public-webrtc-logs@w3.org from September 2022 by subject

[mediacapture-handle] Should the handle be an object? (#68)

[mediacapture-image] Add support for powerline frequency on gUM MediaTrackSupportedConstraints (#294)

[mediacapture-main] Add `SecureContext` attribute to `InputDeviceInfo`. (#901)

[mediacapture-main] Device enumeration spec-ed to hang (#903)

[mediacapture-main] EnumerateDevices() Returns Multiple GroupId for Bluetooth Devices on Windows (#904)

[mediacapture-main] MediaStreamTrack frame rate: configured versus actual? (#826)

[mediacapture-main] new commits pushed by dontcallmedom

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

[mediacapture-main] Pull Request: Add `SecureContext` attribute to `InputDeviceInfo`.

[mediacapture-main] Pull Request: Update to latest ReSpec version 32.2.4

[mediacapture-main] Update enumerateDevices algorithm to make use of device-kind specific exposure checks when building cameraList and microphoneList. (#900)

[mediacapture-output] default audio output should be first in the enumerateDevice returned list (#124)

[mediacapture-output] Go back to the default output (#85)

[mediacapture-output] Spec uses settings object's responsible document which was removed (#132)

[mediacapture-record] Broken references in MediaStream Recording (#215)

[mediacapture-region] Behavior when cropTo races with the track ending (#72)

[mediacapture-region] Need for a predictable error type for unimplemented Element subtypes (#55)

[mediacapture-region] Rejecting cropTo() on ended tracks (#71)

[mediacapture-region] Transferring a track should preserve cropping (#67)

[mediacapture-region] Why expose produceCropTarget at MediaDevices level? (#11)

[mediacapture-screen-share] "current settings object's relevant global object" is not defined (#232)

[mediacapture-screen-share] A CaptureController object for getDisplayMedia() (#230)

[mediacapture-screen-share] Add CaptureController to the spec (#235)

[mediacapture-screen-share] Add spec-hooks for "actively screen-capturing" (#224)

[mediacapture-screen-share] Address nullability of DisplayMediaStreamOptions.controller (#236)

[mediacapture-screen-share] Beef up privacy and security section with regards to the various risks related to the different surfaces (#211)

[mediacapture-screen-share] Capturing audio-only (#100)

[mediacapture-screen-share] Inconsistent formatting in doc (#234)

[mediacapture-screen-share] Markup definitions to make them reusable in other specs (#218)

[mediacapture-screen-share] new commits pushed by eladalon1983

[mediacapture-screen-share] Pull Request: Add CaptureController to the spec

[mediacapture-screen-share] Pull Request: No need to test for presence of member with default value.

[mediacapture-screen-share] Pull Request: s/controller = null;/controller;

[mediacapture-screen-share] s/controller = null;/controller; (#237)

[mediacapture-screen-share] Spec appears to require gDM leak info on availability of audio (#233)

[mediacapture-screen-share] Spec should mention clickjacking concerns (#231)

[mediacapture-transform] Is MediaStreamTrackProcessor for audio necessary? (#29)

[mediacapture-viewport] Revisit: Persisting permissions (#10)

[webrtc-charter] Pull Request: frag id for Success Criteria div

[webrtc-encoded-transform] add mimeType to metadata (#140)

[webrtc-encoded-transform] Add rtp timestamp and capture time to RTCEncodedVideoFrameMetadata (#137)

[webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106)

[webrtc-encoded-transform] describe (some) fields (#155)

[webrtc-encoded-transform] Editorial: generateKeyFrame takes a "rid" argument, but is invoked with "rids" (#143)

[webrtc-encoded-transform] Enqueuing a RTCEncodedVideoFrame/RTCEncodedAudioFrame should transfer the data (#150)

[webrtc-encoded-transform] expose rid (and mid) as metadata on outgoing frames (#147)

[webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)

[webrtc-encoded-transform] generateKeyFrame algorithm leaks pendingKeyFrameTasks (#144)

[webrtc-encoded-transform] generateKeyFrame algorithm makes wrong assumptions about number of encoders (#145)

[webrtc-encoded-transform] generateKeyFrame takes a "rid" argument, but is invoked with "rids" (#143)

[webrtc-encoded-transform] idl: synchronizationsource is a *unsigned* long (#153)

[webrtc-encoded-transform] Interaction with Congestion Control (#31)

[webrtc-encoded-transform] intro: remove funny hats (#148)

[webrtc-encoded-transform] Make RTCRtpSender.generateKeyFrame take a single optional rid parameter. (#146)

[webrtc-encoded-transform] metadata for start and flush (#10)

[webrtc-encoded-transform] Need to create RTCEncodedVideoFrame from scratch (#134)

[webrtc-encoded-transform] new commits pushed by alvestrand

[webrtc-encoded-transform] Protocol Reference for SFrameTransform (#112)

[webrtc-encoded-transform] Pull Request: audio: add rtp sequence number on incoming frames

[webrtc-encoded-transform] Pull Request: describe (some) fields

[webrtc-encoded-transform] Pull Request: idl: synchronizationsource is a *unsigned* long

[webrtc-encoded-transform] Pull Request: intro: remove funny hats

[webrtc-encoded-transform] Pull Request: Make RTCRtpSender.generateKeyFrame take a single optional rid parameter.

[webrtc-encoded-transform] Pull Request: Make use of SFrame WG spec link

[webrtc-encoded-transform] Pull Request: Remove unused RTCInsertableStreams dictionary

[webrtc-encoded-transform] Pull Request: Structure clone frame with passing frame.data as transferables and send frame's clone to packetizer or decoder.

[webrtc-encoded-transform] statistics (#15)

[webrtc-encoded-transform] what metadata is useful? (#9)

[webrtc-extensions] add AES-256 media stream encryption control to peerconnection (#113)

[webrtc-extensions] Add getDataChannels() method on RTCPeerConnection (#110)

[webrtc-extensions] Deprecate audio/video enumeration in getCapabilities in favour of Media Capabilities API (#95)

[webrtc-extensions] enable opus bite rate control by js api instead of SDP munging (#117)

[webrtc-extensions] header extension API: remove setParameters support (#116)

[webrtc-extensions] Integration of congestion control across SCTP and media (#111)

[webrtc-extensions] Mixed-codec simulcast proposal (#43)

[webrtc-extensions] Need to specify behavior of detached RTCDataChannel objects. (#115)

[webrtc-extensions] Pull Request: header extension API: remove setParameters support

[webrtc-extensions] RTCDataChannel transfer and maxMessageSize (#114)

[webrtc-pc] Add RTCPeerConnection.getDataChannels() (#2770)

[webrtc-pc] addTransceiver does not check for missing rid properties (#2734)

[webrtc-pc] addTransceiver does not check for uniqueness of rid (#2733)

[webrtc-pc] Document substantive changes since Rec as candidate amendments (#2713)

[webrtc-pc] Don't let offers to receive simulcast overwrite existing [[SendEncodings]] (#2758)

[webrtc-pc] Handling of simulcast attributes with multiple choices in a version seems to be underspecified (#2769)

[webrtc-pc] Inconsistent initialization of scaleResolutionDownBy (#2730)

[webrtc-pc] Inconsistent rules for rid in RTCRtpEncodingParameters (#2732)

[webrtc-pc] Is "same PT, different FMTP lines" allowed in BUNDLE? (#2766)

[webrtc-pc] Modifications to [[SendEncodings]] from setParameters and sRD can be racy (#2737)

[webrtc-pc] move url from RTCIceEvent to the RTCIceCandidate (#2773)

[webrtc-pc] new commits pushed by aboba

[webrtc-pc] new commits pushed by dontcallmedom

[webrtc-pc] Pull Request: Add RTCPeerConnection.getDataChannels()

[webrtc-pc] Pull Request: move url from RTCIceEvent to the RTCIceCandidate

[webrtc-pc] Pull Request: Remove references to RTCIceCredentialType

[webrtc-pc] Pull Request: Scaleaudio

[webrtc-pc] Pull Request: TypeError on duplicate rids

[webrtc-pc] Pull Request: TypeError unless all or none of encodings have rids

[webrtc-pc] Pull Request: Update link to WebSockets interface

[webrtc-pc] Pull Request: Update to ReSpec version 32.2.4

[webrtc-pc] Remove references to RTCIceCredentialType (#2767)

[webrtc-pc] rtcicecandidate: add relayProtocol (#2763)

[webrtc-pc] Simulcast: Implementations do not fail (and that seems good) (#2762)

[webrtc-pc] TypeError unless all or none of encodings have rids (#2774)

[webrtc-pc] webrtc-pc does not say to clear RTCRtpCodingParameters.rid when sRD rejects simulcast (#2736)

[webrtc-pc] What is the intended behavior of rollback of remote simulcast offer? (#2764)

[webrtc-stats] Add droppedSamplesEvents and synthesizedSamplesEvents (#688)

[webrtc-stats] Add field `fecPacketsSent` to RtcOutboundRtpStreamStats (#692)

[webrtc-stats] add missing fields to candidate stats (#611)

[webrtc-stats] Add powerEfficient[En/De]coder (#666) and fingerprint mitigations (#675). (#670)

[webrtc-stats] Add RTCAudioPlayoutStats, synthesizedSamplesDuration and totalSamplesDuration (#682)

[webrtc-stats] Add totalCaptureDelay and totalSamplesCaptured to RTCAudioSourceStats. (#685)

[webrtc-stats] add video totalFreezesDuration &totalPausesDuration etc to standard getStats (#695)

[webrtc-stats] Adding SVC-related stats fields (#673)

[webrtc-stats] Adds powerEfficientDecoder/powerEfficientEncoder. (#670)

[webrtc-stats] Are "codec" stats per transceiver or per transport? (#614)

[webrtc-stats] clarify nackCount, firCount, pliCount and fix trailing whitespace (#663)

[webrtc-stats] Codec stats reveal hardware information which could be used for fingerprinting (#674)

[webrtc-stats] codec.sdpFmtpLine isn't clear about which description to use (#616)

[webrtc-stats] Confusing round trip time defintion in remote-outbound-rtp (#659)

[webrtc-stats] Delete RTP stream stats when ssrc changes or transceiver stops (#672)

[webrtc-stats] Delete RTP stream stats when ssrc or codec changes (#672)

[webrtc-stats] Do we agree removing "sender", "receiver" and "transceiver" stats is a good idea? (#643)

[webrtc-stats] Don't expose so many RTCCodecStats! (#662)

[webrtc-stats] Duplicated field `kind` in RTCInboundRtpStreamStats (#690)

[webrtc-stats] editorial: remove trailing whitespace (#700)

[webrtc-stats] Expose the usefull experimental stats to a JavaScript layer (#609)

[webrtc-stats] Exposing HW for Cloud Gaming use cases (HW encoder/decoder revisited) (#698)

[webrtc-stats] Fix descriptions according to the upgraded draft (#689)

[webrtc-stats] How many times did capture glitches occur? (Follow-up to #678) (#679)

[webrtc-stats] How many times did glitches occur? (Follow-up to #676) (#677)

[webrtc-stats] How many times did playout glitches occur? (Follow-up to #676) (#677)

[webrtc-stats] Impact of Stereo input and out put on metrics (#686)

[webrtc-stats] Is "same PT, different FMTP lines" allowed in BUNDLE? (#664)

[webrtc-stats] Make RTP creation dependent on SSRC being known rather than packet being sent (#671)

[webrtc-stats] Metrics for capture delay (#681)

[webrtc-stats] Metrics for playout delay (#680)

[webrtc-stats] Move RTCRtpReceivedRtpStreamStats.framesDropped to RTCInboundRtpStreamStats. (#661)

[webrtc-stats] Need metrics for capture glitches (#678)

[webrtc-stats] Need metrics for playout glitches (#676)

[webrtc-stats] new commits pushed by henbos

[webrtc-stats] new commits pushed by vr000m

[webrtc-stats] Only expose RTCCodecStats objects currently in use (#669)

[webrtc-stats] powerEfficientEncoder/powerEfficientDecoder (#666)

[webrtc-stats] Privacy concern: Leaking communication / plain text using patterns in packet size, frequency, etc. (#699)

[webrtc-stats] Pull Request: Add droppedSamplesDuration to RTCAudioSourceStats.

[webrtc-stats] Pull Request: Add droppedSamplesEvents and synthesizedSamplesEvents

[webrtc-stats] Pull Request: Add media-playout to summary table

[webrtc-stats] Pull Request: Add RTCAudioPlayoutStats, synthesizedSamplesDuration and totalSamplesDuration

[webrtc-stats] Pull Request: Add totalCaptureDelay and totalSamplesCaptured to RTCAudioSourceStats.

[webrtc-stats] Pull Request: Add totalPlayoutDelay and totalSamplesCount to RTCAudioPlayoutStats.

[webrtc-stats] Pull Request: Adds powerEfficientDecoder/powerEfficientEncoder.

[webrtc-stats] Pull Request: Delete RTP stream stats when ssrc or codec changes

[webrtc-stats] Pull Request: editorial: remove trailing whitespace

[webrtc-stats] Pull Request: Fix descriptions according to the upgraded draft

[webrtc-stats] Pull Request: Fix recently introduced typos.

[webrtc-stats] Pull Request: inbound-rtp: add frame assembly time

[webrtc-stats] Pull Request: Make RTP creation dependent on SSRC being known rather than packet being sent

[webrtc-stats] Pull Request: Only expose RTCCodecStats objects currently in use

[webrtc-stats] Pull Request: r/"Privacy considerations"/"Procedures for mitigating privacy concerns"

[webrtc-stats] relayProtocol: use RTCIceServerTransportProtocol struct (#657)

[webrtc-stats] remoteTimestamp does not specify how to derive the RTCP SR NTP timestamp (#665)

[webrtc-stats] remove outbound-rtp totalEncodedBytesTarget (#653)

[webrtc-stats] Rename "Privacy considerations" to "Procedures for mitigating privacy concerns" (#696)

[webrtc-stats] Stats API should require additional permission / user opt-in (#550)

[webrtc-stats] Summary table does not list `media-playout` stats (#691)

[webrtc-stats] The stats API allow hardware fingerprinting (encoder, powerEfficient) (#675)

[webrtc-stats] When are RTP stream stats created? (#667)

[webrtc-stats] When are RTP streams destroyed? (#668)

[webrtc-stats] WPT tests are wrong about when "outbound-rtp" and "inbound-rtp" stats appear (#619)

[webrtc-svc] Adding SVC-related stats fields (#72)

[webrtc-svc] Encoding parameters for spatial layers (#14)

[webrtc-svc] Layer drop/add (#4)

[webrtc-svc] S modes and a single active simulcast layer (#73)

[webrtc-svc] should scalabilityMode be an array scalabilityModes? (#50)

[webrtc-svc] SVC getCapabilities() is redundant with Media Capabilities query (#49)

Closed: [mediacapture-main] EnumerateDevices() Returns Multiple GroupId for Bluetooth Devices on Windows (#904)

Closed: [mediacapture-output] default audio output should be first in the enumerateDevice returned list (#124)

Closed: [mediacapture-output] Go back to the default output (#85)

Closed: [mediacapture-region] Why expose produceCropTarget at MediaDevices level? (#11)

Closed: [webrtc-encoded-transform] metadata for start and flush (#10)

Closed: [webrtc-encoded-transform] no funny hats (#122)

Closed: [webrtc-encoded-transform] statistics (#15)

Closed: [webrtc-encoded-transform] what metadata is useful? (#9)

Closed: [webrtc-pc] Enum RTCIceCredentialType with only one value (#2746)

Closed: [webrtc-stats] Add field `fecPacketsSent` to RtcOutboundRtpStreamStats (#692)

Closed: [webrtc-stats] Do we agree removing "sender", "receiver" and "transceiver" stats is a good idea? (#643)

Closed: [webrtc-stats] Don't expose so many RTCCodecStats! (#662)

Closed: [webrtc-stats] Expose the usefull experimental stats to a JavaScript layer (#609)

Closed: [webrtc-stats] How many times did capture glitches occur? (Follow-up to #678) (#679)

Closed: [webrtc-stats] How many times did playout glitches occur? (Follow-up to #676) (#677)

Closed: [webrtc-stats] Metrics for capture delay (#681)

Closed: [webrtc-stats] Metrics for playout delay (#680)

Closed: [webrtc-stats] Need metrics for capture glitches (#678)

Closed: [webrtc-stats] Need metrics for playout glitches (#676)

Closed: [webrtc-stats] powerEfficientEncoder/powerEfficientDecoder (#666)

Closed: [webrtc-stats] Rename "Privacy considerations" to "Procedures for mitigating privacy concerns" (#696)

Closed: [webrtc-stats] Stats API should require additional permission / user opt-in (#550)

Closed: [webrtc-stats] Summary table does not list `media-playout` stats (#691)

Closed: [webrtc-stats] The stats API allow hardware fingerprinting (encoder, powerEfficient) (#675)

Closed: [webrtc-stats] Unclear pliCount, firCount, and nackCount for outbound-rtp (#658)

Closed: [webrtc-stats] When are RTP stream stats created? (#667)

Closed: [webrtc-stats] When are RTP streams destroyed? (#668)

Closed: [webrtc-svc] Adding SVC-related stats fields (#72)

Last message date: Friday, 30 September 2022 21:24:38 UTC