- From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
- Date: Tue, 16 Jan 2018 17:01:06 +0000
- To: public-webrtc@w3.org
- Message-Id: <E1ebUbi-0005Uz-3Y@uranus.w3.org>
Issues
------
* w3c/webrtc-pc (+5/-3/💬69)
5 issues created:
- Editorial: PendingRemoteDescription is set by.... (by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1735
- RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats (by na-g)
https://github.com/w3c/webrtc-pc/issues/1734
- WHATWG streams for data channels (by lgrahl)
https://github.com/w3c/webrtc-pc/issues/1732
- Do not consider direction at "need negotiation" evaluation for replaceTrack (by stefhak)
https://github.com/w3c/webrtc-pc/issues/1730
- "track" event will fire extra times if applying multiple remote offers (by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1729
15 issues received 69 new comments:
- #1705 specify legacy onaddstream? (12 by foolip, fippo, alvestrand, henbos, youennf)
https://github.com/w3c/webrtc-pc/issues/1705
- #1732 WHATWG streams for data channel messages (10 by lgrahl, martinthomson, domenic)
https://github.com/w3c/webrtc-pc/issues/1732
- #1729 "track" event will fire extra times if applying multiple remote offers (9 by taylor-b, jan-ivar, docfaraday, alvestrand, stefhak)
https://github.com/w3c/webrtc-pc/issues/1729
- #1690 RTCRtpContributingSource.timestamp needs a clearer definition (6 by jan-ivar, bzbarsky, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1690
- #1240 editorial: media api introduction (5 by fippo, taylor-b)
https://github.com/w3c/webrtc-pc/issues/1240
- #1721 canTrickleIceCandidiates question (5 by fippo, cdh4u, taylor-b, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1721
- #1700 Maximum message size slightly incorrect (4 by foolip, adam-be)
https://github.com/w3c/webrtc-pc/issues/1700
- #1125 Should the spec describe addStream/onaddstream as legacy API? (4 by foolip, youennf)
https://github.com/w3c/webrtc-pc/issues/1125
- #1728 Possibly racy replaceTrack() (4 by henbos, taylor-b, stefhak)
https://github.com/w3c/webrtc-pc/issues/1728
- #1734 RTCRtpContributingSource.audioLevel has different type and range than similar fields in webrtc-stats (3 by taylor-b, dontcallmedom, jan-ivar)
https://github.com/w3c/webrtc-pc/issues/1734
- #1718 "a=msid" line should contain sender/receiver IDs, not track IDs (3 by henbos, ibc, taylor-b)
https://github.com/w3c/webrtc-pc/issues/1718
- #1644 Adding more values to RTCIceTransportPolicy Enum (1 by jianjunz)
https://github.com/w3c/webrtc-pc/issues/1644
- #1689 Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable? (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1689
- #1662 addTransceiver woes (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1662
- #1695 Effect of mute/disable on on-the-wire framerate is not described (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1695
3 issues closed:
- Maximum message size slightly incorrect https://github.com/w3c/webrtc-pc/issues/1700
- RTCDataChannel.bufferedAmount description confusing https://github.com/w3c/webrtc-pc/issues/1680
- Adding more values to RTCIceTransportPolicy Enum https://github.com/w3c/webrtc-pc/issues/1644
* w3c/webrtc-stats (+5/-4/💬16)
5 issues created:
- Rename "objectDeleted" to something else. (by henbos)
https://github.com/w3c/webrtc-stats/issues/296
- Define timestamps in terms of performance.origin (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/295
- Rename RTCRTPStreamStats to RTCRtpStreamStats? (by henbos)
https://github.com/w3c/webrtc-stats/issues/294
- RTCRTPStreamStats should have senderId/receiverId (by henbos)
https://github.com/w3c/webrtc-stats/issues/293
- Enable continuous publication (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/292
11 issues received 16 new comments:
- #235 Is keeping stats around a memory problem? (4 by lgrahl, vr000m, jan-ivar, alvestrand)
https://github.com/w3c/webrtc-stats/issues/235
- #292 Enable continuous publication (2 by vivienlacourba, alvestrand)
https://github.com/w3c/webrtc-stats/issues/292
- #293 RTCRTPStreamStats should have senderId/receiverId (2 by henbos, taylor-b)
https://github.com/w3c/webrtc-stats/issues/293
- #289 WiFi Stats. (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/289
- #133 Need DSCP information for outgoing RTP streams (1 by jan-ivar)
https://github.com/w3c/webrtc-stats/issues/133
- #230 RTCMediaStreamTrackStats is four dictionaries in one (1 by henbos)
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
- #238 Add stat to reflect the redundancy of FEC/RED data (1 by vr000m)
https://github.com/w3c/webrtc-stats/issues/238
- #271 Add stat for inputAudioLevel, before the audio filter (1 by vr000m)
https://github.com/w3c/webrtc-stats/issues/271
- #275 Add per layer stats for SVC (1 by ssilkin)
https://github.com/w3c/webrtc-stats/issues/275
- #281 "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type (1 by henbos)
https://github.com/w3c/webrtc-stats/issues/281
4 issues closed:
- Need DSCP information for incoming RTP streams https://github.com/w3c/webrtc-stats/issues/135
- Enable continuous publication https://github.com/w3c/webrtc-stats/issues/292
- We need "sender" and "receiver" stats, not "track" stats https://github.com/w3c/webrtc-stats/issues/231
- RTCMediaStreamTrackStats is four dictionaries in one https://github.com/w3c/webrtc-stats/issues/230
* w3c/webrtc-charter (+1/-0/💬1)
1 issues created:
- Update milestones table (by dontcallmedom)
https://github.com/w3c/webrtc-charter/issues/21
1 issues received 1 new comments:
- #20 Add language on enabling Workers to use WebRTC constructs (1 by alvestrand)
https://github.com/w3c/webrtc-charter/issues/20
Pull requests
-------------
* w3c/webrtc-pc (+2/-3/💬2)
2 pull requests submitted:
- Make audioLevel and voiceActivityFlag optional and non-nullable. (by jan-ivar)
https://github.com/w3c/webrtc-pc/pull/1733
- RTCRtpSender: Specify advise about framerate for disabled/muted tracks (by adam-be)
https://github.com/w3c/webrtc-pc/pull/1731
2 pull requests received 2 new comments:
- #1696 replaceTrack: Clarify how the UA determines if negotiation is needed (1 by alvestrand)
https://github.com/w3c/webrtc-pc/pull/1696
- #1719 Define pc.getTransceivers() et al to be in insertion order. (1 by alvestrand)
https://github.com/w3c/webrtc-pc/pull/1719
3 pull requests merged:
- Remove reference from RTPSender to RTCPeerConnection
https://github.com/w3c/webrtc-pc/pull/1716
- Validation of reordered readonly parameters in setParameters
https://github.com/w3c/webrtc-pc/pull/1714
- Rephrase RTCDataChannel.bufferedAmount description
https://github.com/w3c/webrtc-pc/pull/1692
* w3c/webrtc-stats (+0/-3/💬13)
6 pull requests received 13 new comments:
- #273 Add "sender" and "receiver" stats. (6 by henbos, jan-ivar, alvestrand)
https://github.com/w3c/webrtc-stats/pull/273
- #291 Introduction to RTP stream statistics (2 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/pull/291
- #272 Split RTCMediaStreamTrackStats into four dictionaries. (2 by jan-ivar, alvestrand)
https://github.com/w3c/webrtc-stats/pull/272
- #288 Additional description of audioLevel (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/288
- #290 Adds per-DSCP packet counters to RTP streams (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/290
- #284 moved bytes and packets received counters (1 by vr000m)
https://github.com/w3c/webrtc-stats/pull/284
3 pull requests merged:
- Adds per-DSCP packet counters to RTP streams
https://github.com/w3c/webrtc-stats/pull/290
- Introduction to RTP stream statistics
https://github.com/w3c/webrtc-stats/pull/291
- Add "sender" and "receiver" stats.
https://github.com/w3c/webrtc-stats/pull/273
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, 16 January 2018 17:01:21 UTC