PSA: WebRTC M84 Release Notes

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Anatoli Davidson

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Jun 3, 2020, 8:59:50 AM6/3/20
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WebRTC M84 Release Notes


WebRTC M84 branch (cut at r31262)

Summary


WebRTC M84, currently available in Chrome's beta channel, contains 4 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.

PSAs

Simulcast-compatible getStats(): outbound-rtp stats are now split by ssrc

When simulcast is used and multiple RTP streams are produced by the same RTCRtpSender, getStats() now produces one “outbound-rtp” stats object per RTP stream which can be identified by their ssrc. This means that counters such as packets and frame rate are now split per layer rather than aggregated into a single “outbound-rtp” object. This only affects the spec-compliant (promise-based) getStats() API; the legacy (callback-based) getStats() API will continue to report aggregate values. See PSA.

Deprecations


Issue

Description

Component

11450

Remove deprecated ssl_identity methods

Network

PSA

Disabling echo detector V1

Audio, Stats

Features and Bugfixes


Type

Issue

Description

Component

Feature

11251

webrtc voip api

Audio

Feature

11392

Simplify adaptation counting

Stats,Video

Feature

10773

Do not reset video encoder on max/min bitrate change

Video

Bug

11487

AEC3 sometimes applies too conservative suppression for higher frequencies

Audio

Bug

1067597

AEC3 sometimes applies too conservative suppression for higher frequencies

Blink>Media>Audio

Bug

1068705

WebRTC-TCP is broken in Canary in Dev, but only when peer is in LAN

Blink>WebRTC

Bug

628676

ICE restart between Chrome 52 and 53 has chance of completely breaking connection.

Blink>WebRTC>Network

Bug

1065838

webrtc insertable streams: does not work for simulcast

Blink>WebRTC>PeerConnection

Bug

1068468

webrtc insertable streams: receiver chunk.type is always "key"

Blink>WebRTC>PeerConnection

Bug

1069278

webrtc insertable streams: timestamp field in sender audio frames is missing the RTP start timestamp offset

Blink>WebRTC>PeerConnection

Bug

1080789

googTargetEncBitrate doesn't ramp back up when higher simulcast streams are enabled by changing RtpEncodingParameters on the sender

Blink>WebRTC>Video

Bug

11427

Do not include non-Opus audio encoders in bitrate allocation even when audio-twcc has been negotiated

BWE

Bug

11423

Peer reflexive TCP candidates missing tcptype.

Network>ICE

Bug

11480

The MedianSlopeEstimator class is not used

Stats

Bug

9547

getstats results for simulcast are wrong and don't split up per ssrc

Stats,Video

Bug

11477

Early media of SSRC signaled on another "m= section" leads to no video

Video

Bug

11562

Switching degradation preference from MAINTAIN_RESOLUTION to MAINTAIN_FRAMERATE can result in halting adaptations

Video

Bug

11521

[ResourceAdaptation] QualityScalerResource handles feedback from adaptation asynchronously

Video

Bug

11222

Refactor VideoSendStream to allow other resource adaptation modules

Video

Bug

11521

[ResourceAdaptation] QualityScalerResource handles feedback from adaptation asynchronously

Video

Bug

11542

[Adaptation] Move ResourceAdaptationProcessor to a separate task queue (Unblock Call-Level Mitigation Strategies)

Video

Bug

11520

[ResourceAdaptation] Resources run on their own task queue

Video

Bug

11341

Implement injectable video encoder selector.

Video

Bug

11490

ReceiveStatisticsProxy's CallStatsObserver and avg_rtt_ms_ == dead code?

Video

Bug

11490

ReceiveStatisticsProxy's CallStatsObserver and avg_rtt_ms_ == dead code?

Video


Philipp Hancke

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Jun 3, 2020, 9:33:13 AM6/3/20
to discuss...@googlegroups.com
also note that GCM cipher suites for DTLS-SRTP are enabled by default now. They're not preferred so there should be no change in Chrome-Chrome calls.

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Michael McKenzie

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Jun 16, 2020, 5:11:35 AM6/16/20
to discuss-webrtc
Why is a bug from 2016 that relates to interop problems between Chrome 52 and 53 listed in these release notes?
https://crbug.com/628676
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