| commit | 5dd42fd849062373cab757605824254fc16db6b6 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Mon May 02 23:20:01 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon May 02 23:20:08 2016 |
| tree | c7d1c62ba6efaa9fdb3cc4def38148ea3be58aa9 | |
| parent | acf143128fa0d5c728742f3018e075bcec35ae20 [diff] |
Fixing a segfault that can occur when changing the track of an RtpSender. The reference to the old track needs to be kept alive until SetAudioSend/ SetSource is called, because otherwise it could be deleted while the audio/ video engine is still trying to use the track. BUG=webrtc:5796 Review-Url: https://codereview.webrtc.org/1894283002 Cr-Commit-Position: refs/heads/master@{#12598}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.