| commit | 8034614b817810f33ad193d8fa5c460bc6c28276 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Wed Apr 27 14:17:10 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Apr 27 21:17:15 2016 +0000 |
| tree | 5a2d7737b416255b18030eac3f31f54f61ccf299 | |
| parent | 081254f2c62037d016f9fc961764c6f01cb095da [diff] |
Cap the send bitrate for opus and iSAC before passing down to VoE. The voice engine expects send bitrates no more than the maximum for the codec. For example, 510kbps for opus. So if "b=AS" sets a maximum above the codec maximum, WebRtcVoiceEngine needs to cap it. BUG=603690 Review-Url: https://codereview.webrtc.org/1920123002 Cr-Commit-Position: refs/heads/master@{#12537}
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.