| commit | fd5a253688902a3d8fe047853316faa1d59f8a0b | [log] [tgz] | 
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Mon Mar 31 09:20:06 2025 | 
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Mar 31 10:13:54 2025 | 
| tree | fb9d0188d483ead8794d9d4b01e9f04a9bb608a1 | |
| parent | f2337b4eb4191273f55727018ecd5b0d81ebddd6 [diff] | 
Delete deprecated VoipEngineConfig::audio_processing Users should have migrated to audio_processing_builder instead. Bug: webrtc:369904700 Change-Id: I815485ae7b7d41a5fefad90a27158838639719a4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383700 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44264}
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.