commit | cd277b84dad534145445d3586145ff25aa5637ee | [log] [tgz] |
---|---|---|
author | Gustaf Ullberg <gustaf@webrtc.org> | Mon Aug 19 10:15:39 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Aug 19 11:05:21 2019 |
tree | cdc3ffc679011f0d4a4e8a653ab4ef7779e23a57 | |
parent | 17f9ee5358274ff79e6cfafa64c6ed3a073335d9 [diff] |
AEC3: Fix computation of audio buffer delay This change fixes a bug where the initial delay could be set incorrectly. Bug: webrtc:10896 Change-Id: I66b2234b69c46639488f4561e973384001230861 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149820 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Ã…hgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28894}
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.