| commit | d4161a3c9d459709ea92df8b73f7fd06038ec1fe | [log] [tgz] |
|---|---|---|
| author | Alessio Bazzica <alessiob@webrtc.org> | Fri Aug 31 08:41:37 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Aug 31 15:27:50 2018 |
| tree | b8dd9a2cf03bc1d5f2650ba8babcd75c7511663f | |
| parent | 8a3c166fffe6f03afb36ee6681e24f87fe29e73b [diff] |
Moving LappedTransform, Blocker and AudioRingBuffer. 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.
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.