PSA: WebRTC M78 Release Notes

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Chakri Munagala

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Oct 9, 2019, 4:30:23 AM10/9/19
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WebRTC M78 Release Notes


WebRTC M78 branch (cut at r29078)

Summary


WebRTC M78, currently available in Chrome's beta channel, contains 5 new features and over 52 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. Change log is available here.


Features

Stats updates

New stats:

  • encoderImplementation and decoderImplementation (issue).

  • selectedCandidatePairChanges (issue).

  • silentConcealedSamples, insertedSamplesForDeceleration and removedSamplesForAcceleration exposed in JavaScript and back-merged into M77. (These were implemented for M76 in C++ but were not then exposed in Chrome due to a bug.)


chrome://webrtc-internals/ updates:

  • Add graph for audio energy in RMS for sender-side audio levels (issue).

  • Add graph for [framesReceived - framesDecoded] (issue).

  • Display which codec is used by an RTP stream, this was previously filtered out from the internals page (issue).

  • Refactoring: Replace [targetEncodedBytes/s] with [totalEncodedBytesTarget/s] (issue).


Features and Bug Fixes


Type

Issue

Description

Component

Feature

10900

Add support for RTCTransportStats.selectedCandidatePairChanges

Stats

Feature

10915

Make min video target bitrate configurable.

Video

Feature

10650

Add rtpTimestamp to contributing sources

Network>RTP

Feature

10796

Set max video bitrate to value provided by encoder caps if it is not set by app

Video

Bug

10091

Add UMA metrics to track usage on mDNS host candidates

Stats

Bug

10661

GetSources should return sources in descending timestamp order

Network>RTP

Bug

10775

Add totalDecodeTime to RTCInboundRTPStreamStats

Stats

Bug

990318

Add [framesReceived - framesDecoded] to RTCMediaStreamTrack stats

Blink>WebRTC>Video

Bug

10922

sctp_transport.cc ignores usrsctp_sendv() result

DataChannel

Bug

10890

[Standard GetStats] Expose encoder and decoder implementation name

Stats

Bug

1981

Re-enable functionality lost when removing unused main function in unit_test.cc.

Audio

Bug

10896

AEC3: Incorrect computation of audio buffer delay

Audio

Bug

10942

Add support of AudioRecord.Builder in the ADM for Android

Mic, Audio

Bug

10884

Add reporting of audio device underrun counter

Audio

Bug

10903

Some audio track stats are not being picked up by RTCStats::Members()

Stats

Bug

997673

apprtc test files should not live directly in out/

Infra>Client>WebRTC

Bug

10545

RTCRtpContributingSource is specified to store information about the most recent packet DELIVERED to the RTCRtpReceiver's MediaStreamTrack but is implemented to track the most recent RECEIVED packet.

Video, Audio

Bug

10838

Add reporting of decoding_codec_plc events

Audio

Bug

10928

Allow configuration of playout audio buffer

Audio

Bug

9295

`v8_win64_unwinding_info` makes process freeze on abort


Bug

10907

Reuse the AEC3 highpass filter inside APM

Audio

Bug

8671

code deserialized from code cache may not emit code create events


Bug

968161

peer-reflexive candidates can leak ip through getStats

Blink>WebRTC>PeerConnection

Bug

10843

Request key frame if packet buffer is cleared

Video

Bug

993878

RTCIceTransport returns unsanitized candidate pairs in getSelectedCandidatePair()

Blink>WebRTC>Network

Bug

10848

CallStats unittest doesn't call OnRttUpdate on the right thread.


Bug

10877

WebRtcSpl_FilterAR is updating the wrong state vector.


Bug

988542

neteq_signal_fuzzer: Timeout in neteq_signal_fuzzer

Blink>WebRTC>Audio

Bug

10880

VideoSendStreamTest.RespectsMinTransmitBitrate(AfterContentSwitch) is flaky on Mac64 Debug

Video

Bug

7285

VideoSendStreamTest.DoesUtilizeUlpfecForVp9WithNackEnabled is flaky

Video

Bug

989856

rtp_frame_reference_finder_fuzzer: Fatal-signal in rtc::webrtc_checks_impl::FatalLog

Blink>WebRTC>Video

Bug

10633

Let PacedSender own that packets it will send

Network>RTP

Bug

10846

Debugging failed checks on Windows doesn't work well due to use of abort()


Bug

10679

Unify audio and video stats stacks: Decide on getters or callbacks

Stats

Bug

10847

Call::GetStats() called from incorrect threads in various tests


Bug

10206

Move parts of RTC event log to api/

Cleanup

Bug

10842

DCHECK failure when PeerConnection has two pending DataChannels and GetStats is called

PeerConnection

Bug

10868

Adding remote candidates are not correctly reported for non-trickled sessions


Bug

6594

Broken float<->int sample conversion functions

Audio

Bug

9381

Green border line is displayed on left side of the screen during video call

Video, HardwareCodec

Bug

10886

Disable audio/video sync_groups if min_playout_delay == 0

Video

Bug

10872

VideoReceiveStreamTestWithFakeDecoder.RenderedFrameUpdatesGetSources is flaky on iOS

Video

Bug

952910

See about Brio 4K camera improvements in upstream / v4.19

Blink>GetUserMedia>Webcam

Bug

978885

webrtc::CroppingWindowCapturerWin wrongly detects occlusion

Blink>GetUserMedia>Desktop, Internals>WebRTC

Bug

964463

Use per-frame task runner in HtmlVideoElementCapturerSource::StartCapture

Blink>GetUserMedia

Bug

991954

DesktopCapture: Z-ordering of contents border is incorrect on Windows.

Blink>GetUserMedia>Desktop


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