dba1f945cfb65e7d47dd72d0cbe1cd3049048594

Added error checking in AudioIngress and AudioEgress to detect situations where codecs have not been set; added additional unit tests for VoipCore Bug: webrtc:11251 Change-Id: Ibd57e518892c76e2865b844ba866e380a565dd6b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180229 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Tim Na <natim@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31874}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
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