Harald Alvestrand a52fc6f940 Increase loss tolerance for RenegotiateManyVideo test.
Also switch to "LT(meaured, limit)" format as this is easier to read.

There have been bot runs that exceeded the old limit.

Bug: webrtc:13354
Change-Id: I1c19c98e1c1777177e7dc14ae4679765c1c40550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237342
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35317}
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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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