This is step to allow migration of Test ADM to the AudioDeviceModuleImpl as a base class to include AudioDeviceBuffer into SUT. Also it will allow to remove WaitForRecordingEnd() method from Test ADM Bug: b/272350185, webrtc:15081 Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39849}
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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
- 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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