| commit | f58ded7cf086700208e50a6b382b46ee70a5f7ad | [log] [tgz] |
|---|---|---|
| author | Tommi <tommi@webrtc.org> | Thu May 30 11:29:11 2024 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu May 30 13:07:32 2024 |
| tree | 3c5f1f80b016b79192c6992c7acad483b4a12341 | |
| parent | a97c292a05aa32fd7334ed0470704f95b911529f [diff] |
Use audio views in Interleave() and Deinterleave() Interleave and Deinterleave now accept two parameters, one for the interleaved buffer and another for the deinterleaved one. The previous versions of the functions still need to exist for test code that uses ChannelBuffer. Bug: chromium:335805780 Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42412}
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.