[mediacapture-fromelement] CD publication of specification (#47)
[mediacapture-fromelement] Clarify if CanvasCaptureMediaStreamTrack mute, unmute, and ended events are expected to be fired (#82)
[mediacapture-fromelement] define behaviors of the common ConstrainablePattern Interfaces (#48)
[mediacapture-fromelement] Inconsistency: Taint, not mute cross-origin element tracks (#83)
[mediacapture-fromelement] Review mute/unmute/ended and constraints on tracks from canvas.captureStream() (#99)
[mediacapture-fromelement] Review mute/unmute/ended and constraints on tracks from element.captureStream() (#98)
[mediacapture-fromelement] WG CR review for mediacapture-fromelement (#50)
[mediacapture-fromelement] What happens to the audio being rendered to a Media Element when it gets captureStream()ed (#34)
[mediacapture-image] Use constrainable pattern for ImageCapture (second take) (#150)
[mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Philipp Hancke via GitHub (Wednesday, 27 March)
- François Beaufort via GitHub (Friday, 8 March)
- François Beaufort via GitHub (Friday, 8 March)
- Sam Sneddon via GitHub (Thursday, 7 March)
- François Beaufort via GitHub (Thursday, 7 March)
- Karl Dubost via GitHub (Thursday, 7 March)
- Elad Alon via GitHub (Wednesday, 6 March)
- Philipp Hancke via GitHub (Wednesday, 6 March)
- Karl Dubost via GitHub (Wednesday, 6 March)
- Karl Dubost via GitHub (Wednesday, 6 March)
- Philipp Hancke via GitHub (Wednesday, 6 March)
- Karl Dubost via GitHub (Wednesday, 6 March)
[mediacapture-main] Clarify each source is responsible for specifying mute/unmute/ended and constraints behavior (#984)
[mediacapture-main] Fix export of track-muted and set-track-muted. (#986)
[mediacapture-main] Implementers must not refuse to open sources set as default at the machine: "DOMException: Could not start audio source" is not in the specification (#708)
[mediacapture-main] new commits pushed by dontcallmedom
[mediacapture-main] new commits pushed by youennf
[mediacapture-main] Pull Request: Add class="informative" to "Legacy GetUserMedia interface" section
[mediacapture-main] Pull Request: Update to latest ReSpec version 34.5.0
[mediacapture-main] risk model of stored permissions and constraint opportunities (#991)
[mediacapture-output] Persisting deviceIds Across Sessions (#127)
[mediacapture-output] Set Default Audio Output Device in MediaDevices (#141)
[mediacapture-record] mimeType ambiguity: "video/webm;codecs=vp8" means? (#194)
[mediacapture-region] Behavior when full screened (#81)
[mediacapture-region] is this still an experimental feature? (#82)
[mediacapture-screen-share-extensions] Auto-pause capture when user switches captured content (#4)
[mediacapture-screen-share] Define the windowAudio option (#283)
[mediacapture-screen-share] Review mute/unmute/ended and constraints on tracks from getDisplayMedia() (#298)
[mediacapture-transform] Is MediaStreamTrackProcessor for audio necessary? (#29)
[mediacapture-transform] Review mute/unmute/ended and constraints on new VideoTrackGenerator().track (#109)
[webrtc-encoded-transform] Constructor frame (#223)
[webrtc-encoded-transform] Expose captureTimestamp to RTCEncodedAudioFrame and RTCEncodedVideoFrame (#159)
[webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
[webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)
[webrtc-encoded-transform] Should we add captureTimestamp senderCaptureTimeOffset to the RTCEncodedXXXFrameMetadata if abs-capture-time is used? (#225)
[webrtc-extensions] Add API to control jitterBufferTarget handling (#199)
[webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Eldar Rello via GitHub (Thursday, 21 March)
- Eldar Rello via GitHub (Thursday, 21 March)
- Jakob Ivarsson via GitHub (Thursday, 21 March)
- Eldar Rello via GitHub (Thursday, 21 March)
- Eldar Rello via GitHub (Tuesday, 19 March)
- Jakob Ivarsson via GitHub (Tuesday, 19 March)
- Eldar Rello via GitHub (Tuesday, 19 March)
- Jakob Ivarsson via GitHub (Tuesday, 19 March)
- Eldar Rello via GitHub (Tuesday, 19 March)
- Jakob Ivarsson via GitHub (Tuesday, 19 March)
- Eldar Rello via GitHub (Tuesday, 19 March)
- Jakob Ivarsson via GitHub (Tuesday, 19 March)
- Jakob Ivarsson via GitHub (Tuesday, 19 March)
[webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- dontcallmedom-bot via GitHub (Wednesday, 27 March)
- Jan-Ivar Bruaroey via GitHub (Thursday, 21 March)
- youennf via GitHub (Thursday, 21 March)
- youennf via GitHub (Thursday, 21 March)
- Jan-Ivar Bruaroey via GitHub (Wednesday, 20 March)
- guidou via GitHub (Wednesday, 20 March)
- Jan-Ivar Bruaroey via GitHub (Monday, 18 March)
- youennf via GitHub (Monday, 18 March)
- guidou via GitHub (Monday, 18 March)
- Bernard Aboba via GitHub (Thursday, 7 March)
- guidou via GitHub (Tuesday, 5 March)
[webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)
[webrtc-extensions] Peer Connection and back/forward cache (#200)
[webrtc-extensions] Pull Request: Clean references and fix links
[webrtc-extensions] Pull Request: Move jitterBufferTarget to main spec
[webrtc-pc] Alternative storage for RTCCertificates needed (#2944)
[webrtc-pc] Clarify unmute event must fire on receiver.track AFTER sRD(offer) succeeds (#2880)
[webrtc-pc] Consider making RTCIceCandidatePair an interface (#2930)
[webrtc-pc] Document test updates associated with amendments (#2910)
[webrtc-pc] Incorporate jitterBufferTarget (#2952)
[webrtc-pc] Missing tests for candidate amendments (#2950)
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] new commits pushed by jan-ivar
[webrtc-pc] new commits pushed by youennf
[webrtc-pc] Proposing setCodecPreferences to deal with both send and recv codecs (#2939)
[webrtc-pc] Pull Request: Clarify when RTCIceCandidate's relayProtocol and url members are null.
[webrtc-pc] Pull Request: Fix bugs in amendments
[webrtc-pc] Pull Request: Integrate jitterBufferTarget as candidate addition
[webrtc-pc] Pull Request: Make RTCIceCandidatePair members required
[webrtc-pc] Pull Request: Update to ReSpec version 34.5.0
[webrtc-pc] Queue two tasks upon finishing ICE gathering, and fire gatheringstatechange & icegatheringstatechange in same task (#2894)
[webrtc-pc] Review mute/unmute/ended and constraints on RTCRtpReceiver's track. (#2942)
[webrtc-pc] RTCIceCandidate's relayProtocol and url members can be null but not absent (#2947)
[webrtc-pc] Update the accessibility section 14 to include RFC 8865 for real-time text in WebRTC data channel (#2931)
[webrtc-pc] WebRTC spec should explicitly specify all causes of a PeerConnection-sourced track receiving mute/unmute (#2915)
[webrtc-rtptransport] Add "motivation" section, largely copied from IceController explainer (#19)
[webrtc-rtptransport] new commits pushed by aboba
- Bernard Aboba via GitHub (Thursday, 28 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Tuesday, 26 March)
- Bernard Aboba via GitHub (Sunday, 10 March)
- Bernard Aboba via GitHub (Saturday, 9 March)
- Bernard Aboba via GitHub (Saturday, 9 March)
- Bernard Aboba via GitHub (Saturday, 9 March)
- Bernard Aboba via GitHub (Saturday, 9 March)
- Bernard Aboba via GitHub (Friday, 8 March)
- Bernard Aboba via GitHub (Friday, 8 March)
- Bernard Aboba via GitHub (Friday, 8 March)
- Bernard Aboba via GitHub (Friday, 8 March)
[webrtc-rtptransport] Pull Request: Add "motivation" section, largely copied from IceController explainer
[webrtc-rtptransport] Pull Request: Add "Use Case 2"
[webrtc-rtptransport] Pull Request: Cleanup Use Case 1
[webrtc-stats] new commits pushed by jan-ivar
[webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785)
[webrtc-stats] RTCStats.timestamp - fingerprinting and since epoch (#786)
[webrtc-svc] Dependency on AV1 RTP payload specification (#102)
Closed: [mediacapture-fromelement] CD publication of specification (#47)
Closed: [mediacapture-fromelement] WG CR review for mediacapture-fromelement (#50)
Closed: [mediacapture-fromelement] What happens to the audio being rendered to a Media Element when it gets captureStream()ed (#34)
Closed: [mediacapture-output] Persisting deviceIds Across Sessions (#127)
Closed: [mediacapture-region] Behavior when full screened (#81)
Closed: [webrtc-pc] ICE finished gathering algorithm touches state after firing an event (#2893)
Closed: [webrtc-pc] Put a=end-of-candidates into description(s) (#2896)
Closed: [webrtc-pc] Redundant queuing in gathering state computation risks eliding events (#2892)
Closed: [webrtc-pc] RTCIceCandidate's relayProtocol and url members can be null but not absent (#2947)
Last message date: Sunday, 31 March 2024 19:02:57 UTC