == -1) {
first_report_time_ms_ = now_ms;
} else if (IsInStartPhase(now_ms)) {
initially_lost_packets_ += (fraction_loss * number_of_packets) >> 8;
} else if (!uma_updated_) {
uma_updated_ = true;
} else {
UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
}
}
void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
int rtt,
int lost_packets) {
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;
RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", rtt, 0, 2000, 50);
RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
(bitrate_ + 500) / 1000,
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_ - static_cast<int>((bitrate_ + 500) / 1000),
0);
RTC_HISTOGRAM_COUNTS(
"WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
}
}

7
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
View File

@ -43,8 +43,12 @@ class SendSideBandwidthEstimation {
void SetMinBitrate(uint32_t min_bitrate);
private:
enum UmaState { kNoUpdate, kFirstDone, kDone };
bool IsInStartPhase(int64_t now_ms) const;
void UpdateUmaStats(int64_t now_ms, int rtt, int lost_packets);
// Returns the input bitrate capped to the thresholds defined by the max,
// min and incoming bandwidth.
uint32_t CapBitrateToThresholds(uint32_t bitrate);
@ -72,7 +76,8 @@ class SendSideBandwidthEstimation {
uint32_t time_last_decrease_ms_;
int64_t first_report_time_ms_;
int initially_lost_packets_;
bool uma_updated_;
int bitrate_at_2_seconds_kbps_;
UmaState uma_update_state_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_