- 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- #1642 OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION (1 by misi)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