- 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
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* 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