This reverts commit 9572b2fa5850da6d319b9efb5ee36290e2895f7f. Reason for revert: Breaks internal tests Original change's description: > 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} Bug: webrtc:357776213 Change-Id: I5c36ecc2fd9ab672f61cd6b15398452cbd5e98a8 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372200 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43608}
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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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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