This is part of the project that makes RTC rendering more smooth. We've already finished the developement of the frame selection algorithm in WebMediaPlayerMS, where we managed a frame pool, and based on the vsync interval, we actively select the best frame to render in order to maximize the rendering smoothness. Thus the frame timeline control in IncomingVideoStream is no longer needed, because with sophisticated frame selection algorithm in WebMediaPlayerMS, the time control in IncomingVideoStream will do nothing but add some extra delay. BUG=514873 Review URL: https://codereview.webrtc.org/1419673014 Cr-Commit-Position: refs/heads/master@{#10781}
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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