90ecee1ed9
In order for WebRTC to be able to include packet overhead in its bitrate calculations, the AudioEncoder::GetFrameLengthRange() function must be implemented by all audio encoders. Making this member function pure virtual as per the following PSA: https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0 Bug: webrtc:11427 Change-Id: I30d297ef05f57453bfc257624729559057cad118 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517 Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31127}
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