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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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