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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Wed, 08 Mar 2017 09:46:30 +0000
To: public-webrtc@w3.org
Message-Id: <E1clYAw-0002JA-4T@uranus.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

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