| commit | 280054f2e63a50cb56a79172f15fccabb09ec986 | [log] [tgz] |
|---|---|---|
| author | Harald Alvestrand <hta@webrtc.org> | Tue Nov 10 13:12:53 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Nov 10 14:41:45 2020 |
| tree | de5897a225320583c82675758a70d729a891b62b | |
| parent | 5aaec9d0a495b813bdd1e415232d59de091abb30 [diff] |
Eliminate sigslot from RtpTransmissionManager at the cost of adding a WeakPointerFactory. Moves the RtpTransceiver "NegotiationNeeded" signal to a callback function that is passed as a constructor argument. Bug: webrtc:11943 Change-Id: I37b2027379acce38dbaf0f396daebdb3e579ee54 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192540 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32575}
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.