W3C home > Mailing lists > Public > public-webrtc@w3.org > January 2018

Weekly github digest (WebRTC WG specifications)

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

This archive was generated by hypermail 2.3.1 : Tuesday, 16 January 2018 17:01:21 UTC