Rasmus Brandt f0820ffd88 Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay
* https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbuffertargetdelay
* https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay

Tested: https://jsfiddle.net/pfgzj0yo/17/

Bug: webrtc:14244
Change-Id: I3d949ba63c8339b3881f5d00356559d5789d283d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304404
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40157}
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WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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