- From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
- Date: Wed, 08 Mar 2017 09:46:30 +0000
- To: public-webrtc@w3.org
Issues
------
* w3c/webrtc-pc (+11/-11/💬28)
11 issues created:
- 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
- Section 11.6: Issue 6 (by aboba)
https://github.com/w3c/webrtc-pc/issues/1053
- Section 10.3: Issue 5 (by aboba)
https://github.com/w3c/webrtc-pc/issues/1052
- Section 10.3: mute signal (by aboba)
https://github.com/w3c/webrtc-pc/issues/1051
- Section 5.6: Generation of candidates (by aboba)
https://github.com/w3c/webrtc-pc/issues/1050
- Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP (by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1049
- Range checking needed for iceCandidatePoolSize (by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1048
- RTCDataChannelInit.id should use [EnforceRange] (by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1046
- Section 12.2.1.1: RTCError errorDetailEnum (by aboba)
https://github.com/w3c/webrtc-pc/issues/1044
- Section 13: Connecting event (by aboba)
https://github.com/w3c/webrtc-pc/issues/1043
11 issues closed:
- Section 10.3: https://github.com/w3c/webrtc-pc/issues/1062
- Clarify reasoning behind and mitigation of privacy issues (PING review) https://github.com/w3c/webrtc-pc/issues/687
- Guidance for extending objects vs extending Stats needed https://github.com/w3c/webrtc-pc/issues/295
- Align getAlgorithm return value with Web Crypto https://github.com/w3c/webrtc-pc/issues/881
- Advanced Peer-to-peer Example https://github.com/w3c/webrtc-pc/issues/959
- RTCStats timestamp source ambiguous https://github.com/w3c/webrtc-pc/issues/729
- Need to specify precisely when MID generation happens https://github.com/w3c/webrtc-pc/issues/578
- Integrate RTCRtpTransceiver into set local/remote steps https://github.com/w3c/webrtc-pc/issues/787
- Specify when a data channel's ID is assigned, and what the `id` attribute returns when no ID is assigned. https://github.com/w3c/webrtc-pc/issues/795
- Need to describe that codecs can be removed or reordered, but not modified https://github.com/w3c/webrtc-pc/issues/1024
- Describe what happens when media changes https://github.com/w3c/webrtc-pc/issues/305
13 issues received 28 new comments:
- #1051 Section 10.3: mute signal (5 by taylor-b, aboba)
https://github.com/w3c/webrtc-pc/issues/1051
- #881 Align getAlgorithm return value with Web Crypto (5 by foolip, aboba, alvestrand)
https://github.com/w3c/webrtc-pc/issues/881
- #1048 Range checking needed for iceCandidatePoolSize (3 by foolip, fluffy, taylor-b)
https://github.com/w3c/webrtc-pc/issues/1048
- #1052 Section 10.3: Issue 5 (3 by taylor-b, aboba)
https://github.com/w3c/webrtc-pc/issues/1052
- #1043 Section 13: Connecting event (2 by aboba)
https://github.com/w3c/webrtc-pc/issues/1043
- #1053 Section 11.6: Issue 6 (2 by fippo, aboba)
https://github.com/w3c/webrtc-pc/issues/1053
- #1020 Don't fire events on a closed peer connection (2 by adam-be, alvestrand)
https://github.com/w3c/webrtc-pc/issues/1020
- #1050 Section 5.6: Generation of candidates (1 by aboba)
https://github.com/w3c/webrtc-pc/issues/1050
- #295 Guidance for extending objects vs extending Stats needed (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/295
- #687 Clarify reasoning behind and mitigation of privacy issues (PING review) (1 by dontcallmedom)
https://github.com/w3c/webrtc-pc/issues/687
- #959 Advanced Peer-to-peer Example (1 by adam-be)
https://github.com/w3c/webrtc-pc/issues/959
- #849 Specify an AllowUnverifiedMedia RTCConfiguration property (1 by alvestrand)
https://github.com/w3c/webrtc-pc/issues/849
- #1022 sdpFmtpLine isn't very convenient (1 by taylor-b)
https://github.com/w3c/webrtc-pc/issues/1022
* w3c/webrtc-stats (+3/-3/💬16)
3 issues created:
- RoundTripTime not defined when the underlying stream can't calculate it (by jesup)
https://github.com/w3c/webrtc-stats/issues/180
- Packets or frames discarded on send? (by henbos)
https://github.com/w3c/webrtc-stats/issues/178
- Discuss caching and consistency of getStats() return (by alvestrand)
https://github.com/w3c/webrtc-stats/issues/177
3 issues closed:
- 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
- Track stats: track or attachment? https://github.com/w3c/webrtc-stats/issues/130
7 issues received 16 new comments:
- #180 RoundTripTime not defined when the underlying stream can't calculate it (8 by henbos, vr000m, jan-ivar)
https://github.com/w3c/webrtc-stats/issues/180
- #134 Timestamp in the getStats (2 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/134
- #150 Stat for how much time it takes to encode video (2 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/150
- #160 Stat for adaptation reason (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/160
- #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
- #153 Stat for likelihood of echo (1 by alvestrand)
https://github.com/w3c/webrtc-stats/issues/153
Pull requests
-------------
* w3c/webrtc-pc (+12/-8/💬19)
12 pull requests submitted:
- Remove connecting event from Event summary (by aboba)
https://github.com/w3c/webrtc-pc/pull/1061
- ended event fired on the track (by aboba)
https://github.com/w3c/webrtc-pc/pull/1060
- mute signal (by aboba)
https://github.com/w3c/webrtc-pc/pull/1059
- Generation of candidates (by aboba)
https://github.com/w3c/webrtc-pc/pull/1058
- Add clockrate, channels, sdpFmtpLine to codec capability (by aboba)
https://github.com/w3c/webrtc-pc/pull/1057
- Switching to new, consistent terminology when talking about exceptions. (by taylor-b)
https://github.com/w3c/webrtc-pc/pull/1056
- Changing iceCandidatePoolSize to an octet and adding EnforceRange. (by taylor-b)
https://github.com/w3c/webrtc-pc/pull/1055
- Throw InvalidModificationError if changing pool size after SLD. (by taylor-b)
https://github.com/w3c/webrtc-pc/pull/1054
- Adding "[EnforceRange]" to RTCDataChannelInit.id. (by taylor-b)
https://github.com/w3c/webrtc-pc/pull/1047
- Clarifying exactly what "sdpFmtpLine" represents. (by taylor-b)
https://github.com/w3c/webrtc-pc/pull/1045
- Add clockrate, channels, sdpFmtpLine to codec capability (by aboba)
https://github.com/w3c/webrtc-pc/pull/1042
- Label 'Warm-up example' as 'advanced p2p example' (by adam-be)
https://github.com/w3c/webrtc-pc/pull/1041
8 pull requests merged:
- Don't fire events on a closed peer connection
https://github.com/w3c/webrtc-pc/pull/1031
- Label 'Warm-up example' as 'advanced p2p example'
https://github.com/w3c/webrtc-pc/pull/1041
- Clarify which timestamp RTCStats.timestamp represents.
https://github.com/w3c/webrtc-pc/pull/1039
- Specify when random mid generation happens
https://github.com/w3c/webrtc-pc/pull/1037
- Specify how transceivers get their mids in setLocal/Remote
https://github.com/w3c/webrtc-pc/pull/1036
- Make RTCDataChannel.id nullable and describe when it's set.
https://github.com/w3c/webrtc-pc/pull/1038
- Mention that codecs can be reordered or removed but not modified.
https://github.com/w3c/webrtc-pc/pull/1025
- Specify how media is centered, cropped, and scaled. Fixes #305
https://github.com/w3c/webrtc-pc/pull/1023
15 pull requests received 19 new comments:
- #1042 Add clockrate, channels, sdpFmtpLine to codec capability (2 by ibc, aboba)
https://github.com/w3c/webrtc-pc/pull/1042
- #1045 Clarifying exactly what "sdpFmtpLine" represents. (2 by henbos, aboba)
https://github.com/w3c/webrtc-pc/pull/1045
- #1055 Changing iceCandidatePoolSize to an octet and adding EnforceRange. (2 by jan-ivar, aboba)
https://github.com/w3c/webrtc-pc/pull/1055
- #1023 Specify how media is centered, cropped, and scaled. Fixes #305 (2 by fluffy, aboba)
https://github.com/w3c/webrtc-pc/pull/1023
- #1025 Mention that codecs can be reordered or removed but not modified. (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1025
- #1030 Add stats selection algorithm based on sender or receiver of selector. (1 by jan-ivar)
https://github.com/w3c/webrtc-pc/pull/1030
- #1031 Don't fire events on a closed peer connection (1 by adam-be)
https://github.com/w3c/webrtc-pc/pull/1031
- #1035 Parameters for packetization interval (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1035
- #1037 Specify when random mid generation happens (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1037
- #1038 Make RTCDataChannel.id nullable and describe when it's set. (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1038
- #1039 Clarify which timestamp RTCStats.timestamp represents. (1 by alvestrand)
https://github.com/w3c/webrtc-pc/pull/1039
- #1041 Label 'Warm-up example' as 'advanced p2p example' (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1041
- #1047 Adding "[EnforceRange]" to RTCDataChannelInit.id. (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1047
- #1054 Throw InvalidModificationError if changing pool size after SLD. (1 by aboba)
https://github.com/w3c/webrtc-pc/pull/1054
- #990 Add an explicit stats selection algorithm. (1 by alvestrand)
https://github.com/w3c/webrtc-pc/pull/990
* w3c/webrtc-stats (+2/-3/💬4)
2 pull requests submitted:
- Adds definitions for RTCDataChannelStats members. (by alvestrand)
https://github.com/w3c/webrtc-stats/pull/179
- Add link from datachannel to transport (by alvestrand)
https://github.com/w3c/webrtc-stats/pull/176
3 pull requests merged:
- 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
- Make a RTCMediaStreamTrackStats object per track attachment
https://github.com/w3c/webrtc-stats/pull/138
3 pull requests received 4 new comments:
- #176 Add link from datachannel to transport (2 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/176
- #164 Adding remoteTimestamp to RTCRtpStreamStats. (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/164
- #165 Remove separation of received consent and connectivity requests. (1 by alvestrand)
https://github.com/w3c/webrtc-stats/pull/165
Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Wednesday, 8 March 2017 09:46:36 UTC