8c40d510c8090b393bfc64748cf2b0dfdf3e6b94

This will allow us to enable receive-side RTT without having to recreate all AudioReceiveStream objects. Bug: webrtc:12951 Change-Id: I1227297ec4ebeea9ba15fe2ed904349829b2e669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225262 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34464}
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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.
Development
See here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
Description
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