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diff --git a/src/mdaVocoder.cpp b/src/mdaVocoder.cpp
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+/*
+ Copyright 2008-2011 David Robillard <http://drobilla.net>
+ Copyright 1999-2000 Paul Kellett (Maxim Digital Audio)
+
+ This is free software: you can redistribute it and/or modify it
+ under the terms of the GNU General Public License as published by
+ the Free Software Foundation, either version 3 of the License,
+ or (at your option) any later version.
+
+ This software is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
+ See the GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this software. If not, see <http://www.gnu.org/licenses/>.
+*/
+
+#include "mdaVocoder.h"
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <float.h>
+#include <math.h>
+
+AudioEffect *createEffectInstance(audioMasterCallback audioMaster)
+{
+ return new mdaVocoder(audioMaster);
+}
+
+mdaVocoderProgram::mdaVocoderProgram() ///default program settings
+{
+ param[0] = 0.0f; //input select
+ param[1] = 0.50f; //output dB
+ param[2] = 0.40f; //hi thru
+ param[3] = 0.40f; //hi band
+ param[4] = 0.16f; //envelope
+ param[5] = 0.55f; //filter q
+ param[6] = 0.6667f;//freq range
+ param[7] = 0.0f; //num bands
+ strcpy(name, "Vocoder");
+}
+
+
+mdaVocoder::mdaVocoder(audioMasterCallback audioMaster): AudioEffectX(audioMaster, NPROGS, NPARAMS)
+{
+ setNumInputs(2);
+ setNumOutputs(2);
+ setUniqueID("mdaVocoder"); ///identify plug-in here
+ //canMono();
+ canProcessReplacing();
+
+ programs = new mdaVocoderProgram[NPROGS];
+ setProgram(0);
+
+ ///differences from default program...
+ programs[1].param[7] = 1.0f;
+ strcpy(programs[1].name,"16 Band Vocoder");
+ programs[2].param[2] = 0.00f;
+ programs[2].param[3] = 0.00f;
+ programs[2].param[6] = 0.50f;
+ strcpy(programs[2].name,"Old Vocoder");
+ programs[3].param[3] = 0.00f;
+ programs[3].param[5] = 0.70f;
+ programs[3].param[6] = 0.50f;
+ strcpy(programs[3].name,"Choral Vocoder");
+ programs[4].param[4] = 0.78f;
+ programs[4].param[6] = 0.30f;
+ strcpy(programs[4].name,"Pad Vocoder");
+
+ suspend();
+}
+
+bool mdaVocoder::getProductString(char* text) { strcpy(text, "MDA Vocoder"); return true; }
+bool mdaVocoder::getVendorString(char* text) { strcpy(text, "mda"); return true; }
+bool mdaVocoder::getEffectName(char* name) { strcpy(name, "Vocoder"); return true; }
+
+void mdaVocoder::resume() ///update internal parameters...
+{
+ float * param = programs[curProgram].param;
+ double tpofs = 6.2831853/getSampleRate();
+ double rr, th; //, re;
+ float sh;
+ int32_t i;
+
+ swap = 1; if(param[0]>0.5f) swap = 0;
+ gain = (float)pow(10.0f, 2.0f * param[1] - 3.0f * param[5] - 2.0f);
+
+ thru = (float)pow(10.0f, 0.5f + 2.0f * param[1]);
+ high = param[3] * param[3] * param[3] * thru;
+ thru *= param[2] * param[2] * param[2];
+
+ if(param[7]<=0.0f)
+ {
+ nbnd=8;
+ //re=0.003f;
+ f[1][2] = 3000.0f;
+ f[2][2] = 2200.0f;
+ f[3][2] = 1500.0f;
+ f[4][2] = 1080.0f;
+ f[5][2] = 700.0f;
+ f[6][2] = 390.0f;
+ f[7][2] = 190.0f;
+ }
+ else
+ {
+ nbnd=16;
+ //re=0.0015f;
+ f[ 1][2] = 5000.0f; //+1000
+ f[ 2][2] = 4000.0f; //+750
+ f[ 3][2] = 3250.0f; //+500
+ f[ 4][2] = 2750.0f; //+450
+ f[ 5][2] = 2300.0f; //+300
+ f[ 6][2] = 2000.0f; //+250
+ f[ 7][2] = 1750.0f; //+250
+ f[ 8][2] = 1500.0f; //+250
+ f[ 9][2] = 1250.0f; //+250
+ f[10][2] = 1000.0f; //+250
+ f[11][2] = 750.0f; //+210
+ f[12][2] = 540.0f; //+190
+ f[13][2] = 350.0f; //+155
+ f[14][2] = 195.0f; //+100
+ f[15][2] = 95.0f;
+ }
+
+ if(param[4]<0.05f) //freeze
+ {
+ for(i=0;i<nbnd;i++) f[i][12]=0.0f;
+ }
+ else
+ {
+ f[0][12] = (float)pow(10.0, -1.7 - 2.7f * param[4]); //envelope speed
+
+ rr = 0.022f / (float)nbnd; //minimum proportional to frequency to stop distortion
+ for(i=1;i<nbnd;i++)
+ {
+ f[i][12] = (float)(0.025 - rr * (double)i);
+ if(f[0][12] < f[i][12]) f[i][12] = f[0][12];
+ }
+ f[0][12] = 0.5f * f[0][12]; //only top band is at full rate
+ }
+
+ rr = 1.0 - pow(10.0f, -1.0f - 1.2f * param[5]);
+ sh = (float)pow(2.0f, 3.0f * param[6] - 1.0f); //filter bank range shift
+
+ for(i=1;i<nbnd;i++)
+ {
+ f[i][2] *= sh;
+ th = acos((2.0 * rr * cos(tpofs * f[i][2])) / (1.0 + rr * rr));
+ f[i][0] = (float)(2.0 * rr * cos(th)); //a0
+ f[i][1] = (float)(-rr * rr); //a1
+ //was .98
+ f[i][2] *= 0.96f; //shift 2nd stage slightly to stop high resonance peaks
+ th = acos((2.0 * rr * cos(tpofs * f[i][2])) / (1.0 + rr * rr));
+ f[i][2] = (float)(2.0 * rr * cos(th));
+ }
+}
+
+
+void mdaVocoder::suspend() ///clear any buffers...
+{
+ int32_t i, j;
+
+ for(i=0; i<nbnd; i++) for(j=3; j<12; j++) f[i][j] = 0.0f; //zero band filters and envelopes
+ kout = 0.0f;
+ kval = 0;
+}
+
+
+mdaVocoder::~mdaVocoder() ///destroy any buffers...
+{
+ if(programs) delete [] programs;
+}
+
+
+void mdaVocoder::setProgram(int32_t program)
+{
+ curProgram = program;
+ resume();
+}
+
+
+void mdaVocoder::setParameter(int32_t index, float value)
+{
+ programs[curProgram].param[index] = value;
+ resume();
+}
+float mdaVocoder::getParameter(int32_t index) { return programs[curProgram].param[index]; }
+void mdaVocoder::setProgramName(char *name) { strcpy(programs[curProgram].name, name); }
+void mdaVocoder::getProgramName(char *name) { strcpy(name, programs[curProgram].name); }
+bool mdaVocoder::getProgramNameIndexed (int32_t category, int32_t index, char* name)
+{
+ if ((unsigned int)index < NPROGS)
+ {
+ strcpy(name, programs[index].name);
+ return true;
+ }
+ return false;
+}
+
+void mdaVocoder::getParameterName(int32_t index, char *label)
+{
+ switch(index)
+ {
+ case 0: strcpy(label, "Mod In"); break;
+ case 1: strcpy(label, "Output"); break;
+ case 2: strcpy(label, "Hi Thru"); break;
+ case 3: strcpy(label, "Hi Band"); break;
+ case 4: strcpy(label, "Envelope"); break;
+ case 5: strcpy(label, "Filter Q"); break;
+ case 6: strcpy(label, "Mid Freq"); break;
+ default: strcpy(label, "Quality");
+ }
+}
+
+
+void mdaVocoder::getParameterDisplay(int32_t index, char *text)
+{
+ char string[16];
+ float * param = programs[curProgram].param;
+
+ switch(index)
+ {
+ case 0: if(swap) strcpy(string, "RIGHT"); else strcpy(string, "LEFT"); break;
+ case 1: sprintf(string, "%.1f", 40.0f * param[index] - 20.0f); break;
+ case 4: if(param[index]<0.05f) strcpy(string, "FREEZE");
+ else sprintf(string, "%.1f", (float)pow(10.0f, 1.0f + 3.0f * param[index]));
+ break;
+ case 6: sprintf(string, "%.0f", 800.0f * (float)pow(2.0f, 3.0f * param[index] - 2.0f)); break;
+ case 7: if(nbnd==8) strcpy(string, "8 BAND"); else strcpy(string, "16 BAND"); break;
+
+ default: sprintf(string, "%.0f", 100.0f * param[index]);
+ }
+ string[8] = 0;
+ strcpy(text, (char *)string);
+}
+
+
+void mdaVocoder::getParameterLabel(int32_t index, char *label)
+{
+ switch(index)
+ {
+ case 7:
+ case 0: strcpy(label, ""); break;
+ case 1: strcpy(label, "dB"); break;
+ case 4: strcpy(label, "ms"); break;
+ case 6: strcpy(label, "Hz"); break;
+ default: strcpy(label, "%");
+ }
+}
+
+
+void mdaVocoder::process(float **inputs, float **outputs, int32_t sampleFrames)
+{
+ float *in1 = inputs[0];
+ float *in2 = inputs[1];
+ float *out1 = outputs[0];
+ float *out2 = outputs[1];
+ float a, b, c, d, o=0.0f, aa, bb, oo=kout, g=gain, ht=thru, hh=high, tmp;
+ int32_t i, k=kval, sw=swap, nb=nbnd;
+
+ --in1;
+ --in2;
+ --out1;
+ --out2;
+ while(--sampleFrames >= 0)
+ {
+ a = *++in1; //speech
+ b = *++in2; //synth
+ c = out1[1];
+ d = out2[1];
+ if(sw==0) { tmp=a; a=b; b=tmp; } //swap channels
+
+ tmp = a - f[0][7]; //integrate modulator for HF band and filter bank pre-emphasis
+ f[0][7] = a;
+ a = tmp;
+
+ if(tmp<0.0f) tmp = -tmp;
+ f[0][11] -= f[0][12] * (f[0][11] - tmp); //high band envelope
+ o = f[0][11] * (ht * a + hh * (b - f[0][3])); //high band + high thru
+
+ f[0][3] = b; //integrate carrier for HF band
+
+ if(++k & 0x1) //this block runs at half sample rate
+ {
+ oo = 0.0f;
+ aa = a + f[0][9] - f[0][8] - f[0][8]; //apply zeros here instead of in each reson
+ f[0][9] = f[0][8]; f[0][8] = a;
+ bb = b + f[0][5] - f[0][4] - f[0][4];
+ f[0][5] = f[0][4]; f[0][4] = b;
+
+ for(i=1; i<nb; i++) //filter bank: 4th-order band pass
+ {
+ tmp = f[i][0] * f[i][3] + f[i][1] * f[i][4] + bb;
+ f[i][4] = f[i][3];
+ f[i][3] = tmp;
+ tmp += f[i][2] * f[i][5] + f[i][1] * f[i][6];
+ f[i][6] = f[i][5];
+ f[i][5] = tmp;
+
+ tmp = f[i][0] * f[i][7] + f[i][1] * f[i][8] + aa;
+ f[i][8] = f[i][7];
+ f[i][7] = tmp;
+ tmp += f[i][2] * f[i][9] + f[i][1] * f[i][10];
+ f[i][10] = f[i][9];
+ f[i][9] = tmp;
+
+ if(tmp<0.0f) tmp = -tmp;
+ f[i][11] -= f[i][12] * (f[i][11] - tmp);
+ oo += f[i][5] * f[i][11];
+ }
+ }
+ o += oo * g; //effect of interpolating back up to Fs would be minimal (aliasing >16kHz)
+
+ *++out1 = c + o;
+ *++out2 = d + o;
+ }
+
+ kout = oo;
+ kval = k & 0x1;
+ if(fabs(f[0][11])<1.0e-10) f[0][11] = 0.0f; //catch HF envelope denormal
+
+ for(i=1;i<nb;i++)
+ if(fabs(f[i][3])<1.0e-10 || fabs(f[i][7])<1.0e-10)
+ for(k=3; k<12; k++) f[i][k] = 0.0f; //catch reson & envelope denormals
+
+ if(fabs(o)>10.0f) suspend(); //catch instability
+}
+
+
+void mdaVocoder::processReplacing(float **inputs, float **outputs, int32_t sampleFrames)
+{
+ float *in1 = inputs[0];
+ float *in2 = inputs[1];
+ float *out1 = outputs[0];
+ float *out2 = outputs[1];
+ float a, b, o=0.0f, aa, bb, oo=kout, g=gain, ht=thru, hh=high, tmp;
+ int32_t i, k=kval, sw=swap, nb=nbnd;
+
+ --in1;
+ --in2;
+ --out1;
+ --out2;
+ while(--sampleFrames >= 0)
+ {
+ a = *++in1; //speech
+ b = *++in2; //synth
+ if(sw==0) { tmp=a; a=b; b=tmp; } //swap channels
+
+ tmp = a - f[0][7]; //integrate modulator for HF band and filter bank pre-emphasis
+ f[0][7] = a;
+ a = tmp;
+
+ if(tmp<0.0f) tmp = -tmp;
+ f[0][11] -= f[0][12] * (f[0][11] - tmp); //high band envelope
+ o = f[0][11] * (ht * a + hh * (b - f[0][3])); //high band + high thru
+
+ f[0][3] = b; //integrate carrier for HF band
+
+ if(++k & 0x1) //this block runs at half sample rate
+ {
+ oo = 0.0f;
+ aa = a + f[0][9] - f[0][8] - f[0][8]; //apply zeros here instead of in each reson
+ f[0][9] = f[0][8]; f[0][8] = a;
+ bb = b + f[0][5] - f[0][4] - f[0][4];
+ f[0][5] = f[0][4]; f[0][4] = b;
+
+ for(i=1; i<nb; i++) //filter bank: 4th-order band pass
+ {
+ tmp = f[i][0] * f[i][3] + f[i][1] * f[i][4] + bb;
+ f[i][4] = f[i][3];
+ f[i][3] = tmp;
+ tmp += f[i][2] * f[i][5] + f[i][1] * f[i][6];
+ f[i][6] = f[i][5];
+ f[i][5] = tmp;
+
+ tmp = f[i][0] * f[i][7] + f[i][1] * f[i][8] + aa;
+ f[i][8] = f[i][7];
+ f[i][7] = tmp;
+ tmp += f[i][2] * f[i][9] + f[i][1] * f[i][10];
+ f[i][10] = f[i][9];
+ f[i][9] = tmp;
+
+ if(tmp<0.0f) tmp = -tmp;
+ f[i][11] -= f[i][12] * (f[i][11] - tmp);
+ oo += f[i][5] * f[i][11];
+ }
+ }
+ o += oo * g; //effect of interpolating back up to Fs would be minimal (aliasing >16kHz)
+
+ *++out1 = o;
+ *++out2 = o;
+ }
+
+ kout = oo;
+ kval = k & 0x1;
+ if(fabs(f[0][11])<1.0e-10) f[0][11] = 0.0f; //catch HF envelope denormal
+
+ for(i=1;i<nb;i++)
+ if(fabs(f[i][3])<1.0e-10 || fabs(f[i][7])<1.0e-10)
+ for(k=3; k<12; k++) f[i][k] = 0.0f; //catch reson & envelope denormals
+
+ if(fabs(o)>10.0f) suspend(); //catch instability
+}
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