diff --git a/src/jack.cxx b/src/jack.cxx
index e2ba6d6..51c45d2 100644
--- a/src/jack.cxx
+++ b/src/jack.cxx
@@ -176,7 +176,7 @@ Jack::Jack( std::string name ) :
                           0 );
   */
   
-  masterVol = 0.0f;
+
   returnVol = 1.0f;
   
   inputToMixEnable  = false;
@@ -211,6 +211,8 @@ Jack::Jack( std::string name ) :
   /// setup DSP instances
   inputVol = 1.0f;
   masterVol = 0.75f;
+  masterVolLag =0.75f;
+  masterVolDiff =0.0f;
   masterMeter = new DBMeter( buffers.samplerate );
   inputMeter  = new DBMeter( buffers.samplerate );
   
@@ -466,11 +468,14 @@ void Jack::processFrames(int nframes)
     
     buffers.audio[Buffers::SIDECHAIN_SIGNAL][i] += input * inputToXSideVol;
     
+    //compute master volume lag;
+    if(fabs(masterVol-masterVolLag)>=fabs(masterVolDiff/10.0))
+        masterVolLag+=masterVolDiff/10.0;
     /// mixdown returns into master buffers
     // FIXME: Returns broken, due to metronome glitch in master output: buffer
     // writing issue or such. See #95 on github
-    buffers.audio[Buffers::JACK_MASTER_OUT_L][i] = L * masterVol;// (L + returnL*returnVol) * masterVol;
-    buffers.audio[Buffers::JACK_MASTER_OUT_R][i] = R * masterVol;// (R + returnR*returnVol) * masterVol;
+    buffers.audio[Buffers::JACK_MASTER_OUT_L][i] = L * masterVolLag;// (L + returnL*returnVol) * masterVol;
+    buffers.audio[Buffers::JACK_MASTER_OUT_R][i] = R * masterVolLag;// (R + returnR*returnVol) * masterVol;
     
     /// write SEND content to JACK port
     buffers.audio[Buffers::JACK_SEND_OUT][i] = buffers.audio[Buffers::SEND][i];
@@ -541,6 +546,7 @@ void Jack::clearInternalBuffers(int nframes)
 void Jack::masterVolume(float vol)
 {
   masterVol = vol;
+  masterVolDiff=masterVol-masterVolLag;
 }
 
 void Jack::returnVolume(float vol)
diff --git a/src/jack.hxx b/src/jack.hxx
index 1c8f71e..f02d901 100644
--- a/src/jack.hxx
+++ b/src/jack.hxx
@@ -133,7 +133,11 @@ class Jack
     DBMeter* masterMeter;
     
     float inputVol;
+    /// _toMasterLag is a volume that lags behind _toMaster when setMaster() is called
+    /// This prohibits audible jumps when rapidly changing the volume
     float masterVol;
+    float masterVolLag;
+    float masterVolDiff;
     float returnVol;
     
     float inputToMixVol;
diff --git a/src/trackoutput.cxx b/src/trackoutput.cxx
index 9238c33..c1e64ba 100644
--- a/src/trackoutput.cxx
+++ b/src/trackoutput.cxx
@@ -34,7 +34,8 @@ TrackOutput::TrackOutput(int t, AudioProcessor* ap) :
   dbMeter = new DBMeter( jack->getSamplerate() );
   
   _toMaster        = 0.8;
-  
+  _toMasterLag     = 0.8;
+  _toMasterDiff    = 0;
   _toReverb        = 0.0;
   _toSidechain     = 0.0;
   _toPostSidechain = 0.0;
@@ -48,6 +49,7 @@ TrackOutput::TrackOutput(int t, AudioProcessor* ap) :
 void TrackOutput::setMaster(float value)
 {
   _toMaster = value;
+  _toMasterDiff=_toMaster-_toMasterLag;
 }
 
 float TrackOutput::getMaster()
@@ -102,6 +104,9 @@ void TrackOutput::setSend( int send, float value )
 
 void TrackOutput::process(unsigned int nframes, Buffers* buffers)
 {
+    //compute master volume lag;
+    if(fabs(_toMaster-_toMasterLag)>=fabs(_toMasterDiff/10.0))
+        _toMasterLag+=_toMasterDiff/10.0;
   // get & zero track buffer
   float* trackBuffer = buffers->audio[Buffers::TRACK_0 + track];
   memset( trackBuffer, 0, sizeof(float)*nframes );
@@ -114,8 +119,8 @@ void TrackOutput::process(unsigned int nframes, Buffers* buffers)
   
   if (uiUpdateCounter > uiUpdateConstant )
   {
-    float l = dbMeter->getLeftDB() * _toMaster;
-    float r = dbMeter->getRightDB() * _toMaster;
+    float l = dbMeter->getLeftDB() * _toMasterLag;
+    float r = dbMeter->getRightDB() * _toMasterLag;
     EventTrackSignalLevel e( track, l, r );
     writeToGuiRingbuffer( &e );
     uiUpdateCounter = 0;
@@ -137,14 +142,14 @@ void TrackOutput::process(unsigned int nframes, Buffers* buffers)
     float tmp = trackBuffer[i];
     
     // post-sidechain *moves* signal between "before/after" ducking, not add!
-    masterL[i]       += tmp * _toMaster * (1-_toPostSidechain);
-    masterR[i]       += tmp * _toMaster * (1-_toPostSidechain);
+    masterL[i]       += tmp * _toMasterLag * (1-_toPostSidechain);
+    masterR[i]       += tmp * _toMasterLag * (1-_toPostSidechain);
     
     if ( _toPostfaderActive )
-      reverb[i]        += tmp * _toReverb * _toMaster;
+      reverb[i]        += tmp * _toReverb * _toMasterLag;
     
     if ( _toXSideActive )
-      postSidechain[i] += tmp * _toPostSidechain * _toMaster;
+      postSidechain[i] += tmp * _toPostSidechain * _toMasterLag;
     
     // turning down an element in the mix should *NOT* influence sidechaining
     if ( _toKeyActive )
diff --git a/src/trackoutput.hxx b/src/trackoutput.hxx
index 55c2fa0..a4382bb 100644
--- a/src/trackoutput.hxx
+++ b/src/trackoutput.hxx
@@ -59,9 +59,12 @@ class TrackOutput : public AudioProcessor
     int track;
     
     bool  _recordArm;
-    
+    /// _toMasterLag is a volume that lags behind _toMaster when setMaster() is called
+    /// This prohibits audible jumps when rapidly changing the volume
     float _toMaster;
-    
+    float _toMasterLag;
+    float _toMasterDiff;
+
     float _toReverb;
     float _toSidechain;
     float _toPostSidechain;