This is a reland of 4770fd935ac92400487bddd3b755753572e6d692 Original change's description: > Move JsepTransport from p2p/base to pc/. > > The JsepTransport class is moved to pc/ and the utility methods and > enums are moved to where they are used. > > With JsepTransport moved to pc/, JsepTransport can depend on objects in > pc/ including RtpTranport, SrtpTransport etc. > > Forked from https://webrtc-review.googlesource.com/c/src/+/31762/7 > > Bug: webrtc:8636 > Change-Id: I4e8569fe3012946e87deb280f6139f0fd98de34d > Reviewed-on: https://webrtc-review.googlesource.com/33701 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21333} Bug: webrtc:8636 Change-Id: Ibce42be898b96dd8e0266b595611d2ffc86581a8 Reviewed-on: https://webrtc-review.googlesource.com/34586 Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21371}
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.
Authoritative list of directories that contain the native API header files.
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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