commit | 80b2806250aa05e94c3974519e5d2809d8478a3b | [log] [tgz] |
---|---|---|
author | Henrik Lundin <henrik.lundin@webrtc.org> | Mon Nov 25 09:21:00 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Nov 25 12:16:30 2019 |
tree | 931ae851e956caec5af2756e49350d943151756a | |
parent | 00cc836fcfa031a16d9c62375d5aa490519c3ac6 [diff] |
Fixing a buffer overflow in Merge::Downsample In the unlikely event that the decoded audio is really short, the downsampling would read outside of the decoded audio vector. This CL fixes that, and adds a unit test that verifies the fix (when running with ASan). Bug: chromium:1016506 Change-Id: Ifb8071ce0550111cd66e7f7c1bed7f17b33f93c5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160304 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29898}
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.