[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] Pull Request: Improve description of amendment 36
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] Pull Request: Improve and complete description of amendments 16, 18, 29, 49
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-pc] Pull Request: Fix description of amendments 23, 31, 47
[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)
[mediacapture-output] new commits pushed by dontcallmedom
[mediacapture-output] Pull Request: Mark Justin as former editor
Closed: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142)
[mediacapture-output] new commits pushed by jan-ivar
[webrtc-pc] new commits pushed by jan-ivar
[webrtc-pc] Pull Request: Don't validate codec input to setParameters by preferred receive codecs
[webrtc-extensions] Pull Request: Add restrictedResolution (aka scaleResolutionDownTo) to RTCRtpEncodingParameters
[mediacapture-main] Should a muted video track be allowed to deliver black frames to its sinks? (#1011)
[webrtc-pc] Pull Request: Reference the codec parameter in algorithms.
[webrtc-pc] codec input to setParameters shouldn't be validated by receive preferred codecs (#2989)
Re: [mediacapture-main] Avoid circular definition of muted. (#982)
Re: [webrtc-extensions] RTCRtpEncodingParameters: scaleResolutionTo (#159)
Closed: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785)
Re: [webrtc-stats] RTCCodecStats.clockRate - media sampling rate or the codec clock rate? (#785)
[mediacapture-region] Product manager founder (#83)
Re: [mediacapture-screen-share-extensions] Want to restrict the options of the getDisplayMedia API (#1)
[mediacapture-output] Pull Request: Allow prompt bypass for miked speakers exposed through getUserMedia()
Re: [mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142)
Closed: [webrtc-extensions] Migrate RTCRtpEncodingParameters.codec to main spec (#212)
[webrtc-extensions] new commits pushed by Orphis
[webrtc-pc] new commits pushed by Orphis
[webrtc-extensions] Pull Request: Move transferable data channels to main spec
Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
- Re: [mediacapture-output] The first "audiooutput" `MediaDeviceInfo` returned from `enumerateDevices()` is not the default device when the default device is not exposed (#133)
[webrtc-pc] Pull Request: Make data channels transferable to DedicatedWorker
[webrtc-rtptransport] new commits pushed by pthatcher
[webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
- Re: [webrtc-pc] RTCRtpParameters.codec matching is probably too strict (#2987)
[webrtc-extensions] Pull Request: Move RTCRtpEncodingParameters.codec to main spec
Re: [webrtc-extensions] Support ICE Continuous Gathering flag in RTCConfiguration (#121)
[webrtc-pc] new commits pushed by dontcallmedom
[webrtc-svc] new commits pushed by aboba
[webrtc-svc] new commits pushed by aboba
[webrtc-svc] new commits pushed by aboba
[webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
- Re: [webrtc-pc] Consider merging transferable RTCDataChannel from webrtc-extensions (#2986)
[webrtc-pc] Pull Request: Merge RTCRtpEncodingParameters.codec from webrtc-extensions
[webrtc-encoded-transform] Clarification requested (#230)
- Re: [webrtc-encoded-transform] Clarification requested (#230)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)
- Re: [webrtc-encoded-transform] Clarification on "not processing video packets" requested (#230)
[webrtc-rtptransport] Pull Request: Create RTCRtpTransportProcessor and move high freq fields there
[webrtc-rtptransport] Pull Request: Remove unmotivated addRtpXStreams methods
[webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Re: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
- Closed: [webrtc-rtptransport] Unsignalled streams aren't needed by any existing usecase (#66)
[webrtc-rtptransport] Pull Request: Make RTCRtpSendStream per mid, add rid to RTCRtpPacket/Init
Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106)
- Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106)
- Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106)
- Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106)
- Re: [webrtc-encoded-transform] Add use cases that require one-ended encoded streams (#106)
Re: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
Closed: [mediacapture-main] (webkit|moz)getUserMedia usage and webcompat issues (#992)
Re: [mediacapture-main] Define getMuteReasons() (#979)
[webrtc-priority] Migrate RTCDataChannel.priority to main spec (#24)
Re: [webrtc-extensions] Migrate adaptivePtime to main spec (#211)
Re: [webrtc-extensions] Implementation commitments for transferable data channels (#214)
[webrtc-rtptransport] new commits pushed by aboba
[webrtc-rtptransport] new commits pushed by aboba
Re: [webrtc-extensions] Use RTCIceCandidatePair interface in RTCIceTransport (#205)
Re: [webrtc-extensions] Clarify when `icecandidatepairremove` is fired. (#204)
[webrtc-extensions] Pull Request: Fix RTCIceCandidatePair links to suppress ambiguity warnings.
[webrtc-extensions] ReSpec errors from duplicate definitions of RTCIceCandidatePair (#217)
[webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64)
- Re: [webrtc-rtptransport] RTCRtpSender.replaceSendStreams() and simulcast issues (#64)
[webrtc-pc] Unspecified whether the data channel error event fires from SCTP ABORT (#2984)
Re: [webrtc-stats] (jitterBufferDelay/jitterBufferEmittedCount * 1000) from pc.getStats is not equal to jitterBufferDelay/jitterBufferEmittedCount_in_ms in chrome://webrtc-internal (#590)
Re: [webrtc-encoded-transform] Refactor spec to introduce media thread (#107)
Re: [webrtc-encoded-transform] Document legacy API that was removed in #64 (#111)
[mst-content-hint] new commits pushed by alvestrand
Re: [webrtc-rtptransport] Make RtpTransportProcessor transferable (#33)
Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)
- Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)
- Re: [webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)
[webrtc-rtptransport] Pull Request: Add byte length fields for BYOB methods
- Re: [webrtc-rtptransport] Add byte length fields for BYOB methods (#63)
- Re: [webrtc-rtptransport] Add byte length fields for BYOB methods (#63)