commit | cd0eedb2483b8a1cb07c953f0c06aeec8ce6f144 | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Thu Oct 10 11:52:26 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Oct 10 13:20:50 2019 |
tree | ad5ff96a6e634d9ed5468e69f76cb6a5d1806fc4 | |
parent | 9afdddfed0fa76b102d295db1a893522f1340c6d [diff] |
Don't allocate audio if we have no transport sequence number. Bug: chromium:1002875 Change-Id: I597184e59cf7b5f47b2025d26408069199ada2c2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156305 Reviewed-by: Ali Tofigh <alito@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29432}
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.