diff --git a/audiohandler.cpp b/audiohandler.cpp
index 65cf057..7b097cf 100644
--- a/audiohandler.cpp
+++ b/audiohandler.cpp
@@ -96,10 +96,17 @@ bool audioHandler::init(audioSetup setupIn)
     // plugin->setupPlugin((char*)"");
     // plugin->setupPlugin((char*)"");
 
+    //char* pname = (char*)"http://gareus.org/oss/lv2/fil4#mono";
+
+    plugin->setupPlugin((char*)"http://drobilla.net/plugins/mda/Dynamics"); // gate and compressor together. Gate is too rough though.
+
+
+
+
     //plugin->setupPlugin((char*)"http://eq10q.sourceforge.net/gate"); // works well
 
     //plugin->setupPlugin((char*)"http://plugin.org.uk/swh-plugins/gate"); // works well
-    plugin->setupPlugin((char*)"http://eq10q.sourceforge.net/compressor"); // works very well
+    //plugin->setupPlugin((char*)"http://eq10q.sourceforge.net/compressor"); // works very well
 
     //plugin->setupPlugin((char*)"http://eq10q.sourceforge.net/eq/eq1qm"); // malloc double-free error
     //plugin->setupPlugin((char*)"http://eq10q.sourceforge.net/eq/eq6qm"); // malloc double-free error
@@ -118,7 +125,7 @@ bool audioHandler::init(audioSetup setupIn)
 
 
     //plugin->setupPlugin((char*)"http://plugin.org.uk/swh-plugins/dj_eq_mono"); // loads ok
-    //plugin->setupPlugin((char*)"http://plugin.org.uk/swh-plugins/sc2"); // loads ok
+    //plugin->setupPlugin((char*)"http://plugin.org.uk/swh-plugins/sc4"); // loads ok
     //plugin->setupPlugin((char*)"http://plugin.org.uk/swh-plugins/amp");
 
     //plugin->setupPlugin((char*)"http://plugin.org.uk/swh-plugins/fake_fail_plugin"); // fail, of course...
diff --git a/audioplugin.cpp b/audioplugin.cpp
index 1f6ed3d..ad81d7c 100644
--- a/audioplugin.cpp
+++ b/audioplugin.cpp
@@ -508,16 +508,16 @@ void audioPlugin::convertInputBuffer()
 
     // externalSourceBuffer --> scale --> inputBuffer
 
-    float max = 65535.0f;
-    float mid = max / 2.0f;
+    float max = 65535.0f / 2.0f;
+    //float mid = max / 2.0f;
     float scalingFactor = 1.0f / max;
 
     // Note, we have already cast the 8MSB, 8LSB type data into a 16-bit holder
     // So accessing one member should result in a full 16-bit number.
 
     for (int n = 0; n < sourceBufferSampleCount; n += 2) {
-        inputBuffer[0][n/2] = (  externalSourceBuffer[n] - mid) * scalingFactor;
-        inputBuffer[1][n/2] = (  externalSourceBuffer[n+1] - mid) * scalingFactor;
+        inputBuffer[0][n/2] = (  externalSourceBuffer[n]) * scalingFactor;
+        inputBuffer[1][n/2] = (  externalSourceBuffer[n+1]) * scalingFactor;
 
 //        inputBuffer[0][n/2] = (externalSourceBuffer[n] - mid) * scalingFactor;
 //        inputBuffer[1][n/2] = (externalSourceBuffer[n+1] - mid) * scalingFactor;
@@ -554,14 +554,14 @@ void audioPlugin::convertOutputBuffer()
     if(forceOutputMono)
     {
         for (int n = 0; n+1 < sourceBufferSampleCount; n += 2) {
-            externalSinkBuffer[n] = (outputBuffer[0][n/2] + 1) * mid; // L
-            externalSinkBuffer[n+1] = (outputBuffer[0][n/2] + 1) * mid; // Copy L
+            externalSinkBuffer[n] = (outputBuffer[0][n/2]) * mid; // L
+            externalSinkBuffer[n+1] = (outputBuffer[0][n/2]) * mid; // Copy L
         }
     } else {
 
         for (int n = 0; n+1 < sourceBufferSampleCount; n += 2) {
-            externalSinkBuffer[n] = (outputBuffer[0][n/2] + 1) * mid; // L
-            externalSinkBuffer[n+1] = (outputBuffer[1][n/2] + 1) * mid; // R
+            externalSinkBuffer[n] = (outputBuffer[0][n/2]) * mid; // L
+            externalSinkBuffer[n+1] = (outputBuffer[1][n/2]) * mid; // R
         }
     }