| commit | b7261fd3ae7c31269b8c282e4c4c55dd3946db45 | [log] [tgz] |
|---|---|---|
| author | kwiberg <kwiberg@webrtc.org> | Wed Feb 24 09:34:29 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Feb 24 09:34:33 2016 |
| tree | e5f7ce29a85a2ac60730fbf97ea1c234bbdffe1e | |
| parent | b01c7816a8c4ef34102ead1a3eca4f389e7a8f43 [diff] |
iSAC float: Check for end of input buffer while decoding Previously, we relied on the encoded stream to come to an end before the end of the buffer. This is a bad idea, since it is possible to craft a stream that fills the buffer while decoding to less than the expected amount of data; without the new checks introduced here, this causes the decoder to read past the end of the input buffer. BUG=chromium:582471, chromium:587852 Review URL: https://codereview.webrtc.org/1721593004 Cr-Commit-Position: refs/heads/master@{#11734}
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.