commit | 45b9192ad981dcdc12ad4aef087fff2195bd030c | [log] [tgz] |
---|---|---|
author | Johannes Kron <kron@webrtc.org> | Thu May 28 13:09:19 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Thu May 28 21:12:49 2020 |
tree | a5dac40209ffdd19b79d91c4aa36a48c7725c85b | |
parent | 1a4975642bf962ae059674b1b89cde4d785aba5b [diff] |
Add trace of enqueued and sent RTP packets This is useful in debugging the latency from a packet is enqueued until it's sent. Bug: webrtc:11617 Change-Id: Ic2f194334a2e178de221df3a0838481035bb3505 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176231 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31381}
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.