Files
platform-external-webrtc/webrtc/modules/audio_coding
henrik.lundin@webrtc.org 3ab57c514c Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets
This is a relanding of r5725, now with a fix for the failing tests.

BUG=2935
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5738 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-20 15:09:38 +00:00
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