initial commit
This commit is contained in:
Christopher Herb
100
clip/aw_cliponly2.h
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100
clip/aw_cliponly2.h
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#pragma once
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#include <cstdlib>
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namespace trnr::lib::clip {
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// Clipper based on ClipOnly2 by Chris Johnson
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class aw_cliponly2 {
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public:
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aw_cliponly2() {
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samplerate = 44100;
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lastSampleL = 0.0;
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wasPosClipL = false;
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wasNegClipL = false;
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lastSampleR = 0.0;
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wasPosClipR = false;
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wasNegClipR = false;
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for (int x = 0; x < 16; x++) {intermediateL[x] = 0.0; intermediateR[x] = 0.0;}
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//this is reset: values being initialized only once. Startup values, whatever they are.
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}
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void set_samplerate(double _samplerate) {
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samplerate = _samplerate;
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}
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void process_block(double** inputs, double** outputs, long sample_frames) {
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= samplerate;
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int spacing = floor(overallscale); //should give us working basic scaling, usually 2 or 4
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if (spacing < 1) spacing = 1; if (spacing > 16) spacing = 16;
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while (--sample_frames >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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//begin ClipOnly2 stereo as a little, compressed chunk that can be dropped into code
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if (inputSampleL > 4.0) inputSampleL = 4.0; if (inputSampleL < -4.0) inputSampleL = -4.0;
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if (wasPosClipL == true) { //current will be over
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if (inputSampleL<lastSampleL) lastSampleL=0.7058208+(inputSampleL*0.2609148);
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else lastSampleL = 0.2491717+(lastSampleL*0.7390851);
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} wasPosClipL = false;
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if (inputSampleL>0.9549925859) {wasPosClipL=true;inputSampleL=0.7058208+(lastSampleL*0.2609148);}
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if (wasNegClipL == true) { //current will be -over
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if (inputSampleL > lastSampleL) lastSampleL=-0.7058208+(inputSampleL*0.2609148);
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else lastSampleL=-0.2491717+(lastSampleL*0.7390851);
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} wasNegClipL = false;
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if (inputSampleL<-0.9549925859) {wasNegClipL=true;inputSampleL=-0.7058208+(lastSampleL*0.2609148);}
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intermediateL[spacing] = inputSampleL;
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inputSampleL = lastSampleL; //Latency is however many samples equals one 44.1k sample
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for (int x = spacing; x > 0; x--) intermediateL[x-1] = intermediateL[x];
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lastSampleL = intermediateL[0]; //run a little buffer to handle this
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if (inputSampleR > 4.0) inputSampleR = 4.0; if (inputSampleR < -4.0) inputSampleR = -4.0;
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if (wasPosClipR == true) { //current will be over
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if (inputSampleR<lastSampleR) lastSampleR=0.7058208+(inputSampleR*0.2609148);
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else lastSampleR = 0.2491717+(lastSampleR*0.7390851);
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} wasPosClipR = false;
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if (inputSampleR>0.9549925859) {wasPosClipR=true;inputSampleR=0.7058208+(lastSampleR*0.2609148);}
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if (wasNegClipR == true) { //current will be -over
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if (inputSampleR > lastSampleR) lastSampleR=-0.7058208+(inputSampleR*0.2609148);
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else lastSampleR=-0.2491717+(lastSampleR*0.7390851);
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} wasNegClipR = false;
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if (inputSampleR<-0.9549925859) {wasNegClipR=true;inputSampleR=-0.7058208+(lastSampleR*0.2609148);}
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intermediateR[spacing] = inputSampleR;
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inputSampleR = lastSampleR; //Latency is however many samples equals one 44.1k sample
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for (int x = spacing; x > 0; x--) intermediateR[x-1] = intermediateR[x];
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lastSampleR = intermediateR[0]; //run a little buffer to handle this
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//end ClipOnly2 stereo as a little, compressed chunk that can be dropped into code
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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in1++;
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in2++;
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out1++;
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out2++;
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}
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}
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private:
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double samplerate;
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double lastSampleL;
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double intermediateL[16];
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bool wasPosClipL;
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bool wasNegClipL;
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double lastSampleR;
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double intermediateR[16];
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bool wasPosClipR;
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bool wasNegClipR; //Stereo ClipOnly2
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//default stuff
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};
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}
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97
clip/aw_clipsoftly.h
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97
clip/aw_clipsoftly.h
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#pragma once
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#include <cstdlib>
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#include <stdint.h>
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namespace trnr::lib::clip {
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// soft clipper based on ClipSoftly by Chris Johnson
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class aw_clipsoftly {
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public:
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aw_clipsoftly() {
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samplerate = 44100;
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lastSampleL = 0.0;
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lastSampleR = 0.0;
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for (int x = 0; x < 16; x++) {intermediateL[x] = 0.0; intermediateR[x] = 0.0;}
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fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
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fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
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//this is reset: values being initialized only once. Startup values, whatever they are.
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}
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void set_samplerate(double _samplerate) {
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samplerate = _samplerate;
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}
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void process_block(double** inputs, double** outputs, long sample_frames) {
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= samplerate;
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int spacing = floor(overallscale); //should give us working basic scaling, usually 2 or 4
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if (spacing < 1) spacing = 1; if (spacing > 16) spacing = 16;
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while (--sample_frames >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
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if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
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double softSpeed = fabs(inputSampleL);
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if (softSpeed < 1.0) softSpeed = 1.0; else softSpeed = 1.0/softSpeed;
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if (inputSampleL > 1.57079633) inputSampleL = 1.57079633;
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if (inputSampleL < -1.57079633) inputSampleL = -1.57079633;
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inputSampleL = sin(inputSampleL)*0.9549925859; //scale to what cliponly uses
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inputSampleL = (inputSampleL*softSpeed)+(lastSampleL*(1.0-softSpeed));
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softSpeed = fabs(inputSampleR);
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if (softSpeed < 1.0) softSpeed = 1.0; else softSpeed = 1.0/softSpeed;
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if (inputSampleR > 1.57079633) inputSampleR = 1.57079633;
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if (inputSampleR < -1.57079633) inputSampleR = -1.57079633;
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inputSampleR = sin(inputSampleR)*0.9549925859; //scale to what cliponly uses
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inputSampleR = (inputSampleR*softSpeed)+(lastSampleR*(1.0-softSpeed));
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intermediateL[spacing] = inputSampleL;
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inputSampleL = lastSampleL; //Latency is however many samples equals one 44.1k sample
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for (int x = spacing; x > 0; x--) intermediateL[x-1] = intermediateL[x];
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lastSampleL = intermediateL[0]; //run a little buffer to handle this
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intermediateR[spacing] = inputSampleR;
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inputSampleR = lastSampleR; //Latency is however many samples equals one 44.1k sample
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for (int x = spacing; x > 0; x--) intermediateR[x-1] = intermediateR[x];
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lastSampleR = intermediateR[0]; //run a little buffer to handle this
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//begin 64 bit stereo floating point dither
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//int expon; frexp((double)inputSampleL, &expon);
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fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
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//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//frexp((double)inputSampleR, &expon);
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fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
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//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//end 64 bit stereo floating point dither
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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in1++;
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in2++;
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out1++;
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out2++;
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}
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}
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private:
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double samplerate;
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double lastSampleL;
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double intermediateL[16];
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double lastSampleR;
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double intermediateR[16];
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uint32_t fpdL;
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uint32_t fpdR;
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//default stuff
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};
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}
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193
clip/aw_tube2.h
Normal file
193
clip/aw_tube2.h
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@@ -0,0 +1,193 @@
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#pragma once
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#include <cstdlib>
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#include <stdint.h>
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namespace trnr::lib::clip {
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// modeled tube preamp based on tube2 by Chris Johnson
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class aw_tube2 {
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public:
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aw_tube2() {
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samplerate = 44100;
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A = 0.5;
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B = 0.5;
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previousSampleA = 0.0;
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previousSampleB = 0.0;
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previousSampleC = 0.0;
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previousSampleD = 0.0;
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previousSampleE = 0.0;
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previousSampleF = 0.0;
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fpdL = 1.0; while (fpdL < 16386) fpdL = rand()*UINT32_MAX;
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fpdR = 1.0; while (fpdR < 16386) fpdR = rand()*UINT32_MAX;
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//this is reset: values being initialized only once. Startup values, whatever they are.
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}
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void set_input(double value) {
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A = clamp(value);
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}
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void set_tube(double value) {
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B = clamp(value);
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}
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void set_samplerate(double _samplerate) {
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samplerate = _samplerate;
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}
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void process_block(double **inputs, double **outputs, long sampleframes) {
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double* in1 = inputs[0];
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double* in2 = inputs[1];
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double* out1 = outputs[0];
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double* out2 = outputs[1];
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double overallscale = 1.0;
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overallscale /= 44100.0;
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overallscale *= samplerate;
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double inputPad = A;
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double iterations = 1.0-B;
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int powerfactor = (9.0*iterations)+1;
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double asymPad = (double)powerfactor;
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double gainscaling = 1.0/(double)(powerfactor+1);
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double outputscaling = 1.0 + (1.0/(double)(powerfactor));
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while (--sampleframes >= 0)
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{
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double inputSampleL = *in1;
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double inputSampleR = *in2;
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if (fabs(inputSampleL)<1.18e-23) inputSampleL = fpdL * 1.18e-17;
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if (fabs(inputSampleR)<1.18e-23) inputSampleR = fpdR * 1.18e-17;
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if (inputPad < 1.0) {
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inputSampleL *= inputPad;
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inputSampleR *= inputPad;
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}
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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleA; previousSampleA = stored; inputSampleL *= 0.5;
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stored = inputSampleR;
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inputSampleR += previousSampleB; previousSampleB = stored; inputSampleR *= 0.5;
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} //for high sample rates on this plugin we are going to do a simple average
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if (inputSampleL > 1.0) inputSampleL = 1.0;
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if (inputSampleL < -1.0) inputSampleL = -1.0;
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if (inputSampleR > 1.0) inputSampleR = 1.0;
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if (inputSampleR < -1.0) inputSampleR = -1.0;
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//flatten bottom, point top of sine waveshaper L
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inputSampleL /= asymPad;
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double sharpen = -inputSampleL;
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if (sharpen > 0.0) sharpen = 1.0+sqrt(sharpen);
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else sharpen = 1.0-sqrt(-sharpen);
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inputSampleL -= inputSampleL*fabs(inputSampleL)*sharpen*0.25;
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//this will take input from exactly -1.0 to 1.0 max
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inputSampleL *= asymPad;
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//flatten bottom, point top of sine waveshaper R
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inputSampleR /= asymPad;
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sharpen = -inputSampleR;
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if (sharpen > 0.0) sharpen = 1.0+sqrt(sharpen);
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else sharpen = 1.0-sqrt(-sharpen);
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inputSampleR -= inputSampleR*fabs(inputSampleR)*sharpen*0.25;
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//this will take input from exactly -1.0 to 1.0 max
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inputSampleR *= asymPad;
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//end first asym section: later boosting can mitigate the extreme
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//softclipping of one side of the wave
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//and we are asym clipping more when Tube is cranked, to compensate
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//original Tube algorithm: powerfactor widens the more linear region of the wave
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double factor = inputSampleL; //Left channel
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for (int x = 0; x < powerfactor; x++) factor *= inputSampleL;
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if ((powerfactor % 2 == 1) && (inputSampleL != 0.0)) factor = (factor/inputSampleL)*fabs(inputSampleL);
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factor *= gainscaling;
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inputSampleL -= factor;
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inputSampleL *= outputscaling;
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factor = inputSampleR; //Right channel
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for (int x = 0; x < powerfactor; x++) factor *= inputSampleR;
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if ((powerfactor % 2 == 1) && (inputSampleR != 0.0)) factor = (factor/inputSampleR)*fabs(inputSampleR);
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factor *= gainscaling;
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inputSampleR -= factor;
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inputSampleR *= outputscaling;
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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleC; previousSampleC = stored; inputSampleL *= 0.5;
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stored = inputSampleR;
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inputSampleR += previousSampleD; previousSampleD = stored; inputSampleR *= 0.5;
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} //for high sample rates on this plugin we are going to do a simple average
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//end original Tube. Now we have a boosted fat sound peaking at 0dB exactly
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//hysteresis and spiky fuzz L
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double slew = previousSampleE - inputSampleL;
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if (overallscale > 1.9) {
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double stored = inputSampleL;
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inputSampleL += previousSampleE; previousSampleE = stored; inputSampleL *= 0.5;
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} else previousSampleE = inputSampleL; //for this, need previousSampleC always
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if (slew > 0.0) slew = 1.0+(sqrt(slew)*0.5);
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else slew = 1.0-(sqrt(-slew)*0.5);
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inputSampleL -= inputSampleL*fabs(inputSampleL)*slew*gainscaling;
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//reusing gainscaling that's part of another algorithm
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if (inputSampleL > 0.52) inputSampleL = 0.52;
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if (inputSampleL < -0.52) inputSampleL = -0.52;
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inputSampleL *= 1.923076923076923;
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//hysteresis and spiky fuzz R
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slew = previousSampleF - inputSampleR;
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if (overallscale > 1.9) {
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double stored = inputSampleR;
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inputSampleR += previousSampleF; previousSampleF = stored; inputSampleR *= 0.5;
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} else previousSampleF = inputSampleR; //for this, need previousSampleC always
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if (slew > 0.0) slew = 1.0+(sqrt(slew)*0.5);
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else slew = 1.0-(sqrt(-slew)*0.5);
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inputSampleR -= inputSampleR*fabs(inputSampleR)*slew*gainscaling;
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//reusing gainscaling that's part of another algorithm
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if (inputSampleR > 0.52) inputSampleR = 0.52;
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if (inputSampleR < -0.52) inputSampleR = -0.52;
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inputSampleR *= 1.923076923076923;
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//end hysteresis and spiky fuzz section
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//begin 64 bit stereo floating point dither
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//int expon; frexp((double)inputSampleL, &expon);
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fpdL ^= fpdL << 13; fpdL ^= fpdL >> 17; fpdL ^= fpdL << 5;
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//inputSampleL += ((double(fpdL)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//frexp((double)inputSampleR, &expon);
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fpdR ^= fpdR << 13; fpdR ^= fpdR >> 17; fpdR ^= fpdR << 5;
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//inputSampleR += ((double(fpdR)-uint32_t(0x7fffffff)) * 1.1e-44l * pow(2,expon+62));
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//end 64 bit stereo floating point dither
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*out1 = inputSampleL;
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*out2 = inputSampleR;
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in1++;
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in2++;
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out1++;
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out2++;
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}
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}
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private:
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double samplerate;
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double previousSampleA;
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double previousSampleB;
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double previousSampleC;
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double previousSampleD;
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double previousSampleE;
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double previousSampleF;
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uint32_t fpdL;
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uint32_t fpdR;
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//default stuff
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float A;
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float B;
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double clamp(double& value) {
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if (value > 1) {
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value = 1;
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} else if (value < 0) {
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value = 0;
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}
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return value;
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}
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};
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}
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