This CL adds interfaces for the new video codec factories and wires them up in WebRtcVideoEngine. The default behavior is unmodified however, and the new code is currently unused except for the tests. A follow-up CL will be uploaded for exposing them in the PeerConnectionFactory API: https://codereview.webrtc.org/3004353002/. BUG=webrtc:7925 R=andersc@webrtc.org, stefan@webrtc.org Review-Url: https://codereview.webrtc.org/3007073002 . Cr-Commit-Position: refs/heads/master@{#19828}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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.
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://webrtc.googlesource.com/src
- 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
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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