commit | 59574ca6d36f4769ecc734a5a52cdbb23f04d971 | [log] [tgz] |
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author | Tommi <tommi@webrtc.org> | Tue Sep 05 07:21:57 2023 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Sep 05 08:50:11 2023 |
tree | 438760f8c73eaa1e30d0866c9ebe9c29770c6b56 | |
parent | 7cc1ca26c8350b27ffc7252387b60e80b3e1b56f [diff] |
Add absl::AnyInvocable to SSLStreamAdapter::Create Remove internal use of SignalSSLHandshakeError and prepare removal of sigslot dependency from SSLStreamAdapter. Bug: webrtc:11943 Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40695}
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.