PSA: WebRTC M94 Release Notes

[Huib Kleinhout]

WebRTC M94 Release Notes

Branch: WebRTC M94 branch

Summary

WebRTC M94, currently available in Chrome's beta channel, contains 1 new features and over 19 bug fixes, enhancements and stability/performance improvements. We encourage all developers to run versions of Chrome on the Canary, Dev, and Beta channels frequently and quickly report any issues found. Please take a look at this page, for some pointers on how to file a good bug report. The help we have received has been invaluable!

Note that the WebRTC release notes only cover WebRTC specific changes. Follow the Chromium and Chrome releases blog for further updates on important changes in Chrome releases.

We strongly recommend WebRTC developers to fully test their services in Chrome beta to ensure stability for end-users.

The Chrome release schedule can be found here

PSAs & Features

Title	Description
MediaTrack Insertable Streams	The Insertable streams API for MediaStreamTrack is now available as a stable web api and no longer needs an Origin Trial! The API can be used to directly access and modify audio/video streams. Read more about it here: https://web.dev/mediastreamtrack-insertable-media-processing/
nonstandard RTCConfiguration.offerExtmapAllowMixed option has been removed from Chrome	the non-standard offerExtmapAllowMixed in the RTCPeerConnection constructor was removed in M94. See also the intent to remove https://groups.google.com/a/chromium.org/g/blink-dev/c/Plik-x6biZ0/m/eJ8P1iy0AQAJ for details and how to do SDP manipulation before calling setRemoteDescription where required.

Features and Bugfixes

Type	Issue	Description	Component
Bug	http://crbug.com/1084702	Mobile Chrome on Pixel 3 has video corruption for non-16-aligned resolutions in WebRTC calls : Hardware VP8 encoder bug	Blink>WebRTC>Video,Internals>GPU>Video,Internals>Media>Capture
Feature	http://crbug.com/1225701	Enable DCHECKS by default for developer builds	Build
Bug	http://crbug.com/1232358	Broken Dependency Descriptor (DD) RTP extension when enabling AV1-SVC in WebRTC	Blink>WebRTC
Bug	http://crbug.com/1234779	Peer Connection with a=inactive on the remoteDescription still sends RTP.	Blink>WebRTC>PeerConnection
Bug	http://crbug.com/1236202	Merge request: PeerConnection assumes all new m= sections will go in the first BUNDLE group	Blink>WebRTC>PeerConnection
Bug	http://crbug.com/999886	code base vs clang r370594: -Wfinal-dtor-non-final-class	Build,Tools>LLVM
Bug	http://bugs.webrtc.org/11640	Support for RFC 2198 redundancy	Audio
Bug	http://bugs.webrtc.org/12470	Recurrent and empty RTP packets with gaps in sequence number	Network>RTP
Bug	http://bugs.webrtc.org/12837	PC emits candidates when renegotiating	PeerConnection
Bug	http://bugs.webrtc.org/12906	PeerConnection assumes all new m= sections will go in the first BUNDLE group	PeerConnection
Bug	http://bugs.webrtc.org/12975	video_replay's aspect ratio is hard-coded.	Tools
Bug	http://bugs.webrtc.org/12980	Jitter stats for video are much higher than expected	Stats
Bug	http://bugs.webrtc.org/12988	Prepare WebRTC for Chromium's switch of dcheck_always_on default value	Build
Bug	http://bugs.webrtc.org/12989	NackModule2 during 50 tiles can potentially cause 2.5 kHz idle wakeups	Internals
Bug	http://bugs.webrtc.org/12991	AVC parsing error when hardware acceleration is turned on in Chrome Windows	Video
Bug	http://bugs.webrtc.org/12995	FrameEncryptorInterface is unusable with video	PeerConnection
Bug	http://bugs.webrtc.org/13037	Encoded framerate stats per layer is incorrect.	Stats,Video
Bug	http://bugs.webrtc.org/13053	iSAC tests on Mac ARM64 started failing after chromium roll	Audio,Build
Bug	http://bugs.webrtc.org/4299	Remove a=ice-options:google-ice from generated offers	PeerConnection
Bug	http://crbug.com/1231698	display-capture (permission policy)	Blink>GetDisplayMedia