| commit | ac937d03b0de60f38fd4b6fad4b5c175a111303e | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Thu Jun 11 12:01:16 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Jun 11 13:56:11 2020 |
| tree | e93c6d8ebaa1fc89ff5e416b36058deef5825574 | |
| parent | 2899b3bc3da018822ff34196b706d69c808fdb93 [diff] |
Fix for potential infinite loop in TCP traffic simulator. For stream sizes that were not multiple of 4, we could end up causing a size_t wraparound which resulted in an infinite loop. Bug: webrtc:9510 Change-Id: Ie3fe5345e1477efa6a4ec338bd9f9b00225e688e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177005 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31503}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.