- 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