| commit | 7c19a706b0a1501900c80a50b13ca7ab1d51b280 | [log] [tgz] |
|---|---|---|
| author | Alessio Bazzica <alessiob@webrtc.org> | Thu Nov 07 13:22:00 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Nov 07 13:33:09 2019 +0000 |
| tree | 4f9cc9b2d73d53ef182177f63d491b83da73defa | |
| parent | 1cd6fbc2a43e9986c2be2e3d93d63c98ef9193db [diff] |
Audio Processing Module: add play-out audio device runtime information Add a runtime setting that notifies play-out audio device changes. The payload is a pair indicating a device id and its maximum play-out volume. kPlayoutVolumeChange is now forwarded not only to capture, but also render (required by render_pre_processor). Bug: webrtc:10608 Change-Id: I8997c207422c1dcd1d53775397d6290939ef3db8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159002 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29725}
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.