PSA: WebRTC M93 Release Notes

[Huib Kleinhout]

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