| commit | e7e9292fe8b5872500854db695ffa44cb17785c7 | [log] [tgz] |
|---|---|---|
| author | Hanna Silen <silen@webrtc.org> | Thu Jul 08 15:26:31 2021 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Sat Jul 10 13:13:46 2021 |
| tree | 29bc29a3f8b35f16755f46b624068c882f300b5f | |
| parent | a88d5f6733bd42b49da0d977c5f195b587994969 [diff] |
Analog AGC: Add clipping rate metrics Add a histogram WebRTC.Audio.Agc.InputClippingRate and logging of max clipping rate in AgcManagerDirect. Bug: webrtc:12774 Change-Id: I4a72119b65ad032fc50672e2a8fb4a4d55e1ff24 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225264 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34450}
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.