From 8c091c2a050601d13d316ec8a3ddb70afb98f48e Mon Sep 17 00:00:00 2001
From: Gordon JC Pearce <gordonjcp@gmail.com>
Date: Sun, 21 Dec 2025 22:08:09 +0000
Subject: [PATCH] sine-based, getting close to real though

---
 plugin/chorus.cpp | 58 ++++++++++++++++++++---------------------------
 plugin/chorus.hpp |  7 +++---
 plugin/voice.cpp  |  2 +-
 3 files changed, 29 insertions(+), 38 deletions(-)

diff --git a/plugin/chorus.cpp b/plugin/chorus.cpp
index 1c7bb36..2acadcd 100644
--- a/plugin/chorus.cpp
+++ b/plugin/chorus.cpp
@@ -30,17 +30,16 @@ Chorus::Chorus() {
     ram = new float[DELAYSIZE];  // probably needs to be calculated based on sample rate
 
     fastPhase = 0;
-    slowPhase = 0;
 
-    postFilter1l = new SVF(POSTCUTOFF, .546);
-    postFilter2l = new SVF(POSTCUTOFF, 1.324);
-    postFilter1r = new SVF(POSTCUTOFF, .546);
-    postFilter2r = new SVF(POSTCUTOFF, 1.324);
+    // not quite Butterworth but you'd never hear the difference
+    postFilter1l = new SVF(9688, .549);
+    postFilter2l = new SVF(10377, 1.291);
+    postFilter1r = new SVF(9688, .549);
+    postFilter2r = new SVF(10377, 1.291);
 
     // lfo values taken from a rough simulation
-    fastOmega = 6.283 * 5.7 / sampleRate;  // approximate, can be adjusted
-    slowOmega = 6.283 * 0.7 / sampleRate;  // again approximate
-
+    fastOmega = 6.283 * 0.7 / sampleRate;  // approximate, can be adjusted
+    
     // zero out the delay buffer
     memset(ram, 0, sizeof(float) * DELAYSIZE);
     memset(lpfOut1, 0, sizeof(float) * bufferSize);
@@ -62,7 +61,7 @@ void Chorus::run(const float* input, float** outputs, uint32_t frames) {
     // actual effects here
 
     // now run the DSP
-    float out0 = 0, out120 = 0, out240 = 0, s0 = 0, s1 = 0;
+    float s0 = 0, s1 = 0;
     float lfoMod, dly1, frac;
     uint16_t tap, delay;
 
@@ -70,16 +69,14 @@ void Chorus::run(const float* input, float** outputs, uint32_t frames) {
         // run a step of LFO
         fastPhase += fastOmega;
         if (fastPhase > 6.283) fastPhase -= 6.283;
-        slowPhase += slowOmega;
-        if (slowPhase > 6.283) slowPhase -= 6.283;
 
         ram[delayptr] = input[i];
 
 #define BASE 0.05
-#define AMT 0.00175
+#define AMT 0.003175
 
         // 0 degree delay line
-        lfoMod = 0.203 * sin(fastPhase) + 0.835 * sin(slowPhase);
+        lfoMod = 0.603 * sin(fastPhase);
         dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
         delay = (int)dly1;
         frac = dly1 - delay;
@@ -87,39 +84,34 @@ void Chorus::run(const float* input, float** outputs, uint32_t frames) {
         tap = delayptr - delay;
         s1 = ram[(tap - 1) & 0x3ff];
         s0 = ram[tap & 0x3ff];
-        out0 = ((s1 - s0) * frac) + s0;
+        lpfOut1[i] = ((s1 - s0) * frac) + s0;
 
-        // 120 degree delay line
-        lfoMod = 0.248 * sin(fastPhase + 2.09) + 0.745 * sin(slowPhase + 2.09);
-        dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
+        dly1 = (BASE - (AMT * lfoMod)) * sampleRate;
         delay = (int)dly1;
         frac = dly1 - delay;
 
         tap = delayptr - delay;
         s1 = ram[(tap - 1) & 0x3ff];
         s0 = ram[tap & 0x3ff];
-        out120 = ((s1 - s0) * frac) + s0;
+        lpfOut2[i] = ((s1 - s0) * frac) + s0;
 
-        // 240 degree delay line
-        lfoMod = 0.252 * sin(fastPhase + 4.18) + 0.809 * sin(slowPhase + 4.18);
-        dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
-        delay = (int)dly1;
-        frac = dly1 - delay;
-
-        tap = delayptr - delay;
-        s1 = ram[(tap - 1) & 0x3ff];
-        s0 = ram[tap & 0x3ff];
-        out240 = ((s1 - s0) * frac) + s0;
-
-        lpfOut1[i] = (out0 + (out120 * 0.66) + (out240 * 0.33));
-        lpfOut2[i] = (out0 + (out120 * 0.33) + (out240 * 0.66));
+       // lpfOut1[i] = input[i] + 1.2 * out0; //(out0 + (out120 * 0.66) + (out240 * 0.33));
+      //  lpfOut2[i] = input[i] + 1.2 * out120; //(out0 + (out120 * 0.33) + (out240 * 0.66));
 
         delayptr++;
         delayptr &= 0x3ff;
     }
 
     postFilter1l->runSVF(lpfOut1, lpfOut1, frames);
-    postFilter2l->runSVF(lpfOut1, outputs[0], frames);
+    postFilter2l->runSVF(lpfOut1, lpfOut1, frames);
     postFilter1r->runSVF(lpfOut2, lpfOut2, frames);
-    postFilter2r->runSVF(lpfOut2, outputs[1], frames);
+    postFilter2r->runSVF(lpfOut2, lpfOut2, frames);
+
+    for (uint32_t i = 0; i < frames; i++) {
+        float y = input[i];
+        outputs[0][i] = y + (gain * lpfOut1[i]);
+        outputs[1][i] = y + (gain * lpfOut2[i]);
+    }
+
+
 }
diff --git a/plugin/chorus.hpp b/plugin/chorus.hpp
index e936201..3a4cdee 100644
--- a/plugin/chorus.hpp
+++ b/plugin/chorus.hpp
@@ -36,11 +36,10 @@ class Chorus {
     void run(const float* input, float** outputs, uint32_t frames);
 
    private:
-    double fastPhase, fastOmega;
-    double slowPhase, slowOmega;
-    double fastLfo, slowLfo;
+    double fastPhase = 0, fastOmega = 0;
+    float gain = 1.2;
 
-    uint16_t delayptr;
+    uint16_t delayptr = 0;
 
     float* ram;
     float* lpfIn;
diff --git a/plugin/voice.cpp b/plugin/voice.cpp
index 924264f..b734c13 100644
--- a/plugin/voice.cpp
+++ b/plugin/voice.cpp
@@ -52,7 +52,7 @@ void Voice::off() {
 
 void Voice::run(Module* m, float* buffer, uint32_t samples) {
     // carry out per-voice calculations for each block of samples
-    float out, t, fb, res;
+    float out, t, fb;
 
     //float cut = 0.00513 + 0.0000075*env;