PSA: WebRTC M70 Release Notes

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Anatoli Davidson

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Oct 10, 2018, 2:55:48 AM10/10/18
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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

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Oct 11, 2018, 9:43:54 AM10/11/18
to discuss...@googlegroups.com
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

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Oct 11, 2018, 10:45:13 AM10/11/18
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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

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Oct 11, 2018, 11:44:45 AM10/11/18
to discuss...@googlegroups.com
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

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Oct 11, 2018, 12:55:46 PM10/11/18
to discuss...@googlegroups.com
Perhaps you need to update your branch. See more about getting and updating the source code here.

-Anatoli

Mike K

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Oct 23, 2018, 8:56:08 AM10/23/18
to discuss-webrtc
Hello i still have a problem with domain icecandidates issue4165.  Anyone else or do you need to chnage the JS code?

icecand.PNG



br M



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

3440...@qq.com

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May 7, 2019, 4:20:06 AM5/7/19
to discuss-webrtc
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:
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