Consolidate FakeAudioStream and FakeVideoStream classes for tests. Bug: webrtc:42233500 Change-Id: I31fc6151e38087436227ed21f4a219d784b9f651 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/455241 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47108}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.