commit | 2bd54a1bd9c485b009b93b9bc936d525396044a1 | [log] [tgz] |
---|---|---|
author | Mirko Bonadei <mbonadei@webrtc.org> | Wed Feb 13 08:07:55 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 13 12:44:02 2019 |
tree | f8c695e6bea4de9d83bf2b6dc2119d3f0b84515b | |
parent | 6aca0b743e58b7d4e95cc28b2c296c98d4281f48 [diff] |
Ensure TestPeers are destroyed at the end of Run. In order to correctly close audio dump files, TestPeers have to be destroyed after the call is finished. Bug: webrtc:10138 Change-Id: I948e4e1844dfbffd1eef7926a4dd4d7631dbe632 Reviewed-on: https://webrtc-review.googlesource.com/c/122301 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26661}
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.