| commit | cad50b5a1445a344dffaae0f57d5fbde95910b98 | [log] [tgz] |
|---|---|---|
| author | Tommi <tommi@webrtc.org> | Tue May 20 18:54:36 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu May 22 09:34:52 2025 |
| tree | 5f9809d2b01c5f803fc84c02f9632d8b7d2bfbde | |
| parent | 54c8230a472603aa97770e34b5e891eba4dde7b8 [diff] |
[ResamplerHelper] Avoid resampling muted audio frames. Bug: none Change-Id: I1787b5ea6b9b33a0d37992287a4adc287fbb2ce5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392662 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44729}
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.