c1d5fda22c
This CL increases the test coverage for webrtc::SimualtedNetwork, adds some more comments to the class and the interface it implements and simplify the logic around capacity and delay management in the simulated network. More CLs will follow to continue the refactoring but this is the ground work to make this more modular in the future. Bug: webrtc:14525, b/243202138 Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38388}
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