commit | 8313a6fa8f78d18e0f5e2bff31086da01d0db4bf | [log] [tgz] |
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author | brandtr <brandtr@webrtc.org> | Fri Jan 13 07:41:19 2017 -0800 |
committer | Commit bot <commit-bot@chromium.org> | Fri Jan 13 15:41:19 2017 +0000 |
tree | 4098057a7aacf8d03c0a41a6207883719a718270 | |
parent | 44b3ef65ed562aec042c0f1d29a983207228e40e [diff] |
Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config. That object will be used when we enable RTCP reporting from FlexfecReceiveStream. Other related changes: - Stop using FlexfecConfig (from config.h) at receive side in WebRtcVideoEngine2. - Add a IsCompleteAndEnabled() method to FlexfecReceiveStream::Config, to be used in WebRtcVideoEngine2. - Centralize the construction of the FlexfecReceiveStream::Config in unit tests. This will make future additions to the unit tests cleaner. - Simplify setup for receiving FlexFEC in VideoQualityTest. BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2589713003 Cr-Commit-Position: refs/heads/master@{#16059}
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.