Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets

BUG=2935
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5725 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/interface/neteq.h b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
index 6173930..466882a 100644
--- a/webrtc/modules/audio_coding/neteq4/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
@@ -102,7 +102,7 @@
     kSyncPacketNotAccepted
   };
 
-  static const int kMaxNumPacketsInBuffer = 240;  // TODO(hlundin): Remove.
+  static const int kMaxNumPacketsInBuffer = 50;  // TODO(hlundin): Remove.
   static const int kMaxBytesInBuffer = 113280;  // TODO(hlundin): Remove.
 
   // Creates a new NetEq object, starting at the sample rate |sample_rate_hz|.