PSA: WebRTC M68 Release Notes

[Anatoli Davidson]

WebRTC M68 Release Notes

WebRTC M68 branch (cut at r23360)

Summary

WebRTC M68, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 20 new features and over 40 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

Unified Plan SDP testing flag

As announced earlier, Chrome will transition from Plan B SDP format to Unified Plan. Web developers can experiment with the new format by specifying {sdpSemantics: "unified-plan"} in the RTCPeerConnection constructor. Note that certain corresponding API’s are not completely implemented yet. More details about the transition plan can be found on webrtc.org.

WebRTC ObjC SDK now ships without SW video codecs by default

The WebRTC ObjC SDK is now built without SW video codecs by default. The purpose is to save binary size for clients that don't need SW video codecs. Link to PSA.

Features

Simulcast Screenshare

Any developer with an app that has simulcast configured, uses screenshare, and has the x-google-flag:conference flag set, might be affected by a new mode where two video streams with the same resolution but different bitrates and potentially different frame rates are sent. If you are running an SFU/MCU type service, make sure you can reliably distinguish between the high and low quality encodings, as just looking at resolution will no longer be enough. More details can be found in the original PSA.

Experimental support for OS-level echo cancellation on Mac and Windows

Once signed for the Origin Trial, application developers can evaluate using system-specific echo canceller implementations on macOS and Windows. These will be exposed though a new, experimental getUserMedia constraint named echoCancellationType. More information will be available on the Google Developer Blog once the Origin Trial is launched. You can also try it locally by starting Chrome with the command line argument --enable-blink-features=ExperimentalHardwareEchoCancellation.

Deprecations

Deprecation of Android MediaStream.label()

In order to be consistent and spec compliant, we have updated the media stream’s usage of “label” to “ID” (see bug 8977). The Android MediaStream has been updated from label() to getId() for accessing this field. Please update accordingly. This was communicated earlier in a separate PSA.

Deleting RTPFragmentationHeader on the video decode path

Webrtc's public video decoder api is defined by the interface webrtc::VideoDecoder, declared in api/video_codecs/video_decoder.h. The Decode method of this interface, and the corresponding objc api, used to have an argument of type RTPFragmentationHeader*. Since the RTPFragmentationHeader is not useful for video decoding, we have deleted those arguments. This affects applications injecting their own codecs. See details in the original PSA.

Issue	Description	Component
9071	Remove dependency on openmax_dl	Audio
6983	Remove WebRtcVideoSendStream2::VideoSink inheritance.	Video
9228	Remove internal factories from FakeWebRtcVideoCodecFactories	Video
6548	Move AudioFrame to webrtc/api	Audio

Features and Bugfixes

Type	Issue	Description	Component
Feature	9126	Add UMA histograms for NetEq's expand rates	Audio
Feature	7404	Move aecdump file IO from real-time audio thread to low-prio task queue	Audio
Feature	9193	Improvements on transparency for AEC3 under stationary noises at the farend side	Audio
Feature	9252	unpack_aecdump should unpack render/capture call order	Audio
Feature	846270	Add echo canceller type constraint	Blink>WebRTC>Audio
Feature	845187	Support Windows native echo canceller	Blink>WebRTC>Audio
Feature	9185	Optimize WebRTC log calls to use fewer bytes in the binary	Internals
Feature	9168	Support VPN adapter type in WebRTC	Network>ICE
Feature	9122	Handle Receiver Reference Time Report from multiple receivers	Network>RTP
Feature	8977	Update Media Stream naming of label to id	PeerConnection
Feature	9242	Minimum bit rate thresholds too low for 720p in VP9 SVC	Video
Feature	74261	Mitigate RTP pic ID errors in VP9 SVC non-flexible mode (for VP9 SVC enablement).	Video
Feature	76640	Limit inter-layer prediction to key pictures (for VP9 KSVC enablement).	Video
Feature	73685	Fill drops with last decoded frame (for VP9 SVC enablement).	Video
Feature	73440	Decode base reference frame if current layer was dropped  (for VP9 SVC enablement).	Video
Feature	71500	Add control for inter-layer prediction mode (for VP9 KSVC enablement).	Video
Feature	71663	Fix calculation of target bitrate of VP9 spatial layer (for VP9 SVC enablement).	Video
Feature	9260	Add field trial for configuring default number of temporal layers in a vp8 simulcast call	Video
Feature	9165	Reduce FEC table size and generate some FEC masks at runtime	BWE
Feature	9154	Revise examples/peerconnection to work with Unified Plan	PeerConnection
Feature	811683	WebRTC: Add UMA counters for Unified Plan	Blink>WebRTC>PeerConnection
Bug	9159	The AEC3 echo removal is too aggressive for low-level echoes during nearend-speech	Audio
Bug	9246	AEC3 echo leakage for closed headsets with non-negligible echo	Audio
Bug	9256	The AEC3 transparent mode is too aggressive	Audio
Bug	9206	The computation of the spectra for the render signal in AEC3 has an unintended side-effect	Audio
Bug	9163	The AEC3 initial alignment may sometimes be off, causing short periods of echo leakage initially.	Audio
Bug	9283	The AEC3 API call skew tolerance limit is set too tight	Audio
Bug	9204	The leakage behavior in some AEC3 FFT estimates do not match	Audio
Bug	9255	Incorrect usage of the echo path gain in the AEC3 nonlinear mode	Audio
Bug	836979	[desktopCapture OSX] screen capture stops internally on some MBP	Blink>GetUserMedia>Desktop
Bug	841193	The AEC3 transparent mode is too aggressive	Blink>WebRTC>Audio
Bug	839379	AEC3: Suppressor too aggressive during double talk	Blink>WebRTC>Audio
Bug	833801	The AEC3 echo removal is too aggressive for low-level echoes during nearend-speech	Blink>WebRTC>Audio
Bug	837563	The leakage behavior in some AEC3 FFT estimates do not match	Blink>WebRTC>Audio
Bug	840347	AEC3 echo leakage for closed headsets with non-negligible echo	Blink>WebRTC>Audio
Bug	9232	Call::OnTargetTransferRate sometimes tries to use deleted task queue member from another thread.	BWE
Bug	7112	RtpCodecParameters should expose channels and parameters.	Audio, Stats, Video
Bug	9192	AEC3: Attenuation of narrow banded peaks is too aggressive	Audio
Bug	9230	AEC3: Suppressor too aggressive during double talk	Audio
Bug	9241	AEC3: External delay is not applied correctly	Audio
Bug	4844	Clean codes relate to Speex	Audio
Bug	9119	The AEC3 handling of saturated echoes is overly conservative	Audio
Bug	9243	Stream delay reporting is enforced when the AEC3 echo cancellation solution is used	Audio
Bug	9109	AudioFrame::empty_data() adds 7KB of zeros to the binary size	Audio
Bug	796112	Sharing virtual desktop applications may not be captured	Blink>GetUserMedia>Desktop
Bug	831756	Add UMA for screenshare track ended	Blink>GetUserMedia>Desktop
Bug	819629	SetLocalDescription returns wrong error when called in wrong state	Blink>WebRTC>PeerConnection
Bug	9178	[desktopCapture OSX] cursor is wrongly composed in even if cursor is out of the captured screen/window	DesktopCapture
Bug	8652	Use the given IOSurface to reach higher capture framerate	DesktopCapture
Bug	9269	rtc::Event on top of pthreads should use monotonic clock	Internals
Bug	9112	Memory contention by threads in PeerConnection when invoking methods of PortAllocator	Network, PeerConnection
Bug	4050	Implement RTP/RTCP extensions for mid value transport	Network
Bug	9235	GetAdapterTypeFromName maps "eth" to ADAPTER_TYPE_UNKNOWN	Network>ICE
Bug	8772	RtpSenderInterface::SetParameters should fail with RTCError, not bool	PeerConnection
Bug	9171	Wrong payload size set to NetEq simulator in rtc_event_log visualizer.	Tools
Bug	9249	Receiver can't decode VP9 stream with partially enabled SVC	Video
Bug	9245	VP9 SVC: extra reference in GOF	Video
Bug	9151	SVC bitrate limits are used when encoding in single layer	Video
Bug	9217	VP9 SVC: encoder uses base layer for temporal prediction	Video
Bug	839860	AEC3: External delay is not applied correctly	Blink>WebRTC>Audio
Bug	836729	AEC3: Attenuation of narrow banded peaks is too aggressive	Blink>WebRTC>Audio
Bug	590625	SDP s= line (session name) must admit a single space as value	Blink>WebRTC>PeerConnection
Bug	826206	chrome://webrtc-logs doesn't refresh; must be restarted to refresh	Blink>WebRTC>Tools

[Anatoli Davidson]

Also see this PSA about support for RTCRtpSender.getParameters() and setParameters() which landed in M68.

-Anatoli