| commit | a4ac4786a8ca503ac3cb15dfb0a7f6a62a9bdd52 | [log] [tgz] |
|---|---|---|
| author | kwiberg <kwiberg@webrtc.org> | Fri Apr 29 08:00:22 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Apr 29 15:00:28 2016 +0000 |
| tree | 6d512bc6a1fd1ec12aba4982712808b9a7e97191 | |
| parent | ef8b61e11062295365f11b9942f18a08a8b3ec60 [diff] |
Define rtc::BufferT, like rtc::Buffer but for any trivial type And redefine rtc::Buffer as using Buffer = BufferT<uint8_t>; (In the long run, I'd like to remove the type alias and rename the template to just rtc::Buffer, but that requires all current users of Buffer to start saying Buffer<uint8_t> instead, and since Buffer is used in the API, we can't do that in one step.) The immediate reason for the new template is that we'd like to use BufferT<int16_t> in the AudioDecoder interface. BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/1929903002 Cr-Commit-Position: refs/heads/master@{#12564}
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.