| commit | 918b5547897a70ae13053fb0672c5f1a48dd714a | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@webrtc.org> | Mon Sep 19 13:44:09 2016 |
| committer | henrika <henrika@webrtc.org> | Mon Sep 19 13:44:22 2016 |
| tree | 513f1c43130f8493712dbf5fff3d7e64595b2c57 | |
| parent | 27e177c8d81b04d56b999d93e3c4fc77217835aa [diff] |
Adds support for OpenSL ES based audio capture on Android. NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that OpenSL ES is not accidentally activated in existing clients. There are still some unresolved issues to sort out before it can be utilized. Enables possibility to use OpenSL ES based audio in both directions for WebRTC. All unit tests and demo clients have been tested with the new implementation but the new support is behind a flag (see above). More testing is needed before it can be used in the field and additional support for hardware effects is still missing. BUG=webrtc:5925 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/2119633004 . Cr-Commit-Position: refs/heads/master@{#14290}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.