public-webrtc-logs@w3.org from February 2019 by subject

[mediacapture-depth] Pull Request: Remove depth map calibration parameters

[mediacapture-depth] Remove depth map calibration parameters (#168)

[mediacapture-main] Add clarification for overconstrained constraints evaluation. (#558)

[mediacapture-main] Add video latency constrainable property (#572)

[mediacapture-main] Define the three collective forms of constraints, to clarify prose (Editorial). (#569)

[mediacapture-main] exclude ConstrainablePattern from reffy. (#570)

[mediacapture-main] Mitigate fingerprinting from OverconstrainedError in gUM(). (#564)

[mediacapture-main] new commits pushed by alvestrand

[mediacapture-main] new commits pushed by jan-ivar

[mediacapture-main] OverconstrainedError should be a DOMException (#568)

[mediacapture-main] Pull Request: Add video latency constrainable property

[mediacapture-main] Pull Request: Define the three collective forms of constraints, to clarify prose.

[mediacapture-main] Pull Request: exclude ConstrainablePattern from reffy.

[mediacapture-main] Pull Request: Make MediaDeviceInfo interface [SecureContext] also.

[mediacapture-main] What constraint name should be exposed in case of a getUserMedia query with multiple failing constraints (#562)

[mediacapture-screen-share] Clarify what audio is captured and what "application" means in the context of restrictOwnAudio (#102)

[mediacapture-screen-share] Which audio constraints from mediacapture-streams are applicable? (#103)

[webrtc-pc] "Create an RTP sender" doesn't handle creating from SDP (#2072)

[webrtc-pc] `ssrc` in `RTCRtpEncodingParameters` is inconsistent with ORTC (#1174)

[webrtc-pc] Add RTCRtpSender.getSynchronizationSources() to expose audioLevel (#2103)

[webrtc-pc] Add video latency constrainable property (#2109)

[webrtc-pc] Add video latency description (#2110)

[webrtc-pc] Add video latency description (fixes: #2109) (#2110)

[webrtc-pc] addTransceiver-transceivers should get associated with offered m= sections (#2108)

[webrtc-pc] Block replaceTrack on pc.close() like the others, not tc.stop(). (#2107)

[webrtc-pc] Clarifications of certificate expiry (#2045)

[webrtc-pc] Clarifying how the simulcast envelope is created. (#2081)

[webrtc-pc] Define a step-by-step generateCertificate algorithm (#2105)

[webrtc-pc] detecting a remote ICE restart? (#1918)

[webrtc-pc] Differences between pc.removeTrack(sender) and sender.replaceTrack(null) (#2024)

[webrtc-pc] Don't require RTCRtpContributingSource.audioLevel to be present in some cases (#2046)

[webrtc-pc] Effect of RTCRtpSendParameters on Simulcast (#1964)

[webrtc-pc] How to handle Glare (#2095)

[webrtc-pc] Inconsistent setting of receiver.track.readyState violates Mediacapture (#1575)

[webrtc-pc] Make RTCErrorEvent's error member non-nullable (#2099)

[webrtc-pc] new commits pushed by aboba

[webrtc-pc] new commits pushed by alvestrand

[webrtc-pc] Obtain user consent for one-way media and data use cases (#2012)

[webrtc-pc] Order of RTCError parameters (#2111)

[webrtc-pc] Pull Request: Add transceiver created by SRD to the set of transceiver

[webrtc-pc] Pull Request: Add video latency description (fixes: #2109)

[webrtc-pc] Pull Request: Block replaceTrack on pc.close() like the others, not tc.stop().

[webrtc-pc] Pull Request: Define a step-by-step generateCertificate algorithm

[webrtc-pc] Pull Request: Fix duplicate ids for overloaded RTCDataChannel.send

[webrtc-pc] Pull Request: Make RTCErrorEvent's error member non-nullable

[webrtc-pc] Pull Request: Remove stopped transceivers from the set of transceivers

[webrtc-pc] Pull Request: Upgrade document to respec 24.3.0

[webrtc-pc] Pull Request: User agents MUST send audioLevel. If other endpoint is legacy, don't calculate it.

[webrtc-pc] Remove stopped transceivers from getTransceivers() (#2092)

[webrtc-pc] Remove stopped transceivers from the set of transceivers (#2102)

[webrtc-pc] Rollback to address Glare (#2095)

[webrtc-pc] RTCErrorEvent constructor's RTCErrorEventInit parameter should be optional (#2070)

[webrtc-pc] RTCErrorEvent/RTCErrorEventInit: Error should be required (#2094)

[webrtc-pc] RTCIceTransport.component is at the wrong abstraction layer (#1916)

[webrtc-pc] RTCRtpReceiver.getParameters() clarification (#2032)

[webrtc-pc] RTCRtpReceiver.getStreams() and RTCRtpSender.getStreams() (#1921)

[webrtc-pc] RTCRtpTransceivers created by SDL/SRD are never added to the set of transceivers (#2100)

[webrtc-pc] transceiver.stop() shouldn't block in-progress replaceTrack from succeeding (#2106)

[webrtc-pc] typo on 4.4.1.1 section 3: "a the [[Origin]]" (#2093)

[webrtc-pc] User agents MUST send audioLevel. If other endpoint is legacy, don't calculate it. (#2104)

[webrtc-pc] WebRTC-PC: respec errors (#2096)

[webrtc-pc] What happens when an answerer stops a transceiver that others are "bundled" on? (#1858)

[webrtc-pc] WHATWG streams for data channel messages (#1732)

[webrtc-stats] [DataChannels] Use RTT from sctp in the stats (#376)

[webrtc-stats] add a graph showing the relationship of stats to the spec (#348)

[webrtc-stats] Add RTCInboundStreamStats.fractionLost to obsolete stats (#389)

[webrtc-stats] audioLevel can be removed from "track" stats (#391)

[webrtc-stats] audioLevel can be removed from "track"|"receiver"|"sender" stats (#391)

[webrtc-stats] framesDecoded is on both "receiver" and "inbound-rtp" stats. Redundant. (#392)

[webrtc-stats] Move (fir|pli|nack|sli)Count and qpSum out of baseclass to (in|out)bound (#384)

[webrtc-stats] Move {fir|pli|nack|sli}Count and qpSum out of baseclass to {in|out}bound (Editorial) (#384)

[webrtc-stats] Need to add the fractionLost into the dictionary RTCInboundRtpStreamStats (#388)

[webrtc-stats] new commits pushed by alvestrand

[webrtc-stats] new commits pushed by vr000m

[webrtc-stats] Pull Request: Add RTCInboundStreamStats.fractionLost to obsolete stats

[webrtc-stats] Pull Request: initial commit for webrtc-stats-testing

[webrtc-stats] Rename nackCount to nacksReceived & nacksSent respectively? (#390)

[webrtc-stats] Work through implications of simulcast (#318)

Closed: [mediacapture-screen-share] Constraint to exclude application audio (echo) (#79)

Closed: [webrtc-pc] "Create an RTP sender" doesn't handle creating from SDP (#2072)

Closed: [webrtc-pc] detecting a remote ICE restart? (#1918)

Closed: [webrtc-pc] DTLSTransport: No description of when to fire the error event (#2044)

Closed: [webrtc-pc] Effect of RTCRtpSendParameters on Simulcast (#1964)

Closed: [webrtc-pc] How to handle Glare (#2095)

Closed: [webrtc-pc] It is not clear that RTCErrorDetailType is referencing things defined in another repo (#2083)

Closed: [webrtc-pc] Not clear how to set # of layers when answering an offer with a track (#2014)

Closed: [webrtc-pc] RTCErrorEvent constructor's RTCErrorEventInit parameter should be optional (#2070)

Closed: [webrtc-pc] RTCErrorEvent/RTCErrorEventInit: Error should be required (#2094)

Closed: [webrtc-pc] RTCIceTransport.component is at the wrong abstraction layer (#1916)

Closed: [webrtc-pc] RTCRtpContributingSource.audioLevel should be present, but if it isn't due to legacy endpoint, don't calculate it (#2046)

Closed: [webrtc-pc] RTCRtpReceiver.getParameters() clarification (#2032)

Closed: [webrtc-pc] RTCRtpTransceivers created by SDL/SRD are never added to the set of transceivers (#2100)

Closed: [webrtc-pc] transceiver.stop() shouldn't block in-progress replaceTrack from succeeding (#2106)

Closed: [webrtc-pc] typo on 4.4.1.1 section 3: "a the [[Origin]]" (#2093)

Closed: [webrtc-pc] WebRTC-PC: respec errors (#2096)

Closed: [webrtc-pc] What happens when an answerer stops a transceiver that others are "bundled" on? (#1858)

Closed: [webrtc-stats] Add stat for RTTs between client and STUN/TURN server (#339)

Closed: [webrtc-stats] Need to add the fractionLost into the dictionary RTCInboundRtpStreamStats (#388)

Closed: [webrtc-stats] video packet counters should be moved out from RTCRTPStreamStats to appropriate dictionaries (#312)

Last message date: Thursday, 28 February 2019 23:46:49 UTC