Philipp Hancke 9572b2fa58 srtp: spanify Protect + Unprotect
Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.

Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.

BUG=webrtc:357776213
No-Iwyu: missing include is a private libsrtp header

Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43601}
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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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