| commit | 728012e49fc9f0c4ecda92781abbeda7cfc8a03f | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Fri Feb 19 10:38:32 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Feb 19 10:38:37 2016 |
| tree | 2d9262116554f172f9e27ce6655a385127cda901 | |
| parent | ecdeb4cb94d95a7b0819e0ff5e62f272a799f59c [diff] |
Changed the semantics of Buffer::Clear to not alter the capacity Also added a test for Clear to ensure this invariant holds. With this change, it is easy to empty a Buffer and reuse its storage. Further down the line, code filling data into a Buffer could be written to just append to it, with the caller determining if the Buffer should first be cleared or not. There is currently only one use of Buffer::Clear (in AudioEncoderCopyRed::Reset()) and it should benefit from the change, by not requiring a reallocation after Reset. Review URL: https://codereview.webrtc.org/1707693002 Cr-Commit-Position: refs/heads/master@{#11680}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.