WebRTC 110 Release notes

1,564 views
Skip to first unread message

Philipp Hancke

unread,
Jan 30, 2023, 3:14:23 AM1/30/23
to discuss...@googlegroups.com

WebRTC M110 is going to be released as part of Chrome M110 on February 1st 2023.


Note that the deprecated track and stream statistics are going to be removed soon (but not in M110).

See the intent to deprecate thread on blink-dev which should be shared on this list soon as well.


We also had a couple of PSAs which may require action from developers:

This list of PSAs may still be incomplete, this will improve in future releases.


On Linux, the pipewire-based screen capturer is now enabled by default which means less completely black screen shares.


The following issues were marked as fixed or verified and had at least one commit in M110 (build, test and trivial code changes are not included):


Issue

Summary

Component

chromium:1381982

Deadlock in WebRTC metronome

Blink>WebRTC

chromium:1369050

Only expose decoderImplementation if "HW exposure is allowed"

Blink>WebRTC>
PeerConnection

webrtc:10635

Implement stats for packet send-side delay for audio streams

Stats,Audio

webrtc:14522

Unship non-standard video "track" metrics

Stats

webrtc:13960

scalabilityMode support with VP9

PeerConnection,Video

chromium:1374310

Tab crashes after ICE restart when using TCP TURN

Blink>WebRTC

webrtc:14628

"outbound-rtp.active: false" observed while actively sending VP9 SVC

Stats

webrtc:14593

totalPacketSendDelay is incorrectly implemented

Stats

webrtc:14521

Move non-standard pause and freeze from track to inbound-rtp and make them standard

Stats

webrtc:5773

Bringing AppRTCDemo to the background and back to foreground can cause BWE to get stuck at low levels.

Video

webrtc:14608

Delete disable_ipv6 flag

PeerConnection

chromium:1374436

Improve Capturer Selection Logic on Wayland

Internals>Media>
ScreenCapture

webrtc:14632

StreamInterface::Read/Write should use rtc::ArrayView

Internals

webrtc:14547

Peer connection's `getStats` by RtpSender/Receiver is not available in Android SDK

Mobile

webrtc:14627

Simplify PeerConnectionE2EQualityTestFixture::AddPeer signature

PeerConnection

webrtc:14587

Add googTimingFrameInfo to modern stats API

Stats

webrtc:14596

RTCTransportStats.dtlsCipher not available when building with OpenSSL

Stats

chromium:1379802

TrackControls::requested is redundant with TrackControls::stream_type

Blink>
GetDisplayMedia

chromium:1375235

RTCVideoEncoder error handling inconsistency

Blink>WebRTC>Video

chromium:1215480

Conditional Focus - Tracking Bug

Blink>
GetDisplayMedia

chromium:1379243

getDisplayMedia({video: {frameRate: {max: 0}}}) should fail with OverconstrainedError.

Blink>
GetDisplayMedia

chromium:1103280

Audio issues with insertable streams enabled under load

Blink>WebRTC

chromium:1378667

Implement suppressLocalAudioPlayback

Blink>
GetDisplayMedia

chromium:1381982

Deadlock in WebRTC metronome

Blink>WebRTC

chromium:1381265

Correct default value for selfBrowserSurface

Blink>
GetDisplayMedia



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



Reply all
Reply to author
Forward
0 new messages