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Weekly github digest (WebRTC WG specifications)

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

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