| commit | 03d9e52998a6c9256a052b1c50360795d3cc1688 | [log] [tgz] |
|---|---|---|
| author | Erik Språng <sprang@webrtc.org> | Mon May 25 10:58:03 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon May 25 12:14:44 2020 |
| tree | 56ae15da5171aea6c0dd45adad78963d9d147ba3 | |
| parent | c49e9c253f53d7c01ce727ab84b4b321ae745669 [diff] |
Replaces ring buffer in RateStatistics with deque. Since RateStatistics is in practice always used with increasing timestamps, and is often sparesely populated, replace the pre-allocated ring buffer with a simple deque where each element tracks which time it represents. Bug: webrtc:11600 Change-Id: I866d7cfa607228c35452f0f19575825d2e694f75 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175906 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31344}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.