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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 11 Apr 2017 17:00:39 +0000
To: public-webrtc@w3.org
Message-Id: <E1cxz9j-0002Fd-HF@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+8/-6/💬33)
  8 issues created:
  - RTCCertificate.getAlgorithm() wording and serialization (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1121
  - RTCCertificate.getAlgorithm for remote certificates (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1120
  - [TreatNullAs=EmptyString] is not allowed for USVString per Web IDL (by foolip)
    https://github.com/w3c/webrtc-pc/issues/1118
  - RTCRtpContributingSource.getReceiver() (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1117
  - "getParameters" and "setParameters" need more thorough specification (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1116
  - Section 12.2.4: Note (by aboba)
    https://github.com/w3c/webrtc-pc/issues/1113
  - Terminology around "setting" attributes may be incorrect (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1112
  - Describe update strategy on variables (by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1111

  6 issues closed:
  - I think that "gatheringState" of RTCIceTransport should be changed to the name "gathererState". https://github.com/w3c/webrtc-pc/issues/1105
  - Section 12.2.1.1: enum errorDetail definition https://github.com/w3c/webrtc-pc/issues/1044
  - Diagram for RTCSignalingState includes "closed" state, which doesn't exist? https://github.com/w3c/webrtc-pc/issues/1103
  - NetworkError event is not defined and might not be needed https://github.com/w3c/webrtc-pc/issues/526
  - When exactly is an SSRC RTCRtpContributingSource object updated? https://github.com/w3c/webrtc-pc/issues/1091
  - get/setParameters does not have a parameter for packetization interval https://github.com/w3c/webrtc-pc/issues/1021

  12 issues received 33 new comments:
  - #1116 "getParameters" and "setParameters" need more thorough specification (7 by taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/issues/1116
  - #763 Handling of simulcast errors (5 by taylor-b, alvestrand, aboba)
    https://github.com/w3c/webrtc-pc/issues/763
  - #1086 Make legacy API optional to implement (5 by foolip, stefhak, alvestrand, youennf)
    https://github.com/w3c/webrtc-pc/issues/1086
  - #1092 DTLS failures (3 by aboba, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1092
  - #1101 RTCRtpContributingSource naming (3 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1101
  - #1073 Need to specify which members of the encodings in "sendEncodings" are actually used (2 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1073
  - #1111 Describe update strategy on variables (2 by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1111
  - #1117 RTCRtpContributingSource.getReceiver() (2 by henbos, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1117
  - #1121 RTCCertificate.getAlgorithm() wording and serialization (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1121
  - #1105 I think that "gatheringState" of RTCIceTransport should be changed to the name "gathererState". (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1105
  - #1113 Section 12.2.4: Note (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1113
  - #1118 [TreatNullAs=EmptyString] is not allowed for USVString per Web IDL (1 by foolip)
    https://github.com/w3c/webrtc-pc/issues/1118

* w3c/webrtc-stats (+1/-2/💬3)
  1 issues created:
  - RTCMediaStreamTrackStats.audioLevel clarification (by na-g)
    https://github.com/w3c/webrtc-stats/issues/193

  2 issues closed:
  - getStats example is outdated and redundant. https://github.com/w3c/webrtc-stats/issues/117
  - example 8.2: calculating fraction lost vs fractionLost stat https://github.com/w3c/webrtc-stats/issues/190

  3 issues received 3 new comments:
  - #193 RTCMediaStreamTrackStats.audioLevel clarification (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/193
  - #117 getStats example is outdated and redundant. (1 by jan-ivar)
    https://github.com/w3c/webrtc-stats/issues/117
  - #109 RTCCodecStats needs `transportId` and `isRemote` to give it context (1 by taylor-b)
    https://github.com/w3c/webrtc-stats/issues/109



Pull requests
-------------
* w3c/webrtc-pc (+5/-6/💬26)
  5 pull requests submitted:
  - Making legacy methods optional to implement. (by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1119
  - DTLS failures (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1115
  - Mark Identity as a feature at risk (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1114
  - Mark pranswer as a "feature atrisk" (by aboba)
    https://github.com/w3c/webrtc-pc/pull/1110
  - Adding configurable "ptime" member of RTCRtpEncodingParameters. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1109

  6 pull requests merged:
  - Section 12.2.1.1: RTCErrorDetailType Enum definition
    https://github.com/w3c/webrtc-pc/pull/1107
  - Add missing "closed" signaling state.
    https://github.com/w3c/webrtc-pc/pull/1104
  - Always update the RTCRtpContributingSource for SSRCs.
    https://github.com/w3c/webrtc-pc/pull/1099
  - Adding configurable "ptime" member of RTCRtpEncodingParameters.
    https://github.com/w3c/webrtc-pc/pull/1109
  - Eliminate NetworkError
    https://github.com/w3c/webrtc-pc/pull/1011
  - RTP/RTCP non-mux: feature at risk
    https://github.com/w3c/webrtc-pc/pull/1097

  8 pull requests received 26 new comments:
  - #1026 strawman text to show how unverified media would work (8 by pthatcherg, fluffy, taylor-b, rshpount)
    https://github.com/w3c/webrtc-pc/pull/1026
  - #1098 Attempt to update RTCRtpContributingSource objects at playout time. (4 by burnburn, taylor-b, alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1098
  - #1099 Always update the RTCRtpContributingSource for SSRCs. (3 by taylor-b, aboba, alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1099
  - #1110 Mark pranswer as a "feature atrisk" (3 by ekr, alvestrand, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1110
  - #1115 DTLS failures (3 by fluffy, taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/pull/1115
  - #1108 Update structured cloning for recent changes to HTML (2 by alvestrand, domenic)
    https://github.com/w3c/webrtc-pc/pull/1108
  - #1109 Adding configurable "ptime" member of RTCRtpEncodingParameters. (2 by taylor-b, aboba)
    https://github.com/w3c/webrtc-pc/pull/1109
  - #1011 Eliminate NetworkError (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1011

* w3c/webrtc-stats (+6/-1/💬4)
  6 pull requests submitted:
  - RTCMediaStreamTrackStats: framesCaptured added. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/199
  - RTCIceCandidatePairStats: packetsSent/Received added. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/198
  - RTCTransportStats: packetsSent/Received added. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/197
  - RTCIceCandidatePairStats: Update writable, remove readable (by henbos)
    https://github.com/w3c/webrtc-stats/pull/196
  - Adding "codec type" and transportId to RTCCodecStats. (by taylor-b)
    https://github.com/w3c/webrtc-stats/pull/195
  - Adding RTCRTPContributingSourceStats stats report object. (by taylor-b)
    https://github.com/w3c/webrtc-stats/pull/194

  1 pull requests merged:
  - Update example to match webrtc spec's + senders.getStats.
    https://github.com/w3c/webrtc-stats/pull/192

  2 pull requests received 4 new comments:
  - #191 Refactor out isRemote. (3 by jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/191
  - #195 Adding "codec type" and transportId to RTCCodecStats. (1 by taylor-b)
    https://github.com/w3c/webrtc-stats/pull/195


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 11 April 2017 17:00:46 UTC

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