with an intermediate step since Chromium depends on the openssl_stream_adapter.h which will move to the new target. BUG=webrtc:339300437 Change-Id: Iea163e0a6e3923ce8a741a2e11e9a2a1e3f3e7a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350887 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#42362}
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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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