| commit | d350006b70927b19afa43e9c11e0bf3fdb0cdf15 | [log] [tgz] |
|---|---|---|
| author | Tommi <tommi@webrtc.org> | Mon Jun 14 17:39:45 2021 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Jun 14 17:57:57 2021 |
| tree | f0d63d9a5e004a51af35f692dc2cab2bc00cd5f4 | |
| parent | 48420fa947cea4c618d51dc5f87908765a3a69db [diff] |
Add rtp_config() accessor to ReceiveStream. This is a consistent way to get to common config parameters for all receive streams and avoids storing a copy of the extension headers inside of Call. This is needed to get rid of the need of keeping config and copies in sync, which currently is part of why we repeatedly delete and recreate audio receive streams on config changes. Bug: webrtc:11993 Change-Id: Ia356b6cac1425c8c6766abd2e52fdeb73c4a4b4f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222040 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34285}
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.