| commit | fb4d66be9726b849f8af61cafb09ed96bb993604 | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Fri Jun 15 14:16:44 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Jun 15 12:39:59 2018 +0000 |
| tree | bd967f1e1ae0cfb3b8492b82272e0f6d71ec5adc | |
| parent | b9b146c9feaf6fd610c3d96b6ed532c447f4bb89 [diff] |
Improves buffer time calculation in network control tester. The previous solution caused packet reordering if the bandwidth changed with large buffers. To avoid this, the buffer time is tracked instead. This means the the bandwidth is applied per packet and can't be retroactively changed for packets already handled. Bug: webrtc:8415 Change-Id: Ib6c97ba9b948220e88c79776aa8d96de289dcfb5 Reviewed-on: https://webrtc-review.googlesource.com/83723 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23629}
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.
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.