- From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
- Date: Tue, 23 Jan 2018 17:00:54 +0000
- To: public-webrtc@w3.org
- Message-Id: <E1ee1wM-0003is-5q@uranus.w3.org>
Issues
------
* w3c/webrtc-pc (+5/-8/💬31)
5 issues created:
- Multiple SRDs may leave streams and tracks in unexpected state. (by jan-ivar)
https://github.com/w3c/webrtc-pc/issues/1744
- Integrate CSP access control into algorithms (by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1742
- Clarify that "disconnected" is transient. (by jan-ivar)
https://github.com/w3c/webrtc-pc/issues/1740
- Private key access? (by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1739
- Add direction change as reason for "mute" event (by stefhak)
https://github.com/w3c/webrtc-pc/issues/1737
16 issues received 31 new comments:
- #1732 WHATWG streams for data channel messages (5 by ricea, lgrahl, aboba, domenic)
https://github.com/w3c/webrtc-pc/issues/1732
- #1729 "track" event will fire extra times if applying multiple remote offers (4 by jan-ivar, stefhak)
https://github.com/w3c/webrtc-pc/issues/1729
- #1699 Data channel closing procedure (3 by lgrahl, taylor-b)
https://github.com/w3c/webrtc-pc/issues/1699
- #1734 RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats (3 by foolip, taylor-b, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1734
- #1737 Add direction change as reason for "mute" event (2 by jan-ivar, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1737
- #1744 Multiple SRDs may leave streams and tracks in unexpected state. (2 by jan-ivar, stefhak)
https://github.com/w3c/webrtc-pc/issues/1744
- #1718 "a=msid" line should contain sender/receiver IDs, not track IDs (2 by taylor-b, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1718
- #1112 Terminology around "setting" attributes may be incorrect (2 by adam-be)
https://github.com/w3c/webrtc-pc/issues/1112
- #1730 Do not consider direction in "need negotiation" evaluation for replaceTrack (1 by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1730
- #230 Add support for WebRTC Data Channel in Workers (1 by benjamingr)
https://github.com/w3c/webrtc-pc/issues/230
- #1707 onmute then onunmute can fire before negotiation completes (1 by stefhak)
https://github.com/w3c/webrtc-pc/issues/1707
- #1739 Private key access? (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1739
- #1742 Integrate CSP access control into algorithms (1 by murillo128)
https://github.com/w3c/webrtc-pc/issues/1742
- #1454 Using setConfiguration() to add application certificates to an RTCPeerConnection post-construction? (1 by stefhak)
https://github.com/w3c/webrtc-pc/issues/1454
- #1499 Processing Remote MediaStreamTracks should pass on receiver when constructing track event (1 by stefhak)
https://github.com/w3c/webrtc-pc/issues/1499
- #1727 WebRTC bypass CSP connect-src policies (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1727
8 issues closed:
- "track" event will fire extra times if applying multiple remote offers https://github.com/w3c/webrtc-pc/issues/1729
- Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? https://github.com/w3c/webrtc-pc/issues/1689
- Should we require a reference from RTPSender to RTCPeerConnection? https://github.com/w3c/webrtc-pc/issues/1712
- order of transceivers, senders/receivers? https://github.com/w3c/webrtc-pc/issues/1669
- RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats https://github.com/w3c/webrtc-pc/issues/1734
- Processing Remote MediaStreamTracks should pass on receiver when constructing track event https://github.com/w3c/webrtc-pc/issues/1499
- Effect of mute/disable on on-the-wire framerate is not described https://github.com/w3c/webrtc-pc/issues/1695
- RTCDtlsTransportState enum descriptions are lacking https://github.com/w3c/webrtc-pc/issues/1601
* w3c/webrtc-stats (+8/-2/💬23)
8 issues created:
- RTCCodecStats in getStats() is redundant. (by karthikbr82)
https://github.com/w3c/webrtc-stats/issues/306
- averageRTCPInterval should be averageRtcpInterval (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/305
- Stats need to mark up which members are required (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/304
- CodecType "encode" / "decude" needs to default to "both" (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/303
- Lifetime of RTPStreamStats (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/302
- terminology: rename mediaType to kind (by fippo)
https://github.com/w3c/webrtc-stats/issues/301
- Missing definition of "deleted". When exactly do stats get deleted? (by jan-ivar)
https://github.com/w3c/webrtc-stats/issues/300
- Add fecPacketsSent/Received (by vr000m)
https://github.com/w3c/webrtc-stats/issues/299
9 issues received 23 new comments:
- #298 Rename sender/receiver/track stats (6 by henbos, vr000m, jan-ivar, alvestrand)
https://github.com/w3c/webrtc-stats/issues/298
- #235 Is keeping stats around a memory problem? (5 by henbos, vr000m, jan-ivar, alvestrand)
https://github.com/w3c/webrtc-stats/issues/235
- #293 RTCRTPStreamStats should have senderId/receiverId (4 by henbos, taylor-b, alvestrand)
https://github.com/w3c/webrtc-stats/issues/293
- #296 Rename "objectDeleted" to something else. (2 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/issues/296
- #300 Missing definition of "deleted". When exactly do stats get deleted? (2 by jan-ivar, alvestrand)
https://github.com/w3c/webrtc-stats/issues/300
- #193 RTCMediaStreamTrackStats.audioLevel clarification (1 by vr000m)
https://github.com/w3c/webrtc-stats/issues/193
- #294 Rename RTCRTPStreamStats to RTCRtpStreamStats? (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/294
- #271 Add stat for inputAudioLevel, before the audio filter (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/271
- #306 RTCCodecStats in every getStats() is redundant. (1 by fippo)
https://github.com/w3c/webrtc-stats/issues/306
2 issues closed:
- Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? https://github.com/w3c/webrtc-stats/issues/241
- Do the "audio level" stats include MediaStreamTrack volume settings? https://github.com/w3c/webrtc-stats/issues/239
* w3c/webrtc-charter (+4/-4/💬6)
4 issues created:
- Revisiting deliverables (by aboba)
https://github.com/w3c/webrtc-charter/issues/27
- WebRTC 1.0 is already in CR, change text? (by stefhak)
https://github.com/w3c/webrtc-charter/issues/26
- License again (by stefhak)
https://github.com/w3c/webrtc-charter/issues/25
- Add WhatWG to "External Organizations" we coordinate with (by stefhak)
https://github.com/w3c/webrtc-charter/issues/23
4 issues received 6 new comments:
- #26 WebRTC 1.0 is already in CR, change text? (3 by dontcallmedom, aboba, stefhak)
https://github.com/w3c/webrtc-charter/issues/26
- #19 Add language on "Using the Stream API to access data channels" (1 by aboba)
https://github.com/w3c/webrtc-charter/issues/19
- #25 License again (1 by dontcallmedom)
https://github.com/w3c/webrtc-charter/issues/25
- #27 Revisiting deliverables (1 by alvestrand)
https://github.com/w3c/webrtc-charter/issues/27
4 issues closed:
- License again https://github.com/w3c/webrtc-charter/issues/25
- Add language on "Using the Stream API to access data channels" https://github.com/w3c/webrtc-charter/issues/19
- Add WhatWG to "External Organizations" we coordinate with https://github.com/w3c/webrtc-charter/issues/23
- Revise IETF liaison text https://github.com/w3c/webrtc-charter/issues/18
Pull requests
-------------
* w3c/webrtc-pc (+4/-8/💬4)
4 pull requests submitted:
- Properly defined direction when creating transceiver. (by stefhak)
https://github.com/w3c/webrtc-pc/pull/1743
- Clarify that "disconnected" is transient (editorial) (by jan-ivar)
https://github.com/w3c/webrtc-pc/pull/1741
- Passing receiver when constructing track event in 'Processing Remote … (by stefhak)
https://github.com/w3c/webrtc-pc/pull/1738
- Convert audioLevel to double. (by jan-ivar)
https://github.com/w3c/webrtc-pc/pull/1736
3 pull requests received 4 new comments:
- #1725 replaceTrack "negotiation needed" clarification (2 by burnburn, stefhak)
https://github.com/w3c/webrtc-pc/pull/1725
- #1720 Defer SRD add/remove tracks until right before firing track events. (1 by jan-ivar)
https://github.com/w3c/webrtc-pc/pull/1720
- #1719 Define pc.getTransceivers() et al to be in insertion order. (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1719
8 pull requests merged:
- Passing receiver when constructing track event in 'Processing Remote …
https://github.com/w3c/webrtc-pc/pull/1738
- Convert audioLevel to double.
https://github.com/w3c/webrtc-pc/pull/1736
- Make audioLevel and voiceActivityFlag optional and non-nullable.
https://github.com/w3c/webrtc-pc/pull/1733
- RTCRtpSender: Specify advise about framerate for disabled/muted tracks
https://github.com/w3c/webrtc-pc/pull/1731
- Define JSEP terms in terminology section
https://github.com/w3c/webrtc-pc/pull/1726
- Add detail to RTCDtlsTransportState descriptions
https://github.com/w3c/webrtc-pc/pull/1724
- Underlying data transport explanation
https://github.com/w3c/webrtc-pc/pull/1722
- Define pc.getTransceivers() et al to be in insertion order.
https://github.com/w3c/webrtc-pc/pull/1719
* w3c/webrtc-stats (+1/-3/💬4)
1 pull requests submitted:
- Remove reference to RTCRtpSynchronizationSource.audioLevel. (by jan-ivar)
https://github.com/w3c/webrtc-stats/pull/297
3 pull requests received 4 new comments:
- #284 moved bytes and packets received counters (2 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/pull/284
- #288 Additional description of audioLevel (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/288
- #297 Remove reference to RTCRtpSynchronizationSource.audioLevel. (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/297
3 pull requests merged:
- moved bytes and packets received counters
https://github.com/w3c/webrtc-stats/pull/284
- Additional description of audioLevel
https://github.com/w3c/webrtc-stats/pull/288
- Remove reference to RTCRtpSynchronizationSource.audioLevel.
https://github.com/w3c/webrtc-stats/pull/297
* w3c/webrtc-charter (+4/-3/💬3)
4 pull requests submitted:
- Revise IETF interaction language (by alvestrand)
https://github.com/w3c/webrtc-charter/pull/29
- Add to "scope" about contexts for use of API (by alvestrand)
https://github.com/w3c/webrtc-charter/pull/28
- Add WhatWG to "External Organizations" we coordinate with (by aboba)
https://github.com/w3c/webrtc-charter/pull/24
- Add data transfer to scope (by aboba)
https://github.com/w3c/webrtc-charter/pull/22
2 pull requests received 3 new comments:
- #22 Add data transfer to scope (2 by dontcallmedom, aboba)
https://github.com/w3c/webrtc-charter/pull/22
- #24 Add WhatWG to "External Organizations" we coordinate with (1 by aboba)
https://github.com/w3c/webrtc-charter/pull/24
3 pull requests merged:
- Add to "scope" about contexts for use of API
https://github.com/w3c/webrtc-charter/pull/28
- Revise IETF interaction language
https://github.com/w3c/webrtc-charter/pull/29
- Add WhatWG to "External Organizations" we coordinate with
https://github.com/w3c/webrtc-charter/pull/24
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, 23 January 2018 17:01:02 UTC