| commit | 5e58bcbf295bf6908cf6d5da1b011aae405b88be | [log] [tgz] |
|---|---|---|
| author | Ilya Nikolaevskiy <ilnik@webrtc.org> | Wed Oct 24 11:34:32 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Oct 24 12:27:09 2018 |
| tree | 8e23c18bbd6063300c8056f31a907b5f03522345 | |
| parent | ece3c228a2cbd1c1b05eee3a7f55dbb6f020acbc [diff] |
Forward audio rtp frequency to Rtcp sender and use it for SR packets Process video rtp frequency in the same way. Bug: webrtc:6458 Change-Id: Ia22768e1242d686c2b3e2b911f3e5e492cf8b895 Reviewed-on: https://webrtc-review.googlesource.com/c/107651 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25334}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.