WebRTC 114 release notes

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Harald Alvestrand

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May 15, 2023, 2:39:13 AM5/15/23
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WebRTC M114 is going to be released as part of Chrome M114 , currently planned for release on  May 30th 2023.


We had no PSAs for M114 but please note that the legacy callbased-based getStats is on the way out and throws an exception 50% of the time in the Beta and Canary channels:


Screen sharing using OpenH264 got a minor upgrade which should help reducing blurriness for static content such as slides. See the original OpenH264 pull request for details.


The following issues were marked as fixed or verified and had at least one commit in M114 (build, test and trivial code changes are not included):


Issue

Summary

Component

webrtc:11547

High DataChannel.send() timing

DataChannel

webrtc:14859

Ensure ReceiveSideCongestionController use modern types

Cleanup,BWE

chromium:1420245

Remove TransformableVideoFrame::GetMetadata in favour of TransformableVideoFrame::Metadata()

Blink>WebRTC

webrtc:15097

AV1 default maximum bitrate limit changed

Video

webrtc:11547

High DataChannel.send() timing

DataChannel

chromium:1430806

Add UMA to estimate how often WebRTC audio mixer mixers more than 3 streams

Blink>WebRTC>Audio

webrtc:14830

Add support for more formats in v4l2 video capture

Video

webrtc:15059

The target bitrate was mistakenly set to be the maximal bitrate when initializing the libaom encoder

Video

webrtc:14334

Improve IPv6 network resolution and candidate creation

Network>ICE

webrtc:15087

PipeWire based video capture doesn't set unique ID for the current device

Video

webrtc:14029

Make sure audio level 127 in the audio level extension header, represents digital silence


webrtc:4711

Improved WebRTC self- and TCP-fairness.

Video

chromium:1385735

Fix validation of TURN and STUN URLs in RTCPeerConnection

Blink>WebRTC>
PeerConnection

webrtc:13427

WebRTC Android's hardware encoder should handle the stride and slice-height of the input buffer.

Video,Mobile

webrtc:15033

Exercise inactive streams paths

Video

webrtc:15015

Get frame average QP from MediaCodec encoder

Video

webrtc:15030

VideoAdapter has an integer overflow scaling very large frames

Video

webrtc:15098

DegradedCall deadlocks if network thread = worker thread

PeerConnection

webrtc:15106

For AV1, disable error resilience on upper temporal layers.

Video

chromium:1428098

VP9 {L3T3_KEY,inactive,inactive} not sending correctly in M113

Blink>WebRTC>Video

webrtc:4299

Remove a=ice-options:google-ice from generated offers

PeerConnection

webrtc:10739

Add support for the abs-capture-time header extension.

Network>RTP

webrtc:11547

High DataChannel.send() timing

DataChannel

webrtc:14801

Add the dependency descriptor to the RTC event log.

Video

chromium:1428006

Encoded Insertable Streams calls UnregisterEncoded{Audio,Video}StreamCallback on incorrect thread for transform errors in a worker

Blink>WebRTC>
PeerConnection

chromium:1242842

Camera capture incorrectly reports frame rate in MediaTrackSettings

Blink>GetUserMedia

chromium:1375217

tracking bug for webrtc-internals ui/ux improvements

Blink>WebRTC>Tools

chromium:1427193

Unstable fps when using MediaStream for video (Starting from Chrome v111)

Blink>MediaStream

chromium:1375217

tracking bug for webrtc-internals ui/ux improvements

Blink>WebRTC>Tools

chromium:1382005

Deprecate the RegionCaptureExperimentalSubtypes feature flag

Blink>GetDisplayMedia>
RegionCapture

chromium:864871

Implement standalone RTCIceTransport API

Blink>WebRTC

chromium:1423413

Improve MediaRecorder standards compliance.

Blink>MediaRecording

chromium:1248479

webrtc insertable streams: too large audio frames are silently dropped

Blink>WebRTC>
PeerConnection

chromium:1429272

When using insertable streams, getDisplayMedia video track `applyConstraints({framerate: { min: 1 }})` not working as expected

Blink>GetUserMedia,
Blink>GetDisplayMedia

chromium:1414363

getStats(): Define stats objects in WebIDL

Blink>WebRTC>
PeerConnection


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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