| commit | ff2a6351e0ad81ef8123c368fc17eeab40e66c71 | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Thu Jan 14 10:00:21 2016 +0100 |
| committer | Stefan Holmer <stefan@webrtc.org> | Thu Jan 14 09:00:34 2016 +0000 |
| tree | 6c1c6c1a71d9cdb3ecf69bdfda454777add7c7de | |
| parent | 709513d4133107d5c02aed34a5ee99444c4d4e25 [diff] |
Add ramp-up tests for transport sequence number with and w/o audio. Also add a perf metric tracking the average network latency. The audio stream test is disabled for now since audio isn't included in bitrate allocation. BUG=webrtc:5263 R=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1582833002 . Cr-Commit-Position: refs/heads/master@{#11244}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.