| commit | b01c7816a8c4ef34102ead1a3eca4f389e7a8f43 | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Wed Feb 24 01:05:56 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Feb 24 09:06:02 2016 +0000 |
| tree | 904ae6abbbd90aafd5ce4239af8033ef2b85c9e2 | |
| parent | 17849fcc8bf22004ab111f61c5e261c39efc24d4 [diff] |
Added functional variants of Buffer::SetData and Buffer::AppendData.
They are invoked with the maximum size of the data to be added, and a
callable that generates that data, like this:
buffer.AppendData(10, [] (rtc::ArrayView<uint8_t> av) {
for (uint8_t i = 0; i != 5; ++i)
av[i] = i;
return 5;
});
The callable returns the number of bytes actually written, and the
final Buffer size will be adjusted accordingly. SetData and AppendData
both return the number of bytes added (i.e. the return value of the
callable).
These versions will be useful when converting AudioEncoder::Encode to use Buffer rather than raw pointers.
Also added a few tests for the new functionality.
Review URL: https://codereview.webrtc.org/1717273002
Cr-Commit-Position: refs/heads/master@{#11733}
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.