commit | 6f3103f23dae1ed80c98318425d9dba04bd08626 | [log] [tgz] |
---|---|---|
author | Hanna Silen <silen@webrtc.org> | Wed Jun 05 12:52:33 2024 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Jun 11 08:44:10 2024 |
tree | 9389dd37da78c9367c07a0e4f4f8561357daae66 | |
parent | 546d15ae2010d0332a945acc009ad7cd115b347b [diff] |
Add AGC2 input volume controller mode in audioproc_f Bug: webrtc:7494 Change-Id: I454f1fcdfe0eff2440b7fba426f8d950250b6a5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353740 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42459}
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.