commit | a837030f8fdb541ab16bc75fb06477316e85fe2f | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Mon Sep 02 15:16:49 2019 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Sep 02 14:04:47 2019 +0000 |
tree | 1c09ebccb860e95512965a02a6d3f35a0a5a47be | |
parent | d112c758015c095895489d21bed198352f689515 [diff] |
Split out RtpSource from libjingle_peerconnection_api And moved declaration into a new api directory, as api/transport/rtp/rtp_source.h. Bug: webrtc:8733 Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29039}
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.