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New webrtc-stats Editor's draft (v20170614)

From: Varun Singh <varun@callstats.io>
Date: Wed, 14 Jun 2017 12:39:08 -0700
Message-ID: <CACHXSv7KGut2g4_M0D92Di002428vQ5OH_yPusUQr868Smiseg@mail.gmail.com>
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
[#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)

Received on Wednesday, 14 June 2017 19:40:02 UTC

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