| commit | b82de300800f82b88a3dda2174a7e8da84e191a7 | [log] [tgz] |
|---|---|---|
| author | Henrik Lundin <henrik.lundin@webrtc.org> | Fri Oct 20 10:38:56 2017 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Oct 20 09:05:54 2017 +0000 |
| tree | 036cdd359b0d5db59d7f59bb83200fbcd719b6da | |
| parent | 6f38d25f1137c7e751be06f23e5bbbbe2497e88a [diff] |
Add new neteq_signal_fuzzer NetEq has been fuzzed by neteq_rtp_fuzzer for some time. That fuzzer hammers the RTP data, but leaves much of the other data alone. This new fuzzer instead alters the encoded audio, packet arrival timing, clock drift, and packet losses. Bug: webrtc:8421 Change-Id: Ie25b77590a66a7451f32a73c6b5b570944244027 Reviewed-on: https://webrtc-review.googlesource.com/13860 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20368}
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.