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Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call.

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

Review URL: https://webrtc-codereview.appspot.com/31899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7593 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2014-11-03 14:42:43 +00:00
parent 96dc685143
commit 548b228c91
2 changed files with 31 additions and 4 deletions
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4
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
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@ -43,6 +43,8 @@ class SendSideBandwidthEstimation {
void SetMinBitrate(uint32_t min_bitrate);
private:
bool IsInStartPhase(int64_t now_ms) const;
// Returns the input bitrate capped to the thresholds defined by the max,
// min and incoming bandwidth.
uint32_t CapBitrateToThresholds(uint32_t bitrate);
@ -69,6 +71,8 @@ class SendSideBandwidthEstimation {
uint32_t bwe_incoming_;
uint32_t time_last_decrease_ms_;
int64_t first_report_time_ms_;
int initially_lost_packets_;
bool uma_updated_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_