Follow-up of https://webrtc-review.googlesource.com/c/src/+/287221, instead of asking GTest for the test suite and the test name, let's propagate the test case passed by the user of PCLF. Bug: b/237982523, webrtc:14757 Change-Id: Ia2a6ed4781f8c53c25b0006b8c7483e08ecead26 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287126 Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38867}
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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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