3c5d5824a3
Add a timestamp for last data sent in Connection. Move calling of rtc::TimeMillis() to Connection and remove it from RateTracker::AddSamples. This timestamp will be used to further improve fail over logic. BUG=None Change-Id: I4cbc7693a0e081277590b9cb13264dc2a998202e No-Try: True Change-Id: I4cbc7693a0e081277590b9cb13264dc2a998202e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197421 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32831}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs