| commit | fe32a76d6001d17c34ab597026fd1e16e5c3b9ea | [log] [tgz] |
|---|---|---|
| author | Henrik Lundin <henrik.lundin@webrtc.org> | Tue Dec 08 11:27:27 2015 +0100 |
| committer | Henrik Lundin <henrik.lundin@webrtc.org> | Tue Dec 08 10:27:34 2015 +0000 |
| tree | e729af7cd14a97a190265dde2070e372d66a1e53 | |
| parent | ffea13c42cc3ebc118b6261be688f88974cc7ab5 [diff] |
Create fuzzer tests for audio decoders This change adds fuzzer tests for iLBC, iSAC fix and float, and Opus. The fuzzer function takes a random input vector and splits it into a number of payloads. The lengths of the payloads is also determined by the random vector. The payloads are decoded with the decoders. BUG=webrtc:5306 R=kjellander@webrtc.org, pbos@webrtc.org Review URL: https://codereview.webrtc.org/1499093002 . Cr-Commit-Position: refs/heads/master@{#10932}
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.