WebRTC 111 Release Notes

[Harald Alvestrand]

WebRTC M111 is going to be released as part of Chrome M111 , currently planned for release on  March 1st 2023.

Two PSAs were published that may require action from developers:

• PSA: the RTCStats timestamp_us method is going away

• PSA: TaskQueueBase PostTask semantics change

In terms of features the highlights for this release are support for the WebRTC SVC API which you can try on Florent’s simulcast playground as well as support for more 444 pixel formats. Make sure to give it a try with --enable-features=RTCSvcScalabilityMode.

Note that the deprecated track and stream statistics are going to be removed soon (but not in M111 either). See the intent to deprecate thread on blink-dev.

The following issues were marked as fixed or verified and had at least one commit in M111 (build, test and trivial code changes are not included):

Issue	Summary	Component
webrtc:10342	Implement updated version of the codec agnostic descriptor (Generic Frame Descriptor).	Network>RTP
webrtc:13826	H264 422 decoding is not supported	Video
chromium:1227566	CPU usage of playing audio increases significantly with minimum mic input	Blink>WebRTC>Audio, Blink>GetUserMedia, Internals>Media> AudioCapture
webrtc:13896	Failure to re-send packet via RTX due to additional space required for MID/RRID header extension	Network>RTP
webrtc:14844	The stats cache is not cleared by CreateDataChannel()	Stats
chromium:1400642	WebRTC transceiver buffers and plays audio which it was receiving before becoming "inactive"	Blink>WebRTC>Audio
webrtc:14809	VP9 encoders sometimes not properly recreated in spatial layer activation	Video
webrtc:12397	Audio RTP timestamps does not increment when RTCRtpEncodingParameters.active is set to false.	Audio
webrtc:11108	Add totalInterFrameDelay to RTCInboundRTPStreamStats	Stats
webrtc:10405	Improve handling RTP (video) packets arriving before VideoReceiveStream has been setup	Video
webrtc:14769	addTrack does not filter duplicate stream ids	PeerConnection
webrtc:11108	Add totalInterFrameDelay to RTCInboundRTPStreamStats	Stats
webrtc:14688	Crash during stream startup Mac native with h264	Video
webrtc:14804	Expect to stop audio encoding if local audio track disabled, for power efficient.	Audio
webrtc:11108	Add totalInterFrameDelay to RTCInboundRTPStreamStats	Stats
webrtc:12790	Stop CNG after a timeout	Audio
webrtc:14807	change stats constructor to take a std::string	Stats
webrtc:12420	implement the rtx-time parameter for RTX	Video
webrtc:14027	Ensure ProbeClusterConfig.target_duration from GoogCC is used when creating probes
webrtc:14808	Codec stats have excessive amounts of data copying	Stats,Internals
webrtc:14811	receiver.GetParameters() and GetSources() don't work if the SSRC is unsignalled	PeerConnection
webrtc:13982	Remove duplicate implementations of sequence number unwrapping
webrtc:14343	Memory leak when creating Windows video capturer	Video
webrtc:14818	Missing 444 10 bits support in vp9 and h264 decoders	Video
chromium:1379243	getDisplayMedia({video: {frameRate: {max: 0}}}) should fail with OverconstrainedError.	Blink>GetDisplayMedia
chromium:1344751	Remove mojo DeviceFactory	Internals>Media> CameraCapture
chromium:1377296	CrossThreadHandle: Convert render/modules/peerconnection/*	Blink>WebRTC, Blink>GarbageCollection
chromium:1406874	Echo cancellation of WebAudio playing to a non-default output device	Blink>WebAudio, Blink>GetUserMedia
chromium:1406227	Refactor MediaStreamVideoTrack in how it access max_frame_rate	Blink>MediaStream
chromium:1375217	tracking bug for webrtc-internals ui/ux improvements	Blink>WebRTC>Tools
chromium:986069	Implement WebRTC SVC API	Blink>WebRTC> PeerConnection
chromium:1409064	SetCanDiscardAlpha calls ignore multiple tracks on source.	Blink>MediaStream, Blink>CaptureFromElement
chromium:1410129	media::AudioProcessor should not push the reference into WebRTC if there is no echo cancellation	Blink>WebRTC>Audio
chromium:1302689	Align lifecycles of MediaStreamSource and MediaStreamComponent with their Platform members	Blink>MediaStream
chromium:1353279	Move MediaConstraints from platform/ to modules/	Blink>MediaStream

For the full list of commits please refer to the git log between this branch and the previous branch.

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.

These release notes were prepared by Philipp Hancke.