News｜WebRTC M93 update

WebRTC M93 has been released in the Chrome beta, including 12 new features and more than 40 bug fixes, functional enhancements, stability and performance improvements.

. We will regularly translate WebRTC related content to help developers get the latest information and stay at the forefront of the industry.

01. Focus on

Change the audio checkbox in the screen capture selector

We now have separate checkboxes instead of the single audio checkbox previously shared between [Entire Screen] and [Chrome Tab]. [Chrome Tab] is now selected by default; [Entire Screen] remains unselected by default.

Issue tracking link:
https://bugs.chromium.org/p/chromium/issues/detail?id=1220450

Improved capture function area

Add a button to allow quick navigation between capturing/captured tabs. For example, when sharing documents in a meeting, the document tab will allow people to quickly switch to the meeting tab and vice versa.

Issue tracking link:
https://bugs.chromium.org/p/chromium/issues/detail?id=1224363

02. PSAs

Title 1:

Chromium will prohibit camera capture after the screen is locked

:

For privacy reasons, camera capture will be disabled 15 seconds after the screen is locked. Most developers and users should not see any changes brought about by this policy. When the camera capture is active, Chromium will keep the screen awake, and the screen lock during the capture process is very rare. However, some systems may be set at the operating system level to prevent screen wakeup locks. For more details, see PSA.

Title 2:

If MID and BUNDLE are negotiated, the demultiplexing function by load type will be disabled.

:

Demultiplexing by payload type is to support traditional endpoints that do not use MID RTP header extensions in data packets and do not send SSRC signals in SDP. Starting from M93, if the MID header extension is negotiated, demultiplexing based on PT (Payload Type) will be disabled.

03. Function and problem repair

No.1
Type: Bug
Issue ID: 1056230
Description: Add captureTimestamp and senderCaptureTimeOffset to RTCRtpContributingSource
Component: Blink>Media

No.2
Type: Feature
Question ID: 1202526
Description: Lost the mouse cursor when sharing the screen in a Linux Wayland session
Component: Internals>Media>ScreenCapture

No.3
Type: Bug
Issue ID: 1203206
Description: WebRTC / LibvpxVp8Encoder: If you switch from I420A format to I420 format during downsampling, frame loss will occur
Components: Blink>WebRTC>Video

No.4
Type: Bug
Question ID: 1212630
Description: Make WebRTC and Chromium consistent with the libyuv scaling filter
Components: Blink>WebRTC>Video

No.5
Type: Bug
Issue ID: 989932
Description: Code base vs Wimplicit-int-float-conversion compilation option
Component: Tools>LLVM

No.6
Type: Bug
Issue ID: 10395
Description: WebRTC's fuzzer

No.7
Type: Feature
Issue ID: 10739
Description: Add support for abs-capture-time header extension
Component: Network>RTP

No.8
Type: Bug
Issue ID: 11581
Description: Frequent polling of several timers in ModuleRtpRtcpImpl
Components: Internals, Perf

No.9
Type: Bug
Issue ID: 11713
Description: RTP header extension encryption is broken
Component: Network>RTP

No.10
Type: Bug
Issue ID: 12194
Description: The value range of dynamic rtp load type has been used up
Component: PeerConnection

No.11
Type: Bug
Issue ID: 12295
Description: When the 96-127 payload type is exhausted, fall back to the rtp payload type 35-63
Component: Network>RTP

No.12
Type: Bug
Issue ID: 12462
Description: Create too many decoders
Component: Video

No.13
Type: Bug
Issue ID: 12510
Description: Add nackCount statistics for outbound audio
Component: Stats

No.14
Type: Bug
Issue ID: 12551
Description: Add DTLSTransport concept document
Component: Documentation

No.15
Type: Feature
Issue ID: 12575
Description: Add --start_timestamp and --stop_timestamp parameters to video_replay
Component: Tools

No.16
Type: Bug
Issue ID: 12630
Description: VP8: no longer zoom the buffer for the inactive layer
Component: Video

No.17
Type: Feature
Issue ID: 12614
Description: dcSCTP library
Component: DataChannel

No.18
Type: Bug
Issue ID: 12713
Description: NACK: Erase data packets that were not received because the data packet ssrc is incorrect
Component: Network>RTP

No.19
Type: Bug
Issue ID: 12770
Description: Fix the echo return statistics in the modern statistics parser
Component: Stats

No.20
Type: Bug
Issue ID: 12773
Description: MediaStreamTrack::enabled() is accessed from a worker thread
Component: PeerConnection

No.21
Type: Feature
Issue ID: 12787
Description: Equipped with Chrome tracking entry point for WebRTC proxy
Component: Internals

No.22
Type: Bug
Issue ID: 12788
Description: Due to the wrong bit rate allocation, when the "active" state is switched, a simulcast encoder is stuck
Component: Video

No.23
Type: Feature
Issue ID: 12793
Description: Implement a round-robin scheduler for the dcSCTP library
Component: DataChannel

No.24
Type: Feature
Issue ID: 12794
Description: Support bufferedAmountLowThreshold in dcSCTP library
Component: DataChannel

No.25
Type: Bug
Issue ID: 12798
Description: DCHECK is invalid in rtc_stats_report.cc,
RTCRemoteInboundRtpAudioStream already exists in this statistical report
Component: Stats

No.26
Type: Bug
Issue ID: 12810
Description: The code comments in frame_dropper.h appear to be incorrect
Components: BWE, Video

No.27
Type: Bug
Issue ID: 12812
Description: dcSCTP may send FORWARD-TSN if the currently sent message is not completed

No.28
Type: Bug
Issue ID: 12814
Description: Disable demultiplexing based on payload type as much as possible
Component: PeerConnection

No.29
Type: Bug
Issue ID: 12815
Description: Add a small cooling time for the creation of no signal ssrc stream

No.30
Type: Feature
Issue ID: 12829
Description: Allow the encoder to specify resolution alignment properties
Component: Video

No.31
Type: Bug
Issue ID: 12832
Description: Due to the round-robin scheduler, dcSCTP may interleave messages

No.32
Type: Bug
Issue ID: 12837
Description: PC sends out candidates when renegotiating
Component: PeerConnection

No.33
Type: Bug
Issue ID: 12839
Description: Simplify the method of determining whether a data packet has been received.

No.34
Type: Bug
Issue ID: 12841
Description: Add conceptual document for RTC event log
Component: Documentation

No.35
Type: Bug
Issue ID: 12850
Description: PeerConnectionObserverJni lacks OnRemoveTrack event

No.36
Type: Bug
Issue ID: 12857
describe:
VideoStreamEncoder::EncodeVideoFrame is called when encoder_ is empty

No.37
Type: Bug
Issue ID: 12866
Description: Hookup adapter: no longer register invalid encoding completion callback.
Component: Video

No.38
Type: Bug
Issue ID: 12867
Description: When the frame rate is not limited, you can set the number of Fps adaptive degradation frames
Component: Video

No.39
Type: Bug
Issue ID: 12868
describe:
Reduce the frequency of statistical reports in ChannelReceive::GetAudioFrameWithInfo
Component: Audio

No.40
Type: Bug
Issue ID: 12896
Description: Record the supported compilers and platforms
Component: Documentation

No.41
Type: Bug
Issue ID: 12910
Description: jitterBufferDelay and jitterBufferDelay of RTCInboundRtpStreamStats
jitterBufferEmittedCount is not defined for audio (but they are defined for video)
Component: Stats

No.42
Type: Bug
Issue ID: 12913
describe:
PlatformThreadTest.MovesHandles is not reliable
Components: Cleanup

No.43
Type: Bug
Issue ID: 12924
Description: VP9 uncompressed header parser may not be able to do what you think it is doing

No.44
Type: Bug
Issue ID: 12925
Description: implemented for audio
RTCInboundRTPStreamStats.nack_count
Component: Stats

No.45
Type: Bug
Issue ID: 12941
Description: The new version of clang causes the bit accuracy test to fail
Component: Audio

No.46
Type: Feature
Issue ID: 12952
Description: dcSCTP resets all streams when only one stream is reset
Component: DataChannel

No.47
Type: Feature
Issue ID: 6458
Description: Use codec rate when generating RTCP for audio
Component: Audio

No.48
Type: Bug
Issue ID: 6779
Description: Stop using assertions
Component: Audio

No.49
Type: Feature
Issue ID: 7952
Description: Make the internal software video codec injectable and optional
Component: PeerConnection

No.50
Type: Bug
Issue ID: 9267
Description: Update the threshold of VideoCodecTests libvpx on Android/iOS
Component: Video

No.51
Type: Feature
Issue ID: 1220009
Description: Change the audio checkbox of the media selector
Component: Blink>GetDisplayMedia

No.52
Type: Feature
Issue ID: 1214485
Description: Improved capture function area
Component: Blink>GetDisplayMedia

Original link:
https://groups.google.com/g/discuss-webrtc/c/ws0_MYHIBOw/m/HZGn07uIAwAJ

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