commit | 63c38e21dae66e8c708d5a35d3d6a9f0049d34c8 | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Tue Aug 06 15:17:43 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Aug 06 16:26:22 2019 |
tree | f18ba2b849dd40bf3a638b3ba336a2c2509ceba8 | |
parent | 7cbee84610a8d4f2bbc86c55d9ee02d25be19f72 [diff] |
Fix for incorrect transport sequence number config for audio in scenario tests. Bug: webrtc:9883 Change-Id: Iafe1db4b4dbfa81c7901640114057806821de760 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148280 Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28778}
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.