commit | 980ad0cd645aada15927e58c580b8b70b374dacf | [log] [tgz] |
---|---|---|
author | Ali Tofigh <alito@webrtc.org> | Tue Aug 09 07:21:17 2022 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Aug 09 13:32:59 2022 |
tree | a3cb437a02e547ed0ecd71a44bc0df36b509b28b | |
parent | 7f0a7acb0a5d24d2d1471562a5b524645c8212b9 [diff] |
Remove unnecessary overloads of AudioProcessing::CreateAndAttachAecDump() Bug: webrtc:13579 Change-Id: I2e121b5fd30de4ac1813483f00a51184ff861760 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269623 Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37723}
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.