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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 16 May 2017 17:00:38 +0000
To: public-webrtc@w3.org
Message-Id: <E1dAfpu-0007oB-8A@uranus.w3.org>

Issues
------
* w3c/webrtc-pc (+25/-6/💬51)
  25 issues created:
  - Define the unit used for framerates and maxframerates (by fluffy)
    https://github.com/w3c/webrtc-pc/issues/1214
  - addIceCandidate end of candidates woes (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1213
  - Can we directly link to WebRTC-Stats for RTCStatsType? (by SaschaNaz)
    https://github.com/w3c/webrtc-pc/issues/1212
  - RTCIceCandidate: how is extensibility handled? (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1211
  - ufrag vs usernameFragment (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1210
  - Setting a remote description may cause discontinuity due to codec switching. (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1207
  - transceiver.direction isn't initialized. (by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1206
  - Sending data channel messages > maxMessageSize (by lgrahl)
    https://github.com/w3c/webrtc-pc/issues/1205
  - setDirection needs an internal slot and an algorithm. (by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1204
  - Need to identify if changes affect JSEP or MSID specs (by stefhak)
    https://github.com/w3c/webrtc-pc/issues/1202
  - what is the component for a icetransport of a datachannel? (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1201
  - editorial: move "set a configuration" to operation section (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1200
  - RTCPeerConnectionIceEvent parameter: RTCPeerConnectionIceEventInit eventInitDict is optional (by rwaldron)
    https://github.com/w3c/webrtc-pc/issues/1199
  - RTCIceServer "represents a TURN server" (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1198
  - Stream IDs MUST be generated according to mediacapture-main, but webrtc-pc contradicts this (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1196
  - iceCandidatePoolSize type mismatch (minor spec bug) (by rwaldron)
    https://github.com/w3c/webrtc-pc/issues/1195
  - audioLevel of tracks, both send and receive? (by henbos)
    https://github.com/w3c/webrtc-pc/issues/1194
  - editorial: createOffer description (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1193
  - editorial: currentLocalDescription getting updates with candidates (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1192
  - update the ice gathering state: firing icecandidate with null is a legacy hack (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1191
  - editorial: add subsubsubsections to operation description? (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1190
  - editorial: 4.3.1 Operation "settled" is not defined (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1189
  - editorial: terminology of Blob and "blob of SDP" (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1186
  - editorial: p2p data transfer is a facet of video conferencing (by fippo)
    https://github.com/w3c/webrtc-pc/issues/1185
  - Race condition between createOffer and setIdentityProvider (by taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1184

  16 issues received 51 new comments:
  - #1166 RTCIceCandidate constructor should not require sdpMid/sdpMLineIndex (13 by foolip, stefhak, taylor-b, soareschen, rwaldron)
    https://github.com/w3c/webrtc-pc/issues/1166
  - #1128 does MSID still work? (was: example 13: getReceivers semantics) (11 by taylor-b, fippo, jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1128
  - #1161 Do we need a "trackremoved" event? (3 by Pehrsons, jan-ivar, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1161
  - #1165 Lack of validation in RTCIceCandidate constructor (3 by foolip, stefhak, rwaldron)
    https://github.com/w3c/webrtc-pc/issues/1165
  - #1184 Race condition between createOffer and setIdentityProvider (3 by martinthomson, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1184
  - #1196 Stream IDs MUST be generated according to mediacapture-main, but webrtc-pc contradicts this (3 by taylor-b, fippo, jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1196
  - #1125 Should the spec describe addStream/onaddstream as legacy API? (3 by foolip, stefhak, youennf)
    https://github.com/w3c/webrtc-pc/issues/1125
  - #1178 Need to describe when ICE and DTLS transport objects are created/changed (2 by fippo, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1178
  - #1182 Spec-compliant way to get remote streams and tracks? (2 by henbos, fippo)
    https://github.com/w3c/webrtc-pc/issues/1182
  - #1201 what is the component for a icetransport of a datachannel? (2 by fippo, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1201
  - #1177 Some JSEP "applying a description" steps can't occur in parallel. (1 by burnburn)
    https://github.com/w3c/webrtc-pc/issues/1177
  - #1181 End removed tracks remotely again; Make receiver.track nullable instead (1 by stefhak)
    https://github.com/w3c/webrtc-pc/issues/1181
  - #1193 editorial: createOffer description (1 by fippo)
    https://github.com/w3c/webrtc-pc/issues/1193
  - #1194 audioLevel of tracks, both send and receive? (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1194
  - #1198 RTCIceServer "represents a TURN server" (1 by fippo)
    https://github.com/w3c/webrtc-pc/issues/1198
  - #1213 addIceCandidate end of candidates woes (1 by fippo)
    https://github.com/w3c/webrtc-pc/issues/1213

  6 issues closed:
  - type of maxFramerate https://github.com/w3c/webrtc-pc/issues/1162
  - editorial: terminology of Blob and "blob of SDP" https://github.com/w3c/webrtc-pc/issues/1186
  - iceCandidatePoolSize type mismatch (minor spec bug) https://github.com/w3c/webrtc-pc/issues/1195
  - legacy createAnswer: support for answerOptions? https://github.com/w3c/webrtc-pc/issues/1152
  - Related attributes should be updated at the same time (in the same microtask checkpoint?) https://github.com/w3c/webrtc-pc/issues/1124
  - what is the component for a icetransport of a datachannel? https://github.com/w3c/webrtc-pc/issues/1201

* w3c/webrtc-stats (+5/-13/💬14)
  5 issues created:
  - Audio/Video sync follow-up (by henbos)
    https://github.com/w3c/webrtc-stats/issues/222
  - Need some way of getting an averaged audio level stat (by taylor-b)
    https://github.com/w3c/webrtc-stats/issues/220
  - Issues raised in TAG review (by dontcallmedom)
    https://github.com/w3c/webrtc-stats/issues/219
  - RTCMediaStreamTrackStats.audioLevel: update description (by henbos)
    https://github.com/w3c/webrtc-stats/issues/205
  - RTCMediaStreamTrackStats.voiceActivityFlag? (by henbos)
    https://github.com/w3c/webrtc-stats/issues/204

  6 issues received 14 new comments:
  - #193 RTCMediaStreamTrackStats.audioLevel clarification (4 by henbos, taylor-b)
    https://github.com/w3c/webrtc-stats/issues/193
  - #133 Need DSCP information for outgoing RTP streams (3 by vr000m, taylor-b, lennart-csio)
    https://github.com/w3c/webrtc-stats/issues/133
  - #222 Audio/Video sync follow-up (3 by henbos)
    https://github.com/w3c/webrtc-stats/issues/222
  - #108 Should document what RTCRTPStreamStats metrics are valid when `isRemote` is true. (2 by vr000m, taylor-b)
    https://github.com/w3c/webrtc-stats/issues/108
  - #204 RTCMediaStreamTrackStats.voiceActivityFlag? (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/204
  - #158 Stats to keep track of sync between audio and video (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/158

  13 issues closed:
  - Stat for audio playout delay https://github.com/w3c/webrtc-stats/issues/151
  - Add stat to RTCTransportStats for ICE role (controlling or controlled) https://github.com/w3c/webrtc-stats/issues/162
  - Stats report for RTCRtpContributingSource objects https://github.com/w3c/webrtc-stats/issues/183
  - Unclear which framerate "framePerSecond" represents https://github.com/w3c/webrtc-stats/issues/141
  - Should document what RTCRTPStreamStats metrics are valid when `isRemote` is true. https://github.com/w3c/webrtc-stats/issues/108
  - Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. https://github.com/w3c/webrtc-stats/issues/189
  - Need priority information for MediaStreamTracks https://github.com/w3c/webrtc-stats/issues/132
  - Explain DomHighResTimestamp once in the introduction. https://github.com/w3c/webrtc-stats/issues/200
  - Make sure DOMHighResTimeStamp has proper reference. https://github.com/w3c/webrtc-stats/issues/118
  - RTCMediaStreamTrackStats.audioLevel: update description https://github.com/w3c/webrtc-stats/issues/205
  - Stats to keep track of sync between audio and video https://github.com/w3c/webrtc-stats/issues/158
  - Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) https://github.com/w3c/webrtc-stats/issues/152
  - RTCCodecStats needs `transportId` and `isRemote` to give it context https://github.com/w3c/webrtc-stats/issues/109



Pull requests
-------------
* w3c/webrtc-pc (+6/-8/💬12)
  6 pull requests submitted:
  - Throw error if data channel's buffer is filled, rather than closing. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1209
  - Add bufferSize attribute to RTCDataChannel. (by taylor-b)
    https://github.com/w3c/webrtc-pc/pull/1208
  - Invalid RTCRtpTransceiverDirection already throws TypeError. (by jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1203
  - Fix the type of iceCandidatePoolSize in description (by foolip)
    https://github.com/w3c/webrtc-pc/pull/1197
  - editorial: introduction: explain signaling process (by fippo)
    https://github.com/w3c/webrtc-pc/pull/1188
  - createAnswer: disambiguate blob (by fippo)
    https://github.com/w3c/webrtc-pc/pull/1187

  10 pull requests received 12 new comments:
  - #1108 Update structured cloning for recent changes to HTML (3 by domenic, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1108
  - #1160 Remove getAlgorithm() (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1160
  - #1163 Update RTCCertificate ref to get serializer from WebIDL (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1163
  - #1171 revert PR#1108 (1 by domenic)
    https://github.com/w3c/webrtc-pc/pull/1171
  - #1172 Adding note about legacy `createAnswer` not supporting options dict. (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1172
  - #1175 Expanding RTCPeerConnection introduction. (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1175
  - #1176 Adding more detail to RTCIceTransportPolicy enum descriptions. (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1176
  - #1180 maxFramerate type double (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1180
  - #1187 createAnswer: disambiguate blob (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1187
  - #1197 Fix the type of iceCandidatePoolSize in description (1 by stefhak)
    https://github.com/w3c/webrtc-pc/pull/1197

  8 pull requests merged:
  - editorial: introduction: explain signaling process
    https://github.com/w3c/webrtc-pc/pull/1188
  - Fix the type of iceCandidatePoolSize in description
    https://github.com/w3c/webrtc-pc/pull/1197
  - createAnswer: disambiguate blob
    https://github.com/w3c/webrtc-pc/pull/1187
  - maxFramerate type double
    https://github.com/w3c/webrtc-pc/pull/1180
  - Adding more detail to RTCIceTransportPolicy enum descriptions.
    https://github.com/w3c/webrtc-pc/pull/1176
  - Expanding RTCPeerConnection introduction.
    https://github.com/w3c/webrtc-pc/pull/1175
  - Adding note about legacy `createAnswer` not supporting options dict.
    https://github.com/w3c/webrtc-pc/pull/1172
  - Add paragraph about RtpContributingSources being updated simultaneously.
    https://github.com/w3c/webrtc-pc/pull/1149

* w3c/webrtc-stats (+15/-11/💬20)
  15 pull requests submitted:
  - Adding audio level stat that can be used to compute averages. (by taylor-b)
    https://github.com/w3c/webrtc-stats/pull/221
  - Adds RTCTransportStats.iceRole (by henbos)
    https://github.com/w3c/webrtc-stats/pull/218
  - jitter buffer delay added to media stream tracks (by vr000m)
    https://github.com/w3c/webrtc-stats/pull/217
  - stats for number of circuit breaker triggered (by vr000m)
    https://github.com/w3c/webrtc-stats/pull/216
  - RTCMediaStreamTrackStats.concealedAudibleSamples added. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/215
  - adding total consent interval (by vr000m)
    https://github.com/w3c/webrtc-stats/pull/214
  - RTCIceCandidatePairStats.last[Request/Response]Timestamp added (by henbos)
    https://github.com/w3c/webrtc-stats/pull/213
  - Issue 132/mediastreamtrack priority (by lennart-csio)
    https://github.com/w3c/webrtc-stats/pull/212
  - timestamps need for #61 (by vr000m)
    https://github.com/w3c/webrtc-stats/pull/211
  - Add paragraph about timestamp and reference DOMHighResTimeStamp (by henbos)
    https://github.com/w3c/webrtc-stats/pull/210
  - definition for 'framesPerSecond' (by karthikbr82)
    https://github.com/w3c/webrtc-stats/pull/209
  - RTCMediaStreamTrackStats.voiceActivityFlag added. (by henbos)
    https://github.com/w3c/webrtc-stats/pull/208
  - RTCMediaStreamTrackStats.audioLevel: Reference RTCRtpSynchronizationSource (by henbos)
    https://github.com/w3c/webrtc-stats/pull/207
  - packetsRetransmitted and bytesRetransmitted to RTCSentRTPStreamStats (by karthikbr82)
    https://github.com/w3c/webrtc-stats/pull/206
  - Issue 178/packets discarded on send (by lennart-csio)
    https://github.com/w3c/webrtc-stats/pull/203

  9 pull requests received 20 new comments:
  - #214 adding total consent interval (4 by henbos, vr000m, taylor-b)
    https://github.com/w3c/webrtc-stats/pull/214
  - #194 Adding RTCRTPContributingSourceStats stats report object. (3 by henbos, vr000m, taylor-b)
    https://github.com/w3c/webrtc-stats/pull/194
  - #217 jitter buffer delay added to media stream tracks (3 by henbos, vr000m)
    https://github.com/w3c/webrtc-stats/pull/217
  - #213 RTCIceCandidatePairStats.last[Request/Response]Timestamp added (3 by henbos, vr000m)
    https://github.com/w3c/webrtc-stats/pull/213
  - #215 RTCMediaStreamTrackStats.concealedAudibleSamples added. (2 by henbos, vr000m)
    https://github.com/w3c/webrtc-stats/pull/215
  - #185 Added RTCInboundRTPStreamStats sample counters. (2 by vr000m, jan-ivar)
    https://github.com/w3c/webrtc-stats/pull/185
  - #207 RTCMediaStreamTrackStats.audioLevel: Reference RTCRtpSynchronizationSource (1 by vr000m)
    https://github.com/w3c/webrtc-stats/pull/207
  - #211 timestamps need for #61 (1 by vr000m)
    https://github.com/w3c/webrtc-stats/pull/211
  - #218 Adds RTCTransportStats.iceRole (1 by vr000m)
    https://github.com/w3c/webrtc-stats/pull/218

  11 pull requests merged:
  - jitter buffer delay added to media stream tracks
    https://github.com/w3c/webrtc-stats/pull/217
  - stats for number of circuit breaker triggered
    https://github.com/w3c/webrtc-stats/pull/216
  - Adds RTCTransportStats.iceRole
    https://github.com/w3c/webrtc-stats/pull/218
  - Adding RTCRTPContributingSourceStats stats report object.
    https://github.com/w3c/webrtc-stats/pull/194
  - definition for 'framesPerSecond'
    https://github.com/w3c/webrtc-stats/pull/209
  - Issue 132/mediastreamtrack priority
    https://github.com/w3c/webrtc-stats/pull/212
  - Add paragraph about timestamp and reference DOMHighResTimeStamp
    https://github.com/w3c/webrtc-stats/pull/210
  - RTCMediaStreamTrackStats.audioLevel: Reference RTCRtpSynchronizationSource
    https://github.com/w3c/webrtc-stats/pull/207
  - RTCMediaStreamTrackStats members to keep track of audio and video sync.
    https://github.com/w3c/webrtc-stats/pull/187
  - Added RTCInboundRTPStreamStats sample counters.
    https://github.com/w3c/webrtc-stats/pull/185
  - Adding "codec type" and transportId to RTCCodecStats.
    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, 16 May 2017 17:00:46 UTC

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