added filter, not properly connected
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202ddac184
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@ -14,6 +14,7 @@ FILES_DSP = \
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controls.cpp \
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ic1.cpp \
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oscillator.cpp \
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filter.cpp \
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ic29.cpp
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include ../dpf/Makefile.plugins.mk
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@ -35,6 +35,15 @@ void Synth::buildTables(double sampleRate) {
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// on the real synth the tuning knob is tweaked a little off to pull it in
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pitchTable[i] = 260.15f * powf(2, (i - 36) / 12.0f) / sampleRate;
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}
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// precompute a table of values to map the filter value to a filter coefficient
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// the ROM preset for adjusting the VCF scale and centre presets cutoff to $31
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// and key scale to $7f, which corresponds to C4 = 248Hz and C6 = 992Hz, B3 and B5
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for (uint16_t i = 0; i< 256; i++) {
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}
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}
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void Synth::run() {
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@ -73,11 +73,13 @@ class Voice {
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Voice();
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uint8_t note = 0x3c; // middle C
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void run(float *buffer, uint32_t pos, uint32_t samples);
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void filter(float *buffer, uint32_t pos, uint32_t samples);
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void update();
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void on(uint8_t note);
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void off();
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private:
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// private:
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Envelope env; // calculated envelope value
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uint16_t pitch = 0x1818; // calculated pitch value with porta and master pitch etc
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float delay = 0; // delay slot for polyblep
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@ -91,6 +93,8 @@ class Voice {
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V_ON } voiceState;
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void calcPitch();
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float fb=0, b1=0, b2=0, b3=0, b4=0, cut=0.1, reso=4;
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};
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class Synth {
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@ -113,6 +117,7 @@ class Synth {
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int8_t portaCoeff;
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bool sustained;
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double pitchTable[104];
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double filterTable[256];
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float *noise;
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@ -30,8 +30,7 @@ static inline float poly3blep1(float t) {
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void Voice::run(float *buffer, uint32_t pos, uint32_t samples) {
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// generate a full block of samples for the oscillator
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float y, out, t;
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float gain = env.level / 16384.0;
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float y, t;
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if (subosc > 0) subosc = sub; else subosc = -sub;
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@ -71,9 +70,8 @@ void Voice::run(float *buffer, uint32_t pos, uint32_t samples) {
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delay += subosc;
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delay += ic29.noise[i+pos];
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out = y * 0.15;
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lastpw = pwrc;
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buffer[i + pos] += out * gain;
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buffer[i + pos] = y * 0.707;
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}
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}
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@ -29,8 +29,6 @@ Peacock::Peacock() : Plugin(paramCount, 0, 0), sampleRate(getSampleRate()) {
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ic29.bufferSize = getBufferSize();
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}
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void Peacock::runMidi(const MidiEvent *ev, uint32_t count, uint32_t timeLimit) {
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// handle MIDI events, starting at lastEvent and continuing until timeLimit
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uint32_t i;
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@ -52,7 +50,6 @@ void Peacock::initAudioPort(bool input, uint32_t index, AudioPort &port) {
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if (!input && index == 1) port.name = "Right Out";
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}
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void Peacock::run(const float **, float **outputs, uint32_t frames, const MidiEvent *midiEvents, uint32_t midiEventCount) {
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// calculate an entire jack period's worth of samples
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// harder than it sounds because for short jack periods there may be many
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@ -73,7 +70,7 @@ void Peacock::run(const float **, float **outputs, uint32_t frames, const MidiEv
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runMidi(midiEvents, midiEventCount, blockLeft);
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// generate a buffer's worth of samples
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memset(outputs[0], 0, sizeof(float) * frames);
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memset(outputs[1], 0, sizeof(float) * frames);
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while (framePos < frames) {
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if (blockLeft == 0) {
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@ -88,6 +85,10 @@ void Peacock::run(const float **, float **outputs, uint32_t frames, const MidiEv
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// run every synth voice into the buffer here FIXME
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for (uint8_t i = 0; i < NUM_VOICES; i++) {
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ic29.voices[i].run(outputs[0], framePos, sizeThisTime);
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ic29.voices[i].filter(outputs[0], framePos, sizeThisTime);
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for (uint8_t j=0; j<sizeThisTime; j++) {
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outputs[1][framePos+j] += outputs[0][framePos+j] * ic29.voices[i].env.level/16384.0;
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}
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}
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framePos += sizeThisTime; // move along the frame
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@ -95,7 +96,7 @@ void Peacock::run(const float **, float **outputs, uint32_t frames, const MidiEv
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blockLeft -= sizeThisTime;
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}
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// output processing goes here
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memcpy(outputs[1], outputs[0], sizeof(float) * frames);
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memcpy(outputs[0], outputs[1], sizeof(float) * frames);
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}
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Plugin *createPlugin() { return new Peacock(); }
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