commit | 0a6510ddf92c08b29a9762b3318a71536bfd85d5 | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Fri Oct 04 07:31:08 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Oct 07 12:58:55 2019 |
tree | 21d03e84fa1ec3c3f388fc4967cdf51f9c971744 | |
parent | 99a20962483581349f8cd1c37ef7b4cc79dd9ca5 [diff] |
Removes rtp_transport checks in AudioSendStream There's already a DCHECK at construction time ensuring that it's set. Bug: webrtC:9883 Change-Id: I9f41b77273bb859626546ab3534d483d9172ea5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155581 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29393}
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.