Type | Issue | Description | Component |
Bug | http://crbug.com/1053756 | Install use counters for use of illegal character in ICE identifiers | Blink>WebRTC |
Bug | http://crbug.com/1324120 | Screenshare: make refresh frame delivery more reliable | Blink>WebRTC>Video,Internals>Media>ScreenCapture |
Bug | http://crbug.com/1335194 | libevent moved to third_party from base/third_party | Internals>Core |
Bug | http://crbug.com/1336952 | Screenshare: apparent 0hz capture freezes due to dropped frames. | Blink>WebRTC>Video,Internals>Media>ScreenCapture |
Bug | http://crbug.com/1342840 | Fix Timeout in congestion_controller_feedback_fuzzer | Tools>Stability>libFuzzer |
Bug | http://bugs.webrtc.org/10809 | Implement a TaskQueue -driven pacer version | Network>RTP |
Bug | http://bugs.webrtc.org/11789 | Stats Add jitterBufferTargetDelay to inbound-rtp | Stats |
Bug | http://bugs.webrtc.org/12701 | rtc::make_ref_counted is now available in api/ | Internals |
Bug | http://bugs.webrtc.org/13765 | RTCInboundRtpStreamStats framesPerSecond is using the received framerate and not the decoded one | Stats |
Bug | http://bugs.webrtc.org/13826 | H264 422 decoding support | Video |
Bug | http://bugs.webrtc.org/14130 | pacing/packet_router: revert to previous transport_seq_ when packet discarded | Network>RTP |
Bug | http://bugs.webrtc.org/14140 | Add check and error message for reserved payload type ids | PeerConnection |
Bug | http://bugs.webrtc.org/14141 | Change googPreferredJitterBufferMs to be independent from configured minimum/maximum buffer size | undefined |
Bug | http://bugs.webrtc.org/14147 | Change RTCInboundRtpStreamStats.jitterBufferTargetDelay from RTCNonStandardStatsMember into RTCStatsMember | Stats |
Bug | http://bugs.webrtc.org/14174 | Implement RTCInboundRtpStreamStats' trackIdentifier | Stats |
Bug | http://bugs.webrtc.org/14191 | Implement outbound and inbound mid stats | Stats |
Bug | http://bugs.webrtc.org/14195 | Add 422 8 and 10 bit support for vp9 decoding. | Video |
Bug | http://bugs.webrtc.org/14204 | "Machine has no networks; no ports will be allocated" in tests on android | Network |
Bug | http://bugs.webrtc.org/14211 | Chrome report OperationError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection' | PeerConnection |
Bug | http://bugs.webrtc.org/14220 | No bots can run linux video_capture_tests for now | undefined |
Bug | http://bugs.webrtc.org/14225 | Remove excessive log spam from modern getStats / receiver.getParameters() | Stats |
Bug | http://bugs.webrtc.org/14247 | DCHECK the number of block-invokes allowed in GetStatsReport to avoid accidental regressions | Stats |
Bug | http://bugs.webrtc.org/14256 | Investigate error logs from FlexfecReceiveStreamTest.RecoversPacket | Network>RTP |
Bug | http://bugs.webrtc.org/14263 | Fuzz audio processing module (APM) sample rates | Audio |
Bug | http://bugs.webrtc.org/14265 | AV in the webrtc::BlankDetectorDesktopCapturerWrapper::OnCaptureResult | DesktopCapture |
Feature | http://bugs.webrtc.org/14273 | Add link capacity to outgoing bitrate graph | undefined |
Bug | http://bugs.webrtc.org/14279 | Setting degradation_preference using RtpSender.SetParameters does not take effect. | PeerConnection |
Bug | http://bugs.webrtc.org/7219 | Move ProcessThread and loop based PlatformThread use over to TaskQueue | Internals |
Bug | http://bugs.webrtc.org/13774 | lld does not embed bitcode | Build |
Bug | http://bugs.webrtc.org/13957 | Add experiment arm to increase precision of slacked pacer when traffic is high | Network>RTP |
Bug | http://crbug.com/1345653 | webrtc-internals: Remove "iceconnectionstatechange (legacy)" event | Blink>WebRTC>PeerConnection |
Bug | http://crbug.com/1345378 | Check expiry of your histograms: Media.ConditionalFocus.* | Blink>GetDisplayMedia |
Bug | http://crbug.com/1344773 | Remove Plan B on Fuchsia. | Blink>WebRTC>PeerConnection |
Bug | http://crbug.com/1344463 | Step "bf_cache_content_browsertests on Ubuntu-18.04" failing on builder "chromium/ci/linux-bfcache-rel" | Blink>WebRTC>Tools |
Bug | http://crbug.com/1344422 | webrtc-internals: json dump drops stats type | Blink>WebRTC>Tools |
Bug | http://crbug.com/1342947 | Audio processing fails for certain sample rates during Chrome-wide echo cancellation on M103 onward | Blink>WebRTC>Audio |
Bug | http://crbug.com/1341687 | Define RTCPeerConnection's mediaConstraints with WebIDL | Blink>WebRTC>PeerConnection |
Feature | http://crbug.com/1339784 | webrtc-internals: allow download as gzip | Blink>WebRTC>Tools |
Bug | http://crbug.com/1339762 | Remove excessive log spam from modern getStats / receiver.getParameters() | Blink>WebRTC>PeerConnection |
Feature | http://crbug.com/1337543 | Created video frame buffer conversions from webrtc I422 and I210 buffers | Blink>WebRTC>Video |
Bug | http://crbug.com/1336988 | Clean up stats allow list to reflect the merging of the track stats | Blink>WebRTC>PeerConnection |
Bug | http://crbug.com/1336952 | Screenshare: apparent 0hz capture freezes due to dropped frames. | Blink>WebRTC>Video,Internals>Media>ScreenCapture |
Bug | http://crbug.com/1336427 | RegionCaptureBrowserTest.CropToAllowedIfTopLevelCropsToElementInEmbedded is flaky | Blink>GetDisplayMedia>RegionCapture |
Bug | http://crbug.com/1334542 | WebRtcTaskQueue is now Metronome only | Blink>WebRTC>PeerConnection |
Bug | http://crbug.com/1327560 | Region Capture Blue border is way off for crop target in iframe | Blink>GetDisplayMedia>RegionCapture,Internals>Media>SurfaceCapture |
Bug | http://crbug.com/1324243 | OpenH264 overshooting target bitrate | Blink>WebRTC |
Bug | http://crbug.com/1303138 | cropTo() Promises resolved prematurely | Blink>GetDisplayMedia |
Bug | http://crbug.com/1298896 | Region Capture Blue border issue when cropping to an invisible target | Blink>GetDisplayMedia>RegionCapture |
Bug | http://crbug.com/1215472 | Reworking HW codec enabling on Android | Blink>Media>WebCodecs,Blink>WebRTC>Video |
Bug | http://crbug.com/1127211 | WebRtcAudioRendererTest.SwitchOutputDevice is flaky | Blink>WebRTC |
