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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 04 Apr 2017 17:00:45 +0000
To: public-webrtc@w3.org
Message-Id: <E1cvRoz-00089o-Aa@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+4/-4/💬16)
  4 issues created:
  - Mark Provisional Answer as a feature at risk? (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1106
  - I think that "gatheringState" of RTCIceTransport should be changed to the name "gathererState". (by gtk2k)
    https://github.com/w3c/webrtc-pc/issues/1105
  - Diagram for RTCSignalingState includes "closed" state, which doesn't exist? (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1103
  - RTCRtpContributingSource naming (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1101

  4 issues closed:
  - Sender/Receiver.rtcpTransport:  feature at risk? https://github.com/w3c/webrtc-pc/issues/1093
  - editorial: Some links to setLocalDescription points to the legacy extensions https://github.com/w3c/webrtc-pc/issues/1095
  - When should RTCRtpContributingSource#audioLevel be null? https://github.com/w3c/webrtc-pc/issues/1090
  - Clean up remaining uses of 'set of receivers' https://github.com/w3c/webrtc-pc/issues/788

  10 issues received 16 new comments:
  - #1073 Need to specify which members of the encodings in "sendEncodings" are actually used (3 by pthatcherg, taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/issues/1073
  - #849 Specify an AllowUnverifiedMedia RTCConfiguration property  (3 by pthatcherg, stefhak)
    https://github.com/w3c/webrtc-pc/issues/849
  - #1077 Candidate from onicecandidate event and addIceCandidate are incompatible (2 by lgrahl, aboba)
    https://github.com/w3c/webrtc-pc/issues/1077
  - #1085 RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout (2 by fippo, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1085
  - #1089 Update for structured cloning changes in HTML (1 by adam-be)
    https://github.com/w3c/webrtc-pc/issues/1089
  - #1101 RTCRtpContributingSource naming (1 by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1101
  - #1103 Diagram for RTCSignalingState includes "closed" state, which doesn't exist? (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1103
  - #1105 I think that "gatheringState" of RTCIceTransport should be changed to the name "gathererState". (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1105
  - #1106 Mark Provisional Answer as a feature at risk? (1 by fippo)
    https://github.com/w3c/webrtc-pc/issues/1106
  - #1086 Make legacy API optional to implement (1 by foolip)
    https://github.com/w3c/webrtc-pc/issues/1086

* w3c/webrtc-stats (+0/-4/💬12)
  4 issues closed:
  - Clarify if stat object's members can change after having been returned https://github.com/w3c/webrtc-stats/issues/121
  - mediaTrackId field missing a description https://github.com/w3c/webrtc-stats/issues/106
  - Stat for how much time it takes to encode video https://github.com/w3c/webrtc-stats/issues/150
  - RoundTripTime not defined when the underlying stream can't calculate it https://github.com/w3c/webrtc-stats/issues/180

  5 issues received 12 new comments:
  - #189 Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. (7 by fluffy, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/189
  - #190 example 8.2: calculating fraction lost vs fractionLost stat (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/190
  - #106 mediaTrackId field missing a description (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/106
  - #109 RTCCodecStats needs `transportId` and `isRemote` to give it context (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/109
  - #117 getStats example is outdated and redundant. (1 by vr000m)
    https://github.com/w3c/webrtc-stats/issues/117



Pull requests
-------------
* w3c/webrtc-pc (+7/-3/💬14)
  7 pull requests submitted:
  - Update structured cloning for recent changes to HTML (by domenic)
    https://github.com/w3c/webrtc-pc/pull/1108
  - Section 12.2.1.1: enum errorDetail definition (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1107
  - Add missing "closed" signaling state. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1104
  - Don't link function names to legacy section (unless intended) (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1102
  - Clarify when RTCRtpContributingSource.audioLevel can be null. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1100
  - Always update the RTCRtpContributingSource for SSRCs. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1099
  - Attempt to update RTCRtpContributingSource objects at playout time. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1098

  3 pull requests merged:
  - Don't link function names to legacy section (unless intended)
    https://github.com/w3c/webrtc-pc/pull/1102
  - Clarify when RTCRtpContributingSource.audioLevel can be null.
    https://github.com/w3c/webrtc-pc/pull/1100
  - Remove last use of 'set of receivers'
    https://github.com/w3c/webrtc-pc/pull/1096

  8 pull requests received 14 new comments:
  - #1026 strawman text to show how unverified media would work (6 by pthatcherg, fluffy, adamroach, rshpount)
    https://github.com/w3c/webrtc-pc/pull/1026
  - #1099 Always update the RTCRtpContributingSource for SSRCs. (2 by fluffy, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1099
  - #1097 RTP/RTCP non-mux: feature at risk (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1097
  - #1098 Attempt to update RTCRtpContributingSource objects at playout time. (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1098
  - #1100 Clarify when RTCRtpContributingSource.audioLevel can be null. (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1100
  - #1102 Don't link function names to legacy section (unless intended) (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1102
  - #1104 Add missing "closed" signaling state. (1 by fluffy)
    https://github.com/w3c/webrtc-pc/pull/1104
  - #1107 Section 12.2.1.1: RTCErrorDetailType Enum definition (1 by fluffy)
    https://github.com/w3c/webrtc-pc/pull/1107

* w3c/webrtc-stats (+2/-4/💬6)
  2 pull requests submitted:
  - Update example to match webrtc spec's + senders.getStats. (by jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/192
  - New 'remote-inbound-rtp' + 'remote-outbound-rtp' (refactor out isRemote) (by jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/191

  4 pull requests merged:
  - Add RTCOutboundRTPStreamStats.totalEncodeTime
    https://github.com/w3c/webrtc-stats/pull/184
  - Change log, and make tidy
    https://github.com/w3c/webrtc-stats/pull/186
  - rtt undefined when no RTCP RR
    https://github.com/w3c/webrtc-stats/pull/188
  - Bandwidth estimations again (Issue 97 redux)
    https://github.com/w3c/webrtc-stats/pull/182

  3 pull requests received 6 new comments:
  - #191 Refactor out isRemote. (4 by jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/191
  - #192 Update example to match webrtc spec's + senders.getStats. (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/192
  - #186 Change log, and make tidy (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/186


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 4 April 2017 17:00:55 UTC

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