The BaseChannel can set the transport directly without depending on TransportController. When initializing the network of the BaseChannel, the ChannelManager will create TransportChannels with the TransportController. When enabling bundling, WebRtcSession will get or create TransportChannels with the TransportController. When a TransportChannel of the BaseChannel needs to be destroyed, it will fire a signal to notify the WebRtcSession. BUG=none. Review-Url: https://codereview.webrtc.org/2614263002 Cr-Commit-Position: refs/heads/master@{#16043}
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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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