PSA: WebRTC M70 Release Notes

[Anatoli Davidson]

WebRTC M70 Release Notes

WebRTC M70 branch (cut at r24472)

Summary

WebRTC M70, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 15 new features and over 55 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

Android SDK builds without video codecs by default

Android SDK doesn’t include video codecs by default anymore. If no video codecs are passed to createPeerConnectionFactory, no video codecs will be included. The purpose is to save binary size for clients that don't need video (e.g. audio or data only). This doesn’t affect clients that are already using injectable codecs. Original PSA here.

Features

Try out getDisplayMedia()

getDisplayMedia is an extension to the Media Capture API, allowing users to capture a user's display, or part thereof, in the form of a video stream. Different than the current capabilities, it doesn’t require any extensions usage. The initial working implementation is now available. Please try out and give feedback. To enable it, you can start Chrome with flag --enable-experimental-web-platform-features or turn it on via chrome://flags/. See bug tracker for details.

RTCPeerConnection.getConfiguration()

getConfiguration() returns the last configuration applied via setConfiguration(), or if setConfiguration() hasn't been called, the configuration the RTCPeerConnection was constructed with. The configuration also has a setting to detect if the RTCPeerConnection defaults to PlanB or Unified Plan which is useful during the transition period. For more details see the Chrome’s feature page.

WebRTC ContentHint attribute

MediaStreamTrack.prototype.contentHint is a property rThe ContentHint interface to tell Chrome about whether you attach more importance to showing motion or detail in your WebRTC video track, as specified in the W3C specification. See the original PSA for more details.

Deprecations

Issue	Description	Component
9409	Deprecate MetricsObserverInterface and use metrics.h macros instead	PeerConnection
9239	Delete deprecated MediaConstraints API	PeerConnection

Features and Bugfixes

Type	Issue	Description	Component
Feature	9479	Avoid the adaptation of the decay in the reverberation model for bad performing filters in AEC3	Audio
Feature	9660	AEC3: State-specific echo suppressor behavior	Audio
Feature	844146	Make WebRTC work with network service	Blink>WebRTC
Feature	8665	Render-side pre-processing in APM	Audio
Feature	9651	The decay parameter estimator in AEC3 can be improved and optimized.	Audio
Feature	9313	Enable the default route when the explicit network binding is not allowed.	Network>ICE
Feature	9655	Allow creating simulcast offer/answer with Plan B semantics	PeerConnection, Video
Feature	9522	Preserve color space information for internal codecs	Video
Feature	9670	Add rtt_mult video error resilience experiment	Video
Feature	8830	Cleanup VideoStreamEncoder api	Video
Feature	9632	Update multiplex encoder to support having augmenting data attached to the video	Video
Feature	653531	Add a method to inform media tracks about their type of content	Blink>MediaStream
Feature	9473	Replace rtc::{Make,Wrap}Unique with their Abseil counterparts	Internals
Feature	9434	Implement and evaluate PCC	BWE
Feature	9650	Add frame rate field to webrtc::SpatialLayer struct.	Video
Bug	9641	AEC3: Very poor transparency during repeated audio buffer issues	Audio
Bug	9591	Render preprocessor integration impacting the residual echo detector	Audio
Bug	9561	AEC3: The filter output is not correctly computed when the AGC gain changes are compensated	Audio
Bug	9565	AEC3: The shadow filter is slow to re-adapt which significantly limits the adaptive filter performance	Audio
Bug	9572	AEC3 transparency during call start and after echo path changes is non as good as it can be	Audio
Bug	9581	For devices with render playout effects AEC3 sometimes leaks echoes	Audio
Bug	9602	AEC3: Temporary mismatch between main and adaptive filter lengths cause DCHECK failure	Audio
Bug	9612	AEC3: Transparency loss when the main filter is inaccurate	Audio
Bug	9614	AEC3: The ability to adjust the AEC3 performance is limited by the filter lengths	Audio
Bug	9663	AEC3: Low level echo leakage for platforms with strong echo path gain above 10 kHz	Audio
Bug	873187	Migrate WebRtcEventLogUploaderImpl to SimpleURLLoader	Blink>WebRTC
Bug	881224	Spammy log from MediaStreamAudioProcessor.	Blink>WebRTC>Audio
Bug	865397	Avoid the adaptation of the decay in the reverberation model for bad performing filters in AEC3	Blink>WebRTC>Audio
Bug	867373	AEC3: The filter output is not correctly computed when the AGC gain changes are compensated	Blink>WebRTC>Audio
Bug	867873	AEC3: The shadow filter is slow to re-adapt which significantly limits the adaptive filter performance	Blink>WebRTC>Audio
Bug	868329	AEC3 transparency during call start and after echo path changes is non as good as it can be	Blink>WebRTC>Audio
Bug	869821	For devices with render playout effects AEC3 sometimes leaks echoes	Blink>WebRTC>Audio
Bug	872201	AEC3: Temporary mismatch between main and adaptive filter lengths cause DCHECK failure	Blink>WebRTC>Audio
Bug	873074	AEC3: Transparency loss when the main filter is inaccurate	Blink>WebRTC>Audio
Bug	873100	AEC3: The ability to adjust the AEC3 performance is limited by the filter lengths	Blink>WebRTC>Audio
Bug	873100	AEC3: The ability to adjust the AEC3 performance is limited by the filter lengths	Blink>WebRTC>Audio
Bug	866374	Merge to M69: WebRTC: Enable simulcast screenshare by default	Blink>WebRTC>Video
Bug	9637	Packets sometimes dropped in UDP socket buffers at very high packet rates	Network
Bug	9608	SimulcastEncoderAdapter should not update max qp for screencast.	Video
Bug	9677	Avoid the decrease of the ERLE estimation during render pauses.	Audio
Bug	9668	Transparency negatively affected by ERLE uncertainty	Audio
Bug	9526	AEC3: No special action is taken when the microphone gain is changed.	Audio
Bug	9647	AEC3: AEC3 does not handle platforms where there reporting of the render signal to the AEC is delayed	Audio
Bug	862395	[Video Capture,Windows] Some capture devices not working with MediaFoundation	Blink>GetUserMedia>Webcam
Bug	862055	WebRTC getUserMedia stream freezes if browser is slept/backgrounded	Blink>GetUserMedia>Webcam
Bug	859610	MediaStreamTrack gets borked after too many start-stop recordings	Blink>MediaStream
Bug	863826	AEC3: No special action is taken when the microphone gain is changed	Blink>WebRTC>Audio
Bug	800212	Mojofy content/renderer/p2p/	Blink>WebRTC>Network
Bug	750512	Newly constructed RTCPeerConnection object has non-null localDescription and remoteDescription properties	Blink>WebRTC>PeerConnection
Bug	866447	chrome://webrtc-internals tracking RTCRtpTransceiver	Blink>WebRTC>PeerConnection
Bug	704356	RTCPeerConnection.getConfiguration	Blink>WebRTC>PeerConnection
Bug	875213	stun/turns server missing from chrome://webrtc-internals	Blink>WebRTC>Tools
Bug	867029	Transfer WebRTC color space information to chromium color space	Blink>WebRTC>Video
Bug	864529	Refactor WebRtcVideoCapturerAdapter	Blink>WebRTC>Video
Bug	875391	173.6% regression in webrtc_perf_tests at 24254:24254	Blink>WebRTC>Video
Bug	9587	BufferTest.TestConstructEmpty fails on UBSan with new clang	Build
Bug	7774	Ignoring EOR flag when receiving data from usrsctp,resulting in loss of message integrity for messages between 65KB and 256KB	DataChannel
Bug	868776	3.3%-8.8% regression in webrtc_perf_tests at 23973:23973	Internals>WebRTC
Bug	4165	ICE candidates doesn't support domain name	Network
Bug	9623	Add TLS Custom Certificate Verifier To Reconfigure Port	Network>DTLS
Bug	9593	SafeSetError() in peerconnection.cc contains use-after-move of webrtc::RTCError variable	PeerConnection
Bug	9537	RTC_DCHECK for valid "mid" is too strict in "RtpDemuxer::AddSink"	PeerConnection, SpecConformance
Bug	9619	Incorrect encode usage percent stats using new cpu adaptation,with undesired downscaling as result	Video
Bug	9643	RTP ref frame finder doesn't handle VP9 flexible mode properly.	Video
Bug	9657	Duplicates in VP9 RTP p_diff list	Video
Bug	9658	No scalability structure in flexible mode	Video
Bug	865193	Video played in Windows Media Player is not captured for presentation	Blink>GetUserMedia>Desktop
Bug	874235	CrOS puts the active window at the bottom in the Windows tab of the picker	Blink>GetUserMedia>Desktop
Bug	743093	Error logs from v4l2_capture_delegate.cc when using webcam	Blink>GetUserMedia>Webcam
Bug	9634	Keyframes may be dropped by vp8 encoder	Video

[Naoki Hiroshima]

Hi there,

I can only see the branches up to `branch-heads/69`, but not `branch-heads/70`.

Am I missing something?

-- N

[Anatoli Davidson]

Hi Naoki,

The link to the M70 branch is at the top of the release notes: WebRTC M70 branch

Does that link work for you?

-Anatoli 

[Naoki Hiroshima]

Hi Anatoli,

Yes, I can see all those commits in the link in the `master` branch, but I don't see `branch-heads/70` branch.

`git branch -r` shows 48 branches and the latest one is `branch-heads/69`. I was expecting 70's branch would show up.

-- N

[Anatoli Davidson]

Perhaps you need to update your branch. See more about getting and updating the source code here.

-Anatoli

[Mike K]

Hello i still have a problem with domain icecandidates issue4165.  Anyone else or do you need to chnage the JS code?

br M

Dne sreda, 10. oktober 2018 08.55.48 UTC+2 je oseba Anatoli Davidson napisala:

[3440...@qq.com]

Which version of cocoapods does webrtc M70 correspond to? thank you

在 2018年10月10日星期三 UTC+8下午2:55:48，Anatoli Davidson写道：