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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 07 Feb 2017 17:00:39 +0000
To: public-webrtc@w3.org
Message-Id: <E1cb98B-0007sJ-Gv@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+3/-6/💬21)
  3 issues created:
  - OpenSource WebRTC IdP  (by Pierpaolo1992)
    https://github.com/w3c/webrtc-pc/issues/1017
  - Inconsistencies in description of RTCDTMFToneChangeEvent.tone (by 
a2sheppy)
    https://github.com/w3c/webrtc-pc/issues/1014
  - "track" event shouldn't fire for a "recvonly" description. (by 
taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1013

  6 issues closed:
  - Ambiguous wording in addIceCandidate 
https://github.com/w3c/webrtc-pc/issues/930
  - offerToReceiveAudio/offerToReceiveVideo remain in implementations 
(likely needed for compat) https://github.com/w3c/webrtc-pc/issues/709
  - Event when a transceiver is stopped via remote action 
https://github.com/w3c/webrtc-pc/issues/962
  - Change SetLocalDescription to require unchanged offer/answer 
string https://github.com/w3c/webrtc-pc/issues/950
  - currentRemoteDescription.sdp -- does it need to match the last SDP
 set via setRemoteDescription? 
https://github.com/w3c/webrtc-pc/issues/921
  - The RTCPeerConnectionIceErrorEvent constructor should have an 
optional init dict https://github.com/w3c/webrtc-pc/issues/857

  12 issues received 21 new comments:
  - #1014 Inconsistencies in description of 
RTCDTMFToneChangeEvent.tone (5 by aboba, alvestrand, a2sheppy)
    https://github.com/w3c/webrtc-pc/issues/1014
  - #687 Clarify reasoning behind and mitigation of privacy issues 
(PING review) (4 by dontcallmedom, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/687
  - #678 Support assertions that identify the recipient (2 by 
Pierpaolo1992, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/678
  - #942 Meta: auto-publish changes to the spec (2 by fluffy, aboba)
    https://github.com/w3c/webrtc-pc/issues/942
  - #930 Ambiguous wording in addIceCandidate (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/930
  - #845 "Throw a FooError" steps not written in a consistent manner 
(1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/845
  - #688 Indicators of usage and data flow (PING review) (1 by 
dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/688
  - #881 Align getAlgorithm return value with Web Crypto (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/881
  - #690 Information available prior to permission prompt (PING 
review) (1 by dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/690
  - #1013 "track" event shouldn't fire for a "recvonly" description. 
(1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1013
  - #954 Receive a track multiple times (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/954
  - #671 Processing remote MediaStreamTracks without MediaStreams info
 (1 by dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/671

* w3c/webrtc-stats (+11/-1/💬33)
  11 issues created:
  - Definitions from MSE need re-targeting (by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/161
  - Stat for adaptation reason (by henbos)
    https://github.com/w3c/webrtc-stats/issues/160
  - Stat for camera input framerate of local video tracks (by henbos)
    https://github.com/w3c/webrtc-stats/issues/159
  - Stats to keep track of sync between audio and video (by henbos)
    https://github.com/w3c/webrtc-stats/issues/158
  - Stat for target and actual encoding bitrate (by henbos)
    https://github.com/w3c/webrtc-stats/issues/155
  - Stat for retransmitted bytes (by henbos)
    https://github.com/w3c/webrtc-stats/issues/154
  - Stat for likelihood of echo (by henbos)
    https://github.com/w3c/webrtc-stats/issues/153
  - Stat for how many audio stream packets are expanded when the user 
is speaking (by henbos)
    https://github.com/w3c/webrtc-stats/issues/152
  - Stat for audio playout delay (by henbos)
    https://github.com/w3c/webrtc-stats/issues/151
  - Stat for how much time it takes to encode video (by henbos)
    https://github.com/w3c/webrtc-stats/issues/150
  - Difference between 'Id' and 'Identifier' fields (by na-g)
    https://github.com/w3c/webrtc-stats/issues/147

  1 issues closed:
  - Stat for target and actual encoding bitrate 
https://github.com/w3c/webrtc-stats/issues/155

  18 issues received 33 new comments:
  - #158 Stats to keep track of sync between audio and video (5 by 
henbos, stefanholmer, icydragons)
    https://github.com/w3c/webrtc-stats/issues/158
  - #160 Stat for adaptation reason (4 by henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/160
  - #141 Unclear which framerate "framePerSecond" represents (3 by 
henbos, taylor-b)
    https://github.com/w3c/webrtc-stats/issues/141
  - #96 Not clear how to differentiate between received connectivity 
checks and consent requests (2 by henbos, taylor-b)
    https://github.com/w3c/webrtc-stats/issues/96
  - #109 RTCCodecStats needs `transportId` and `isRemote` to give it 
context (2 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/109
  - #112 What should 
`availableOutgoingBitrate`/`availableOutgoingBitrate` be for candidate
 pairs not in use? (2 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/112
  - #152 Stat for how many audio stream packets are expanded when the 
user is speaking (2 by henbos)
    https://github.com/w3c/webrtc-stats/issues/152
  - #155 Stat for target and actual encoding bitrate (2 by henbos, 
alvestrand)
    https://github.com/w3c/webrtc-stats/issues/155
  - #159 Stat for camera input framerate of local video tracks (2 by 
henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/159
  - #161 Definitions from MSE need re-targeting (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/161
  - #147 Difference between 'Id' and 'Identifier' fields (1 by 
alvestrand)
    https://github.com/w3c/webrtc-stats/issues/147
  - #146 Stat for audio acceleration rate? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/146
  - #144 Stat for how many adaption changes occur for a video track (1
 by henbos)
    https://github.com/w3c/webrtc-stats/issues/144
  - #105 Why is ssrc a DOMString? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/105
  - #150 Stat for how much time it takes to encode video (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/150
  - #151 Stat for audio playout delay (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/151
  - #153 Stat for likelihood of echo (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/153
  - #154 Stat for retransmitted bytes (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/154



Pull requests
-------------
* w3c/webrtc-pc (+2/-5/💬5)
  2 pull requests submitted:
  - Ensure that "track" event is only fired for "send" direction 
m-sections. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1016
  - Fix inconsistencies in description of RTCDTMFToneChangeEvent.tone 
(by aboba)
    https://github.com/w3c/webrtc-pc/pull/1015

  5 pull requests merged:
  - Add offerToReceive* as legacy extensions
    https://github.com/w3c/webrtc-pc/pull/1005
  - Freeze version of respec-w3c-common.js
    https://github.com/w3c/webrtc-pc/pull/1007
  - errorCode required in RTCPeerConnectionIceErrorEventInit
    https://github.com/w3c/webrtc-pc/pull/1006
  - currentRemoteDescription.sdp need not match remoteDescription.sdp
    https://github.com/w3c/webrtc-pc/pull/1004
  - Change setLocalDescription to require unchanged offer/answer 
string
    https://github.com/w3c/webrtc-pc/pull/1003

  3 pull requests received 5 new comments:
  - #1001 Effect of a BYE on RtpReceiver.track (2 by alvestrand, 
aboba)
    https://github.com/w3c/webrtc-pc/pull/1001
  - #971 Clarify wording on TypeError from addIceCandidate. (2 by 
alvestrand, aboba)
    https://github.com/w3c/webrtc-pc/pull/971
  - #1005 Add offerToReceive* as legacy extensions (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/pull/1005

* w3c/webrtc-stats (+4/-0/💬3)
  4 pull requests submitted:
  - Changes type of RTCRTPStreamStats.ssrc from string to unsigned 
long. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/157
  - frameWidth/frameHeight: use last decoded value (by fippo)
    https://github.com/w3c/webrtc-stats/pull/156
  - Remove references saying "defines an API" (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/149
  - Explanation for "remoteSource" (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/148

  2 pull requests received 3 new comments:
  - #149 Remove references saying "defines an API" (2 by annevk, 
alvestrand)
    https://github.com/w3c/webrtc-stats/pull/149
  - #138 Make a RTCMediaStreamTrackStats object per track attachment 
(1 by henbos)
    https://github.com/w3c/webrtc-stats/pull/138


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 7 February 2017 17:00:47 UTC

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