commit | 8c51f2e9cde4ca72ae2f84bbbe3638540b84e565 | [log] [tgz] |
---|---|---|
author | Gustaf Ullberg <gustaf@webrtc.org> | Tue Oct 22 13:21:31 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Oct 22 14:27:14 2019 |
tree | ee203bcb0d396af95908595c0c7bf93b4a887b40 | |
parent | e76b3abf610bc4acd386dbc5de9aff00a64823a3 [diff] |
AnalyzeReverseStream with StreamConfig Adding a version of AnalyzeReverseStream with audio parameters described by StreamConfig. This is part of preparations for multichannel APM in Chromium. Bug: webrtc:10913 Change-Id: I7c4650eab8bd7fcdec970a7e4a8fa203f09bed9e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157897 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Ã…hgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29573}
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.