commit | 94324f2774bc1816735ead8b1b1a8d26d8424882 | [log] [tgz] |
---|---|---|
author | Harald Alvestrand <hta@webrtc.org> | Wed Jan 13 12:31:53 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 13 13:48:02 2021 |
tree | 2673e55127904d44e7864024a4eebf05e2ade0e7 | |
parent | 0e8dd039bee3d3ca219ae2510b2711a20b743fff [diff] |
Add a test to detect excessive audio delay during renegotiation. This version uses relative_packet_arrival_delay as the target metric. Bug: none Change-Id: Ie6eb575ce4d13fd005f026862892b14bd4fb1135 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201620 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32962}
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.