minyue@webrtc.org 0040a6ef97 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
2014-08-04 14:41:57 +00:00
2014-06-11 13:59:44 +00:00
2014-07-18 13:33:48 +00:00
2014-08-04 14:41:57 +00:00
2014-06-17 08:54:03 +00:00
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
BSD-3-Clause 446 MiB
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