This initial version contains a few tests, testing both lossy, non-lossy and bandwidth limited networks. It uses simulated time, and runs much faster than wall time on release builds, but slower on debug when there is a lot of outstanding data (high throughput) as there are consistency checks on outstanding data. Because of that, some tests had to be disabled in debug build. Bug: webrtc:12943 Change-Id: I9323f2dc99bca4e40558d05a283e99ce7dded7f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225201 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35124}
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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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