d4161a3c9d
LappedTransform is only used in BandwidthAdaptationTest and therefore it should not be anymore a visible target under common_audio. This CL moves LappedTransform and other two classes it depends on (and which are not used elsewhere) to modules/audio_coding/codecs/opus/test. Bug: webrtc:9577, webrtc:5298 Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a Reviewed-on: https://webrtc-review.googlesource.com/96440 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24509}
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