PSA: WebRTC M62 Release Notes

[Anatoli Davidson]

M62

WebRTC M62 branch (cut at r19592)

Summary

WebRTC M62, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 10 new features and over 20 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.

Important PSAs

WebRTC mobile libraries

We have launched prebuilt libraries for mobile development on Android and iOS. The easiest way to get started is using the libraries available at JCenter and cocoapods.org. These libraries are compiled from the tip-of-tree and are meant for development purposes only. The versioning system used is 1.1.cr-commit-position, where cr-commit-position can be used to identify the exact WebRTC revision the .aar/pod was built from. We intent to release the libraries on a weekly basis.

Features

Added SetBitrate to native PeerConnection API

PeerConnection::SetBitrate limits the minimum and maximum bandwidth allocated for all RTP streams sent by a PeerConnection. Other limitations might affect these parameters and are respected (for example “b=AS” in SDP). ObjC and Java bindings are also provided. See webrtc:7395 for details.

Deprecations

RTCPeerConnection#getStreamById

In line with other browsers, the getStreamById method on RTCPeerConnection has now been removed (it was initially announced in this PSA).

RTCAVFoundationVideoSource

The deprecated interface RTCAVFoundationVideoSource is now removed (in iOS). If you have an application that still uses RTCAVFoundationVideoSource, you are strongly encouraged to migrate ASAP. More details in the original PSA.

Platform	Issue	Description	Component
Chrome	8125	Remove deprecated code from DirectTransport	Network

Features and Bugfixes

Chrome

Type	Issue	Description	Component
Feature	762867	Merge to M62: Additional metrics reporting for WebRTC experiments.	Blink>WebRTC>Video
Feature	7907	Implement RTP keep-alive	ORTC, Video
Feature	7218	APM quality assessment toolbox	Audio
Feature	7882	Add audio call duration UMA metric based on packet traffic	Audio, Stats
Feature	8138	Add a bursty frame generator that doesn't rely on files	Video
Feature	7420	Report end-to-end delay in UMA separately for screenshare and realtime video.	Stats, Video
Feature	8076	Implement RTCMediaStreamTrackStats.totalSamplesReceived and concealedSamples, used to calculate "expand rate".	Stats, Audio
Bug	8131	Slow recovery in AEC3 for some variants of capture buffer issues	Audio
Bug	8028	Internal corruption to incoming VP8 stream	Video
Bug	7405	Thread-checkers fail on UT	Audio
Bug	7903	Packet discard rate unimplemented in NetEq.	Audio
Bug	750656	"Enable diagnostic packet and event recording" checkbox state forgotten	Blink>WebRTC>Tools
Bug	7860	REMB packets have audio SSRC	BWE
Bug	7946	STUN / TURN Candidates not gathered when re-creating PeerConnection	Network>ICE
Bug	8137	RTCP receiver should verify sender ssrc when parsing target bitrate	Network>RTP
Bug	7966	OnNegotiationNeeded not called when track is added to stream	PeerConnection
Bug	6783	Turn off error resilience when running without temporal and spatial layers for VP9.	Video
Bug	8037	RtpVideoStreamReceiver::receive_cs_ suspected unnecessary and replaceable by ThreadChecker	Video
Bug	7551	Remove or throttle logging from FrameBuffer2	Video
Bug	7920	VP9 SVC stream freeze.	Video
Bug	8060	Don't clear newer packets from video_coding::PacketBuffer when calling ClearTo	Video

Native Android/iOS

Type	Issue	Description	Component
Feature	8155	Consider exposing EglBase14 and EglBase10 as part of the Android SDK API	Mobile (Android)
Feature	7785	Support more formats in RTCVideoFrame	Mobile (iOS)
Feature	7924	Injectable Obj-C video codecs	Mobile (iOS)
Feature	7177	Refactor iOS video source API	Mobile, Video (iOS)
Feature	7351	Expose RTCAudioSession.h in iOS SDK/Framework build to allow using manual audio	Mobile (iOS)
Feature	7395	Add API for setting min/max BWE to PeerConnection and/or RtpTransportController	PeerConnection (Android, iOS)
Feature	7389	Unity native plugin example	SampleApps (WIndows, Android)
Bug	8110	rtc::CurrentThreadId wrong on Android	Mobile (Android)
Bug	7901	SurfaceViewRenderer does not respond to changes of scaling type	Mobile (Android)
Bug	8035	PeerConnectionFactory crashes if it is reinitialized without hardware codec support	Mobile (Android)
Bug	8124	HardwareVideoDecoder stops working after a resolution change	Mobile (Android)
Bug	7880	ObjcVideoTrackSource::OnCapturedFrame ignores cropX & cropY	Mobile (iOS)
Bug	7898	Receiving video is momentarily shown upside down	Mobile (iOS)
Bug	8022	H264 vt compression session is created with hardware accl disabled always on mac osx.	Mobile (iOS)
Bug	8106	Setting field trial flags in AppRTCMobile not working	Mobile (iOS)
Bug	7976	iOS AppRTCDemo doesn't call PeerConnection close()	Mobile (iOS)
Bug	7703	iPhone fails to send and receive video on T-Mobile network (due to limit of 5 IPv6 interfaces)	Mobile, Network (iOS)
Bug	8043	Issue on exiting the video call on Android	Mobile, PeerConnection (Android)
Bug	7715	When network interface has multiple IPv6 addresses, and OS selects one WebRTC doesn't expect, the port is discarded	Network>ICE (Android)
Bug	8129	SDP parsing will fail on AppRTCMobile when preferred codec is the last in the list.	SampleApps (iOS)
Bug	7829	UIView.layer accessed off main thread.	Mobile (iOS)

[Philipp Hancke]

also https://bugs.chromium.org/p/webrtc/issues/detail?id=7877 has landed so if you don't see one-byte VP8 pictureIds generated by Chrome anymore you can thank Gustavo Garcia!

(or curse him if this breaks things for you but there is really no reason this should break)

--

---
You received this message because you are subscribed to the Google Groups "discuss-webrtc" group.
To unsubscribe from this group and stop receiving emails from it, send an email to discuss-webrtc+unsubscribe@googlegroups.com.
To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/CACxSdkGiS47-7VXcodASEAKBWx5qmwUyjVt19PpVRHUqmi0W0A%40mail.gmail.com.
For more options, visit https://groups.google.com/d/optout.

[Nils Ohlmeier]

And Firefox will do the same from version 58 on so that things are consistent.

To unsubscribe from this group and stop receiving emails from it, send an email to discuss-webrt...@googlegroups.com.
To view this discussion on the web visit https://groups.google.com/d/msgid/discuss-webrtc/CADxkKi%2Bcx-9aow%3DwuMaQdB7XQLBwSNwqoQRvXdqH8L0U%3Dns8Jg%40mail.gmail.com.