commit | 3c918b1af80e93c636233e88ac1ecba016ec30f9 | [log] [tgz] |
---|---|---|
author | Gustaf Ullberg <gustaf@webrtc.org> | Fri Oct 11 11:14:44 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Oct 11 11:57:36 2019 |
tree | 43c799cac5103678665c6f0d9b9acbc43b001c60 | |
parent | 51bf200294622a45444b68ad1498a41f8a860df3 [diff] |
Fix bypass of unnecessary resampling This change fixes an issue with bypass of unnecessary resampling when using ProcessStream(AudioFrame*). Bug: b/130016532 Change-Id: I887f05d55aaa47f21164ba237cf83d0be33a1fd5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156540 Reviewed-by: Per Ã…hgren <peah@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29446}
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.