1B2A3b4j5i via GitHub
Bernard Aboba via GitHub
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) (Monday, 19 August)
- [webrtc-svc] new commits pushed by aboba (Saturday, 17 August)
- [webrtc-svc] new commits pushed by aboba (Saturday, 17 August)
- [webrtc-svc] new commits pushed by aboba (Saturday, 17 August)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Wednesday, 14 August)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) (Saturday, 10 August)
- Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992) (Thursday, 8 August)
- Closed: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992) (Thursday, 8 August)
- Re: [mediacapture-main] Define getMuteReasons() (#979) (Thursday, 8 August)
- Re: [webrtc-extensions] Migrate adaptivePtime to main spec (#211) (Thursday, 8 August)
- Closed: [webrtc-rtptransport] BYOB also needs fields giving the byte length (#62) (Tuesday, 6 August)
- [webrtc-rtptransport] new commits pushed by aboba (Tuesday, 6 August)
- Closed: [webrtc-rtptransport] Allow inserting padding into packets (#60) (Tuesday, 6 August)
- [webrtc-rtptransport] new commits pushed by aboba (Tuesday, 6 August)
- Re: [webrtc-rtptransport] Add byte length fields for BYOB methods (#63) (Thursday, 1 August)
Chen Xing via GitHub
docfaraday via GitHub
Dominique Hazael-Massieux via GitHub
- [webrtc-pc] new commits pushed by dontcallmedom (Friday, 30 August)
- [webrtc-pc] Pull Request: Improve description of amendment 36 (Friday, 30 August)
- [webrtc-pc] new commits pushed by dontcallmedom (Friday, 30 August)
- [webrtc-pc] Pull Request: Improve and complete description of amendments 16, 18, 29, 49 (Friday, 30 August)
- [webrtc-pc] new commits pushed by dontcallmedom (Friday, 30 August)
- [webrtc-pc] Pull Request: Fix description of amendments 23, 31, 47 (Friday, 30 August)
- [mediacapture-output] new commits pushed by dontcallmedom (Thursday, 29 August)
- [mediacapture-output] Pull Request: Mark Justin as former editor (Thursday, 29 August)
- Re: [webrtc-extensions] ReSpec errors from duplicate definitions of RTCIceCandidatePair (#217) (Wednesday, 28 August)
- Closed: [mediacapture-region] Product manager founder (#83) (Tuesday, 27 August)
- [webrtc-extensions] Pull Request: Move transferable data channels to main spec (Wednesday, 21 August)
- [webrtc-pc] Pull Request: Make data channels transferable to DedicatedWorker (Wednesday, 21 August)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) (Monday, 19 August)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) (Monday, 19 August)
- [webrtc-extensions] Pull Request: Move RTCRtpEncodingParameters.codec to main spec (Monday, 19 August)
- [webrtc-pc] new commits pushed by dontcallmedom (Monday, 19 August)
dontcallmedom-bot via GitHub
Florent Castelli via GitHub
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) (Tuesday, 27 August)
- Closed: [webrtc-extensions] Migrate RTCRtpEncodingParameters.codec to main spec (#212) (Thursday, 22 August)
- [webrtc-extensions] new commits pushed by Orphis (Thursday, 22 August)
- [webrtc-pc] new commits pushed by Orphis (Thursday, 22 August)
- Re: [webrtc-extensions] Move RTCRtpEncodingParameters.codec to main spec (#219) (Monday, 19 August)
- [webrtc-pc] Pull Request: Merge RTCRtpEncodingParameters.codec from webrtc-extensions (Thursday, 15 August)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Tuesday, 13 August)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) (Monday, 12 August)
- [webrtc-rtptransport] Pull Request: Make RTCRtpSendStream per mid, add rid to RTCRtpPacket/Init (Friday, 9 August)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) (Friday, 9 August)
- [webrtc-priority] Migrate RTCDataChannel.priority to main spec (#24) (Thursday, 8 August)
- Re: [webrtc-extensions] Implementation commitments for transferable data channels (#214) (Thursday, 8 August)
- [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64) (Tuesday, 6 August)
guidou via GitHub
Harald Alvestrand via GitHub
- Re: [webrtc-pc] codec input to setParameters shouldn't be validated by preferred receive codecs (#2989) (Thursday, 29 August)
- Closed: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785) (Tuesday, 27 August)
- Re: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785) (Tuesday, 27 August)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) (Wednesday, 21 August)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) (Tuesday, 20 August)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) (Tuesday, 20 August)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986) (Monday, 19 August)
- Re: [webrtc-extensions] Support ICE Continuous Gathering flag in RTCConfiguration (#121) (Monday, 19 August)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) (Thursday, 15 August)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) (Thursday, 15 August)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Wednesday, 14 August)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Tuesday, 13 August)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Tuesday, 13 August)
- Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106) (Monday, 12 August)
- Re: [webrtc-stats] (jitterBufferDelay/jitterBufferEmittedCount * 1000) from pc.getStats is not equal to jitterBufferDelay/jitterBufferEmittedCount_in_ms in chrome://webrtc-internal (#590) (Monday, 5 August)
- Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201) (Thursday, 1 August)
- Re: [webrtc-encoded-transform] Document legacy API that was removed in #64 (#111) (Thursday, 1 August)
- [mst-content-hint] new commits pushed by alvestrand (Thursday, 1 August)
henbos via GitHub
Jan-Ivar Bruaroey via GitHub
- Re: [webrtc-pc] Don't validate codec input to setParameters by preferred receive codecs (#2991) (Thursday, 29 August)
- Closed: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142) (Thursday, 29 August)
- [mediacapture-output] new commits pushed by jan-ivar (Thursday, 29 August)
- [webrtc-pc] new commits pushed by jan-ivar (Thursday, 29 August)
- [webrtc-pc] Pull Request: Don't validate codec input to setParameters by preferred receive codecs (Thursday, 29 August)
- [webrtc-pc] Pull Request: Reference the codec parameter in algorithms. (Wednesday, 28 August)
- [webrtc-pc] codec input to setParameters shouldn't be validated by receive preferred codecs (#2989) (Wednesday, 28 August)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) (Tuesday, 27 August)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) (Monday, 26 August)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) (Monday, 26 August)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) (Monday, 26 August)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133) (Saturday, 24 August)
- [mediacapture-output] Pull Request: Allow prompt bypass for miked speakers exposed through getUserMedia() (Friday, 23 August)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) (Friday, 23 August)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987) (Friday, 23 August)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) (Thursday, 15 August)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230) (Thursday, 15 August)
- Re: [webrtc-pc] Merge RTCRtpEncodingParameters.codec from webrtc-extensions (#2985) (Thursday, 15 August)
- [webrtc-pc] Unspecified whether the data channel error event fires from SCTP ABORT (#2984) (Monday, 5 August)
- Re: [webrtc-encoded-transform] Refactor spec to introduce media thread (#107) (Thursday, 1 August)
Johannes Kron via GitHub
Karl Tomlinson via GitHub
Lenny via GitHub
luciaalonsomozo via GitHub
Peter Thatcher via GitHub
Philipp Hancke via GitHub
RobinMa via GitHub
Sameer via GitHub
Thomas Guilbert via GitHub
Tony Herre via GitHub
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Tuesday, 20 August)
- Re: [webrtc-rtptransport] Make RtpTransportProcessor transferable (#33) (Friday, 16 August)
- [webrtc-rtptransport] Pull Request: Create RTCRtpTransportProcessor and move high freq fields there (Tuesday, 13 August)
- [webrtc-rtptransport] Pull Request: Remove unmotivated addRtpXStreams methods (Tuesday, 13 August)
- [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66) (Tuesday, 13 August)
- Re: [webrtc-rtptransport] Add byte length fields for BYOB methods (#63) (Friday, 2 August)
- Re: [webrtc-rtptransport] Make RtpTransportProcessor transferable (#33) (Thursday, 1 August)
- [webrtc-rtptransport] Pull Request: Add byte length fields for BYOB methods (Thursday, 1 August)
- [webrtc-rtptransport] BYOB also needs fields giving the byte length (#62) (Thursday, 1 August)
- [webrtc-rtptransport] Pull Request: Add paddingBytes fields (Thursday, 1 August)
- [webrtc-rtptransport] Allow inserting padding into packets (#60) (Thursday, 1 August)
Varun Singh via GitHub
vitaly-castLabs via GitHub
youennf via GitHub
zhuya1996 via GitHub
Last message date: Friday, 30 August 2024 17:26:30 UTC