| commit | 15389c034d45eeaa81b4390a4959ddbc18e47540 | [log] [tgz] |
|---|---|---|
| author | nisse <nisse@webrtc.org> | Tue Jan 24 02:36:58 2017 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Jan 24 10:36:58 2017 +0000 |
| tree | 49384e3b422e60b887871d3b68f662efe5fca1e3 | |
| parent | 568c9e72d149662dab92a671bd3139e4ae7f167a [diff] |
Drop pacer and retransmission_rate_limiter from RtpStreamReceiver constructor. They were passed on via RtpRtcp::Configuration, but unused for a receive only RtpRtcp module. BUG=webrtc:6847 Review-Url: https://codereview.webrtc.org/2639423007 Cr-Commit-Position: refs/heads/master@{#16234}
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.