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

[mediacapture-depth] Re-add the camera/depth map intrinsics (#174)

[mediacapture-main] A privacy concern of "Media Capture and Streams" (#630)

[mediacapture-main] addtrack and removetrack events only fire when user agent updates stream (#517)

[mediacapture-main] Clarification needed on ended HTMLMediaElement with MediaStream becoming active (#519)

[mediacapture-main] Enforcing user gesture for getUserMedia (#639)

[mediacapture-main] enumerateDevices should only provide device info permission for granted device types (#645)

[mediacapture-main] MediaStreamTrack and b/f cache (#638)

[mediacapture-main] Move enumerateDevices behind permission (#612)

[mediacapture-main] Move request permission steps into the for-loop. (#643)

[mediacapture-main] new commits pushed by alvestrand

[mediacapture-main] new commits pushed by henbos

[mediacapture-main] Only Firefox turns off device on disabled track. Stronger language needed? (#642)

[mediacapture-main] Only reveal labels of devices user has given permission to (#640)

[mediacapture-main] Prompt user to choose unless constraints reduce to 1. (#644)

[mediacapture-main] Pull Request: Move request permission steps into the for-loop.

[mediacapture-main] Pull Request: Prompt user to choose unless constraints reduce to 1.

[mediacapture-main] Pull Request: Replace device-info permission by a browsing context boolean flag

[mediacapture-main] Replace device-info permission by a browsing context boolean flag (#641)

[mediacapture-main] Should enumerateDevices by default return an empty list? (#646)

[mediacapture-main] Specification to capture unmodified audio (#457)

[mediacapture-output] API to request audio output device selection (#86)

[mediacapture-output] Pull Request: API to request audio output device selection

[mediacapture-output] Selecting audio output in case device info permission is not granted (#83)

[mediacapture-output] Setting the audio output for a whole context or page (#87)

[mediacapture-output] Setting the output for a whole context (#87)

[mediacapture-record] Add `audioConstantBitRate` flag to `MediaRecorderOptions`. (#185)

[mediacapture-record] Defer deciding codec until "onstart" fires (#190)

[mediacapture-record] MediaStream does not expose a RecordingState (#181)

[mediacapture-screen-share] Add conversational sharing and assistance use case (#123)

[mediacapture-screen-share] Add the tech support use case. (#123)

[mediacapture-screen-share] Normative security requirements ("ISSUE 1") (#126)

[mediacapture-screen-share] Possible duplication of requirements (#125)

[mediacapture-screen-share] Pull Request: Add explainer.

[mediacapture-screen-share] Pull Request: Add the tech support use case.

[mediacapture-screen-share] Pull Request: Link fixes

[mediacapture-screen-share] Should getDisplayMedia be absent on platforms with no support? (#127)

[mediacapture-screen-share] Should getDisplayMedia be undefined on platforms with no support? (#127)

[mediacapture-screen-share] TAG review (#59)

[mst-content-hint] Add conformance section (#32)

[mst-content-hint] Add description of behaviors triggered by contentHint (#31)

[mst-content-hint] Pull Request: Add conformance section

[webrtc-dscp-exp] Add priority to DSCP spec (#5)

[webrtc-dscp-exp] Pull Request: Add priority to DSCP spec

[webrtc-pc] "Rollback" is under-specified (#2324)

[webrtc-pc] [[ReceiveCodecs]] is only set on answer. How to achieve early media? (#2369)

[webrtc-pc] [[SendCodecs]] is updated based on answers, but [[SendEncodings]] is updated based on remote description (#2313)

[webrtc-pc] Add a perfect negotiation example (#2370)

[webrtc-pc] Add accessibility considerations regarding real-time text (#2328)

[webrtc-pc] Add explicit "rollback" steps (#2384)

[webrtc-pc] Add hardwareAccelerated field to RTCRtpCodecCapability (#2355)

[webrtc-pc] Add hardwareAccelerated field to RTCRtpCodecCapability (#2356)

[webrtc-pc] Add more specificity on the format of RTCStats.ids (#2319)

[webrtc-pc] Add security note about certificates and postMessage. (#2333)

[webrtc-pc] Backwards compat concerns with RTCError (#2330)

[webrtc-pc] Can IceTransportState go checking -> completed? (#2316)

[webrtc-pc] Can simulcast offers renegotiate rids? (#2314)

[webrtc-pc] Clarify what happens to codecs when rolling back (#2368)

[webrtc-pc] Clarify what happens when rolling back an ICE restart (#2367)

[webrtc-pc] Construct RTCErrors with name 'OperationError' instead of 'RTCError' (#2375)

[webrtc-pc] degradationPreference is under-specified (#2248)

[webrtc-pc] DOMException(name=<Exception>) or instances of <Exception>? (#1817)

[webrtc-pc] Editors vs Former editors (#2386)

[webrtc-pc] Find a home for 1.1 features, extensions and features at risk (was: Move features at risk to an extension spec) (#2323)

[webrtc-pc] Firing the right events in "rollback": Which description to use? (#2332)

[webrtc-pc] iceconnectionstatechange event after PeerConnection.close() is called (#2335)

[webrtc-pc] Mark RTCError attributes at risk (#2374)

[webrtc-pc] Move Cullen Jennings from 'Editors' to 'Former editors' (#2387)

[webrtc-pc] Move henbos/webrtc-extensions to w3c/webrtc-extensions (#2378)

[webrtc-pc] Move henbos/webrtc-provisional-stats to w3c/webrtc-provisional-stats (#2379)

[webrtc-pc] Move identity steps to the identity spec (#2364)

[webrtc-pc] Move oauth to webrtc-extensions (#2347)

[webrtc-pc] new commits pushed by alvestrand

[webrtc-pc] new commits pushed by henbos

[webrtc-pc] Peer Connection and b/f cache (#2346)

[webrtc-pc] Peer Connection and back/forward cache (#2346)

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

[webrtc-pc] Prefer variable ptime (audio frame lengths) (#2300)

[webrtc-pc] Pull Request: Add explicit "rollback" steps

[webrtc-pc] Pull Request: Add hardwareAccelerated field to RTCRtpCodecCapability

[webrtc-pc] Pull Request: Construct RTCErrors with name 'OperationError' instead of 'RTCError'

[webrtc-pc] Pull Request: Fix travis error introduced by PR 2380

[webrtc-pc] Pull Request: Mark RTCError attributes "at risk"

[webrtc-pc] Pull Request: Move Cullen Jennings from 'Editors' to 'Former editors'

[webrtc-pc] Pull Request: Move former editors to separate entry

[webrtc-pc] Pull Request: Note that getSynchronizationSources works without sinks

[webrtc-pc] Pull Request: Note that ICE transport state can go to completed

[webrtc-pc] Pull Request: Remove [[WEBIDL-1]] reference in favor of [[WEBIDL]]

[webrtc-pc] Pull Request: Remove codecPayloadType, dtx, ptime and maxFramerate from parameters

[webrtc-pc] Pull Request: Remove getDefaultIceServers reference from "at risk" section

[webrtc-pc] Pull Request: Remove getDefaultIceServers()

[webrtc-pc] Pull Request: Remove getSupportedAlgorithms()

[webrtc-pc] Pull Request: Remove maxFramerate (to be moved to extension spec)

[webrtc-pc] Pull Request: Remove OAuth (to be moved to an extension spec)

[webrtc-pc] Pull Request: Remove RTCRtcpMuxPolicy::negotiate and rtcpTransport

[webrtc-pc] Pull Request: Remove RTCRtpReceiveParameters.encodings

[webrtc-pc] Pull Request: Remove voiceActivityFlag from RTCOfferAnswerOptions

[webrtc-pc] Pull Request: Revert [[ReceiveCodecs]] on rollback

[webrtc-pc] Pull Request: Revert transports on rollback

[webrtc-pc] Pull Request: Set receiver.[[ReceiveCodecs]] on all descriptions.

[webrtc-pc] Pull Request: Update Mandatory stats to match refactoring

[webrtc-pc] Racy setParameters()/getParameters() behavior (#2315)

[webrtc-pc] Remove codecPayloadType, dtx, ptime and maxFramerate from parameters (#2351)

[webrtc-pc] Remove dtx, maxFramerate, ptime and codecPayloadType from RTCRtpEncodingParameters (#2350)

[webrtc-pc] Remove dtx, ptime and codecPayloadType from RTCRtpEncodingParameters (#2350)

[webrtc-pc] Remove getDefaultIceServers() (#2354)

[webrtc-pc] Remove maxFramerate (#2357)

[webrtc-pc] Remove maxFramerate (to be moved to extension spec) (#2361)

[webrtc-pc] Remove negotiate value from RTCRtcpMuxPolicy (#2348)

[webrtc-pc] Remove OAuth (to be moved to an extension spec) (#2362)

[webrtc-pc] Remove Priority from Sender and Datachannel (#2342)

[webrtc-pc] Remove RTCRtpReceiveParameters.encodings (#2382)

[webrtc-pc] Remove VoiceActivityDetection (#2349)

[webrtc-pc] Removing degradationPreference (#2326)

[webrtc-pc] Removing getSupportedAlgorithms() (#2344)

[webrtc-pc] Revert [[ReceiveCodecs]] on rollback (#2373)

[webrtc-pc] Revert transports on rollback (#2372)

[webrtc-pc] RTCCertificate security boundary (#2343)

[webrtc-pc] Section 4.4.1.6: Set the RTCSessionDescription (Object visibility) (#2338)

[webrtc-pc] setCodecPreferences() and removing a codec after initial negotiation (#2371)

[webrtc-pc] Should peer identity specific steps be marked as "Feature at risk" (#2284)

[webrtc-pc] Should SCTP transport enter "closed" state when DTLS transport does? (#2360)

[webrtc-pc] Should we remove getDefaultIceServers? (#2023)

[webrtc-pc] Timing of setRemoteDescription's identity validation is unclear (#2173)

[webrtc-pc] Transports on offer need to be rolled back (#2363)

[webrtc-pc] Update Mandatory stats to match refactoring (#2366)

[webrtc-pc] Upgrade to ReSpec 24.32.1 produced 281 warnings (#2358)

[webrtc-pc] What to do with transports on rollback? (#2363)

[webrtc-stats] Add frameBitDepth to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats (#517)

[webrtc-stats] Add frameBitDepth to RTCInboundRtpStreamStats and RTCOutbountRtpStreamStats (#516)

[webrtc-stats] Add RTCTransportStats.iceUfrag (#502)

[webrtc-stats] description of when roundTripTimeMeasurements increments is wrong (#515)

[webrtc-stats] Exposing RTCIceCandidateStats.networkType might trigger fingerprinting (#374)

[webrtc-stats] Issues with replaceTrack, will statsended fire or give me what I want. (#472)

[webrtc-stats] new commits pushed by henbos

[webrtc-stats] new commits pushed by vr000m

[webrtc-stats] Pull Request: Add frameBitDepth to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats

[webrtc-stats] Pull Request: Fix Travis and reformat Obsolete section

[webrtc-stats] Pull Request: Fix typo in usage of totalInterFrameDelay.

[webrtc-stats] Pull Request: Remove RTCNetworkType and networkType fields

[webrtc-stats] Pull Request: RTCIceServerStats relayProtocol

[webrtc-stats] Remove RTCNetworkType and networkType fields (#521)

[webrtc-stats] RTC[Audio/Video]SenderStats should have mid (#396)

[webrtc-stats] Stats need to mark up which members are required (#304)

[webrtc-stats] What should we do about onstatsended? (#480)

[webrtc-stats] Wrap around issue with bytes and packet counters (#509)

[webrtc-stats] wrong definition of RTCIceServerStats.protocol? (#512)

Closed: [mediacapture-main] addtrack and removetrack events only fire when user agent updates stream (#517)

Closed: [mediacapture-main] Clarification needed on HTMLMediaElement with MediaStream ending playback (#518)

Closed: [mediacapture-main] Clarify request for permission is for both audio *and* video (editorial). (#556)

Closed: [mediacapture-main] Enforcing user gesture for getUserMedia (#639)

Closed: [mediacapture-main] Specification to capture unmodified audio (#457)

Closed: [mediacapture-record] MediaStream does not expose a RecordingState (#181)

Closed: [mediacapture-screen-share] TAG review (#59)

Closed: [webrtc-pc] "Rollback" is under-specified (#2324)

Closed: [webrtc-pc] [[ReceiveCodecs]] is only set on answer. How to achieve early media? (#2369)

Closed: [webrtc-pc] [[SendCodecs]] is updated based on answers, but [[SendEncodings]] is updated based on remote description (#2313)

Closed: [webrtc-pc] Add hardwareAccelerated field to RTCRtpCodecCapability (#2355)

Closed: [webrtc-pc] Add more specificity on the format of RTCStats.ids (#2319)

Closed: [webrtc-pc] Backwards compat concerns with RTCError (#2330)

Closed: [webrtc-pc] Clarify that getSynchronizationSources() should return information even if the track has no sink (<video> tag) (#2240)

Closed: [webrtc-pc] Clarify what happens to codecs when rolling back (#2368)

Closed: [webrtc-pc] Clarify what happens when rolling back an ICE restart (#2367)

Closed: [webrtc-pc] Find a home for 1.1 features, extensions and features at risk (was: Move features at risk to an extension spec) (#2323)

Closed: [webrtc-pc] Mark RTCError attributes at risk (#2374)

Closed: [webrtc-pc] Move henbos/webrtc-extensions to w3c/webrtc-extensions (#2378)

Closed: [webrtc-pc] Move henbos/webrtc-provisional-stats to w3c/webrtc-provisional-stats (#2379)

Closed: [webrtc-pc] Move maxFramerate to extension spec (#2357)

Closed: [webrtc-pc] Move oauth to webrtc-extensions (#2347)

Closed: [webrtc-pc] Move priority to DSCP spec (#2334)

Closed: [webrtc-pc] Prefer variable ptime (audio frame lengths) (#2300)

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

Closed: [webrtc-pc] Remove dtx, ptime and codecPayloadType from RTCRtpEncodingParameters (#2350)

Closed: [webrtc-pc] Remove negotiate value from RTCRtcpMuxPolicy (#2348)

Closed: [webrtc-pc] Remove RTCRtpReceiveParameters.encodings (#2382)

Closed: [webrtc-pc] Remove VoiceActivityDetection (#2349)

Closed: [webrtc-pc] Removing getSupportedAlgorithms() (#2344)

Closed: [webrtc-pc] Section 4.4.1.6: Set the RTCSessionDescription (Object visibility) (#2338)

Closed: [webrtc-pc] Should we remove getDefaultIceServers? (#2023)

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

Closed: [webrtc-pc] Transports on offer need to be rolled back (#2363)

Closed: [webrtc-pc] Update mandatory to implement stats based on recent restructuring (#2302)

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

Closed: [webrtc-stats] Add frameBitDepth to RTCInboundRtpStreamStats and RTCOutbountRtpStreamStats (#516)

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

Closed: [webrtc-stats] Issues with replaceTrack, will statsended fire or give me what I want. (#472)

Closed: [webrtc-stats] RTC[Audio/Video]SenderStats should have mid (#396)

Closed: [webrtc-stats] Should we have transceiver stats? (#478)

Closed: [webrtc-stats] What should we do about onstatsended? (#480)

Last message date: Saturday, 30 November 2019 17:58:01 UTC