PSA: WebRTC M93 Release Notes

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Huib Kleinhout

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Aug 27, 2021, 8:33:35 AMAug 27
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WebRTC M93 Release Notes


Branch: WebRTC M93 branch

Summary

WebRTC M93, currently available in Chrome's beta channel, contains 12 new features and over 40 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. 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

Highlights

Screen Capture Picker Audio Checkbox Changes

Instead of the previous single checkbox for audio, shared between both [Entire Screen] and [Chrome Tab], we now have separate checkboxes. The default state for the [Chrome Tab] is now checked; the default state for [Entire Screen] remains unchecked. Tracking bug

Improved Capture Ribbon

Add a button that allows navigating quickly between the capturing/captured tabs. For instance, when sharing Docs from Meet, the Docs tab would allow one to quickly switch to the Meet tab, and vice versa. Tracking bug

PSAs

Title

Description

Chromium will disable camera capture after screen lock

For privacy reasons, camera capture will be disabled 15s after a screen lock occurs. The majority of developers and users should see no changes at all from this policy. Chromium already keeps the screen awake when capturing is active and screen lock during capture is very rare. However, some systems may be configured to prevent screen wake locks at the OS level. See the PSA for further details

Demuxing by Payload Type is now disabled if MID and BUNDLE is negotiated

Demuxing by Payload Type is used in order to support legacy endpoints that do not use the MID RTP header extension in the packets and do not signal SSRCs in the SDP. As of M93, PT-based demuxing is disabled if the MID header extension is negotiated.



Features and Bugfixes

Type

Issue

Description

Component

Bug

http://crbug.com/1056230

Adding captureTimestamp and senderCaptureTimeOffset to RTCRtpContributingSource.

Blink>Media

Feature

http://crbug.com/1202526

Missing mouse cursor when sharin a screen on Linux Wayland session

Internals>Media>ScreenCapture

Bug

http://crbug.com/1203206

WebRTC / LibvpxVp8Encoder: Frames are dropped if I420A is converted to I420 when downscaled

Blink>WebRTC>Video

Bug

http://crbug.com/1212630

Make WebRTC and Chromium agree on libyuv scaling filter

Blink>WebRTC>Video

Bug

http://crbug.com/989932

Codebase vs Wimplicit-int-float-conversion

Tools>LLVM

Bug

http://bugs.webrtc.org/10395

Fuzzers For WebRTC


Feature

http://bugs.webrtc.org/10739

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

Network>RTP

Bug

http://bugs.webrtc.org/11581

Frequent polling of a few timers in ModuleRtpRtcpImpl

Internals, Perf

Bug

http://bugs.webrtc.org/11713

RTP header extension encryption is broken

Network>RTP

Bug

http://bugs.webrtc.org/12194

the range of dynamic rtp payload types is exhausted

PeerConnection

Bug

http://bugs.webrtc.org/12295

fallback to rtp payload type range 35-63 when 96-127 is exhausted

Network>RTP

Bug

http://bugs.webrtc.org/12462

Too many decoders are created

Video

Bug

http://bugs.webrtc.org/12510

nackCount stats for outbound audio

Stats

Bug

http://bugs.webrtc.org/12551

Add conceptual documentation for DTLSTransport

Documentation

Feature

http://bugs.webrtc.org/12575

add --start_timestamp and --stop_timestamp to video_replay

Tools

Bug

http://bugs.webrtc.org/12603

VP8: Don't scale buffers for inactive layers

Video

Feature

http://bugs.webrtc.org/12614

dcSCTP

DataChannel

Bug

http://bugs.webrtc.org/12713

NACK: erase an unreceived packet because of uncorrect packet ssrc

Network>RTP

Bug

http://bugs.webrtc.org/12770

Fix echo return stats in modern stats parser

Stats

Bug

http://bugs.webrtc.org/12773

MediaStreamTrack::enabled() is accessed from worker thread

PeerConnection

Feature

http://bugs.webrtc.org/12787

Equip WebRTC proxies with Chrome tracing entrypoints.

Internals

Bug

http://bugs.webrtc.org/12788

One of the simulcasting encoders is stuck when `active` toggled due to wrong bitrate allocation

Video

Feature

http://bugs.webrtc.org/12793

Implement Round Robin scheduler for dcSCTP

DataChannel

Feature

http://bugs.webrtc.org/12794

Support bufferedAmountLowThreshold in dcSCTP

DataChannel

Bug

http://bugs.webrtc.org/12798

DCHECK failed in rtc_stats_report.cc that RTCRemoteInboundRtpAudioStream is already present in this stats report

Stats

Bug

http://bugs.webrtc.org/12810

Code comment in frame_dropper.h seems incorrect

BWE,Video

Bug

http://bugs.webrtc.org/12812

dcSCTP might send FORWARD-TSN not finishing the currently sent message


Bug

http://bugs.webrtc.org/12814

Disable PT demuxing whenever possible

PeerConnection

Bug

http://bugs.webrtc.org/12815

Add small cooldown to unsignalled ssrc stream creation


Feature

http://bugs.webrtc.org/12829

Allow encoders to specify resolution alignment properties

Video

Bug

http://bugs.webrtc.org/12832

dcSCTP may interleave messages due to round robin scheduler


Bug

http://bugs.webrtc.org/12837

PC emits candidates when renegotiating

PeerConnection

Bug

http://bugs.webrtc.org/12839

Simplify the way to determine if a packet has been received.


Bug

http://bugs.webrtc.org/12841

Add conceptual documentation for RTC event log

Documentation

Bug

http://bugs.webrtc.org/12850

PeerConnectionObserverJni is missing OnRemoveTrack event


Bug

http://bugs.webrtc.org/12857

VideoStreamEncoder::EncodeVideoFrame called when encoder_ is null


Bug

http://bugs.webrtc.org/12866

Simulcast adapter: Don't register invalid encode complete callbacks.

Video

Bug

http://bugs.webrtc.org/12867

Fps adaptation downgrade count could be set when framerate is unrestricted.

Video

Bug

http://bugs.webrtc.org/12868

Reduce stats reporting frequency in ChannelReceive::GetAudioFrameWithInfo

Audio

Bug

http://bugs.webrtc.org/12896

Document supported compilers and platforms

Documentation

Bug

http://bugs.webrtc.org/12910

RTCInboundRtpStreamStats's jitterBufferDelay and jitterBufferEmittedCount are not defined for audio (but they are for video)

Stats

Bug

http://bugs.webrtc.org/12913

PlatformThreadTest.MovesHandles is flaky

Cleanup

Bug

http://bugs.webrtc.org/12924

VP9 uncompressed header parser may not do what you thought it was doing


Bug

http://bugs.webrtc.org/12925

Implement RTCInboundRTPStreamStats.nack_count for audio

Stats

Bug

http://bugs.webrtc.org/12941

Bit exactness tests failing with a new version of clang

Audio

Bug

http://bugs.webrtc.org/12952

dcSCTP resets all stream when intended to only reset one

DataChannel

Feature

http://bugs.webrtc.org/6458

Use codec rate when generating RTCP for audio

Audio

Bug

http://bugs.webrtc.org/6779

Stop using assert

Audio

Feature

http://bugs.webrtc.org/7925

Make internal software video codecs injectable and optional

PeerConnection

Bug

http://bugs.webrtc.org/9267

Update thresholds for VideoCodecTests libvpx on Android/iOS

Video

Feature

http://crbug.com/https://bugs.chromium.org/p/chromium/issues/detail?id=1220009

Media-Picker Audio Checkbox Changes

Blink>GetDisplayMedia

Feature

http://crbug.com/http://crbug.com/1214485

Improved Capture Ribbon

Blink>GetDisplayMedia


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