commit | 3b80aace615be848a6e9dd47f60ab449b711f16e | [log] [tgz] |
---|---|---|
author | Steve Anton <steveanton@webrtc.org> | Thu Oct 19 17:17:12 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Oct 19 18:01:52 2017 |
tree | 240ab0cce5ef5af2104c89f47193cfe75f00f137 | |
parent | dc9ca9329b920dc3e5a0e9551582cfc22c0fa0ce [diff] |
Fix flaky memory leak in RemoteAudioSource Bug: webrtc:8405 Change-Id: Ie7c89214323678c6ea34e344bb1a5a33ad46b3f0 Reviewed-on: https://webrtc-review.googlesource.com/13401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20362}
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.