| commit | 86cc6ffc7ca14c79d760695e89ab9698ed251873 | [log] [tgz] |
|---|---|---|
| author | mflodman <mflodman@webrtc.org> | Tue Jul 26 11:44:06 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Jul 26 11:44:12 2016 |
| tree | fc3a84794f675afbb29d44fe383a1709c5f7beba | |
| parent | 62d695f6550cb8577191f00af9df76c4d8474afc [diff] |
Variable audio bitrate. This is a first CL wiring up AudioSendStream to BitrateAllocator. This is still experimental and there is a test added for the audio only case, combined audio video variable bitrate test cases will be added as a follow up. BUG=5079 Review-Url: https://codereview.webrtc.org/2165743003 Cr-Commit-Position: refs/heads/master@{#13527}
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.