- From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
- Date: Tue, 14 Mar 2017 17:00:42 +0000
- To: public-webrtc@w3.org
Issues
------
* w3c/webrtc-pc (+6/-12/💬22)
6 issues created:
- Mark Identity as feature at risk? (by dontcallmedom)
https://github.com/w3c/webrtc-pc/issues/1074
- Need to specify which members of the encodings in "sendEncodings" are actually used (by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1073
- Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy (by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1065
- Attaching the same track to multiple RTCRtpSenders: spec contradiction. (by henbos)
https://github.com/w3c/webrtc-pc/issues/1064
- Section 10.3: Freeing resources for incoming stream (by aboba)
https://github.com/w3c/webrtc-pc/issues/1063
- Section 10.3: (by aboba)
https://github.com/w3c/webrtc-pc/issues/1062
12 issues closed:
- Section 5.6: Generation of candidates https://github.com/w3c/webrtc-pc/issues/1050
- Attaching the same track to multiple RTCRtpSenders: spec contradiction. https://github.com/w3c/webrtc-pc/issues/1064
- Codecs supporting multiple clock rates/packetization-mode https://github.com/w3c/webrtc-pc/issues/1040
- Section 13: Connecting event https://github.com/w3c/webrtc-pc/issues/1043
- Section 10.3: mute signal https://github.com/w3c/webrtc-pc/issues/1051
- Section 10.3: Issue 5 https://github.com/w3c/webrtc-pc/issues/1052
- "Throw a FooError" steps not written in a consistent manner https://github.com/w3c/webrtc-pc/issues/845
- Range checking needed for iceCandidatePoolSize https://github.com/w3c/webrtc-pc/issues/1048
- Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP https://github.com/w3c/webrtc-pc/issues/1049
- RTCDataChannelInit.id should use [EnforceRange] https://github.com/w3c/webrtc-pc/issues/1046
- sdpFmtpLine isn't very convenient https://github.com/w3c/webrtc-pc/issues/1022
- Section 10.3: https://github.com/w3c/webrtc-pc/issues/1062
8 issues received 22 new comments:
- #1063 Section 10.3: Freeing resources for incoming stream (6 by Pehrsons, aboba, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1063
- #1053 Section 11.6: Issue 6 (5 by fippo, aboba, alvestrand, stefhak)
https://github.com/w3c/webrtc-pc/issues/1053
- #1051 Section 10.3: mute signal (4 by taylor-b, aboba)
https://github.com/w3c/webrtc-pc/issues/1051
- #1052 Section 10.3: Issue 5 (2 by taylor-b, aboba)
https://github.com/w3c/webrtc-pc/issues/1052
- #1065 Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy (2 by taylor-b, aboba)
https://github.com/w3c/webrtc-pc/issues/1065
- #1064 Attaching the same track to multiple RTCRtpSenders: spec contradiction. (1 by aboba)
https://github.com/w3c/webrtc-pc/issues/1064
- #881 Align getAlgorithm return value with Web Crypto (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/881
- #1021 get/setParameters does not have a parameter for packetization interval (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/1021
* w3c/webrtc-stats (+0/-10/💬25)
10 issues closed:
- RTCPeerConnection.getStats: What to do with 'selector' argument? https://github.com/w3c/webrtc-stats/issues/116
- Make sure timing and responsibility for stat object creation is clear https://github.com/w3c/webrtc-stats/issues/120
- Consider making (aggregate) stats more accessible https://github.com/w3c/webrtc-stats/issues/119
- Expose protocol used to communicate with TURN server https://github.com/w3c/webrtc-stats/issues/83
- Consider putting roundTripTime on RTCInboundStreamStats https://github.com/w3c/webrtc-stats/issues/78
- There is no ICE "cancelled" state https://github.com/w3c/webrtc-stats/issues/66
- Unclear if the request encompasses consent checks or not https://github.com/w3c/webrtc-stats/issues/171
- RTCDataChannelStats should reference RTCTransportStats https://github.com/w3c/webrtc-stats/issues/140
- Not clear how to differentiate between received connectivity checks and consent requests https://github.com/w3c/webrtc-stats/issues/96
- Need descriptions for `label`, `protocol` and `state` members of `RTCDataChannelStats` https://github.com/w3c/webrtc-stats/issues/111
14 issues received 25 new comments:
- #163 RTCIceCandidatePairStats.writable/readable: redundant? (4 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/issues/163
- #180 RoundTripTime not defined when the underlying stream can't calculate it (4 by henbos, vr000m, jesup, jan-ivar)
https://github.com/w3c/webrtc-stats/issues/180
- #160 Stat for adaptation reason (3 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/issues/160
- #171 Unclear if the request encompasses consent checks or not (2 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/issues/171
- #178 Packets or frames discarded on send? (2 by vr000m, alvestrand)
https://github.com/w3c/webrtc-stats/issues/178
- #150 Stat for how much time it takes to encode video (2 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/150
- #161 Definitions from MSE need re-targeting (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/161
- #144 Stat for how many adaptation changes occur for a video track (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/144
- #152 Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/152
- #116 RTCPeerConnection.getStats: What to do with 'selector' argument? (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/116
- #119 Consider making (aggregate) stats more accessible (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/119
- #120 Make sure timing and responsibility for stat object creation is clear (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/120
- #154 Stat for retransmitted bytes (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/154
- #153 Stat for likelihood of echo (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/153
Pull requests
-------------
* w3c/webrtc-pc (+7/-12/💬31)
7 pull requests submitted:
- Remove Section 11.6 note (by aboba)
https://github.com/w3c/webrtc-pc/pull/1072
- Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. (by taylor-b)
https://github.com/w3c/webrtc-pc/pull/1071
- Update Call Flow Browser to Browser (by aboba)
https://github.com/w3c/webrtc-pc/pull/1070
- Freeing resources for incoming stream (by aboba)
https://github.com/w3c/webrtc-pc/pull/1069
- Remove Section 12.2.1.1 errorDetailEnum (by aboba)
https://github.com/w3c/webrtc-pc/pull/1068
- Mark getAlgorithm method at risk (by aboba)
https://github.com/w3c/webrtc-pc/pull/1067
- RtpSender and track (by aboba)
https://github.com/w3c/webrtc-pc/pull/1066
12 pull requests merged:
- Add stats selection algorithm based on sender or receiver of selector.
https://github.com/w3c/webrtc-pc/pull/1030
- Switching to new, consistent terminology when talking about exceptions.
https://github.com/w3c/webrtc-pc/pull/1056
- RtpSender and track
https://github.com/w3c/webrtc-pc/pull/1066
- Remove connecting event from Event summary
https://github.com/w3c/webrtc-pc/pull/1061
- ended event fired on the track
https://github.com/w3c/webrtc-pc/pull/1060
- mute signal
https://github.com/w3c/webrtc-pc/pull/1059
- Generation of candidates
https://github.com/w3c/webrtc-pc/pull/1058
- Add clockrate, channels, sdpFmtpLine to codec capability
https://github.com/w3c/webrtc-pc/pull/1057
- Changing iceCandidatePoolSize to an octet and adding EnforceRange.
https://github.com/w3c/webrtc-pc/pull/1055
- Throw InvalidModificationError if changing pool size after SLD.
https://github.com/w3c/webrtc-pc/pull/1054
- Adding "[EnforceRange]" to RTCDataChannelInit.id.
https://github.com/w3c/webrtc-pc/pull/1047
- Clarifying exactly what "sdpFmtpLine" represents.
https://github.com/w3c/webrtc-pc/pull/1045
17 pull requests received 31 new comments:
- #1033 "Hybrid" OAuth solution. (5 by taylor-b, alvestrand, misi, stefhak)
https://github.com/w3c/webrtc-pc/pull/1033
- #1055 Changing iceCandidatePoolSize to an octet and adding EnforceRange. (4 by jan-ivar, aboba)
https://github.com/w3c/webrtc-pc/pull/1055
- #1054 Throw InvalidModificationError if changing pool size after SLD. (3 by taylor-b, aboba)
https://github.com/w3c/webrtc-pc/pull/1054
- #1071 Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. (3 by taylor-b, aboba, stefhak)
https://github.com/w3c/webrtc-pc/pull/1071
- #1030 Add stats selection algorithm based on sender or receiver of selector. (2 by jan-ivar, alvestrand)
https://github.com/w3c/webrtc-pc/pull/1030
- #1056 Switching to new, consistent terminology when talking about exceptions. (2 by dontcallmedom, alvestrand)
https://github.com/w3c/webrtc-pc/pull/1056
- #1066 RtpSender and track (2 by alvestrand, stefhak)
https://github.com/w3c/webrtc-pc/pull/1066
- #1047 Adding "[EnforceRange]" to RTCDataChannelInit.id. (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1047
- #1058 Generation of candidates (1 by alvestrand)
https://github.com/w3c/webrtc-pc/pull/1058
- #1059 mute signal (1 by alvestrand)
https://github.com/w3c/webrtc-pc/pull/1059
- #1060 ended event fired on the track (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1060
- #1061 Remove connecting event from Event summary (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1061
- #1067 Mark getAlgorithm method at risk (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1067
- #1068 Remove Section 12.2.1.1 errorDetailEnum (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1068
- #1069 Freeing resources for incoming stream (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1069
- #1070 Update Call Flow Browser to Browser (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1070
- #1072 Remove Section 11.6 note (1 by stefhak)
https://github.com/w3c/webrtc-pc/pull/1072
* w3c/webrtc-stats (+1/-4/💬5)
1 pull requests submitted:
- Add section on obsoleted stats. (by alvestrand)
https://github.com/w3c/webrtc-stats/pull/181
4 pull requests merged:
- Adding remoteTimestamp to RTCRtpStreamStats.
https://github.com/w3c/webrtc-stats/pull/164
- Add link from datachannel to transport
https://github.com/w3c/webrtc-stats/pull/176
- Remove separation of received consent and connectivity requests.
https://github.com/w3c/webrtc-stats/pull/165
- Adds definitions for RTCDataChannelStats members.
https://github.com/w3c/webrtc-stats/pull/179
3 pull requests received 5 new comments:
- #164 Adding remoteTimestamp to RTCRtpStreamStats. (2 by taylor-b, alvestrand)
https://github.com/w3c/webrtc-stats/pull/164
- #181 Add section on obsoleted stats. (2 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/181
- #176 Add link from datachannel to transport (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/176
Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 14 March 2017 17:00:50 UTC