WebRTC M105 release notes

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Harald Alvestrand

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Aug 18, 2022, 6:04:28 AM8/18/22
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WebRTC M105 Release Notes


Branch: WebRTC M105 branch

Summary

WebRTC M105, currently available in Chrome's beta channel, contains 3 new features and over 47 bug fixes, enhancements and stability/performance improvements. We encourage all developers to run versions of Chrome on the Canary, Dev, and Beta channels frequently and quickly report any issues found. Please take a look at this page, for some pointers on how to file a good bug report. The help we have received has been invaluable!

Note that the WebRTC release notes only cover WebRTC specific changes. Build, test and trivial code changes are not included. Follow the Chromium and Chrome releases blog for further updates on important changes in Chrome releases.

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

Features and Bugfixes

Type

Issue

Description

Component

Bug

http://crbug.com/1053756

Install use counters for use of illegal character in ICE identifiers

Blink>WebRTC

Bug

http://crbug.com/1324120

Screenshare: make refresh frame delivery more reliable

Blink>WebRTC>Video,Internals>Media>ScreenCapture

Bug

http://crbug.com/1335194

libevent moved to third_party from base/third_party

Internals>Core

Bug

http://crbug.com/1336952

Screenshare: apparent 0hz capture freezes due to dropped frames.

Blink>WebRTC>Video,Internals>Media>ScreenCapture

Bug

http://crbug.com/1342840

Fix Timeout in congestion_controller_feedback_fuzzer

Tools>Stability>libFuzzer

Bug

http://bugs.webrtc.org/10809

Implement a TaskQueue -driven pacer version

Network>RTP

Bug

http://bugs.webrtc.org/11789

Stats Add jitterBufferTargetDelay to inbound-rtp

Stats

Bug

http://bugs.webrtc.org/12701

rtc::make_ref_counted is now available in api/

Internals

Bug

http://bugs.webrtc.org/13765

RTCInboundRtpStreamStats framesPerSecond is using the received framerate and not the decoded one

Stats

Bug

http://bugs.webrtc.org/13826

H264 422 decoding support

Video

Bug

http://bugs.webrtc.org/14130

pacing/packet_router: revert to previous transport_seq_ when packet discarded

Network>RTP

Bug

http://bugs.webrtc.org/14140

Add check and error message for reserved payload type ids

PeerConnection

Bug

http://bugs.webrtc.org/14141

Change googPreferredJitterBufferMs to be independent from configured minimum/maximum buffer size

undefined

Bug

http://bugs.webrtc.org/14147

Change RTCInboundRtpStreamStats.jitterBufferTargetDelay from RTCNonStandardStatsMember into RTCStatsMember

Stats

Bug

http://bugs.webrtc.org/14174

Implement RTCInboundRtpStreamStats' trackIdentifier

Stats

Bug

http://bugs.webrtc.org/14191

Implement outbound and inbound mid stats

Stats

Bug

http://bugs.webrtc.org/14195

Add 422 8 and 10 bit support for vp9 decoding.

Video

Bug

http://bugs.webrtc.org/14204

"Machine has no networks; no ports will be allocated" in tests on android

Network

Bug

http://bugs.webrtc.org/14211

Chrome report OperationError: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection'

PeerConnection

Bug

http://bugs.webrtc.org/14220

No bots can run linux video_capture_tests for now

undefined

Bug

http://bugs.webrtc.org/14225

Remove excessive log spam from modern getStats / receiver.getParameters()

Stats

Bug

http://bugs.webrtc.org/14247

DCHECK the number of block-invokes allowed in GetStatsReport to avoid accidental regressions

Stats

Bug

http://bugs.webrtc.org/14256

Investigate error logs from FlexfecReceiveStreamTest.RecoversPacket

Network>RTP

Bug

http://bugs.webrtc.org/14263

Fuzz audio processing module (APM) sample rates

Audio

Bug

http://bugs.webrtc.org/14265

AV in the webrtc::BlankDetectorDesktopCapturerWrapper::OnCaptureResult

DesktopCapture

Feature

http://bugs.webrtc.org/14273

Add link capacity to outgoing bitrate graph

undefined

Bug

http://bugs.webrtc.org/14279

Setting degradation_preference using RtpSender.SetParameters does not take effect.

PeerConnection

Bug

http://bugs.webrtc.org/7219

Move ProcessThread and loop based PlatformThread use over to TaskQueue

Internals

Bug

http://bugs.webrtc.org/13774

lld does not embed bitcode

Build

Bug

http://bugs.webrtc.org/13957

Add experiment arm to increase precision of slacked pacer when traffic is high

Network>RTP

Bug

http://crbug.com/1345653

webrtc-internals: Remove "iceconnectionstatechange (legacy)" event

Blink>WebRTC>PeerConnection

Bug

http://crbug.com/1345378

Check expiry of your histograms: Media.ConditionalFocus.*

Blink>GetDisplayMedia

Bug

http://crbug.com/1344773

Remove Plan B on Fuchsia.

Blink>WebRTC>PeerConnection

Bug

http://crbug.com/1344463

Step "bf_cache_content_browsertests on Ubuntu-18.04" failing on builder "chromium/ci/linux-bfcache-rel"

Blink>WebRTC>Tools

Bug

http://crbug.com/1344422

webrtc-internals: json dump drops stats type

Blink>WebRTC>Tools

Bug

http://crbug.com/1342947

Audio processing fails for certain sample rates during Chrome-wide echo cancellation on M103 onward

Blink>WebRTC>Audio

Bug

http://crbug.com/1341687

Define RTCPeerConnection's mediaConstraints with WebIDL

Blink>WebRTC>PeerConnection

Feature

http://crbug.com/1339784

webrtc-internals: allow download as gzip

Blink>WebRTC>Tools

Bug

http://crbug.com/1339762

Remove excessive log spam from modern getStats / receiver.getParameters()

Blink>WebRTC>PeerConnection

Feature

http://crbug.com/1337543

Created video frame buffer conversions from webrtc I422 and I210 buffers

Blink>WebRTC>Video

Bug

http://crbug.com/1336988

Clean up stats allow list to reflect the merging of the track stats

Blink>WebRTC>PeerConnection

Bug

http://crbug.com/1336952

Screenshare: apparent 0hz capture freezes due to dropped frames.

Blink>WebRTC>Video,Internals>Media>ScreenCapture

Bug

http://crbug.com/1336427

RegionCaptureBrowserTest.CropToAllowedIfTopLevelCropsToElementInEmbedded is flaky

Blink>GetDisplayMedia>RegionCapture

Bug

http://crbug.com/1334542

WebRtcTaskQueue is now Metronome only

Blink>WebRTC>PeerConnection

Bug

http://crbug.com/1327560

Region Capture Blue border is way off for crop target in iframe

Blink>GetDisplayMedia>RegionCapture,Internals>Media>SurfaceCapture

Bug

http://crbug.com/1324243

OpenH264 overshooting target bitrate

Blink>WebRTC

Bug

http://crbug.com/1303138

cropTo() Promises resolved prematurely

Blink>GetDisplayMedia

Bug

http://crbug.com/1298896

Region Capture Blue border issue when cropping to an invisible target

Blink>GetDisplayMedia>RegionCapture

Bug

http://crbug.com/1215472

Reworking HW codec enabling on Android

Blink>Media>WebCodecs,Blink>WebRTC>Video

Bug

http://crbug.com/1127211

WebRtcAudioRendererTest.SwitchOutputDevice is flaky

Blink>WebRTC


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