| commit | 836fee1e1ad07378c0e48f61ddd7f2f0fdc02cf3 | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Fri Feb 08 15:08:10 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Feb 08 19:33:17 2019 |
| tree | a735247abb508d5716937d6b13c97372fa3043c5 | |
| parent | f6adac87b4b98d33649fb31f7c79c92c375b3bf5 [diff] |
Calculate next process time in simulated network. Currently there's an implicit requirement that users of SimulatedNetwork should call it repeatedly, even if the return value of NextDeliveryTimeUs is unset. With this change, it will indicate that there might be a delivery in 5 ms at any time there are packets in queue. Which results in unchanged behavior compared to current usage but allows new users to expect robust behavior. Bug: webrtc:9510 Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069 Reviewed-on: https://webrtc-review.googlesource.com/c/120402 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26617}
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.