| commit | ac9365ed64d162631565e40fa5f658d24ccf4c15 | [log] [tgz] |
|---|---|---|
| author | Artem Titov <titovartem@webrtc.org> | Wed Mar 28 13:21:54 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Mar 28 15:05:26 2018 |
| tree | 46047bc0ac7b645050f25a794ec74245949558ce | |
| parent | 19c242d119f727bde362caf0dbd1adaf98bd0096 [diff] |
Set safe values to prevent possible sigsegv while using AudioTransport, add doc Bug: webrtc:8946 Change-Id: Ica066a05905894fba6ba24e45af46b0d5951b5d5 Reviewed-on: https://webrtc-review.googlesource.com/65040 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22652}
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.