81c0cf287c
This CL performs a major refactoring and simplification of the AudioBuffer code that. -Removes 7 of the 9 internal buffers of the AudioBuffer. -Avoids the implicit copying required to keep the internal buffers in sync. -Removes all code relating to handling of fixed-point sample data in the AudioBuffer. -Changes the naming of the class methods to reflect that only floating point is handled. -Corrects some bugs in the code. -Extends the handling of internal downmixing to be more generic. Bug: webrtc:10882 Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28928}
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