a8b7c7f4c6
- Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
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