commit | 0adcc2b8bae9c1e4aa3c6c892d899ff3a2741dc6 | [log] [tgz] |
---|---|---|
author | Jakob Ivarsson <jakobi@webrtc.org> | Fri Mar 28 15:14:09 2025 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Mar 28 16:57:44 2025 |
tree | 9166e059f5ff085dc00a1671def70377fb6bdd70 | |
parent | 90f23ed8f9238a3e5b041bdd3c4f7cbcd5e234ce [diff] |
Call the audio encoder asynchronously when updating RTT, loss and bitrate allocation. This avoids blocking the worker thread on audio encoding for these updates. Bug: None Change-Id: I4a3ddbbfd5cfad6190d258d8d173d07d3d86d404 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/383520 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44260}
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.