commit | b24c00f02d90261a97fac77c393304f2c170bcc1 | [log] [tgz] |
---|---|---|
author | Sam Zackrisson <saza@webrtc.org> | Mon Nov 26 16:18:25 2018 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Nov 26 15:52:14 2018 +0000 |
tree | ff015be683723bd141e3f6316eb945b3a5c9bef0 | |
parent | 2918d4e309ed61b51d41c9f3f7a016b6ce5be703 [diff] |
Add AudioProcessingCaptureStats and a level estimator replacement This adds an interface for accessing stats on the capture stream, and adds a level estimator to report one of the stats. Bug: webrtc:9947 Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb Reviewed-on: https://webrtc-review.googlesource.com/c/109587 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Per Ã…hgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25786}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.