- From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
- Date: Tue, 31 Oct 2017 17:00:29 +0000
- To: public-webrtc@w3.org
- Message-Id: <E1e9Ztt-0005yu-OR@uranus.w3.org>
Issues
------
* w3c/webrtc-pc (+4/-4/💬35)
4 issues created:
- Provide changelog for the spec in new editors draft workflow (by dontcallmedom)
https://github.com/w3c/webrtc-pc/issues/1648
- Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs (by soareschen)
https://github.com/w3c/webrtc-pc/issues/1646
- getIdentityAssertion should reject with InvalidAccessError if no identity provider is set (by soareschen)
https://github.com/w3c/webrtc-pc/issues/1645
- Adding more values to RTCIceTransportPolicy Enum (by jianjunz)
https://github.com/w3c/webrtc-pc/issues/1644
10 issues received 35 new comments:
- #1644 Adding more values to RTCIceTransportPolicy Enum (9 by alvestrand, taylor-b, jianjunz, aboba)
https://github.com/w3c/webrtc-pc/issues/1644
- #1533 Clarify whether RTCRtpContributingSource members are live. (7 by fluffy, taylor-b, alvestrand, aboba, na-g)
https://github.com/w3c/webrtc-pc/issues/1533
- #1643 Why is setDirection a method? (6 by taylor-b, adam-be, fippo, jan-ivar, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1643
- #1646 Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs (4 by aboba, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1646
- #1446 RTCSctpTransport.maxMessageSize 0 case (3 by lgrahl, taylor-b)
https://github.com/w3c/webrtc-pc/issues/1446
- #1613 Stats & isolated streams (2 by dontcallmedom, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1613
- #1248 setParameters woes (1 by aboba)
https://github.com/w3c/webrtc-pc/issues/1248
- #1635 Need for Initial Bitrate by the Application/RtpSender? (1 by aboba)
https://github.com/w3c/webrtc-pc/issues/1635
- #942 Meta: auto-publish changes to the spec (1 by dontcallmedom)
https://github.com/w3c/webrtc-pc/issues/942
- #1640 offerToReceiveAudio/Video processing creates sendrecv transceivers (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1640
4 issues closed:
- Section 9.6: protocol type https://github.com/w3c/webrtc-pc/issues/1293
- pc.close() method should close sctp transport https://github.com/w3c/webrtc-pc/issues/1381
- No way to tell when SCTP association is closed https://github.com/w3c/webrtc-pc/issues/1612
- Missing detail on obtaining lastOffer/lastAnswer in setLocalDescription https://github.com/w3c/webrtc-pc/issues/1164
* w3c/webrtc-stats (+0/-5/💬29)
18 issues received 29 new comments:
- #256 Stats for adaptation reason, for realsies (4 by henbos, alvestrand)
https://github.com/w3c/webrtc-stats/issues/256
- #193 RTCMediaStreamTrackStats.audioLevel clarification (3 by henbos, taylor-b, alvestrand)
https://github.com/w3c/webrtc-stats/issues/193
- #246 jitterBufferDelay and concealed samples, DTX/CNG samples (3 by henbos, vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/issues/246
- #258 Add estimatedClockSkew (2 by henbos, alvestrand)
https://github.com/w3c/webrtc-stats/issues/258
- #135 Need DSCP information for incoming RTP streams (2 by taylor-b, alvestrand)
https://github.com/w3c/webrtc-stats/issues/135
- #222 Audio/Video sync follow-up (2 by icydragons, alvestrand)
https://github.com/w3c/webrtc-stats/issues/222
- #229 Interframe delay stat for video receive stream. (2 by ilyanikolaevskiy, alvestrand)
https://github.com/w3c/webrtc-stats/issues/229
- #133 Need DSCP information for outgoing RTP streams (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/133
- #161 Definitions from MSE need re-targeting (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/161
- #101 Move to continuous publication (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/101
- #98 Add "privacy" to security considerations (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/98
- #230 RTCMediaStreamTrackStats is four dictionaries in one (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/230
- #231 We need "sender" and "receiver" stats, not "track" stats (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/231
- #240 Stats for Audio network adaptation (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/240
- #241 Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/241
- #253 packetsLost is unsigned but can be negative according to RFC (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/253
- #254 add packetsDuplicated as a result of #253 (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/254
- #255 What is fractionLost for a local incoming media stream? (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/255
5 issues closed:
- RTCMediaStreamTrackStats.jitterBufferDelay: Specify unit in seconds https://github.com/w3c/webrtc-stats/issues/244
- packetsLost is unsigned but can be negative according to RFC https://github.com/w3c/webrtc-stats/issues/253
- Add "privacy" to security considerations https://github.com/w3c/webrtc-stats/issues/98
- Difference between 'Id' and 'Identifier' fields https://github.com/w3c/webrtc-stats/issues/147
- Move to continuous publication https://github.com/w3c/webrtc-stats/issues/101
Pull requests
-------------
* w3c/webrtc-pc (+3/-3/💬13)
3 pull requests submitted:
- Specify how RTCSctpTransport.maxMessageSize gets its value (by adam-be)
https://github.com/w3c/webrtc-pc/pull/1650
- Add a new value to RTCIceTransportPolicy. (by jianjunz)
https://github.com/w3c/webrtc-pc/pull/1649
- Replace setDirection() with writable direction attribute (by adam-be)
https://github.com/w3c/webrtc-pc/pull/1647
6 pull requests received 13 new comments:
- #1618 format & fixup Example code (6 by adam-be, Jxck)
https://github.com/w3c/webrtc-pc/pull/1618
- #1608 Validate protocol string in IdP operations (3 by soareschen, aboba, stefhak)
https://github.com/w3c/webrtc-pc/pull/1608
- #1650 Specify how RTCSctpTransport.maxMessageSize gets its value (1 by lgrahl)
https://github.com/w3c/webrtc-pc/pull/1650
- #1209 Throw error if data channel's buffer is filled, rather than closing. (1 by lgrahl)
https://github.com/w3c/webrtc-pc/pull/1209
- #1647 Replace setDirection() with writable direction attribute (1 by adam-be)
https://github.com/w3c/webrtc-pc/pull/1647
- #1151 Prepare status of the document for CR publication (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1151
3 pull requests merged:
- Set up automatic PR review request on webrtc.html
https://github.com/w3c/webrtc-pc/pull/1622
- Let setLocalDescription use [[LastAnswer/Offer]] internal slots
https://github.com/w3c/webrtc-pc/pull/1638
- format & fixup Example code
https://github.com/w3c/webrtc-pc/pull/1618
* w3c/webrtc-stats (+5/-3/💬11)
5 pull requests submitted:
- RTCQualityLimitationReason and friends (by henbos)
https://github.com/w3c/webrtc-stats/pull/270
- signed packetsLost (by henbos)
https://github.com/w3c/webrtc-stats/pull/269
- jitterBufferOutput added (by henbos)
https://github.com/w3c/webrtc-stats/pull/268
- jitterBufferDelay in seconds (by henbos)
https://github.com/w3c/webrtc-stats/pull/267
- TAG review: Link to design principles (by alvestrand)
https://github.com/w3c/webrtc-stats/pull/266
5 pull requests received 11 new comments:
- #268 jitterBufferOutput added (4 by henbos, vr000m)
https://github.com/w3c/webrtc-stats/pull/268
- #259 Adding "networkType" field to RTCIceCandidateStats. (3 by taylor-b, alvestrand)
https://github.com/w3c/webrtc-stats/pull/259
- #270 RTCQualityLimitationReason and friends (2 by henbos)
https://github.com/w3c/webrtc-stats/pull/270
- #266 TAG review: Link to design principles (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/266
- #269 signed packetsLost (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/269
3 pull requests merged:
- jitterBufferDelay in seconds
https://github.com/w3c/webrtc-stats/pull/267
- signed packetsLost
https://github.com/w3c/webrtc-stats/pull/269
- Add a paragraph about "identifier" when "id" is occupied
https://github.com/w3c/webrtc-stats/pull/264
Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 31 October 2017 17:00:34 UTC