| commit | 0e7000b20a5a3f7a420dfe1ef4fa7f3347cded75 | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Tue Jul 05 07:53:35 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Jul 05 14:53:45 2016 +0000 |
| tree | 7d4c73fedf246bea784f3636320eeec8a2612a3c | |
| parent | 36a321d2e32bd9d823329958a71d4eba8356522a [diff] |
Changes in UI and minor extra functionality for rtp_analyzer. 1. The tool now displays packet loss in %. 2. It can print header information to stdout like rtp_analyze. 3. It has a command-line switch that lets you override the sample rate guessing. With the flag "--query_sample_rate" the tool asks you to always provide a sample rate. 4. Less decimals are printed for the estimated sample rate. NOTRY=True Review-Url: https://codereview.webrtc.org/2123773002 Cr-Commit-Position: refs/heads/master@{#13385}
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.