This allow to remove the testonly from the rtc_event_log_visualizer binary and the implicit dependency on the path of the default conversational speech file. The binary size of event_log_visualizer passes from 2.1 MB to 4.0 MB. Bug: b/237526033 Change-Id: I71cf647f039f26f30c792c49c752cff5c5b329a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267663 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37453}
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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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