| commit | 0040a6ef97236053d9698470b9d4c095e8019f1c | [log] [tgz] |
|---|---|---|
| author | minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Mon Aug 04 14:41:57 2014 |
| committer | minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Mon Aug 04 14:41:57 2014 |
| tree | 19376d75f30d6614eeef05d4212202eccc598b9f | |
| parent | 84b9e1e9d9407c48ff85a28f0825fe3a23a1f614 [diff] |
This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906 In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate). BUG= R=henrik.lundin@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13099004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d