ed23be91b84b26b6a03b799e51aec833d5ffb052

This should be a separate utility class, rather than internal thing in ReceiveStatisticsProxy. Bug: webrtc:8347 Change-Id: I9c8238e625999dba5776d6038c819732d07e9656 Reviewed-on: https://webrtc-review.googlesource.com/7609 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20267}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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.
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
Description
Languages
C++
88.6%
C
3.3%
Java
3%
Objective-C++
1.9%
Python
1.9%
Other
1%