PSA: WebRTC M90 Release Notes

[Huib Kleinhout]

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Bug	11266	Outdated information about working with branches	Documentation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]

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]

that list looks much better, thanks Huib!

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