public-webrtc-logs@w3.org from August 2024 by subject

[mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)

[mediacapture-main] Avoid circular definition of muted. (#982)

[mediacapture-main] Define getMuteReasons() (#979)

[mediacapture-main] Should a muted video track be allowed to deliver black frames to its sinks? (#1011)

[mediacapture-output] Allow prompt bypass for miked speakers exposed through getUserMedia() (#143)

[mediacapture-output] new commits pushed by dontcallmedom

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

[mediacapture-output] Pull Request: Allow prompt bypass for miked speakers exposed through getUserMedia()

[mediacapture-output] Pull Request: Mark Justin as former editor

[mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)

[mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142)

[mediacapture-region] Product manager founder (#83)

[mediacapture-screen-share-extensions] Want to restrict the options of the getDisplayMedia API (#1)

[mst-content-hint] new commits pushed by alvestrand

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

[webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)

[webrtc-encoded-transform] Clarification requested (#230)

[webrtc-encoded-transform] Document legacy API that was removed in #64 (#111)

[webrtc-encoded-transform] Enum values that ignore naming conventions in WebRTC Encoded Transform (#232)

[webrtc-encoded-transform] No transfer steps defined for RTCEncodedAudioFrame and RTCEncodedVideoFrame (#231)

[webrtc-encoded-transform] Refactor spec to introduce media thread (#107)

[webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)

[webrtc-extensions] Clarify when `icecandidatepairremove` is fired. (#204)

[webrtc-extensions] Implementation commitments for transferable data channels (#214)

[webrtc-extensions] Migrate adaptivePtime to main spec (#211)

[webrtc-extensions] Move RTCRtpEncodingParameters.codec to main spec (#219)

[webrtc-extensions] new commits pushed by Orphis

[webrtc-extensions] Pull Request: Add restrictedResolution (aka scaleResolutionDownTo) to RTCRtpEncodingParameters

[webrtc-extensions] Pull Request: Fix RTCIceCandidatePair links to suppress ambiguity warnings.

[webrtc-extensions] Pull Request: Move RTCRtpEncodingParameters.codec to main spec

[webrtc-extensions] Pull Request: Move transferable data channels to main spec

[webrtc-extensions] ReSpec errors from duplicate definitions of RTCIceCandidatePair (#217)

[webrtc-extensions] RTCRtpEncodingParameters: scaleResolutionTo (#159)

[webrtc-extensions] Support ICE Continuous Gathering flag in RTCConfiguration (#121)

[webrtc-extensions] Use RTCIceCandidatePair interface in RTCIceTransport (#205)

[webrtc-pc] codec input to setParameters shouldn't be validated by preferred receive codecs (#2989)

[webrtc-pc] codec input to setParameters shouldn't be validated by receive preferred codecs (#2989)

[webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)

[webrtc-pc] Don't validate codec input to setParameters by preferred receive codecs (#2991)

[webrtc-pc] Merge RTCRtpEncodingParameters.codec from webrtc-extensions (#2985)

[webrtc-pc] new commits pushed by dontcallmedom

[webrtc-pc] new commits pushed by jan-ivar

[webrtc-pc] new commits pushed by Orphis

[webrtc-pc] Pull Request: Don't validate codec input to setParameters by preferred receive codecs

[webrtc-pc] Pull Request: Fix description of amendments 23, 31, 47

[webrtc-pc] Pull Request: Improve and complete description of amendments 16, 18, 29, 49

[webrtc-pc] Pull Request: Improve description of amendment 36

[webrtc-pc] Pull Request: Make data channels transferable to DedicatedWorker

[webrtc-pc] Pull Request: Merge RTCRtpEncodingParameters.codec from webrtc-extensions

[webrtc-pc] Pull Request: Reference the codec parameter in algorithms.

[webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)

[webrtc-pc] Unspecified whether the data channel error event fires from SCTP ABORT (#2984)

[webrtc-priority] Migrate RTCDataChannel.priority to main spec (#24)

[webrtc-rtptransport] Add byte length fields for BYOB methods (#63)

[webrtc-rtptransport] Allow inserting padding into packets (#60)

[webrtc-rtptransport] BYOB also needs fields giving the byte length (#62)

[webrtc-rtptransport] Make RtpTransportProcessor transferable (#33)

[webrtc-rtptransport] new commits pushed by aboba

[webrtc-rtptransport] new commits pushed by pthatcher

[webrtc-rtptransport] Pull Request: Add byte length fields for BYOB methods

[webrtc-rtptransport] Pull Request: Add paddingBytes fields

[webrtc-rtptransport] Pull Request: Create RTCRtpTransportProcessor and move high freq fields there

[webrtc-rtptransport] Pull Request: Make RTCRtpSendStream per mid, add rid to RTCRtpPacket/Init

[webrtc-rtptransport] Pull Request: Remove unmotivated addRtpXStreams methods

[webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64)

[webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)

[webrtc-stats] (jitterBufferDelay/jitterBufferEmittedCount * 1000) from pc.getStats is not equal to jitterBufferDelay/jitterBufferEmittedCount_in_ms in chrome://webrtc-internal (#590)

[webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785)

[webrtc-svc] new commits pushed by aboba

Closed: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)

Closed: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142)

Closed: [mediacapture-region] Product manager founder (#83)

Closed: [webrtc-extensions] Migrate RTCRtpEncodingParameters.codec to main spec (#212)

Closed: [webrtc-rtptransport] Allow inserting padding into packets (#60)

Closed: [webrtc-rtptransport] BYOB also needs fields giving the byte length (#62)

Closed: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)

Closed: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785)

Last message date: Friday, 30 August 2024 17:26:30 UTC