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
This commit is contained in:
minyue@webrtc.org
2014-08-04 14:41:57 +00:00
parent 84b9e1e9d9
commit 0040a6ef97
5 changed files with 110 additions and 16 deletions

View File

@ -28,6 +28,7 @@
],
'sources': [
'interface/opus_interface.h',
'opus_inst.h',
'opus_interface.c',
],
},