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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 12 Dec 2017 17:00:41 +0000
To: public-webrtc@w3.org
Message-Id: <E1eOnv7-0003wN-G3@uranus.w3.org>



Issues
------
* w3c/webrtc-pc (+8/-7/💬36)
  8 issues created:
  - RTCCertificate Interface should (or should not) be backed up. (by steely-glint)
    https://github.com/w3c/webrtc-pc/issues/1694
  - When to fire events triggered by setRemoteDescription. (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1691
  - RTCRtpContributingSource.timestamp needs a clearer definition (by bzbarsky)
    https://github.com/w3c/webrtc-pc/issues/1690
  -  Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? (by bzbarsky)
    https://github.com/w3c/webrtc-pc/issues/1689
  - Why is RTCRtpContributingSource.byte required-but-nullable? (by bzbarsky)
    https://github.com/w3c/webrtc-pc/issues/1688
  - Testing guidelines: Comments or help link? (by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1687
  - offerToReceive: Current text can't handle two options to createOffer (by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1685
  - should offerToReceive* in createOfffer be marked feature-at-risk? (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1682

  15 issues received 36 new comments:
  - #1688 Why is RTCRtpContributingSource.byte required-but-nullable? (6 by jan-ivar, bzbarsky)
    https://github.com/w3c/webrtc-pc/issues/1688
  - #1689  Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? (6 by jan-ivar, bzbarsky)
    https://github.com/w3c/webrtc-pc/issues/1689
  - #1677 replaceTrack and removeTrack: Synchronous? (5 by henbos, jan-ivar, aboba)
    https://github.com/w3c/webrtc-pc/issues/1677
  - #1586 behaviour of offerToReceive* set to false when there is a local track (4 by fippo, jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1586
  - #1690 RTCRtpContributingSource.timestamp needs a clearer definition (4 by jan-ivar, bzbarsky, na-g)
    https://github.com/w3c/webrtc-pc/issues/1690
  - #1662 addTransceiver woes (2 by jan-ivar, stefhak)
    https://github.com/w3c/webrtc-pc/issues/1662
  - #1317 More granular data channel error reporting (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1317
  - #1446 RTCSctpTransport.maxMessageSize 0 case (1 by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1446
  - #1642 OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION (1 by misi)
    https://github.com/w3c/webrtc-pc/issues/1642
  - #1533 Clarify whether RTCRtpContributingSource members are live. (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1533
  - #1680 RTCDataChannel.bufferedAmount description confusing (1 by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1680
  - #1477 Minor inconsistencies around RTCSessionDescription vs. RTCSessionDescriptionInit (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1477
  - #1685 offerToReceive: Current text can't handle two options to createOffer (1 by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1685
  - #1467 RTCTrackEvent's type parameter (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1467
  - #1599 Ordering of stream "addtrack"/"removetrack" events vs. "track" event (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1599

  7 issues closed:
  - offerToReceive: Current text can't handle two options to createOffer https://github.com/w3c/webrtc-pc/issues/1685
  - RTCSctpTransport.maxMessageSize 0 case https://github.com/w3c/webrtc-pc/issues/1446
  - MTI specification of crypto for certs https://github.com/w3c/webrtc-pc/issues/1258
  - More granular data channel error reporting https://github.com/w3c/webrtc-pc/issues/1317
  - Review issues reported to last in September  https://github.com/w3c/webrtc-pc/issues/1269
  - Clarify whether RTCRtpContributingSource members are live. https://github.com/w3c/webrtc-pc/issues/1533
  - RTCTrackEvent's type parameter https://github.com/w3c/webrtc-pc/issues/1467

* w3c/webrtc-stats (+3/-5/💬21)
  3 issues created:
  - "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type (by henbos)
    https://github.com/w3c/webrtc-stats/issues/281
  - Add TLS version and EC group id to stats (by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/279
  - What happens when a partial keyFrames is received?  (by vr000m)
    https://github.com/w3c/webrtc-stats/issues/278

  12 issues received 21 new comments:
  - #229 Interframe delay stat for video receive stream. (3 by vr000m, ilyanikolaevskiy, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/229
  - #202 RTCMediaStreamTrackStats.concealedAudibleSamples (3 by henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/202
  - #240 Stats for Audio network adaptation (2 by henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/240
  - #235 Is keeping stats around a memory problem? (2 by jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/235
  - #278 What happens when a partial keyFrames is received?  (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/278
  - #279 Add TLS version and EC group id to stats (2 by ekr, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/279
  - #281 "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type (2 by henbos, vr000m)
    https://github.com/w3c/webrtc-stats/issues/281
  - #258 Add estimatedClockSkew (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/258
  - #161 Definitions from MSE need re-targeting (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/161
  - #239 Do the "audio level" stats include MediaStreamTrack volume settings? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/239
  - #275 Add per layer stats for SVC (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/275
  - #222 Audio/Video sync follow-up (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/222

  5 issues closed:
  - Consistent marker for "non-active" object? https://github.com/w3c/webrtc-stats/issues/131
  - Stats for adaptation reason, for realsies https://github.com/w3c/webrtc-stats/issues/256
  - RTCMediaStreamTrackStats.concealedAudibleSamples https://github.com/w3c/webrtc-stats/issues/202
  - Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. https://github.com/w3c/webrtc-stats/issues/248
  - jitterBufferDelay and concealed samples, DTX/CNG samples https://github.com/w3c/webrtc-stats/issues/246

* w3c/webrtc-charter (+0/-0/💬2)
  2 issues received 2 new comments:
  - #14 License for our specs (1 by stefhak)
    https://github.com/w3c/webrtc-charter/issues/14
  - #15 Adopt test as you commit policy in the charter (1 by stefhak)
    https://github.com/w3c/webrtc-charter/issues/15



Pull requests
-------------
* w3c/webrtc-pc (+6/-7/💬15)
  6 pull requests submitted:
  - fix offerToReceive(Audio|Video): false (by fippo)
    https://github.com/w3c/webrtc-pc/pull/1693
  - Rephrase RTCDataChannel.bufferedAmount description (by lgrahl)
    https://github.com/w3c/webrtc-pc/pull/1692
  - offerToReceive: Rewrite to handle two options to createOffer (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1686
  - Fire removetrack/addtrack events before track events. (by jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1684
  - fix offerToReceive(Audio|Video): false (by fippo)
    https://github.com/w3c/webrtc-pc/pull/1683
  - offerToReceive: s/transceiver type/transceiver kind (by fippo)
    https://github.com/w3c/webrtc-pc/pull/1681

  4 pull requests received 15 new comments:
  - #1686 offerToReceive: Rewrite to handle two options to createOffer (6 by adam-be, fippo)
    https://github.com/w3c/webrtc-pc/pull/1686
  - #1683 fix offerToReceive(Audio|Video): false (4 by fippo, jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1683
  - #1681 offerToReceive: s/transceiver type/transceiver kind/ (and do it for kinds, not mediaTypes) (3 by fippo)
    https://github.com/w3c/webrtc-pc/pull/1681
  - #1693 fix offerToReceive(Audio|Video): false (2 by fippo, alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1693

  7 pull requests merged:
  - offerToReceive: Rewrite to handle two options to createOffer
    https://github.com/w3c/webrtc-pc/pull/1686
  - Fix offerToReceive* bug introduced by PR #1672
    https://github.com/w3c/webrtc-pc/pull/1679
  - Editorial: Add jib as editor
    https://github.com/w3c/webrtc-pc/pull/1676
  - Return UnknownError on RTCPeerConnection constructor failure
    https://github.com/w3c/webrtc-pc/pull/1674
  - Set muted before SRD resolves, using new set muted algorithm.
    https://github.com/w3c/webrtc-pc/pull/1667
  - Add testing guideline for naming test files and adding comments
    https://github.com/w3c/webrtc-pc/pull/1664
  - RTCSctpTransport: Specify special cases for maxMessageSize
    https://github.com/w3c/webrtc-pc/pull/1656

* w3c/webrtc-stats (+2/-5/💬32)
  2 pull requests submitted:
  - qualityLimitationDurations record with DOMString key (by henbos)
    https://github.com/w3c/webrtc-stats/pull/282
  - Use () instead of <> for record qualityLimitationDurations (by henbos)
    https://github.com/w3c/webrtc-stats/pull/280

  9 pull requests received 32 new comments:
  - #272 Split RTCMediaStreamTrackStats into four dictionaries. (6 by henbos, vr000m, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/272
  - #277 Add packetsDuplicated (6 by henbos, vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/277
  - #270 RTCQualityLimitationReason and friends (5 by henbos, vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/270
  - #262 Added 'objectDeleted' attribute (3 by vr000m, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/262
  - #268 jitterBufferEmittedCount added (jitterBufferOutput) (3 by henbos, vr000m)
    https://github.com/w3c/webrtc-stats/pull/268
  - #276 Adding dtlsCipher and srtpCipher (3 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/276
  - #282 qualityLimitationDurations record with DOMString key (3 by henbos, dontcallmedom)
    https://github.com/w3c/webrtc-stats/pull/282
  - #273 Pivot from "track" to "sender" and "receiver" stats. (2 by vr000m, jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/273
  - #280 Use underlines instead of record<a,b> for qualityLimitationDurations (1 by henbos)
    https://github.com/w3c/webrtc-stats/pull/280

  5 pull requests merged:
  - Added 'objectDeleted' attribute
    https://github.com/w3c/webrtc-stats/pull/262
  - Use underlines instead of record<a,b> for qualityLimitationDurations
    https://github.com/w3c/webrtc-stats/pull/280
  - RTCQualityLimitationReason and friends
    https://github.com/w3c/webrtc-stats/pull/270
  - Adding dtlsCipher and srtpCipher
    https://github.com/w3c/webrtc-stats/pull/276
  - jitterBufferEmittedCount added (jitterBufferOutput)
    https://github.com/w3c/webrtc-stats/pull/268

* w3c/webrtc-charter (+1/-0/💬0)
  1 pull requests submitted:
  - Work item updates (by aboba)
    https://github.com/w3c/webrtc-charter/pull/17


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
* https://github.com/w3c/webrtc-charter
Received on Tuesday, 12 December 2017 17:00:50 UTC

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