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

[mediacapture-fromelement] Don't run the steps for MediaStreamTrack.stop() when ending a track (#81)

[mediacapture-fromelement] Is mute event expected to be dispatched at MediaStreamTrack when enabled is set to false? (#80)

[mediacapture-image] Extend getPhotoSettings() to include additional EXIF data (#209)

[mediacapture-main] Clarification needed on HTMLMediaElement attributes that carry over (#599)

[mediacapture-main] Clarify language around element attributes after stream removal. (#600)

[mediacapture-main] Echo Cancellation (#597)

[mediacapture-main] Interpret the empty string as if the constraint is not specified (#595)

[mediacapture-main] MediaStream in HTMLMediaElement: Does it have a first frame? (#555)

[mediacapture-main] MediaTrackSettings lacks support of channel layout (#605)

[mediacapture-main] new commits pushed by aboba

[mediacapture-main] new commits pushed by alvestrand

[mediacapture-main] oldList definition in enumerateDevices is unclear (#604)

[mediacapture-main] Pull Request: Add definition to track-ending algorithm

[mediacapture-main] Pull Request: Create FUNDING.yml

[mediacapture-main] Pull Request: Filter enumerateDevices by feature policy.

[mediacapture-main] Pull Request: Reference to WHATWG HTML LS

[mediacapture-main] Recommend UUID for deviceId and groupId (#591)

[mediacapture-main] Should deviceId be partitioned by top level origin and document origin (#598)

[mediacapture-main] Specify a way for WebDriver to add/remove/setup capture devices (#554)

[mediacapture-main] When an HTMLMediaElement has srcObject set to a MediaStream where the MediaStreamTrack is muted the HTMLMediaElement muted attribute should be set to muted to reflect muted input and muted output (#583)

[mediacapture-output] Normative reference to HTML 5.2 should be changed to HTML LS (#82)

[mediacapture-record] Clarification needed for mime type handling (#170)

[mediacapture-record] Clarify MediaRecorder's mimeType handling (#171)

[mediacapture-record] Define behavior for all bitsPerSecond attributes (#175)

[mediacapture-record] Do not discard data in case track set change (#162)

[mediacapture-record] Input video track width and height MUST be recorded and playable (#173)

[mediacapture-record] Input width, height MUST be recorded and playable (#172)

[mediacapture-record] Pull Request: (Editorial) Make timeslice unsigned everywhere

[mediacapture-record] Pull Request: Clarify MediaRecorder's mimeType handling

[mediacapture-record] Pull Request: Define behavior for all bitsPerSecond attributes

[mediacapture-record] Pull Request: Input width, height MUST be recorded and playable

[mediacapture-record] What are the default bitrate attribute values? (#169)

[mediacapture-record] Why is start 5.3 in the specification? (#168)

[mediacapture-screen-share] Provide a means to select only part of a screen to capture (#105)

[mediacapture-screen-share] Pull Request: Remove See Work Flow and bad links

[mediacapture-screen-share] Remove See Work Flow and bad links (#109)

[mediacapture-screen-share] Screenshot (still image) capability (#107)

[mediacapture-screen-share] Should the permissions UI be the first captured frame? (#108)

[webrtc-pc] Add a security/privacy note about remote SDP (#2193)

[webrtc-pc] Add reject() + make stop() safe. New stopping state affects createOffer only. (#2220)

[webrtc-pc] Add security note about remote SDP (#2201)

[webrtc-pc] Add support for WebRTC Data Channel in Workers (#230)

[webrtc-pc] Clarify/strengthen timing invariants of candidates not firing before SLD(offer) (#2199)

[webrtc-pc] DataChannel max value for "id" before connecting? (#2158)

[webrtc-pc] Define which constraints are applicable in WebRTC (#2188)

[webrtc-pc] Don't rely on event.streams in Example 11. (#2218)

[webrtc-pc] Encode and Decode error event (#2219)

[webrtc-pc] Example 11 expects event.streams[0] from streamless tracks (#2217)

[webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213)

[webrtc-pc] Make DOMException references cite [WEBIDL] (#2200)

[webrtc-pc] new commits pushed by aboba

[webrtc-pc] new commits pushed by alvestrand

[webrtc-pc] new commits pushed by henbos

[webrtc-pc] new commits pushed by jan-ivar

[webrtc-pc] Order of multiple RTCDataChannel.send calls in case of blobs (#2215)

[webrtc-pc] Permission API for receive-only media and data use cases (#2175)

[webrtc-pc] Pull Request: Add reject() + make stop() safe. New stopping state affects createOffer only.

[webrtc-pc] Pull Request: Add security note about remote SDP

[webrtc-pc] Pull Request: Add setRemoteDescription(description, {rollback: true}) option.

[webrtc-pc] Pull Request: Clarify that setting active to false does not send RTCP BYE

[webrtc-pc] Pull Request: Define an "unknown" value for RTCIceRole

[webrtc-pc] Pull Request: Don't rely on event.streams in Example 11.

[webrtc-pc] Pull Request: Ensure icecandidate events happen after SLD(offer).

[webrtc-pc] Pull Request: Fix outdated information in mandatory stats list.

[webrtc-pc] Pull Request: Make DOMException references cite [WEBIDL]

[webrtc-pc] Pull Request: Read [[LastCreatedOffer/Answer]] at the right time in SLD

[webrtc-pc] Pull Request: Specify that "id" is ignored when negotiated=false.

[webrtc-pc] Pull Request: Use WHATWG references for HTML

[webrtc-pc] Read [[LastCreatedOffer/Answer]] at the right time in SLD (#2209)

[webrtc-pc] Remove [WEBIDL-1] reference in favor of [WEBIDL] (#2207)

[webrtc-pc] RTCDataChannel.bufferedAmount value (#1823)

[webrtc-pc] RTCIceTransport role before start() called (#2210)

[webrtc-pc] Setting active attribute to false triggers BYE? (#2202)

[webrtc-pc] SLD without {sdp} reads [[LastCreatedOffer/Answer]] at the wrong time (#2208)

[webrtc-pc] Specify when RTCIceRole is updated (#2214)

[webrtc-pc] transceiver.stop() needs more work (avoid BUNDLE footgun) (#2150)

[webrtc-pc] {iceRestart: true} works poorly with negotiationneeded (#2167)

[webrtc-stats] Add receiving "track" stats to the obsolete section (#452)

[webrtc-stats] Add remoteSource to "media-source" (#453)

[webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#450)

[webrtc-stats] Audio samples and channels (#448)

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

[webrtc-stats] Clarify qualityLimitationReason when limited by multiple reasons (#440)

[webrtc-stats] Clarify that remote-[in/out]bound-rtp.ssrc refers to the localId's ssrc (#454)

[webrtc-stats] Clarify the lifetime of remote-* stats objects (#439)

[webrtc-stats] Correlating RTCInboundRtpStreamStats with simulcast streams (#393)

[webrtc-stats] Deprecate remoteSource (#442)

[webrtc-stats] Detecting glitches (#443)

[webrtc-stats] Detecting Video Playback glitches (#443)

[webrtc-stats] Move audio levels to "media-source" and "receiver" (#451)

[webrtc-stats] new commits pushed by vr000m

[webrtc-stats] Outbound audio levels (#449)

[webrtc-stats] Pull Request: Add RTCInboundRtpStreamStats.totalDecodeTime

[webrtc-stats] Pull Request: Deprecate remoteSource

[webrtc-stats] Pull Request: Move audio levels to "media-source" and "receiver"

[webrtc-stats] Pull Request: WIP: Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded

[webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399)

[webrtc-stats] Replace RTCVideoSenderStats::keyFramesSent with RTCOutboundRtpStreamStats::keyFramesEncoded (#447)

[webrtc-stats] Replace RTCVideo{Sender, Receiver}Stats::keyFrames{Sent, Received} with RTC{Out, In}boundRtpStreamStats::keyFrames{En, De}coded (#447)

[webrtc-stats] RTCInboundRtpStreamStats.fractionLost (#446)

[webrtc-stats] Type stunserverconnection should be stun-server-connection (#444)

[webrtc-stats] We should move RTCCodecStats.implementation to RTC[In/Out]boundRtpStreamStats (#445)

[webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446)

Closed: [mediacapture-fromelement] Should tracks captured from a media element fire "ended" when ending? (#77)

Closed: [mediacapture-main] Clarification needed on HTMLMediaElement attributes that carry over (#599)

Closed: [mediacapture-main] Clarify that enumerateDevices must not expose devices that a given context cannot use through getUserMedia (#549)

Closed: [mediacapture-main] Echo Cancellation (#597)

Closed: [mediacapture-main] MediaStream in HTMLMediaElement: Does it have a first frame? (#555)

Closed: [mediacapture-main] MediaTrackSettings lacks support of channel layout (#605)

Closed: [mediacapture-screen-share] Ask for user gesture to call getDisplayMedia (#104)

Closed: [webrtc-pc] Add a security/privacy note about remote SDP (#2193)

Closed: [webrtc-pc] DataChannel max value for "id" before connecting? (#2158)

Closed: [webrtc-pc] DatachannelInit "id" attribute description inconsistent with procedure (#2157)

Closed: [webrtc-pc] Example 11 expects event.streams[0] from streamless tracks (#2217)

Closed: [webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213)

Closed: [webrtc-pc] RTCIceTransport role before start() called (#2210)

Closed: [webrtc-pc] Setting active attribute to false triggers BYE? (#2202)

Closed: [webrtc-pc] Update WebIDL reference in Terminology (#2091)

Closed: [webrtc-stats] Add RTCInboundRtpStreamStats.totalDecodeTime (#434)

Closed: [webrtc-stats] Clarify the lifetime of remote-* stats objects (#439)

Closed: [webrtc-stats] Remove RTC[Audio/Video][Sender/Receiver]Stats.remoteSource (#399)

Closed: [webrtc-stats] What should RTCInboundRtpStreamStats.fractionLost be if RTCP is disabled? (#446)

Last message date: Saturday, 29 June 2019 00:05:21 UTC