PSA: WebRTC M71 Release Notes

[Anatoli Davidson]

WebRTC M71 Release Notes

WebRTC M71 branch (cut at r25118)

Summary

WebRTC M71, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains 20 new features and over 45 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. Native libraries for Android and iOS are built on weekly basis and are available on JCenter and CocoaPods; the Changelog is available here.

PSAs

Unified Plan M71 Experiment; M72 Target - Applications May Break! Please Update Your Code.

“Unified Plan” is the standards compliant SDP format, which in many cases is incompatible with the current “Plan B” SDP format. Switching the default behavior to Unified Plan is a breaking change. Firefox is already using Unified Plan, and Safari Technology Preview has a flag for it. An experiment is currently running that make Chrome Canary/Dev clients change the default on startup with high probability.

The target for rolling out Unified Plan is M72. Developers need to test their applications, they may need to be updated or explicitly set the desired SDP format with “sdpSemantics”. For more information, including links to transition guides, see the PSA for the Chrome change and this PSA for the native transition guide.

Deprecations

Removing C++ interfaces webrtc::EchoCancellation and webrtc::EchoControlMobile

Functionality in those interfaces is partly deprecated, partly moved into the webrtc::AudioProcessing::Config.

Moving the AEC configuration into webrtc::AudioProcessing::Config simplifies the interface for clients. Additionally, these interfaces hinder internal refactoring of the audio processing module.

Any client that uses these interfaces will need to update their code as detailed in the PSA, with the exception that custom AECs (implementing webrtc::EchoControl) will continue overriding the config enable/disable flag for now.

Adding numberOfSimulcastStreams to Java VideoEncoder.Settings constructor

A new parameter numberOfSimulcastStreams has been added to the constructor of Java VideoEncoder.Settings and the old constructor has been removed. More details in the bug.

Issue	Description	Component
9689	Remove the intelligibility enhancer in APM.	Audio

Features and Bugfixes

Type	Issue	Description	Component
Feature	880075	Add blink::UserMediaClient::IsCapturing	Blink>GetUserMedia
Feature	9848	Expose a parameter for setting a lower bound for the packet loss rate in Opus encoder.	Audio
Feature	888971	Log if audio service or process are used	Blink>WebRTC>Audio
Feature	895741	Show AudioService information to chrome://media-internals	Blink>WebRTC>Audio
Feature	9142	Build and publish macOS framework	Build
Feature	8415	Integrate and test WebRTC-adapted BBR.	BWE
Feature	8968	Enable round robin packet queue by default.	Network, Video
Feature	9680	Support attaching RTP extensions differently to the first packet of a video frame layer	Network>RTP
Feature	9823	Video bitrate allocations with new layer structure should be signaled as soon as possible	Network>RTP
Feature	9711	Add field trial for disabling the frame dropper	Video
Feature	8288	Implement round-robin sending in PacedSender	Video
Feature	9809	Make libvpx into an interface inside vp8 encoder wrapper for easier testing	Video
Feature	9803	Add 'number of video freezes per minute' metric	Video
Feature	9632	Update multiplex encoder to support having augmenting data attached to the video	Video
Feature	8136	Get rid of kVideoCodecUnknown	Cleanup
Feature	9346	Field trial parser	BWE
Feature	9582	Add picture_id to generic RTP packetizer format.	Video
Feature	9829	Enable multithreading in libvpx VP9 decoder	Video
Feature	9669	Move VP9 frame rate controller to separate class	Video
Feature	9682	VP9 SVC: control frame rate per spatial layer	Video
Bug	9816	Incorrect filter alignment causing echo leakage	Audio
Bug	9694	AEC3: Allow controlled usage of the shadow filter output	Audio
Bug	890040	[Video Capture Service] Uninitialized field error when using service from ChromeOS	Blink>GetUserMedia>Webcam
Bug	879451	AEC3: Allow controlled usage of the shadow filter output	Blink>WebRTC>Audio
Bug	9811	API level check for ConnectivityManager.getActiveNetwork	Network
Bug	9828	Freeze metric should be based on render time and not decode time	Video
Bug	9761	Simulcast screenshare at high bitrate/low resolution might caused encoder init to fail	Video
Bug	9745	screenshare layers (vp8) might incorrectly adjust max qp setting	Video
Bug	9734	Potential bad performance with simulcast screensharing	Video
Bug	9715	VP9 SVC sent resolution is reported per-layer.	Video
Bug	9740	VP9 enc wrapper doesn't not account for VideoCodec::maxFramerate	Video
Bug	9776	Too much fading when the linear filter has already converged	Audio
Bug	9697	Poor speech quality at the beginning of the call under stationary noises at the render signal.	Audio
Bug	9685	Echo leaks when the limits for the ERLE estimator are increased	Audio
Bug	9805	Echo leaks when the preamplifier gain changes	Audio
Bug	9762	Echo leaks at volume decreases in AEC3	Audio
Bug	9773	AEC3: Delay estimator saturation detection is overly cautious	Audio
Bug	9741	AEC3: Bug in the transition between main and shadow filter output	Audio
Bug	9746	AEC3: Bad transparency at low echo levels	Audio
Bug	9746	AEC3: Bad transparency at low echo levels	Audio
Bug	879264	[Video Capture Service] Support padded yuv frames for virtual devices	Blink>GetUserMedia>Webcam
Bug	892043	Gain too high in WebRTC adaptive digital gain controller 2	Blink>WebRTC>Audio
Bug	878319	TSan reports a data race in webrtc::RtpVideoStreamReceiver::OnRtpPacket	Blink>WebRTC>Network
Bug	883888	webrtc: sdpSemantics: unified-plan breaks onaddstream	Blink>WebRTC>PeerConnection
Bug	884164	2.4%-15.6% regression in webrtc_perf_tests at 24671:24674	Blink>WebRTC>Video
Bug	9812	Log spam when remote sequence number starts high.	BWE
Bug	9703	Desktop capture frame can have corrupted data when using multiple screens	DesktopCapture
Bug	9856	The logic in rtc::Buffer::OnMovedFrom is backwards w.r.t. RTC_DCHECK_IS_ON	Internals
Bug	9684	PhysicalSocketServer should use monotonic time for timeouts	Internals
Bug	9802	Update rtc_json JSON parsing utility code	Internals
Bug	9112	Memory contention by threads in PeerConnection when invoking methods of PortAllocator	Network, PeerConnection
Bug	9839	SRTP decryptor logs several errors every second if the received packets can't be decrypted.	Network>DTLS
Bug	9832	AudioTransport::NeedMorePlayData not exposing NTP value	Network>RTP
Bug	9770	GetSources() for VideoRtpReceiver	Network>RTP, PeerConnection
Bug	9712	webrtcsdp.cc::HasAttribute() only matches beginning of line	PeerConnection
Bug	7600	RTCRtpTransceiver API	SpecConformance
Bug	9674	getStats: mediaType was renamed to kind	Stats
Bug	9785	Add num_temporal_layers to RtpEncodingParameters struct.	Video
Bug	9597	Add support for RtpEncodingParameters max_framerate.	Video
Bug	9791	Temporal dependencies used in layers checker could be tightened for the 2 TL case	Video
Bug	9747	SimulcastEncoderAdapter is initially setting all streams to active	Video
Bug	9782	Low spatial layer frame is marked as non-reference when number of activated layers is < max layers	Video
Bug	882789	AEC3: Bug in the transition between main and shadow filter output	Blink>WebRTC>Audio
Bug	883264	AEC3: Bad transparency at low echo levels	Blink>WebRTC>Audio
Bug	9634	Keyframes may be dropped by vp8 encoder	Video
Bug	cl 103983	Use zeros for rtp packet padding	Network>RTP