Tommi f58ded7cf0 Use audio views in Interleave() and Deinterleave()
Interleave and Deinterleave now accept two parameters, one for the
interleaved buffer and another for the deinterleaved one.

The previous versions of the functions still need to exist for test
code that uses ChannelBuffer.

Bug: chromium:335805780
Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42412}
2024-05-30 13:07:32 +00:00
2024-05-30 05:57:43 +00:00
2024-05-27 12:51:11 +00:00
2023-10-30 14:56:36 +00:00
2024-05-24 13:08:35 +00:00
2022-02-20 14:22:13 +00:00
2022-12-02 09:21:47 +00:00
2024-05-22 13:32:41 +00:00
2023-09-25 15:56:09 +00:00
2024-05-29 12:18:44 +00:00
2024-05-27 12:46:57 +00:00

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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%