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

[EXTERNAL] [minutes] TPAC meetings

[EXTERNAL] RTCSender still transmits audio packet even when a=inactive is received in SDP

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

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

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

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

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

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

[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] Rejecting cropTo() on ended tracks (#71)

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

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

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

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

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

[mediacapture-streams] YOUR TOPIC HERE

[minutes] TPAC meetings

[Reminder] Call for Exclusions: Region Capture

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

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

[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-nv-use-cases] Pull Request: Add requirements for low latency streaming use case

[webrtc-nv-use-cases] Support for End to End Encryption (E2EE) (#76)

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

AW: [EXTERNAL] [minutes] TPAC meetings

Event Updated: Web Real-Time Communications Working Group and Media Working Group and Media and Entertainment Interest Group - Joint meeting

Fwd: VOTE by 2022-09-21/22: Proposed Charter for the Web Real-Time Communications Working Group

Media pipeline architecture repository for cross-group issues

RTCSender still transmits audio packet even when a=inactive is received in SDP

TPAC 2022 Schedule

Weekly github digest (WebRTC WG specifications)

Last message date: Friday, 30 September 2022 13:45:52 UTC