diff --git a/lib/synth/include/channel.h b/lib/synth/include/channel.h
index 44fe43c..0918593 100644
--- a/lib/synth/include/channel.h
+++ b/lib/synth/include/channel.h
@@ -53,8 +53,8 @@ public:
 
         std::fill(out, out + bufferSize, 0.f);
         for(Voice** voice = voices; *voice != NULL; ++voice) {
-            float voiceOut[numFrames];
-            (*voice)->tick(voiceOut, numFrames, pitchMod, fltMod);
+            float voiceOut[bufferSize];
+            (*voice)->tick(voiceOut, bufferSize, pitchMod, fltMod);
 
             for(size_t i = 0; i < bufferSize; ++i) {
                 out[i] += voiceOut[i];
diff --git a/lib/synth/include/dsp/filter.h b/lib/synth/include/dsp/filter.h
index 714e7c5..ea2f29b 100644
--- a/lib/synth/include/dsp/filter.h
+++ b/lib/synth/include/dsp/filter.h
@@ -52,6 +52,46 @@ public:
     }
 };
 
+class SVF12Stereo {
+protected:
+    float hp[2], bp[2], lp[2];
+    float as1[2], as2[2];
+
+public:
+    SVF12Stereo() : hp{0, 0}, bp{0, 0}, lp{0, 0}, as1{0, 0}, as2{0, 0} {}
+
+    inline void tick(float *in, float *hpo, float *bpo, float *lpo, size_t bufferSize, float *kK, float *kQ) {
+        size_t frameCount = bufferSize / 2;
+
+        float kQ_1[frameCount];
+        for(size_t i = 0; i < frameCount; ++i) {
+            kQ_1[i] = 1.f / kQ[i];
+        }
+        float kMul[frameCount];
+        for(size_t i = 0; i < frameCount; ++i) {
+            kMul[i] = 1.f / (1.f + kK[i]*kQ_1[i] + kK[i]*kK[i]);
+        }
+        for(size_t i = 0, j = 0; i < bufferSize; ++i) {
+            size_t ch = i & 1;
+            
+            hp[ch] = (in[i] - (kQ_1[j] + kK[j]) * as1[ch] - as2[ch]) * kMul[j];
+            float au = hp[ch] * kK[j];
+            bp[ch] = au + as1[ch];
+            as1[ch] = au + bp[ch];
+            au = bp[ch] * kK[j];
+            lp[ch] = au + as2[ch];
+            as2[ch] = au + lp[ch];
+
+            hpo && (hpo[i] = hp[ch]);
+            bpo && (bpo[i] = bp[ch]);
+            lpo && (lpo[i] = lp[ch]);
+
+            j += ch;
+        }
+    }
+};
+
+
 // 12 and 24 dB/oct
 class Filter {
 public:
@@ -120,5 +160,74 @@ public:
     }
 };
 
+// 12 and 24 dB/oct
+class FilterStereo {
+public:
+    enum Type {
+        TYPE_LP = 0,
+        TYPE_BP,
+        TYPE_HP
+    };
+
+    enum Slope {
+        SLOPE_12 = 0,
+        SLOPE_24,
+    };
+
+    typedef struct {
+        float freq;
+        float res;
+        Type type;
+        Slope slope;
+    } Settings;
+
+private:
+    Settings const * settings;
+    SVF12Stereo fltA, fltB;
+    float freq, res;
+
+public:
+    void assign(Settings const * settings);
+    
+    inline void tick(float *in, float *out, size_t bufferSize, float *freqMod) {
+        size_t frameCount = bufferSize / 2;
+
+        float dt = 1.f / (float) frameCount;
+
+        float kK[frameCount];
+        float dFreq = dt * (settings->freq - freq);
+        for(size_t i = 0; i < frameCount; i++) {
+            kK[i] = fastCVtoK(freq + freqMod[i]);
+            freq += dFreq;
+        }
+        float kQ[frameCount];
+        float dRes = dt * (settings->res - res);
+        for(size_t i = 0; i < frameCount; i++) {
+            kQ[i] = (float) M_SQRT1_2 + res;
+            res += dRes;
+        }
+
+        switch(settings->type) {
+            case TYPE_LP:
+                fltA.tick(in, NULL, NULL, out, bufferSize, kK, kQ);
+                if(settings->slope == SLOPE_24) {
+                    fltB.tick(out, NULL, NULL, out, bufferSize, kK, kQ);
+                }
+                break;
+            case TYPE_BP:
+                fltA.tick(in, NULL, out, NULL, bufferSize, kK, kQ);
+                if(settings->slope == SLOPE_24) {
+                    fltB.tick(out, NULL, out, NULL, bufferSize, kK, kQ);
+                }
+                break;
+            case TYPE_HP:
+                fltA.tick(in, out, NULL, NULL, bufferSize, kK, kQ);
+                if(settings->slope == SLOPE_24) {
+                    fltB.tick(out, out, NULL, NULL, bufferSize, kK, kQ);
+                }
+                break;
+        }
+    }
+};
 
 #endif
\ No newline at end of file
diff --git a/lib/synth/include/dsp/oscillator.h b/lib/synth/include/dsp/oscillator.h
index 99b7a58..1b1f1e0 100644
--- a/lib/synth/include/dsp/oscillator.h
+++ b/lib/synth/include/dsp/oscillator.h
@@ -67,8 +67,8 @@ public:
                 value += polyBlep(phaseBuf[i], phaseStepBuf[i]); // Layer output of Poly BLEP on top (flip)
                 value -= polyBlep(fmodf(phaseBuf[i] + 0.5f, 1.0f), phaseStepBuf[i]); // Layer output of Poly BLEP on top (flop)
 
-                out[j++] = gain[j & 1] * value;
-                out[j++] = gain[j & 1] * value;
+                out[j++] += gain[j & 1] * value;
+                out[j++] += gain[j & 1] * value;
             }
         }
     }
diff --git a/lib/synth/include/synth.h b/lib/synth/include/synth.h
index d2746e5..e956cf8 100644
--- a/lib/synth/include/synth.h
+++ b/lib/synth/include/synth.h
@@ -44,12 +44,14 @@ public:
             }
         }
 
+        /*
         for(size_t i = 0; i < bufferSize; i += 2) {
             frame f = {reverbBus[i], reverbBus[i + 1]};
             f = reverb.tick(f);
             out[i] += f.l;
             out[i + 1] += f.r;
         }
+        */
         
         for(size_t i = 0; i < bufferSize; ++i) {
             out[i] *= 0.125f;
diff --git a/lib/synth/include/voice.h b/lib/synth/include/voice.h
index 9ad311f..45f1377 100644
--- a/lib/synth/include/voice.h
+++ b/lib/synth/include/voice.h
@@ -8,7 +8,7 @@
 #include "dsp/filter.h"
 #include "dsp/adsr.h"
 
-#define MAX_OSCS 8
+#define MAX_OSCS 16
 
 class Voice {
 public:
@@ -23,7 +23,7 @@ public:
 
         float noiseMix;
 
-        Filter::Settings filter;
+        FilterStereo::Settings filter;
 
         ADSR::Envelope ampEnv;
         ADSR::Envelope modEnv;
@@ -43,7 +43,7 @@ private:
     ADSR adsrAmp;
     ADSR adsrMod;
     Oscillator osc[MAX_OSCS];
-    Filter filter;
+    FilterStereo filter;
 
 public:
     int note;
@@ -64,12 +64,12 @@ public:
             basePitchBuf[i] = note + pitchMod[i];
         }
 
-        float oscCVBuf[MAX_OSCS][frameCount];
+        float oscPitchBuf[MAX_OSCS][frameCount];
         float detune = -settings->oscDetune * 0.5f * (settings->unison - 1);
         for(size_t i = 0; i < settings->unison; ++i) {
             float transpose = (i & 1) ? settings->osc2Pitch : 0;
             for(size_t j = 0; j < frameCount; ++j) {
-                oscCVBuf[i][j] = noteToCV(basePitchBuf[i] + detune + transpose);
+                oscPitchBuf[i][j] = noteToCV(basePitchBuf[j] + detune + transpose);
             }
             detune += settings->oscDetune;
         }
@@ -88,10 +88,10 @@ public:
         for(size_t i = 0; i < settings->unison; ++i) {
             float baseGain = (i & 1) ? settings->oscMix : (1 - settings->oscMix);
             float gain[2];
-            gain[0] = sinf((1.f - pan) * (float) M_PI_2);
-            gain[1] = sinf(pan * (float) M_PI_2);
+            gain[0] = baseGain * sinf((1.f - pan) * (float) M_PI_2);
+            gain[1] = baseGain * sinf(pan * (float) M_PI_2);
             pan += panStep;
-            osc[i].tick(oscBuf, bufferSize, oscCVBuf[i], gain);
+            osc[i].tick(oscBuf, bufferSize, oscPitchBuf[i], gain);
         }
 
         float noiseBuf[bufferSize];
@@ -117,6 +117,8 @@ public:
         for(size_t i = 0; i < frameCount; ++i) {
             fltModBuf[i] = settings->modEnvFltGain * adsrMod.tick() + keyTrackVal + fltMod[i];
         }
+
+        //std::copy(oscBuf, oscBuf + bufferSize, out);
         filter.tick(oscBuf, out, bufferSize, fltModBuf);
         for(size_t i = 0; i < bufferSize; ++i) {
             out[i] *= gain * adsrAmp.tick();
diff --git a/lib/synth/src/channel.cpp b/lib/synth/src/channel.cpp
index dced626..fb5f529 100644
--- a/lib/synth/src/channel.cpp
+++ b/lib/synth/src/channel.cpp
@@ -34,7 +34,7 @@ inline float ccToLFOFltMod(int x) {
 }
 
 void Channel::loadPreset(const Preset * preset) {
-    settings.unison = 1 + floorf(preset->unison * 8.f / 128.f);
+    settings.unison = 1 + floorf(preset->unison * 16.f / 128.f);
 
     settings.osc1Mode = Oscillator::Mode(preset->osc1Mode);
     settings.osc2Mode = Oscillator::Mode(preset->osc2Mode);
@@ -43,8 +43,8 @@ void Channel::loadPreset(const Preset * preset) {
     settings.oscMix = preset->oscMix / 127.f;
     settings.noiseMix = preset->noiseMix / 127.f;
 
-    settings.filter.type = Filter::Type(floorf(3.f * preset->filter.type / 128.f));
-    settings.filter.slope = Filter::Slope(floorf(2.f * preset->filter.slope / 128.f));
+    settings.filter.type = FilterStereo::Type(floorf(3.f * preset->filter.type / 128.f));
+    settings.filter.slope = FilterStereo::Slope(floorf(2.f * preset->filter.slope / 128.f));
     settings.filter.freq = (float) FILTER_CV_MAX * preset->filter.freq / 127.f;
     settings.filter.res = preset->filter.Q / 31.75f;
 
@@ -71,7 +71,7 @@ void Channel::noteOn(int note, int velocity) {
     //printf("on: midi=%d note=%d\n", note, midiToNote(note));
     Voice* const voice = voiceManager->get(number);
     voice->assign(&settings);
-    voice->noteOn(midiToNote(note), velocity / 127.f, 1.f, 0.5f);
+    voice->noteOn(midiToNote(note), velocity / 127.f, 1.f);
 }
 
 void Channel::noteOff(int note) {
@@ -159,7 +159,7 @@ void Channel::control(int code, int value) {
             break;
 
         case CC_UNISON: // Unison Amount
-            settings.unison = 1 + floorf(value * 8.f / 128.f);
+            settings.unison = 1 + floorf(value * 16.f / 128.f);
             break;
 
         case CC_OSC_MIX: // Oscillator 1/2 Mix
@@ -175,11 +175,11 @@ void Channel::control(int code, int value) {
             break;
 
         case CC_FLT_TYP: // Filter Type
-            settings.filter.type = Filter::Type(floorf(3.f * value / 128.f));
+            settings.filter.type = FilterStereo::Type(floorf(3.f * value / 128.f));
             break;
 
         case CC_FLT_SLP: // Filter Slope
-            settings.filter.slope = Filter::Slope(floorf(2.f * value / 128.f));
+            settings.filter.slope = FilterStereo::Slope(floorf(2.f * value / 128.f));
             break;
 
         case CC_RVB_SND:
diff --git a/lib/synth/src/dsp/filter.cpp b/lib/synth/src/dsp/filter.cpp
index b983d74..e4fb0cf 100644
--- a/lib/synth/src/dsp/filter.cpp
+++ b/lib/synth/src/dsp/filter.cpp
@@ -4,4 +4,10 @@ void Filter::assign(Settings const * settings) {
     this->settings = settings;
     freq = settings->freq;
     res = settings->res;
+}
+
+void FilterStereo::assign(Settings const * settings) {
+    this->settings = settings;
+    freq = settings->freq;
+    res = settings->res;
 }
\ No newline at end of file
diff --git a/lib/synth/src/voice.cpp b/lib/synth/src/voice.cpp
index 23b70ed..94100d0 100644
--- a/lib/synth/src/voice.cpp
+++ b/lib/synth/src/voice.cpp
@@ -19,11 +19,10 @@ void Voice::assign(Settings const * settings) {
     filter.assign(&settings->filter);
 }
 
-void Voice::noteOn(int note, float velocity, float gain, float pan) {
+void Voice::noteOn(int note, float velocity, float gain) {
     this->note = note;
     this->velocity = velocity;
     this->gain = gain;
-    this->pan = pan;
     adsrAmp.noteOn();
     adsrMod.noteOn();
 }