PSA: WebRTC M62 Release Notes

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Anatoli Davidson

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Sep 29, 2017, 11:02:05 AM9/29/17
to discuss...@googlegroups.com

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

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Oct 24, 2017, 4:48:39 AM10/24/17
to WebRTC-discuss
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)

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Nils Ohlmeier

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Oct 24, 2017, 1:40:39 PM10/24/17
to discuss-webrtc
And Firefox will do the same from version 58 on so that things are consistent.

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