Philipp Hancke 3719a0c4e8 stats: use decoded framerate for inbound-rtp framesPerSecond
instead of the framerate received on the network. This is specified in
  https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framespersecond

BUG=webrtc:13765

Change-Id: I9a0a89d29de49ac5257254deae9b7e5212e09363
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267409
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37422}
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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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