... As opposed to DtlsTransportInternal. The code is suboptimal right now, storing two pointers to the different interfaces. This will all be cleaned up when we have an "RtpTransport" abstraction that BaseChannel can use. This CL also cleans up the "fake transport" classes a bit, and gives them their own header files. BUG=None Review-Url: https://codereview.webrtc.org/2648233003 Cr-Commit-Position: refs/heads/master@{#16258}
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
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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