| commit | 1f39ba1cd916d97e6f203ba9da717ae85aeb4577 | [log] [tgz] |
|---|---|---|
| author | philipel <philipel@webrtc.org> | Wed Sep 21 09:27:47 2016 |
| committer | philipel <philipel@webrtc.org> | Wed Sep 21 09:27:56 2016 |
| tree | adafca389b762ec1491b63f37db18b0f9fea96b2 | |
| parent | 66492210e57ee8efce2ad4d45a8781df1fcaa5e3 [diff] |
Copy payload data when inserting packets into video_coding::PacketBuffer. The payload pointed to by |dataPtr| is volatile and needs to be copied to its own buffer. BUG=webrtc:5514 R=brandtr@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2302763002 . Cr-Commit-Position: refs/heads/master@{#14321}
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.