WebRTC 116 Release Notes

1,623 views
Skip to first unread message

Harald Alvestrand

unread,
Aug 1, 2023, 9:34:22 AM8/1/23
to discuss...@googlegroups.com

WebRTC M116 is going to be released as part of Chrome M116, currently planned for release on August 15th 2023.


We had two PSAs for M116:

Be careful with your ids, they are supposed to be unique identifiers.


The legacy callbased-based getStats is still on the way out (and going to be removed by default in M117) and throws an exception 50% of the time in the Beta and Canary channels:


The following issues were marked as fixed or verified and had at least one commit in M116 (build, test and trivial code changes are not included):


Issue

Summary

Component

webrtc:14205

Echo leaks when early reverberation occurs before the strongest echo

Audio

webrtc:15085

The video implementation of RTCInboundRtpStreamStats.jitterBufferDelay does not match the spec

Stats

webrtc:15241

Add missing definition for StatsReport::Value::id_val

Stats

chromium:1463451

Cloned or sender RTCEncodedVideoFrames fed into a decoder break playback delay buffering

Blink>WebRTC
>PeerConnection

chromium:1291247

[Remoting host] Tracking bug to support remote desktop on linux/wayland

Services>Chromoting

webrtc:15203

Use a constant for invalid PipeWire file descriptor

DesktopCapture

webrtc:15182

Allow WebRTC to send padding packets with 255 bytes

BWE

webrtc:14985

TemporalLayers/PictureIdTest.IncreasingAfterRecreateStream* tests are flaking

Video,Network

chromium:1455428

WebRTC + MediaFoundation = RtpPayloadParams::Vp9ToGeneric produces wrong metadata resulting in 30 keyframes/s

Blink>WebRTC
>Video

webrtc:15258

DataChannel hanging in "connecting" state if createOffer used before createAnswer

DataChannel

webrtc:14227

Need a process for release notes

Documentation

webrtc:14244

Implement RTCInboundRtpStreamStats.jitterBufferTargetDelay for video streams

Stats

chromium:1453226

Transformable Encoded Audio frames should always inherit from TransformableAudioFrameInterface regardless of direction

Blink>WebRTC
>PeerConnection

webrtc:15162

Cleanup NonStandardGroupId, StatExposureCriteria and RTCNonStandardStatsMember

Stats

webrtc:15185

Missing RTP packet info after parsing RED packets.

Audio

webrtc:15245

Consider deleting RTCStats enum values

Stats

webrtc:15198

Delete "track_id" stats member temporarily added to assist a roll

Stats

webrtc:14884

Support VP9 simulcast

Video

webrtc:15202

PipeWire camera - split portal and PipeWire implementations

Video

webrtc:14175

Deprecate and remove RTCMediaStreamTrackStats

Stats

webrtc:15240

Allow setting a custom randomness source

Internals

webrtc:12194

the range of dynamic rtp payload types is exhausted

PeerConnection

webrtc:15257

use RGB565 camera on windows maybe crash


chromium:1455847

serialize RTCDataChannelInit

Blink>WebRTC>Tools

chromium:1447318

Add "Copy video frame" to video element's context menu

Internals>Media>UI

webrtc:14175

Deprecate and remove RTCMediaStreamTrackStats

Stats

chromium:1451036

MediaRecorder throws with mimeType video/webm and audio-only stream

Blink
>MediaRecording

chromium:1443983

Cleanup WebRtcStatsReportIdl experiment

Blink>WebRTC
>PeerConnection

chromium:1450496

Confusing DOMException message mentioning "setFocusBehavior()"

Blink
>GetDisplayMedia

chromium:1429996

MediaStreamAudioTrackUnderlyingSource::OnData allocates memory on RT thread


chromium:1448816

WebRTC low bitrate scenario: HW encoder worse than SW in QVGA

Blink>WebRTC
>Video

chromium:1453226

Transformable Encoded Audio frames should always inherit from TransformableAudioFrameInterface regardless of direction

Blink>WebRTC
>PeerConnection

chromium:1451358

Encoded transforms for video have an unexpected dependency on SSRC

Blink>WebRTC
>PeerConnection

chromium:1448046

VideoTrackAdapter drops too many frames

Blink>WebRTC
>Video

chromium:1375217

tracking bug for webrtc-internals ui/ux improvements

Blink>WebRTC>Tools

chromium:1450844

structuredClone() on RTCEncodedVideoFrame and RTCEncodedAudioFrame only performs shallow copy

Blink>WebRTC
>PeerConnection

chromium:1454398

[WebRTC] Add flag to control if forcing SW should include or exclude 360p

Blink>WebRTC
>Video



For the full list of commits please refer to the git log between this branch and the previous branch. See here for a description of what the release notes contain.


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.




Reply all
Reply to author
Forward
0 new messages