WebRTC 111 Release Notes

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Harald Alvestrand

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Feb 17, 2023, 8:12:45 AM2/17/23
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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:


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.



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