| commit | c59a576c860a2f9cbbeeaac6fb18ced2c5a59072 | [log] [tgz] |
|---|---|---|
| author | Per Åhgren <peah@webrtc.org> | Mon Dec 11 21:34:19 2017 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Dec 11 21:09:56 2017 +0000 |
| tree | af9ac1211085585df3d9af71543aa9e9e741b2bb | |
| parent | efc5fbd8e0d272064364d67b0f98a0b3d087630f [diff] |
Corrections of the render buffering scheme in AEC3 to ensure causality This CL modifies the refactored render buffering scheme in AEC3 so that: -A non-causal state can never occur which means that situations with nonrecoverable echo should not occur. -For a stable audio pipeline with a predefined API call jitter, render overruns and underruns can never occur. Bug: webrtc:8629,chromium:793305 Change-Id: I06ba1c368f92db95274090b08475dd02dbb85145 Reviewed-on: https://webrtc-review.googlesource.com/29861 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21215}
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.