commit | 7307952a5bf63311e5f9a2a90089a06177e42716 | [log] [tgz] |
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author | Stefan Holmer <stefan@webrtc.org> | Thu Jan 14 13:15:45 2016 +0100 |
committer | Stefan Holmer <stefan@webrtc.org> | Thu Jan 14 12:15:56 2016 +0000 |
tree | e7eb779bbd533f2c4d203ba21eaa105ac931f88e | |
parent | 268493a96b93d6a11a595b3272c5a4cd7a1fdc47 [diff] |
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it. BUG=4173 R=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1577873003 . Cr-Commit-Position: refs/heads/master@{#11247}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.