diff --git a/MicroDexed.ino b/MicroDexed.ino
index ad2e1e2..b3c29ae 100644
--- a/MicroDexed.ino
+++ b/MicroDexed.ino
@@ -1814,6 +1814,9 @@ void _softRestart(void)
 
 float pseudo_log_curve(float value)
 {
+// ideas from here: https://forum.pjrc.com/threads/56015-Changing-Volume-Levels-based-on-pot-wheel
+
+    
   const float _pseudo_log = 1048575 / (float)(1 << 20);
   /*
     #ifdef DEBUG
diff --git a/config.h b/config.h
index db59fd0..22b8fc9 100644
--- a/config.h
+++ b/config.h
@@ -84,7 +84,7 @@
 //*************************************************************************************************
 //* DEBUG OUTPUT SETTINGS
 //*************************************************************************************************
-#define DEBUG 1
+//#define DEBUG 1
 #define SERIAL_SPEED 230400
 #define SHOW_XRUN 1
 #define SHOW_CPU_LOAD_MSEC 5000
diff --git a/dexed.cpp b/dexed.cpp
index 1034cf8..233f15f 100644
--- a/dexed.cpp
+++ b/dexed.cpp
@@ -149,7 +149,7 @@ void Dexed::getSamples(uint16_t n_samples, int16_t* buffer)
         for (j = 0; j < _N_; ++j)
         {
           //sumbuf[i + j] += static_cast<float>(signed_saturate_rshift(audiobuf.get()[j] >> 5, 24, 9)) / 0x8000;
-          sumbuf[i + j] += signed_saturate_rshift(audiobuf.get()[j] >> 4, 24, 9) / 32768.0;
+          sumbuf[i + j] += signed_saturate_rshift(audiobuf.get()[j] >> 5, 24, 9) / 32768.0;
           audiobuf.get()[j] = 0;
         }
       }
@@ -800,3 +800,82 @@ void Dexed::setPortamentoMode(uint8_t portamento_mode, uint8_t portamento_glissa
 
   controllers.refresh();
 }
+
+// https://www.musicdsp.org/en/latest/Effects/169-compressor.html#
+void compress
+    (
+        float*  wav_in,     // signal
+        int     n,          // N samples
+        double  threshold,  // threshold (percents)
+        double  slope,      // slope angle (percents)
+        int     sr,         // sample rate (smp/sec)
+        double  tla,        // lookahead  (ms)
+        double  twnd,       // window time (ms)
+        double  tatt,       // attack time  (ms)
+        double  trel        // release time (ms)
+    )
+{
+    typedef float   stereodata[2];
+    stereodata*     wav = (stereodata*) wav_in; // our stereo signal
+    threshold *= 0.01;          // threshold to unity (0...1)
+    slope *= 0.01;              // slope to unity
+    tla *= 1e-3;                // lookahead time to seconds
+    twnd *= 1e-3;               // window time to seconds
+    tatt *= 1e-3;               // attack time to seconds
+    trel *= 1e-3;               // release time to seconds
+
+    // attack and release "per sample decay"
+    double  att = (tatt == 0.0) ? (0.0) : exp (-1.0 / (sr * tatt));
+    double  rel = (trel == 0.0) ? (0.0) : exp (-1.0 / (sr * trel));
+
+    // envelope
+    double  env = 0.0;
+
+    // sample offset to lookahead wnd start
+    int     lhsmp = (int) (sr * tla);
+
+    // samples count in lookahead window
+    int     nrms = (int) (sr * twnd);
+
+    // for each sample...
+    for (int i = 0; i < n; ++i)
+    {
+        // now compute RMS
+        double  summ = 0;
+
+        // for each sample in window
+        for (int j = 0; j < nrms; ++j)
+        {
+            int     lki = i + j + lhsmp;
+            double  smp;
+
+            // if we in bounds of signal?
+            // if so, convert to mono
+            if (lki < n)
+                smp = 0.5 * wav[lki][0] + 0.5 * wav[lki][1];
+            else
+                smp = 0.0;      // if we out of bounds we just get zero in smp
+
+            summ += smp * smp;  // square em..
+        }
+
+        double  rms = sqrt (summ / nrms);   // root-mean-square
+
+        // dynamic selection: attack or release?
+        double  theta = rms > env ? att : rel;
+
+        // smoothing with capacitor, envelope extraction...
+        // here be aware of pIV denormal numbers glitch
+        env = (1.0 - theta) * rms + theta * env;
+
+        // the very easy hard knee 1:N compressor
+        double  gain = 1.0;
+        if (env > threshold)
+            gain = gain - (env - threshold) * slope;
+
+        // result - two hard kneed compressed channels...
+        float  leftchannel = wav[i][0] * gain;
+        float  rightchannel = wav[i][1] * gain;
+    }
+}
+