- From: Varun Singh <varun@callstats.io>
- Date: Wed, 14 Jun 2017 12:39:08 -0700
- To: WebRTC WG <public-webrtc@w3.org>
Hi all, A new dated version of the Editors' draft is available. Dated version: https://w3c.github.io/webrtc-stats/archives/20170614/webrtc-stats.html Living document: https://w3c.github.io/webrtc-stats/ Changes include: [#184] Stat for how much time it takes to encode video [#185] Added sample counters to RTCMediaStreamTrackStats [#187] Metrics to keep track of audio and video sync added to RTCMediaStreamTrackStats members [#191] Refactor out isRemote [#192] Update example to match webrtc spec's + senders.getStats [#194] Adding RTCRTPContributingSourceStats stats report object [#195] Adding "codec type" and transportId to RTCCodecStats [#196] Removed writable/readable, and added consentExpiredTimestamp to RTCIceCandidatePairStats [#197] packetsSent/Received added to RTCTransportStats [#198] packetsSent/Received added to RTCIceCandidatePairStats [#199] framesCaptured added to RTCMediaStreamTrackStats [#203] Packets discarded on send [#208] voiceActivityFlag added to RTCMediaStreamTrackStats [#209] definition for 'framesPerSecond' [#210] Added paragraph about timestamp and reference DOMHighResTimeStamp [#211] timestamps needed for circuit breakers and a/v sync [#212] added priority to RTCMediaStreamTrackStats [#213] added last[Request/Response]Timestamp and firstRequestTimestamp to RTCIceCandidatePairStats [#216] stats for number of circuit breaker triggered [#217] added jitter buffer delay added to RTCMediaStreamTrackStats [#218] added iceRole to RTCTransportStats [#221] Added audio level stat that can be used to compute averages Please review and provide feedback. Varun (for the editors) -- http://www.callstats.io
Received on Wednesday, 14 June 2017 19:40:02 UTC