Re: [mediacapture-image] Use constrainable pattern for ImageCapture (second take) (#150)
[mediacapture-output] Set Default Audio Output Device in MediaDevices (#141)
Closed: [webrtc-pc] Put a=end-of-candidates into description(s) (#2896)
Closed: [webrtc-pc] ICE finished gathering algorithm touches state after firing an event (#2893)
[webrtc-pc] new commits pushed by jan-ivar
Closed: [webrtc-pc] Redundant queuing in gathering state computation risks eliding events (#2892)
[webrtc-extensions] Pull Request: Move jitterBufferTarget to main spec
[webrtc-pc] Pull Request: Integrate jitterBufferTarget as candidate addition
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-pc] new commits pushed by youennf
Re: [webrtc-pc] Consider making RTCIceCandidatePair an interface (#2930)
- Re: [webrtc-pc] Consider making RTCIceCandidatePair an interface (#2930)
- Re: [webrtc-pc] Consider making RTCIceCandidatePair an interface (#2930)
- Re: [webrtc-pc] Consider making RTCIceCandidatePair an interface (#2930)
[webrtc-pc] new commits pushed by aboba
Re: [webrtc-pc] Queue two tasks upon finishing ICE gathering, and fire gatheringstatechange & icegatheringstatechange in same task (#2894)
- Re: [webrtc-pc] Queue two tasks upon finishing ICE gathering, and fire gatheringstatechange & icegatheringstatechange in same task (#2894)
- Re: [webrtc-pc] Queue two tasks upon finishing ICE gathering, and fire gatheringstatechange & icegatheringstatechange in same task (#2894)
[webrtc-extensions] Pull Request: Clean references and fix links
[webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)
[webrtc-pc] Incorporate jitterBufferTarget (#2952)
[webrtc-pc] Pull Request: Make RTCIceCandidatePair members required
Re: [webrtc-encoded-transform] Constructor frame (#223)
Re: [mediacapture-fromelement] Inconsistency: Taint, not mute cross-origin element tracks (#83)
Re: [mediacapture-fromelement] define behaviors of the common ConstrainablePattern Interfaces (#48)
Re: [mediacapture-fromelement] Review mute/unmute/ended and constraints on tracks from element.captureStream() (#98)
Re: [mediacapture-main] Clarify each source is responsible for specifying mute/unmute/ended and constraints behavior (#984)
Re: [webrtc-pc] Alternative storage for RTCCertificates needed (#2944)
- Re: [webrtc-pc] Alternative storage for RTCCertificates needed (#2944)
- Re: [webrtc-pc] Alternative storage for RTCCertificates needed (#2944)
Re: [mediacapture-record] mimeType ambiguity: "video/webm;codecs=vp8" means? (#194)
Re: [mediacapture-fromelement] Review mute/unmute/ended and constraints on tracks from canvas.captureStream() (#99)
Re: [mediacapture-fromelement] Clarify if CanvasCaptureMediaStreamTrack mute, unmute, and ended events are expected to be fired (#82)
Re: [mediacapture-transform] Review mute/unmute/ended and constraints on new VideoTrackGenerator().track (#109)
Re: [webrtc-pc] WebRTC spec should explicitly specify all causes of a PeerConnection-sourced track receiving mute/unmute (#2915)
Re: [webrtc-pc] Review mute/unmute/ended and constraints on RTCRtpReceiver's track. (#2942)
Re: [mediacapture-screen-share] Review mute/unmute/ended and constraints on tracks from getDisplayMedia() (#298)
[webrtc-rtptransport] Pull Request: Add "motivation" section, largely copied from IceController explainer
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] Pull Request: Add "Use Case 2"
[webrtc-rtptransport] Pull Request: Cleanup Use Case 1
[webrtc-pc] Missing tests for candidate amendments (#2950)
[webrtc-pc] Pull Request: Fix bugs in amendments
Re: [webrtc-pc] Clarify unmute event must fire on receiver.track AFTER sRD(offer) succeeds (#2880)
Re: [mediacapture-main] Fix export of track-muted and set-track-muted. (#986)
[webrtc-pc] new commits pushed by jan-ivar
[webrtc-pc] new commits pushed by jan-ivar
Re: [mediacapture-screen-share] Define the windowAudio option (#283)
[webrtc-pc] Pull Request: Clarify when RTCIceCandidate's relayProtocol and url members are null.
[webrtc-pc] RTCIceCandidate's relayProtocol and url members can be null but not absent (#2947)
Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
- Re: [webrtc-extensions] Add jitterBufferMaximumDelay attribute to RTCRtpReceiver (#199)
[mediacapture-main] new commits pushed by dontcallmedom
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] Pull Request: Update to ReSpec version 34.5.0
[mediacapture-main] Pull Request: Update to latest ReSpec version 34.5.0
Re: [webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)
- Re: [webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)
- Re: [webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)
- Re: [webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)
Re: [webrtc-pc] Update the accessibility section 14 to include RFC 8865 for real-time text in WebRTC data channel (#2931)
- Re: [webrtc-pc] Update the accessibility section 14 to include RFC 8865 for real-time text in WebRTC data channel (#2931)
- Re: [webrtc-pc] Update the accessibility section 14 to include RFC 8865 for real-time text in WebRTC data channel (#2931)
- Re: [webrtc-pc] Update the accessibility section 14 to include RFC 8865 for real-time text in WebRTC data channel (#2931)
- Re: [webrtc-pc] Update the accessibility section 14 to include RFC 8865 for real-time text in WebRTC data channel (#2931)
Re: [webrtc-pc] Proposing setCodecPreferences to deal with both send and recv codecs (#2939)
Re: [mediacapture-fromelement] WG CR review for mediacapture-fromelement (#50)
Closed: [mediacapture-fromelement] WG CR review for mediacapture-fromelement (#50)
Re: [mediacapture-fromelement] CD publication of specification (#47)
Closed: [mediacapture-fromelement] CD publication of specification (#47)
Re: [mediacapture-fromelement] What happens to the audio being rendered to a Media Element when it gets captureStream()ed (#34)
Closed: [mediacapture-fromelement] What happens to the audio being rendered to a Media Element when it gets captureStream()ed (#34)
Re: [webrtc-extensions] Peer Connection and back/forward cache (#200)
Re: [webrtc-extensions] Add API to control jitterBufferTarget handling (#199)
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[mediacapture-region] is this still an experimental feature? (#82)
[webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
- Re: [webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
- Re: [webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
- Re: [webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
- Re: [webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
- Re: [webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
- Re: [webrtc-encoded-transform] Expose RTCEncoded*Frame interfaces in Worklets (#226)
[webrtc-stats] RTCStats.timestamp - fingerprinting and since epoch (#786)
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-stats] new commits pushed by jan-ivar
Re: [mediacapture-transform] Is MediaStreamTrackProcessor for audio necessary? (#29)
Re: [webrtc-pc] Document test updates associated with amendments (#2910)
[mediacapture-main] new commits pushed by youennf
Re: [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)
Re: [webrtc-encoded-transform] Expose captureTimestamp to RTCEncodedAudioFrame and RTCEncodedVideoFrame (#159)
[mediacapture-region] Behavior when full screened (#81)
- Re: [mediacapture-region] Behavior when full screened (#81)
- Closed: [mediacapture-region] Behavior when full screened (#81)
- Re: [mediacapture-region] Behavior when full screened (#81)
[mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
- Re: [webrtc-extensions] Add RTCRtpEncodedSource and explainer (#198)
Re: [mediacapture-output] Persisting deviceIds Across Sessions (#127)
Closed: [mediacapture-output] Persisting deviceIds Across Sessions (#127)
[mediacapture-main] risk model of stored permissions and constraint opportunities (#991)
- Re: [mediacapture-main] risk model of stored permissions and constraint opportunities (#991)
- Re: [mediacapture-main] risk model of stored permissions and constraint opportunities (#991)