Issue | Summary | Component |
webrtc:11547 | High DataChannel.send() timing | DataChannel |
webrtc:14859 | Ensure ReceiveSideCongestionController use modern types | Cleanup,BWE |
chromium:1420245 | Remove TransformableVideoFrame::GetMetadata in favour of TransformableVideoFrame::Metadata() | Blink>WebRTC |
webrtc:15097 | AV1 default maximum bitrate limit changed | Video |
webrtc:11547 | High DataChannel.send() timing | DataChannel |
chromium:1430806 | Add UMA to estimate how often WebRTC audio mixer mixers more than 3 streams | Blink>WebRTC>Audio |
webrtc:14830 | Add support for more formats in v4l2 video capture | Video |
webrtc:15059 | The target bitrate was mistakenly set to be the maximal bitrate when initializing the libaom encoder | Video |
webrtc:14334 | Improve IPv6 network resolution and candidate creation | Network>ICE |
webrtc:15087 | PipeWire based video capture doesn't set unique ID for the current device | Video |
webrtc:14029 | Make sure audio level 127 in the audio level extension header, represents digital silence |
|
webrtc:4711 | Improved WebRTC self- and TCP-fairness. | Video |
chromium:1385735 | Fix validation of TURN and STUN URLs in RTCPeerConnection | Blink>WebRTC> PeerConnection |
webrtc:13427 | WebRTC Android's hardware encoder should handle the stride and slice-height of the input buffer. | Video,Mobile |
webrtc:15033 | Exercise inactive streams paths | Video |
webrtc:15015 | Get frame average QP from MediaCodec encoder | Video |
webrtc:15030 | VideoAdapter has an integer overflow scaling very large frames | Video |
webrtc:15098 | DegradedCall deadlocks if network thread = worker thread | PeerConnection |
webrtc:15106 | For AV1, disable error resilience on upper temporal layers. | Video |
chromium:1428098 | VP9 {L3T3_KEY,inactive,inactive} not sending correctly in M113 | Blink>WebRTC>Video |
webrtc:4299 | Remove a=ice-options:google-ice from generated offers | PeerConnection |
webrtc:10739 | Add support for the abs-capture-time header extension. | Network>RTP |
webrtc:11547 | High DataChannel.send() timing | DataChannel |
webrtc:14801 | Add the dependency descriptor to the RTC event log. | Video |
chromium:1428006 | Encoded Insertable Streams calls UnregisterEncoded{Audio,Video}StreamCallback on incorrect thread for transform errors in a worker | Blink>WebRTC> PeerConnection |
chromium:1242842 | Camera capture incorrectly reports frame rate in MediaTrackSettings | Blink>GetUserMedia |
chromium:1375217 | tracking bug for webrtc-internals ui/ux improvements | Blink>WebRTC>Tools |
chromium:1427193 | Unstable fps when using MediaStream for video (Starting from Chrome v111) | Blink>MediaStream |
chromium:1375217 | tracking bug for webrtc-internals ui/ux improvements | Blink>WebRTC>Tools |
chromium:1382005 | Deprecate the RegionCaptureExperimentalSubtypes feature flag | Blink>GetDisplayMedia> RegionCapture |
chromium:864871 | Implement standalone RTCIceTransport API | Blink>WebRTC |
chromium:1423413 | Improve MediaRecorder standards compliance. | Blink>MediaRecording |
chromium:1248479 | webrtc insertable streams: too large audio frames are silently dropped | Blink>WebRTC> PeerConnection |
chromium:1429272 | When using insertable streams, getDisplayMedia video track `applyConstraints({framerate: { min: 1 }})` not working as expected | Blink>GetUserMedia, Blink>GetDisplayMedia |
chromium:1414363 | getStats(): Define stats objects in WebIDL | Blink>WebRTC> PeerConnection |