globals for samplerate and buffersize

plugin/chorus.cpp

@@ -23,12 +23,7 @@
 
 #include <cstdio>
 
-//extern double sampleRate;
-//extern uint32_t bufferSize;
-
-double sampleRate = 48000;
-uint32_t bufferSize = 1024;
-Chorus::Chorus() {  // no parameters, programs, or states
+Chorus::Chorus() { 
 
     lpfOut1 = new float[bufferSize];
     lpfOut2 = new float[bufferSize];
@@ -53,6 +48,7 @@ Chorus::Chorus() {  // no parameters, programs, or states
 }
 
 Chorus::~Chorus() {
+    printf("called chorus destructor\n");
     delete lpfOut1;
     delete lpfOut2;
     delete ram;
@@ -70,6 +66,8 @@ void Chorus::run(const float *input, float **outputs, uint32_t frames) {
     float lfoMod, dly1, frac;
     uint16_t tap, delay;
 
+    fastOmega = sampleRate*4;
+
     for (uint32_t i = 0; i < frames; i++) {
         // run a step of LFO
         fastPhase += fastOmega;

plugin/chorus.hpp

@@ -16,11 +16,14 @@
    CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
 */
 
-#ifndef Chorus_HPP
-#define Chorus_HPP
+#ifndef __CHORUS_HPP
+#define __CHORUS_HPP
 
 #include "svf.hpp"
 
+extern uint32_t bufferSize;
+extern double sampleRate;
+
 // total size of delay line buffer
 #define DELAYSIZE 1028
 

plugin/module.cpp

@@ -84,7 +84,7 @@ void Module::run(Voice* voice) {
 
         px *= (patchRam.switch1 & 0x07);
 
-        v->omega = px / 192000.0f;  // fixme use proper scaler
+        v->omega = px / (sampleRate * 4.0f);  // fixme use proper scaler
 
         // per voice we need to calculate the key follow amount and envelope amount
         v->vcfCut = (patchRam.vcfFreq << 7) + ((vcf * v->env) >> 16);

plugin/module.hpp

@@ -23,15 +23,16 @@
 
 #include "DistrhoPluginInfo.h"
 
+extern double sampleRate;
+
 class Voice;
 
 class Module {
-
    public:
     Module();
 
     void run(Voice* voice);
-    // Voice voices[NUM_VOICES];
+
     float res = 0;
     // precomputed values for all voices
     float pw;  //, saw, square, sub;
@@ -66,11 +67,11 @@ class Module {
 
    private:
     // controls
-    float subRC = 0, outRC = 0, pwmRC = 0, resRC = 0, noiseRC = 0;
 };
 
 class Voice {
     friend Module;
+
    public:
     Voice();
     void on(uint8_t midiNote);

plugin/peacock.cpp

@@ -18,22 +18,27 @@
 
 #include "peacock.hpp"
 
+double sampleRate;
+uint32_t bufferSize;
+
 START_NAMESPACE_DISTRHO
 
 Peacock::Peacock() : Plugin(parameterCount, 0, 0) {
     d_debug("peacock constructor\n");
+
     sampleRate = getSampleRate();
-    m = new Module;
+    bufferSize = getBufferSize();
+    m = new Module();
     ic1 = new Assigner;
     ic1->voice = voice;
-    chorus = new Chorus;
+    chorus = new Chorus();
+
 
 }
 
 Peacock::~Peacock() {
-    free(m);
-    free(ic1);
-    free(chorus);
+    printf("peacock destructor\n");
+
 }
 
 void Peacock::initAudioPort(bool input, uint32_t index, AudioPort& port) {

plugin/peacock.hpp

@@ -21,8 +21,11 @@
 
 #include "DistrhoPlugin.hpp"
 #include "assigner.hpp"
-#include "module.hpp"
 #include "chorus.hpp"
+#include "module.hpp"
+
+extern double sampleRate;
+extern uint32_t bufferSize;
 
 START_NAMESPACE_DISTRHO
 
@@ -31,7 +34,6 @@ class Peacock : public Plugin {
     Peacock();
     ~Peacock();
 
- 
    protected:
     const char* getLabel() const override { return "peacock-8"; }
     const char* getDescription() const override {
@@ -59,8 +61,6 @@ class Peacock : public Plugin {
     Module* m;
     Chorus* chorus;
 
-    uint32_t sampleRate;
-
     // variables for breaking up the 4.3 millisecond module board ticks
     uint32_t lastEvent = 0;  // event number of last MIDI event processed in a chunk
     uint32_t framesLeft = 0, blockLeft = 0;

plugin/svf.hpp

@@ -16,8 +16,8 @@
    CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
 */
 
-#ifndef SVF_HPP
-#define SVF_HPP
+#ifndef __SVF_HPP
+#define __SVF_HPP
 
 #include <stdint.h>