aa6fbc156e
* This replaces the video stream methods for forcing adaptation with a mock resource that triggers overuse. * Resources can now be injected to the Module using the AddResource function. * Resources now have tests for adding and removing callbacks. * Quality/EncoderUse% resources are tracked in the Resource list of the adaptation module. * The adaptation module ties all resources to a reason to keep stats working as expected. BUG=webrtc:11377 Change-Id: I1f5902f7416dc41b4915c0072e6f0da2bb3bb2b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168948 Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30610}
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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct