commit | ecb3ed7a76e952f636641f18e3f46afca17dd35d | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Wed Oct 16 18:09:09 2024 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Oct 17 11:12:40 2024 |
tree | bea84be4aa79dc3030a41e5e590eab7104bda869 | |
parent | b280cb95c69090096d64ed0fd165e3fa26783d06 [diff] |
Migrate CreateVoipEngine to take audio_processing_factory instead of audio_processing This would allow users of the voip engine to migrate away from the AudioProcessingBuilder Bug: webrtc:369904700 Change-Id: Ie4f6f4579e185ff6366333a3f37e6aaa23b892b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365920 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43255}
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.