PSA: WebRTC M88 Release Notes

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Huib Kleinhout

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Dec 16, 2020, 1:43:20 AM12/16/20
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WebRTC M88 Release Notes


WebRTC 4324 branch (cut at daab6896e2938e28f01e305ce2fff038f47554c4)

Summary


WebRTC M88 currently available in Chrome's beta channel, contains over 29 bug fixes, enhancements and stability/performance improvements. As with previous releases, 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! 


The Chrome release schedule can be found here.


Features

Optimized video capture using NV12 VideoFrame

Both VP8 and VP9 libvpx wrappers now support encoding NV12 video frames. Video sources that emit NV12 video frames can thus be encoded directly without requiring conversion to I420. Chrome is making use of this by avoiding unnecessary conversions when NV12 frames can be encoded.


Deprecations

RTP Datachannels are going away

As of M88, use of RTP datachannels will cause a deprecation warning. We expect to remove them altogether in M90. Use SCTP-based datachannels instead. More info in the PSA.




Platform

Issue

Description

Component

webrtc

12054

Deprecate and remove GetRemoteAudioSSLCertificate



Features and Bugfixes


Type

Issue

Description

Component

Bug

1111273

MSE freezes and comes to life after 30 minutes

Blink>Media,Internals>Media>Hardware,Internals>Media>Video

Bug

1132299

[macOS Capture] Implement efficient scaling/conversion with VTPixelTransferSession and libyuv-into-IOSurface

Internals>Media>Capture

Bug

1134686

Possible regression in webrtc createOffer, sometimes fails with warning about unknown transceiver when didn't before

Blink>WebRTC

Bug

1137936

Possible deadlock when handling SCTP restart

Blink>WebRTC>PeerConnection

Bug

1139052

"Failed to demux RTP packet" errors with two video tracks on M86

Blink>WebRTC>PeerConnection

Bug

1142626

webrtc stats audioLevel incorrect on inbound-rtp

Blink>WebRTC>PeerConnection

Bug

10155

Excessive video delay with pacing factor 1.0

Video

Bug

10232

The AEC3 transparency is poor initially in the call when headsets are used

Audio

Bug

10676

Implement early loss detection using transport feedback

Network>RTP

Bug

11622

NetEq statistics are reset when new getStats() is called

Audio

Feature

11635

Reduce frame copies in libvpx

Video

Bug

11643

opus speech detection ignores SILK vad results

Audio

Feature

11769

Add new parameter for H.264

Video

Bug

11974

Add NV12 Support for VP9

Video

Bug

11975

Add NV12 Support for VP8

Video

Feature

11976

Avoid converting VideoFrameBuffers to I420 before passing them into a VideoEncoder

Video

Bug

11977

NV12 frames held by kNative VideoFrameBuffers can be accessed

Video

Bug

11995

Split SDP offer/answer handling from the rest of PeerConnection

Blink>WebRTC>PeerConnection

Bug

1200

audio is broken on android since r3226

Audio

Bug

12026

VideoStreamTest.ResolutionAdaptsToAvailableBandwidth is flaky

Video

Bug

12035

Get rid of memory allocations in webrtc::AudioMixerImpl::GetAudioFromSources()

Audio

Bug

12036

Get rid of memory allocation in webrtc::AudioMixerImpl::CalculateOutputFrequency()

Audio

Bug

12051

Fix logging in desktop_capture module on Windows

DesktopCapture

Bug

12102

Sort out webrtc usage of reader-writer locks

Internals

Bug

12121

Enable continuous audio polling from ADM after StopPlay in VoIP API

Audio

Bug

3513

SDP parsing: malformed m-lines are ignored

PeerConnection

Feature

6762

Include packet overhead in BWE

BWE

Bug

1140452

AEC3: New transparent mode classifier can cause low level echo leakage

Blink>WebRTC>Audio

Bug

12054

Deprecate and remove GetRemoteAudioSSLCertificate

PeerConnection


Philipp Hancke

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Dec 16, 2020, 1:58:03 AM12/16/20
to discuss...@googlegroups.com
Is there a reason it takes four weeks to generate the release notes after the branch cut?

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