[mediacapture-main] Get raw data from MediaStream (#327)
[mediacapture-output] Why prompt for a subset of stored speakers or speakers setSinkId already accepts? (#142)
[mediacapture-record] MediaRecorder BlobEvent timecode clarification (#222)
[mediacapture-screen-share] Avoid implicit task queuing with promises, to clarify setFocusBehavior (#304)
[mediacapture-transform] First timestamp generated by mediastream fails (#111)
[mediacapture-transform] Out-of-main-thread processing by default (#23)
[webrtc-charter] add missing /ul, more accurate Changes summary (#85)
[webrtc-charter] Pull Request: add missing /ul, more accurate Changes summary
[webrtc-charter] Representing Meetings' Transcripts and Minutes (#84)
[webrtc-encoded-transform] Does Chromium require anything in SDP or RTP Header to make this work? (#37)
[webrtc-encoded-transform] Generalize ScriptTransform constructor to allow main-thread processing (#89)
[webrtc-encoded-transform] IDL changes for setMetadata for 3.2.2 webrtc nv use case (#202)
[webrtc-encoded-transform] Remove restriction on streams being limited to only one PC (#201)
[webrtc-extensions] Clarify status of RTP Header Extension for Absolute Capture Time (#201)
[webrtc-extensions] Clarify when `icecandidatepairremove` is fired. (#204)
[webrtc-extensions] Normative reference to googlesource.com (#216)
[webrtc-pc] Convert RTCIceCandidatePair dictionary to an interface (#2961)
[webrtc-pc] Pull Request: Update localDescription etc. attributes rather than just their sdp.
[webrtc-rtptransport] Add batch interfaces for reading RtpSents and RtpAcks (#42)
[webrtc-rtptransport] Add use case 3 for custom NACK/RTX (#59)
[webrtc-rtptransport] Custom BWE without pacing is not well supported. (#56)
[webrtc-rtptransport] Make RtpTransport transferable (#33)
[webrtc-rtptransport] Make RtpTransportProcessor transferable (#33)
[webrtc-rtptransport] new commits pushed by pthatcher
[webrtc-rtptransport] No signal when transport path change. (#57)
[webrtc-rtptransport] Pull Request: Add use case 3 for custom NACK/RTX
[webrtc-rtptransport] Pull Request: Cross stream and transport wide RTP/RTCP
[webrtc-rtptransport] RtpSendStreamInit and RtpReceiveStreamInit are not defined in explainer-use-case-1.md (#35)
[webrtc-stats] new commits pushed by alvestrand
Closed: [mediacapture-transform] First timestamp generated by mediastream fails (#111)
Closed: [webrtc-charter] Representing Meetings' Transcripts and Minutes (#84)
Closed: [webrtc-extensions] Normative reference to googlesource.com (#216)
Last message date: Thursday, 18 July 2024 23:45:18 UTC