- 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