commit | ebc7d1eda6f23c4fa070f527fa557ba750fa30b0 | [log] [tgz] |
---|---|---|
author | Colin Cross <ccross@android.com> | Tue Apr 12 10:42:31 2022 -0700 |
committer | Colin Cross <ccross@android.com> | Tue Apr 12 12:11:06 2022 -0700 |
tree | 08c8e85c5b1496b8c65086275934b2dec90e2090 | |
parent | 7d5c00bbb9ff7b14de493bfa0ea56d4993f2e187 [diff] |
Remove alsa and pulseaudio sources ALSA and pulseaudio are not used by either of the clients of webrtc, remove them from the sources to avoid having to have the ALSA and pulseaudio headers. Bug: 190084016 Test: m USE_HOST_MUSL=true webRTC libaudiopreprocessing Change-Id: I1e74c64a71fa54db69e7d2e9d118dfd374501cb7
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.