a4ac4786a8
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc