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

Friday, 30 August 2024

[webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
[webrtc-pc] Pull Request: Improve description of amendment 36 Dominique Hazael-Massieux via GitHub
[webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
[webrtc-pc] Pull Request: Improve and complete description of amendments 16, 18, 29, 49 Dominique Hazael-Massieux via GitHub
[webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
[webrtc-pc] Pull Request: Fix description of amendments 23, 31, 47 Dominique Hazael-Massieux via GitHub
[webrtc-encoded-transform] Enum values that ignore naming conventions in WebRTC Encoded Transform (#232) dontcallmedom-bot via GitHub
[webrtc-encoded-transform] No transfer steps defined for RTCEncodedAudioFrame and RTCEncodedVideoFrame (#231) luciaalonsomozo via GitHub

Thursday, 29 August 2024

[mediacapture-output] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
[mediacapture-output] Pull Request: Mark Justin as former editor Dominique Hazael-Massieux via GitHub
Re: [webrtc-pc] Don't validate codec input to setParameters by preferred receive codecs (#2991) Jan-Ivar Bruaroey via GitHub
Closed: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142) Jan-Ivar Bruaroey via GitHub
[mediacapture-output] new commits pushed by jan-ivar Jan-Ivar Bruaroey via GitHub
[webrtc-pc] new commits pushed by jan-ivar Jan-Ivar Bruaroey via GitHub
[webrtc-pc] Pull Request: Don't validate codec input to setParameters by preferred receive codecs Jan-Ivar Bruaroey via GitHub
[webrtc-extensions] Pull Request: Add restrictedResolution (aka scaleResolutionDownTo) to RTCRtpEncodingParameters henbos via GitHub
Re: [webrtc-extensions] RTCRtpEncodingParameters: scaleResolutionTo (#159) henbos via GitHub
Re: [mediacapture-main] Should a muted video track be allowed to deliver black frames to its sinks? (#1011) youennf via GitHub
[mediacapture-main] Should a muted video track be allowed to deliver black frames to its sinks? (#1011) youennf via GitHub
Re: [webrtc-pc] codec input to setParameters shouldn't be validated by preferred receive codecs (#2989) Harald Alvestrand via GitHub

Wednesday, 28 August 2024

[webrtc-pc] Pull Request: Reference the codec parameter in algorithms. Jan-Ivar Bruaroey via GitHub
[webrtc-pc] codec input to setParameters shouldn't be validated by receive preferred codecs (#2989) Jan-Ivar Bruaroey via GitHub
Re: [webrtc-extensions] ReSpec errors from duplicate definitions of RTCIceCandidatePair (#217) Dominique Hazael-Massieux via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) dontcallmedom-bot via GitHub
Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) dontcallmedom-bot via GitHub
Re: [mediacapture-output] Allow prompt bypass for miked speakers exposed through getUserMedia() (#143) dontcallmedom-bot via GitHub
Re: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142) dontcallmedom-bot via GitHub
Re: [mediacapture-main] Avoid circular definition of muted. (#982) dontcallmedom-bot via GitHub
Re: [webrtc-extensions] RTCRtpEncodingParameters: scaleResolutionTo (#159) dontcallmedom-bot via GitHub

Tuesday, 27 August 2024

Closed: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785) Harald Alvestrand via GitHub
Re: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785) Harald Alvestrand via GitHub
Re: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785) Varun Singh via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Florent Castelli via GitHub
Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) Jan-Ivar Bruaroey via GitHub
Closed: [mediacapture-region] Product manager founder (#83) Dominique Hazael-Massieux via GitHub
[mediacapture-region] Product manager founder (#83) 1B2A3b4j5i via GitHub
Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) Karl Tomlinson via GitHub

Monday, 26 August 2024

Saturday, 24 August 2024

Re: [mediacapture-screen-share-extensions] Want to restrict the options of the getDisplayMedia API (#1) RobinMa via GitHub
Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) Jan-Ivar Bruaroey via GitHub

Friday, 23 August 2024

[mediacapture-output] Pull Request: Allow prompt bypass for miked speakers exposed through getUserMedia() Jan-Ivar Bruaroey via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Jan-Ivar Bruaroey via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Jan-Ivar Bruaroey via GitHub

Thursday, 22 August 2024

Re: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142) youennf via GitHub
Closed: [webrtc-extensions] Migrate RTCRtpEncodingParameters.codec to main spec (#212) Florent Castelli via GitHub
[webrtc-extensions] new commits pushed by Orphis Florent Castelli via GitHub
[webrtc-pc] new commits pushed by Orphis Florent Castelli via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) youennf via GitHub

Wednesday, 21 August 2024

[webrtc-extensions] Pull Request: Move transferable data channels to main spec Dominique Hazael-Massieux via GitHub
Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) guidou via GitHub
[webrtc-pc] Pull Request: Make data channels transferable to DedicatedWorker Dominique Hazael-Massieux via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Harald Alvestrand via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) youennf via GitHub

Tuesday, 20 August 2024

Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Harald Alvestrand via GitHub
Closed: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Peter Thatcher via GitHub
[webrtc-rtptransport] new commits pushed by pthatcher Peter Thatcher via GitHub
Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Tony Herre via GitHub
Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Peter Thatcher via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Harald Alvestrand via GitHub

Monday, 19 August 2024

Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Philipp Hancke via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Thomas Guilbert via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Philipp Hancke via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) Bernard Aboba via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
[webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) docfaraday via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Dominique Hazael-Massieux via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Harald Alvestrand via GitHub
Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Dominique Hazael-Massieux via GitHub
Re: [webrtc-extensions] Move RTCRtpEncodingParameters.codec to main spec (#219) Florent Castelli via GitHub
Re: [webrtc-extensions] Support ICE Continuous Gathering flag in RTCConfiguration (#121) Harald Alvestrand via GitHub
[webrtc-extensions] Pull Request: Move RTCRtpEncodingParameters.codec to main spec Dominique Hazael-Massieux via GitHub
Re: [webrtc-extensions] Support ICE Continuous Gathering flag in RTCConfiguration (#121) Lenny via GitHub
[webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub

Saturday, 17 August 2024

[webrtc-svc] new commits pushed by aboba Bernard Aboba via GitHub
[webrtc-svc] new commits pushed by aboba Bernard Aboba via GitHub
[webrtc-svc] new commits pushed by aboba Bernard Aboba via GitHub

Friday, 16 August 2024

Re: [webrtc-rtptransport] Make RtpTransportProcessor transferable (#33) Tony Herre via GitHub

Thursday, 15 August 2024

[webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) Thomas Guilbert via GitHub
Re: [webrtc-extensions] Implementation commitments for transferable data channels (#214) Thomas Guilbert via GitHub
Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) Philipp Hancke via GitHub
Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) Jan-Ivar Bruaroey via GitHub
Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) Jan-Ivar Bruaroey via GitHub
Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) guidou via GitHub
Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) Harald Alvestrand via GitHub
Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) Harald Alvestrand via GitHub
Re: [webrtc-pc] Merge RTCRtpEncodingParameters.codec from webrtc-extensions (#2985) Jan-Ivar Bruaroey via GitHub
[webrtc-pc] Pull Request: Merge RTCRtpEncodingParameters.codec from webrtc-extensions Florent Castelli via GitHub
Re: [webrtc-encoded-transform] Clarification requested (#230) luciaalonsomozo via GitHub
[webrtc-encoded-transform] Clarification requested (#230) luciaalonsomozo via GitHub

Wednesday, 14 August 2024

Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Harald Alvestrand via GitHub
Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Bernard Aboba via GitHub

Tuesday, 13 August 2024

Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Harald Alvestrand via GitHub
Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Florent Castelli via GitHub
[webrtc-rtptransport] Pull Request: Create RTCRtpTransportProcessor and move high freq fields there Tony Herre via GitHub
Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Harald Alvestrand via GitHub
[webrtc-rtptransport] Pull Request: Remove unmotivated addRtpXStreams methods Tony Herre via GitHub
[webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) Tony Herre via GitHub

Monday, 12 August 2024

Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) Florent Castelli via GitHub
Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106) Harald Alvestrand via GitHub

Saturday, 10 August 2024

Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) Bernard Aboba via GitHub
Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106) vitaly-castLabs via GitHub

Friday, 9 August 2024

[webrtc-rtptransport] Pull Request: Make RTCRtpSendStream per mid, add rid to RTCRtpPacket/Init Florent Castelli via GitHub
Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) Florent Castelli via GitHub
Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106) Philipp Hancke via GitHub
Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106) vitaly-castLabs via GitHub
Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106) vitaly-castLabs via GitHub

Thursday, 8 August 2024

Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992) Bernard Aboba via GitHub
Closed: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992) Bernard Aboba via GitHub
Re: [mediacapture-main] Define getMuteReasons() (#979) Bernard Aboba via GitHub
[webrtc-priority] Migrate RTCDataChannel.priority to main spec (#24) Florent Castelli via GitHub
Re: [webrtc-extensions] Migrate adaptivePtime to main spec (#211) Bernard Aboba via GitHub
Re: [webrtc-extensions] Implementation commitments for transferable data channels (#214) Florent Castelli via GitHub

Tuesday, 6 August 2024

Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) Peter Thatcher via GitHub
Closed: [webrtc-rtptransport] BYOB also needs fields giving the byte length (#62) Bernard Aboba via GitHub
[webrtc-rtptransport] new commits pushed by aboba Bernard Aboba via GitHub
Closed: [webrtc-rtptransport] Allow inserting padding into packets (#60) Bernard Aboba via GitHub
[webrtc-rtptransport] new commits pushed by aboba Bernard Aboba via GitHub
Re: [webrtc-extensions] Use RTCIceCandidatePair interface in RTCIceTransport (#205) Sameer via GitHub
Re: [webrtc-extensions] Clarify when `icecandidatepairremove` is fired. (#204) Sameer via GitHub
[webrtc-extensions] Pull Request: Fix RTCIceCandidatePair links to suppress ambiguity warnings. Sameer via GitHub
[webrtc-extensions] ReSpec errors from duplicate definitions of RTCIceCandidatePair (#217) Sameer via GitHub
[webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) Florent Castelli via GitHub

Monday, 5 August 2024

[webrtc-pc] Unspecified whether the data channel error event fires from SCTP ABORT (#2984) Jan-Ivar Bruaroey via GitHub
Re: [webrtc-stats] (jitterBufferDelay/jitterBufferEmittedCount * 1000) from pc.getStats is not equal to jitterBufferDelay/jitterBufferEmittedCount_in_ms in chrome://webrtc-internal (#590) Harald Alvestrand via GitHub

Sunday, 4 August 2024

Re: [webrtc-stats] (jitterBufferDelay/jitterBufferEmittedCount * 1000) from pc.getStats is not equal to jitterBufferDelay/jitterBufferEmittedCount_in_ms in chrome://webrtc-internal (#590) zhuya1996 via GitHub

Friday, 2 August 2024

Re: [webrtc-rtptransport] Add byte length fields for BYOB methods (#63) Tony Herre via GitHub

Thursday, 1 August 2024

Re: [webrtc-rtptransport] Add byte length fields for BYOB methods (#63) Bernard Aboba via GitHub
Re: [webrtc-encoded-transform] Refactor spec to introduce media thread (#107) Jan-Ivar Bruaroey via GitHub
Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201) Harald Alvestrand via GitHub
Re: [webrtc-encoded-transform] Document legacy API that was removed in #64 (#111) Harald Alvestrand via GitHub
[mst-content-hint] new commits pushed by alvestrand Harald Alvestrand via GitHub
Re: [webrtc-rtptransport] Make RtpTransportProcessor transferable (#33) Tony Herre via GitHub
Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201) Chen Xing via GitHub
Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201) Johannes Kron via GitHub
[webrtc-rtptransport] Pull Request: Add byte length fields for BYOB methods Tony Herre via GitHub
[webrtc-rtptransport] BYOB also needs fields giving the byte length (#62) Tony Herre via GitHub
[webrtc-rtptransport] Pull Request: Add paddingBytes fields Tony Herre via GitHub
[webrtc-rtptransport] Allow inserting padding into packets (#60) Tony Herre via GitHub

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