WebRTC M90 is currently available in Chrome beta, with 2 new features and more than 29 bug fixes, as well as enhancements, stability and performance.
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01. Public service announcements
Plan B The SDP is disabled
Note: Plan B SDP is deprecated and will be removed completely in the future.
See time line: groups.google.com/g/discuss-w
Function of 02.
MediaStreamTrack Insertable Streams source trial
This API is an extension of the MediaStream and WebCodecs APIS, allowing applications to:
- Access raw data in MediaStreamTrack;
- Define new custom MediaStreamTracks.
These two functions can be combined, for example, to create media effects (e.g., “funny Hats “).
The API relies on WebCodecs Raw Media interfaces and WHATWG Streams APIS. This feature is part of the WebCodecs source trial.
GetCurrentBrowsingContextMedia source trial version
This is a new experimental API for retrieving current Tab content that is currently under development. Implementation can be used as a trial version for the first time, for more information, see: docs.google.com/document/d/…
03. Function and problem repair
Can be found at: bugs.chromium.org/p/webrtc/is… Enter the problem ID to query the bug details.
No.1
- Type: Bug
- Problem ID: 1138888
- Description: WebRTC low latency renderer
- Components: Blink > WebRTC > Video
No.2
- Type: Bug
- Problem ID: 1155477
- Description: AEC3: Linear filter gradually diverges over a long call
- Components: Blink > WebRTC > Audio
No.3
- Type: Bug
- Problem ID: 1170699
- Description: WebRTC AV1 encoding initialization failed
- Components: Blink > WebRTC
No.4
- Type: Feature
- Problem ID: 516700
- WebRTC Chromium clock poor
- Components: Blink > WebRTC
No.5
- Type: Bug
- Problem ID: 10675
- Description: Support recording raw RTP in Text2pCAP format
- Components: Network > RTP
No.6
- Type: Bug
- Problem ID: 11031
- Description: Retransmission may fail after MID negotiation is complete [Unified Plan]
- Components: Network > RTP
No.7
- Type: Feature
- Problem ID: 11989
- Description: Provides VoipStatistics interface for VoIP APIs for media statistics
- Components: Audio
No.8
- Type: Bug
- Problem ID: 12265
- Description: AEC3: Linear filter gradually diverges over a long call
- Components: Audio
No.9
- Type: Bug
- Problem ID: 12279
- Description: (network.cc:908): 10051 connection failures occur every 2 seconds
- Components: PeerConnection, Tools
No.10
- Type: Bug
- Problem ID: 12380
- Description: When receiving Opus stream, the comfort noise of DTX packet changes the energy value suddenly every time it refreshes
- Components: Audio
No.11
- Type: Bug
- Problem ID: 12383
- Description: Collect statistics about bundle usage
No.12
- Type: Bug
- Problem ID: 12384
- Description: Registry-Key- mmDevices-audio Handles is added for every Audio call on a Windows client
- Components: Audio
No.13
- Type: Bug
- Problem ID: 12398
- Description: AV1 encoder seG error when SVC is used and width/height is odd
- Components: Video
No.14
- Type: Bug
- Problem ID: 12407
- Description: SEA creates and initializes the encoder for the still layer
- Components: Video
No.15
- Type: Bug
- Problem ID: 12426
- Description: Multithreaded access to JsepTransport::jsep_transports_by_name_ is not protected
No.16
- Type: Bug
- Problem ID: 12427
- PeerConnetion Orchestrates JsepTransportController events between different threads
No.17
- Type: Bug
- Problem ID: 12430
- Description: RtpBitrateConfigurator TSAN report
No.18
- Type: Bug
- Problem ID: 12431
- Description: RTC event log visualization does not work in Python3
- Components: the Tools
No.19
- Type: Feature
- Problem ID: 12432
- Description: Visualize RTCP BYE messages in RTC event logs
- Components: the Tools
No.20
- Type: Bug
- Problem ID: 12439
- Description: If the system time goes back, traditional getStats will stop working
- Components: Stats
No.21
- Type: Bug
- Problem ID: 12445
- Description: JsepTransportController: : mid_to_transport_ for protection
No.22
- Type: Bug
- Problem ID: 12448
- Description: ULPFEC: Abnormal arrival order and too long arrival delay
- Components: Video
No.23
- Type: Bug
- Problem ID: 12455
- Webrtc: : AudioSendStream: : Config: : ToString () call fails in M90 version
- Components: Audio
No.24
- Type: Feature
- Problem ID: 12459
- Description: Allows clipping resolution when limiting the maximum number of layers
- Components: Video
No.25
- Type: Bug
- Problem ID: 12487
- Description: Implement jitter statistics of video RTP streams
The original link: groups.google.com/g/discuss-w…
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