Type | Issue | Description | Component |
Feature | 10650 | Add rtpTimestamp to contributing sources | Network>RTP |
Feature | 10579 | Add FrameMarking RTP Header Extension support in H.264 receiver | Video |
Feature | 10542 | Add the possibility to limit the delay based bandwidth estimator to increase | Network |
Feature | 10495 | Create superframe index header and append it to frame buffer | Video |
Feature | 10456 | [standard stats] Implement stats for roundTripTime of RTP streams of kind video | Stats |
Feature | 10455 | [standard stats] Implement stats for roundTripTime of RTP streams of kind audio | Stats |
Feature | 10453 | [standard stats] Implement stats for resolution and framerate pre-encoding | Stats |
Feature | 10451 | [standard stats] Implement stats for quality limitation: qualityLimitationReason | Stats |
Feature | 10450 | [standard stats] Implement jitterBufferDelay and jitterBufferEmittedCount for video | Stats |
Feature | 10447 | [standard stats] Implement counters for retransmitted bytes | Stats |
Feature | 10446 | [standard stats] Implement stats for target encode bitrate | Stats |
Feature | 10444 | [standard stats] Implement stats for error correction of RTP streams | Stats |
Feature | 10443 | [standard stats] Implement stats for audible/silent concealed samples | Stats |
Feature | 10442 | [standard stats] Implement stats for accelerating/decelerating playout speed | Stats |
Feature | 9934 | Makes send packet information non optional for feedback reports. | BWE |
Feature | 9801 | Split voe::Channel into send and receive classes for audio rtp transport. | Audio, Network>RTP |
Feature | 9777 | Implement RTCRtpTransceiver::setCodecPreferences() | PeerConnection |
Feature | 9545 | Implement most of RTCRemoteInboundRtpStreamStats | Stats |
Feature | 4612 | SCTP SDP m-lines: Convert to sending new draft SDP spec | PeerConnection |
Feature | 10506 | [standard stats] Implement stats for packet send-side delay | Stats |
Feature | 965994 | Add RTP timestamp to RTCRtpReceiver::RTCRtpContributingSource | Blink>WebRTC>Network |
Feature | 930186 | [Video Capture, Feature] Dynamic Screen Capture | Blink>WebRTC>Video |
Feature | 891556 | Implement RTCRtpTransceiver.setCodecPreferences() | Blink>WebRTC>PeerConnection |
Feature | 878465 | mDNS service for IP handling in WebRTC | Blink>WebRTC>Network |
Feature | 844386 | Implement RTCRtpSender.setStreams() | Blink>WebRTC>PeerConnection |
Feature | 818643 | Implement RTCSctpTransport | Blink>WebRTC>PeerConnection |
Bug | 10693 | packetization mode should be checked when selecting H264 as send video codec | PeerConnection |
Bug | 936715 | VP8 Decoder: Quality expectation and improvements for Accelerated Decoders in chromium | Blink>WebRTC>Video |
Bug | 10607 | Make sure packets in the pacer queue are preserved | Network>RTP |
Bug | 10604 | Potential overflow in sequence number map tracking loss vectors | Network>RTP |
Bug | 955416 | Merge to M75: Write VP9 RTP SS on key frames of each independently coded spatial layer. | Blink>WebRTC>Video |
Bug | 10571 | Potentially unnecessary scaling in LibvpxVp8Encoder::Encode() |
|
Bug | 10565 | In sumulcast mode VP9 sender doesn't write scalability structure on key frames of high spatial layers | Video |
Bug | 10564 | Duplicate calls to OnSentPacket() breaks ALR detection | BWE |
Bug | 10551 | Incoming offer for simulcast does not generate video | PeerConnection |
Bug | 10546 | The runtime-settings in aecdumps for the pre-amplifier gain cannot be overruled in audioproc_f | Audio |
Bug | 10543 | Color space not parsed correctly on receiver side | Network>RTP |
Bug | 10462 | Acknowledged bitrate estimate can get stuck at low bandwidth. |
|
Bug | 10460 | Populate meta-information fields on VideoFrame from EncodedImage in one place | Video |
Bug | 943976 | Surface max number of channels in SctpTransport interface | Blink>WebRTC>PeerConnection |
Bug | 943975 | Surface max message size in RTCSctpTransport | Blink>WebRTC>PeerConnection |
Bug | 943972 | Surface remote certificates in RtcDtlsTransport | Blink>WebRTC>PeerConnection |
Bug | 10373 | CriticalSection doesn't play well with audio callback threads on MacOS | Audio, Internals |
Bug | 10368 | RTT based backoff is not capped below. |
|
Bug | 10358 | SCTP: Compute max message size and max channels correctly | DataChannel |
Bug | 10270 | Clock implementation hides mutable behavior hidden under const. |
|
Bug | 10222 | Bandwidth toggles between two estimates in StartUpPhase. | BWE |
Bug | 10533 | The minimum comfort noise level in AEC3 is too high | Audio |
Bug | 8434 | Excessive AEC suppression | Audio |
Bug | 6855 | PeerConnectionInterface doesn't expose any useful error information | PeerConnection |
Bug | 9410 | Need a way to add unstandardized stats for native applications | Stats |
Bug | 5948 | rfc6184, rfc6185 sprop-parameter-sets | Video |
Bug | 9688 | Simulcast streams will send one key frame on all spatial layers for each FIR with different SSRC | Video |
Bug | 4484 | SDP parsing: addIceCandidate with candidate priority 0 is not rejected | Network |
Bug | 10643 | Make unpack_aecdump unpack RuntimeSettings | Audio |
Bug | 10609 | Let RuntimeSetting store either int or float | Audio |
Bug | 10608 | Add RuntimeSetting for volume change | Audio |
Bug | 964332 | MediaPicker: don't use TabbedPane when there's only one tab | Blink>GetUserMedia>Desktop |
Bug | 944731 | Webcam.js Error: Webcam is not loaded yet | Blink>GetUserMedia>Webcam |
Bug | 868026 | Unable to access microphone on Huawei Matebook X Pro | Blink>GetUserMedia>Mic |
Bug | 627793 | MediaDeviceInfo object of kind videoinput is missing groupId | Blink>GetUserMedia |
Bug | 967231 | Merge to M75: VP9 low-fps screen share fixes | Blink>WebRTC>Video |
Bug | 965483 | addIceCandidate(new RTCIceCandidate({candidate, sdpMid})) no longer works | Blink>WebRTC>PeerConnection |
Bug | 963818 | Distorted sound when using Web Audio API to mux audio sources in WebRTC on Mac | Blink>WebRTC |
Bug | 962860 | Chrome uses obsolete format for SCTP data channels | Blink>WebRTC>PeerConnection |
Bug | 961269 | Merge Request for [ Ensure that we always set values for min and max audio bitrate.] | Blink>WebRTC>Audio |
Bug | 960736 | Unreasonable IO buffer size on Mac audio output when unplugging device | Blink>Media>Audio, Blink>WebRTC>Audio, Internals>Media>Audio |
Bug | 960161 | Tab mirroring audio quality is significantly worse with audio service enabled on M75, 76 | Blink>WebRTC>Audio, Internals>Cast>Streaming, Internals>Media>Audio, Internals>Media>Capture |
Bug | 959128 | iceConnectionState not going past "checking" in M75 | Blink>WebRTC |
Bug | 956634 | Merge to M75: Expand UsagePattern and private IP address definition | Blink>WebRTC>Network |
Bug | 956525 | Merge to M75: Parse color space only in last packet of key frame | Blink>WebRTC>Video |
Bug | 956472 | Remove generic error from WebRTC event log collection | Blink>WebRTC>Tools |
Bug | 953512 | Invoking getStats with an invalid second argument (such as errorCallback) is no longer equivalent to getStats(successCallback) | Blink>WebRTC |
Bug | 944451 | WebRtcRemoteEventLogManager does not always upload over WiFi | Blink>WebRTC>Tools |
Bug | 740501 | RTCPeerConnection.onnegotiationneeded can sometimes fire multiple times in a row | Blink>WebRTC>PeerConnection |
Bug | 962731 | Microphone doesn't work | Blink>GetUserMedia>Mic, OS>Kernel>Camera, Platform>Apps>Hangouts |
Bug | 820961 | Video feed from Brio 4K camera is flickering on Jaq device | Blink>GetUserMedia>Webcam, OS>Kernel>Camera |
