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This CL adds calculation and logging of average excess buffer delay and number of delayed packet outage events per minute. The first is the average of time spent in the packet buffer for all packets. The average is calculated for intervals of one minute, and the result is logged to the UMA stat WebRTC.Audio.AverageExcessBufferDelayMs. The second is a counter of delayed packet outage events that is restarted every minute, and the result is logged to the UMA stat WebRTC.Audio.DelayedPacketOutageEventsPerMinute. For a description of delayed packet outages, see previous CL implementing a duration log for these events. BUG=webrtc:4915, chromium:488124 R=minyue@webrtc.org Review URL: https://codereview.webrtc.org/1287333005 . Cr-Commit-Position: refs/heads/master@{#9731}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.