PSA: WebRTC M90 Release Notes

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Huib Kleinhout

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Apr 7, 2021, 10:28:02 AMApr 7
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WebRTC M90 Release Notes

better late than never


WebRTC 4430 branch (cut at bb52bdf09516ca548c4aff50526eda561f239bc0)

Summary


WebRTC M90, currently available in Chrome's beta channel, contains 2 new features and 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.

PSAs


Plan B SDP Deprecation

Reminder: Plan B SDP is deprecated and will be removed. The timeline is available here.


Features


MediaStreamTrack Insertable Streams Origin Trial

An extension to the MediaStream and WebCodecs API that allow applications to 1) have access to the raw data contained in a MediaStreamTrack, and 2) define new custom MediaStreamTracks. Both capabilities can be used in combination, for example, to create media effects (e.g. "funny hats"). The API relies on the WebCodecs raw media interfaces and the WHATWG Streams API. This feature is part of the WebCodecs origin trial.


getCurrentBrowsingContextMedia Origin Trial

A new experimental API for capturing the current tab is under development. A first implementation is available as an origin trial. Read the explainer here.


Features and Bugfixes


Type

Issue

Description

Component

Bug

1152841

Browser hangs occasionally when closing share target picker

Internals>Media>ScreenCapture

Bug

1155459

Default STUN attribute length limit is too small

Blink>WebRTC>Network

Bug

943975

Surface max message size in RTCSctpTransport

Blink>WebRTC>PeerConnection

Feature

10439

Provide common interface for bitstream parsers

Video

Feature

10480

Improve RNN VAD efficiency and code quality

Audio

Bug

10675

support for logging raw rtp in text2pcap format

Network>RTP

Feature

10897

Add TURN_LOGGING_ID

Network>ICE,PeerConnection

Bug

11266

Outdated information about working with branches

Documentation

Bug

11495

Don't allocate 1 unused byte for sequence checkers in non-debug mode

Internals

Bug

11767

[Stats] Reduce the number of blocking-invokes from 2 to 1.

Stats

Feature

12111

VoipVolumeControl interface for VoIP API

Audio

Bug

12148

AV1 active decode target mask is not set properly

Video

Bug

12167

AV1 packetizer sets mark bit on each spatial layer

Network>RTP

Bug

12181

transportId is missing from RTCCodecStats

Stats

Bug

12185

Incorrect range of GetLinearAecOutput output

Audio

Feature

12193

VoIP API result types and enforcement

Audio

Bug

12194

the range of dynamic rtp payload types is exhausted

PeerConnection

Bug

12204

fix broken video_replay threading

Tools

Feature

12208

Create documentation for network emulation framework

Documentation

Bug

12215

SetLocalDescription/SetRemoteDescription calling CreateSessionDescription thrice.

PeerConnection

Bug

12216

Enable initial frame drop for one active simulcast stream

Video

Bug

12217

Robotic audio heard when the connection is using TLSv1.2

Audio,Network

Bug

12238

RTCPeerConnection Create function should return an error code

PeerConnection

Bug

12249

rtc::GetProcessCpuTimeNanos behaves strange in DVQA tests on Windows

undefined

Bug

12265

AEC3: Linear filters can gradually diverge in long calls

Audio

Bug

12274

Libvpx VP9 codec wrapper is hard to test

Video

Bug

12297

VideoReceiveStream2: remove unneeded PostTask

Perf

Bug

12323

JsepSessionDescription::Clone() does not copy ICE candidates

PeerConnection

Bug

9424

SrtpTransport::OnWritableState incorrectly calculates writability

PeerConnection

Huib Kleinhout

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Apr 12, 2021, 3:44:48 AMApr 12
to discuss...@googlegroups.com
UPDATE: The list of bugfixes was incorrect. The right list follows here:

Type

Issue

Description

Component

Bug

1138888

Low-latency renderer for WebRTC

Blink>WebRTC>Video

Bug

1155477

AEC3: Linear filters can gradually diverge in long calls

Blink>WebRTC>Audio

Bug

1170699

Failed to initialize the WEBRCT av1 encode

Blink>WebRTC

Feature

516700

WebRTC Chromium Clock Difference

Blink>WebRTC

Bug

10675

support for logging raw rtp in text2pcap format

Network>RTP

Bug

11031

Retransmission can fail when MID has been negotiated [Unified Plan]

Network>RTP

Feature

11989

VoipStatistics interface on VoIP APIs for media statistics

Audio

Bug

12265

AEC3: Linear filters can gradually diverge in long calls

Audio

Bug

12279

(network.cc:908): Connect failed with 10051 every 2 sec

PeerConnection,Tools

Bug

12380

The comfort noise suddenly change its energy for each refresh DTX packet when receiving an Opus stream.

Audio

Bug

12383

Collect stats on bundle usage


Bug

12384

Registry-Key-MMDevices-Audio-Handles are increasing in the windows Client by every audio call

Audio

Bug

12398

seg fault on av1 encoder when using svc and odd width/heigh values

Video

Bug

12407

SEA creates and initializes encoders for deactivated layers

Video

Bug

12426

JsepTransport::jsep_transports_by_name_ accessed from multiple threads without protection


Bug

12427

JsepTransportController events marshalled between threads for PeerConnection


Bug

12430

TSAN report for RtpBitrateConfigurator


Bug

12431

RTC event log visualization doesn't work with python3

Tools

Feature

12432

Visualize RTCP BYE messages in RTC event logs

Tools

Bug

12439

Legacy getStats stops working if system time goes backwards

Stats

Bug

12445

JsepTransportController::mid_to_transport_ unprotected


Bug

12448

ULPFEC: unexpected arrival order and very delay arrival

Video

Bug

12455

webrtc::AudioSendStream::Config::ToString() fails on m90

Audio

Feature

12459

Allow cropped resolution when limiting max layers.

Video

Bug

12487

Implement jitter stats for video RTP stream



Thanks to Philipp Hancke for noticing!

Huib

Philipp Hancke

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Apr 12, 2021, 3:46:15 AMApr 12
to discuss...@googlegroups.com
that list looks much better, thanks Huib!

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