| commit | 660b3fa98184e4f47b8fa7ef390f23914146e395 | [log] [tgz] |
|---|---|---|
| author | Harald Alvestrand <hta@webrtc.org> | Thu Oct 16 10:12:25 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Oct 17 07:03:06 2025 |
| tree | a0203011f8047a70531518946fd283d78f7e35a1 | |
| parent | 53a7bb255dc00f7ef3f782ce99a3fcc2f4338726 [diff] |
Remove sigslot::has_slots from JsepTransportController All signals are now connected via slots in other objects. Bug: webrtc:42222066 Change-Id: I4e6fdea52aa0a0c21ff24698fe33a6263e279e5f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/417421 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45960}
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.