+ Monosynth: add inertia for pitch wheel

[calf.git] / src / calf / audio_fx.h

/* Calf DSP Library
 * Reusable audio effect classes.
 *
 * Copyright (C) 2001-2007 Krzysztof Foltman
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This program 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General
 * Public License along with this program; if not, write to the
 * Free Software Foundation, Inc., 59 Temple Place, Suite 330,
 * Boston, MA 02111-1307, USA.
 */
#ifndef __CALF_AUDIOFX_H
#define __CALF_AUDIOFX_H

#include <complex>
#include <iostream>
#include <calf/biquad.h>
#include "primitives.h"
#include "delay.h"
#include "fixed_point.h"
#include "inertia.h"

namespace dsp {
#if 0
}; to keep editor happy
#endif

/**
 * Audio effect base class. Not really useful until it gets more developed.
 */
class audio_effect
{
public:
    virtual void setup(int sample_rate)=0;
    virtual ~audio_effect() {}
};

class modulation_effect: public audio_effect
{
protected:
    int sample_rate;
    float rate, wet, dry, odsr;
    gain_smoothing gs_wet, gs_dry;
public:
    fixed_point<unsigned int, 20> phase, dphase;
    float get_rate() {
        return rate;
    }
    void set_rate(float rate) {
        this->rate = rate;
        dphase = rate/sample_rate*4096;        
    }
    float get_wet() {
        return wet;
    }
    void set_wet(float wet) {
        this->wet = wet;
        gs_wet.set_inertia(wet);
    }
    float get_dry() {
        return dry;
    }
    void set_dry(float dry) {
        this->dry = dry;
        gs_dry.set_inertia(dry);
    }
    void reset_phase(float req_phase)
    {
        phase = req_phase * 4096.0;
    }
    void inc_phase(float req_phase)
    {
        phase += fixed_point<unsigned int, 20>(req_phase * 4096.0);
    }
    void setup(int sample_rate)
    {
        this->sample_rate = sample_rate;
        this->odsr = 1.0 / sample_rate;
        phase = 0;
        set_rate(get_rate());
    }
};

/**
 * A monophonic phaser. If you want stereo, combine two :)
 * Also, gave up on using template args for signal type.
 */
template<int MaxStages>
class simple_phaser: public modulation_effect
{
protected:
    float base_frq, mod_depth, fb;
    float state;
    int cnt, stages;
    dsp::onepole<float, float> stage1;
    float x1[MaxStages], y1[MaxStages];
public:
    simple_phaser()
    {
        set_base_frq(1000);
        set_mod_depth(1000);
        set_fb(0);
        state = 0;
        cnt = 0;
        stages = 0;
        set_stages(6);
    }
    float get_base_frq() {
        return base_frq;
    }
    void set_base_frq(float _base_frq) {
        base_frq = _base_frq;
    }
    int get_stages() {
        return stages;
    }
    void set_stages(int _stages) {
        if (_stages > stages)
        {
            for (int i = stages; i < _stages; i++)
            {
                x1[i] = x1[stages-1];
                y1[i] = y1[stages-1];
            }
        }
        stages = _stages;
    }
    float get_mod_depth() {
        return mod_depth;
    }
    void set_mod_depth(float _mod_depth) {
        mod_depth = _mod_depth;
    }
    float get_fb() {
        return fb;
    }
    void set_fb(float fb) {
        this->fb = fb;
    }
    virtual void setup(int sample_rate) {
        modulation_effect::setup(sample_rate);
        reset();
    }
    void reset()
    {
        cnt = 0;
        state = 0;
        phase.set(0);
        for (int i = 0; i < MaxStages; i++)
            x1[i] = y1[i] = 0;
        control_step();
    }
    inline void control_step()
    {
        cnt = 0;
        int v = phase.get() + 0x40000000;
        int sign = v >> 31;
        v ^= sign;
        // triangle wave, range from 0 to INT_MAX
        double vf = (double)((v >> 16) * (1.0 / 16384.0) - 1);
        
        float freq = base_frq * pow(2.0, vf * mod_depth / 1200.0);
        freq = dsp::clip<float>(freq, 10.0, 0.49 * sample_rate);
        stage1.set_ap_w(freq * (M_PI / 2.0) * odsr);
        phase += dphase * 32;
        for (int i = 0; i < stages; i++)
        {
            dsp::sanitize(x1[i]);
            dsp::sanitize(y1[i]);
        }
        dsp::sanitize(state);
    }
    void process(float *buf_out, float *buf_in, int nsamples) {
        for (int i=0; i<nsamples; i++) {
            cnt++;
            if (cnt == 32)
                control_step();
            float in = *buf_in++;
            float fd = in + state * fb;
            for (int j = 0; j < stages; j++)
                fd = stage1.process_ap(fd, x1[j], y1[j]);
            state = fd;
            
            float sdry = in * gs_dry.get();
            float swet = fd * gs_wet.get();
            *buf_out++ = sdry + swet;
        }
    }
    float freq_gain(float freq, float sr)
    {
        typedef std::complex<double> cfloat;
        freq *= 2.0 * M_PI / sr;
        cfloat z = 1.0 / exp(cfloat(0.0, freq)); // z^-1
        
        cfloat p = cfloat(1.0);
        cfloat stg = stage1.h_z(z);
        
        for (int i = 0; i < stages; i++)
            p = p * stg;
        
        p = p / (cfloat(1.0) - cfloat(fb) * p);        
        return std::abs(cfloat(gs_dry.get_last()) + cfloat(gs_wet.get_last()) * p);
    }
};

/**
 * Base class for chorus and flanger. Wouldn't be needed if it wasn't
 * for odd behaviour of GCC when deriving templates from template
 * base classes (not seeing fields from base classes!).
 */
class chorus_base: public modulation_effect
{
protected:
    int min_delay_samples, mod_depth_samples;
    float min_delay, mod_depth;
    sine_table<int, 4096, 65536> sine;
public:
    float get_min_delay() {
        return min_delay;
    }
    void set_min_delay(float min_delay) {
        this->min_delay = min_delay;
        this->min_delay_samples = (int)(min_delay * 65536.0 * sample_rate);
    }
    float get_mod_depth() {
        return mod_depth;
    }
    void set_mod_depth(float mod_depth) {
        this->mod_depth = mod_depth;
        // 128 because it's then multiplied by (hopefully) a value of 32768..-32767
        this->mod_depth_samples = (int)(mod_depth * 32.0 * sample_rate);
    }
};

/**
 * Single-tap chorus without feedback.
 * Perhaps MaxDelay should be a bit longer!
 */
template<class T, int MaxDelay=512>
class simple_chorus: public chorus_base
{
protected:
    simple_delay<MaxDelay,T> delay;
public:    
    simple_chorus() {
        rate = 0.63f;
        dry = 0.5f;
        wet = 0.5f;
        min_delay = 0.005f;
        mod_depth = 0.0025f;
        setup(44100);
    }
    void reset() {
        delay.reset();
    }
    virtual void setup(int sample_rate) {
        modulation_effect::setup(sample_rate);
        delay.reset();
        set_min_delay(get_min_delay());
        set_mod_depth(get_mod_depth());
    }
    template<class OutIter, class InIter>
    void process(OutIter buf_out, InIter buf_in, int nsamples) {
        int mds = min_delay_samples + mod_depth_samples * 1024 + 2*65536;
        int mdepth = mod_depth_samples;
        for (int i=0; i<nsamples; i++) {
            phase += dphase;
            unsigned int ipart = phase.ipart();
            
            float in = *buf_in++;
            int lfo = phase.lerp_by_fract_int<int, 14, int>(sine.data[ipart], sine.data[ipart+1]);
            int v = mds + (mdepth * lfo >> 6);
            // if (!(i & 7)) printf("%d\n", v);
            int ifv = v >> 16;
            delay.put(in);
            T fd; // signal from delay's output
            delay.get_interp(fd, ifv, (v & 0xFFFF)*(1.0/65536.0));
            T sdry = in * gs_dry.get();
            T swet = fd * gs_wet.get();
            *buf_out++ = sdry + swet;
        }
    }
};

/**
 * Single-tap flanger (chorus plus feedback).
 */
template<class T, int MaxDelay=1024>
class simple_flanger: public chorus_base
{
protected:
    simple_delay<MaxDelay,T> delay;
    float fb;
    int last_delay_pos, last_actual_delay_pos;
    int ramp_pos, ramp_delay_pos;
public:
    simple_flanger()
    : fb(0) {}
    void reset() {
        delay.reset();
        last_delay_pos = 0;
        ramp_pos = 1024;
    }
    virtual void setup(int sample_rate) {
        this->sample_rate = sample_rate;
        this->odsr = 1.0 / sample_rate;
        delay.reset();
        phase = 0;
        set_rate(get_rate());
        set_min_delay(get_min_delay());
    }
    float get_fb() {
        return fb;
    }
    void set_fb(float fb) {
        this->fb = fb;
    }
    template<class OutIter, class InIter>
    void process(OutIter buf_out, InIter buf_in, int nsamples) {
        if (!nsamples)
            return;
        int mds = this->min_delay_samples + this->mod_depth_samples * 1024 + 2 * 65536;
        int mdepth = this->mod_depth_samples;
        int delay_pos;
        unsigned int ipart = this->phase.ipart();
        int lfo = phase.lerp_by_fract_int<int, 14, int>(this->sine.data[ipart], this->sine.data[ipart+1]);
        delay_pos = mds + (mdepth * lfo >> 6);
        
        if (delay_pos != last_delay_pos || ramp_pos < 1024)
        {
            if (delay_pos != last_delay_pos) {
                // we need to ramp from what the delay tap length actually was, 
                // not from old (ramp_delay_pos) or desired (delay_pos) tap length
                ramp_delay_pos = last_actual_delay_pos;
                ramp_pos = 0;
            }
            
            int64_t dp = 0;
            for (int i=0; i<nsamples; i++) {
                float in = *buf_in++;
                T fd; // signal from delay's output
                dp = (((int64_t)ramp_delay_pos) * (1024 - ramp_pos) + ((int64_t)delay_pos) * ramp_pos) >> 10;
                ramp_pos++;
                if (ramp_pos > 1024) ramp_pos = 1024;
                this->delay.get_interp(fd, dp >> 16, (dp & 0xFFFF)*(1.0/65536.0));
                sanitize(fd);
                T sdry = in * this->dry;
                T swet = fd * this->wet;
                *buf_out++ = sdry + swet;
                this->delay.put(in+fb*fd);

                this->phase += this->dphase;
                ipart = this->phase.ipart();
                lfo = phase.lerp_by_fract_int<int, 14, int>(this->sine.data[ipart], this->sine.data[ipart+1]);
                delay_pos = mds + (mdepth * lfo >> 6);
            }
            last_actual_delay_pos = dp;
        }
        else {
            for (int i=0; i<nsamples; i++) {
                float in = *buf_in++;
                T fd; // signal from delay's output
                this->delay.get_interp(fd, delay_pos >> 16, (delay_pos & 0xFFFF)*(1.0/65536.0));
                sanitize(fd);
                T sdry = in * this->gs_dry.get();
                T swet = fd * this->gs_wet.get();
                *buf_out++ = sdry + swet;
                this->delay.put(in+fb*fd);

                this->phase += this->dphase;
                ipart = this->phase.ipart();
                lfo = phase.lerp_by_fract_int<int, 14, int>(this->sine.data[ipart], this->sine.data[ipart+1]);
                delay_pos = mds + (mdepth * lfo >> 6);
            }
            last_actual_delay_pos = delay_pos;
        }
        last_delay_pos = delay_pos;
    }
    float freq_gain(float freq, float sr)
    {
        typedef std::complex<double> cfloat;
        freq *= 2.0 * M_PI / sr;
        cfloat z = 1.0 / exp(cfloat(0.0, freq)); // z^-1
        
        float ldp = last_delay_pos / 65536.0;
        float fldp = floor(ldp);
        cfloat zn = std::pow(z, fldp); // z^-N
        cfloat zn1 = zn * z; // z^-(N+1)
        // simulate a lerped comb filter - H(z) = 1 / (1 + fb * (lerp(z^-N, z^-(N+1), fracpos))), N = int(pos), fracpos = pos - int(pos)
        cfloat delayed = zn + (zn1 - zn) * cfloat(ldp - fldp);
        cfloat h = cfloat(delayed) / (cfloat(1.0) - cfloat(fb) * delayed);
        // mix with dry signal
        float v = std::abs(cfloat(gs_dry.get_last()) + cfloat(gs_wet.get_last()) * h);
        return v;
    }
};

/**
 * A classic allpass loop reverb with modulated allpass filter.
 * Just started implementing it, so there is no control over many
 * parameters.
 */
template<class T>
class reverb: public audio_effect
{
    simple_delay<2048, T> apL1, apL2, apL3, apL4, apL5, apL6;
    simple_delay<2048, T> apR1, apR2, apR3, apR4, apR5, apR6;
    fixed_point<unsigned int, 25> phase, dphase;
    sine_table<int, 128, 10000> sine;
    onepole<T> lp_left, lp_right;
    T old_left, old_right;
    int type;
    float time, fb, cutoff, diffusion;
    int tl[6], tr[6];
    float ldec[6], rdec[6];
    
    int sr;
public:
    reverb()
    {
        phase = 0.0;
        time = 1.0;
        cutoff = 9000;
        type = 2;
        diffusion = 1.f;
        setup(44100);
    }
    virtual void setup(int sample_rate) {
        sr = sample_rate;
        set_time(time);
        set_cutoff(cutoff);
        phase = 0.0;
        dphase = 0.5*128/sr;
        update_times();
    }
    void update_times()
    {
        switch(type)
        {
        case 0:
            tl[0] =  397 << 16, tr[0] =  383 << 16;
            tl[1] =  457 << 16, tr[1] =  429 << 16;
            tl[2] =  549 << 16, tr[2] =  631 << 16;
            tl[3] =  649 << 16, tr[3] =  756 << 16;
            tl[4] =  773 << 16, tr[4] =  803 << 16;
            tl[5] =  877 << 16, tr[5] =  901 << 16;
            break;
        case 1:
            tl[0] =  697 << 16, tr[0] =  783 << 16;
            tl[1] =  957 << 16, tr[1] =  929 << 16;
            tl[2] =  649 << 16, tr[2] =  531 << 16;
            tl[3] = 1049 << 16, tr[3] = 1177 << 16;
            tl[4] =  473 << 16, tr[4] =  501 << 16;
            tl[5] =  587 << 16, tr[5] =  681 << 16;
            break;
        case 2:
        default:
            tl[0] =  697 << 16, tr[0] =  783 << 16;
            tl[1] =  957 << 16, tr[1] =  929 << 16;
            tl[2] =  649 << 16, tr[2] =  531 << 16;
            tl[3] = 1249 << 16, tr[3] = 1377 << 16;
            tl[4] = 1573 << 16, tr[4] = 1671 << 16;
            tl[5] = 1877 << 16, tr[5] = 1781 << 16;
            break;
        case 3:
            tl[0] = 1097 << 16, tr[0] = 1087 << 16;
            tl[1] = 1057 << 16, tr[1] = 1031 << 16;
            tl[2] = 1049 << 16, tr[2] = 1039 << 16;
            tl[3] = 1083 << 16, tr[3] = 1055 << 16;
            tl[4] = 1075 << 16, tr[4] = 1099 << 16;
            tl[5] = 1003 << 16, tr[5] = 1073 << 16;
            break;
        case 4:
            tl[0] =  197 << 16, tr[0] =  133 << 16;
            tl[1] =  357 << 16, tr[1] =  229 << 16;
            tl[2] =  549 << 16, tr[2] =  431 << 16;
            tl[3] =  949 << 16, tr[3] = 1277 << 16;
            tl[4] = 1173 << 16, tr[4] = 1671 << 16;
            tl[5] = 1477 << 16, tr[5] = 1881 << 16;
            break;
        case 5:
            tl[0] =  197 << 16, tr[0] =  133 << 16;
            tl[1] =  257 << 16, tr[1] =  179 << 16;
            tl[2] =  549 << 16, tr[2] =  431 << 16;
            tl[3] =  619 << 16, tr[3] =  497 << 16;
            tl[4] = 1173 << 16, tr[4] = 1371 << 16;
            tl[5] = 1577 << 16, tr[5] = 1881 << 16;
            break;
        }
        
        float fDec=1000 + 2400.f * diffusion;
        for (int i = 0 ; i < 6; i++) {
            ldec[i]=exp(-float(tl[i] >> 16) / fDec), 
            rdec[i]=exp(-float(tr[i] >> 16) / fDec);
        }
    }
    float get_time() {
        return time;
    }
    void set_time(float time) {
        this->time = time;
        // fb = pow(1.0f/4096.0f, (float)(1700/(time*sr)));
        fb = 1.0 - 0.3 / (time * sr / 44100.0);
    }
    float get_type() {
        return type;
    }
    void set_type(int type) {
        this->type = type;
        update_times();
    }
    float get_diffusion() {
        return diffusion;
    }
    void set_diffusion(float diffusion) {
        this->diffusion = diffusion;
        update_times();
    }
    void set_type_and_diffusion(int type, float diffusion) {
        this->type = type;
        this->diffusion = diffusion;
        update_times();
    }
    float get_fb()
    {
        return this->fb;
    }
    void set_fb(float fb)
    {
        this->fb = fb;
    }
    float get_cutoff() {
        return cutoff;
    }
    void set_cutoff(float cutoff) {
        this->cutoff = cutoff;
        lp_left.set_lp(cutoff,sr);
        lp_right.set_lp(cutoff,sr);
    }
    void reset()
    {
        apL1.reset();apR1.reset();
        apL2.reset();apR2.reset();
        apL3.reset();apR3.reset();
        apL4.reset();apR4.reset();
        apL5.reset();apR5.reset();
        apL6.reset();apR6.reset();
        lp_left.reset();lp_right.reset();
        old_left = 0; old_right = 0;
    }
    void process(T &left, T &right)
    {
        unsigned int ipart = phase.ipart();
        
        // the interpolated LFO might be an overkill here
        int lfo = phase.lerp_by_fract_int<int, 14, int>(sine.data[ipart], sine.data[ipart+1]) >> 2;
        phase += dphase;
        
        left += old_right;
        left = apL1.process_allpass_comb_lerp16(left, tl[0] - 45*lfo, ldec[0]);
        left = apL2.process_allpass_comb_lerp16(left, tl[1] + 47*lfo, ldec[1]);
        float out_left = left;
        left = apL3.process_allpass_comb_lerp16(left, tl[2] + 54*lfo, ldec[2]);
        left = apL4.process_allpass_comb_lerp16(left, tl[3] - 69*lfo, ldec[3]);
        left = apL5.process_allpass_comb_lerp16(left, tl[4] + 69*lfo, ldec[4]);
        left = apL6.process_allpass_comb_lerp16(left, tl[5] - 46*lfo, ldec[5]);
        old_left = lp_left.process(left * fb);
        sanitize(old_left);

        right += old_left;
        right = apR1.process_allpass_comb_lerp16(right, tr[0] - 45*lfo, rdec[0]);
        right = apR2.process_allpass_comb_lerp16(right, tr[1] + 47*lfo, rdec[1]);
        float out_right = right;
        right = apR3.process_allpass_comb_lerp16(right, tr[2] + 54*lfo, rdec[2]);
        right = apR4.process_allpass_comb_lerp16(right, tr[3] - 69*lfo, rdec[3]);
        right = apR5.process_allpass_comb_lerp16(right, tr[4] + 69*lfo, rdec[4]);
        right = apR6.process_allpass_comb_lerp16(right, tr[5] - 46*lfo, rdec[5]);
        old_right = lp_right.process(right * fb);
        sanitize(old_right);
        
        left = out_left, right = out_right;
    }
    void extra_sanitize()
    {
        lp_left.sanitize();
        lp_right.sanitize();
    }
};

class filter_module_iface
{
public:
    virtual void  calculate_filter(float freq, float q, int mode, float gain = 1.0) = 0;
    virtual void  filter_activate() = 0;
    virtual void  sanitize() = 0;
    virtual int   process_channel(uint16_t channel_no, float *in, float *out, uint32_t numsamples, int inmask) = 0;
    virtual float freq_gain(int subindex, float freq, float srate) = 0;

    virtual ~filter_module_iface() {}
};


class biquad_filter_module: public filter_module_iface
{
private:
    dsp::biquad_d1<float> left[3], right[3];
    int order;
    
public:    
    uint32_t srate;
    
    enum { mode_12db_lp = 0, mode_24db_lp = 1, mode_36db_lp = 2, 
           mode_12db_hp = 3, mode_24db_hp = 4, mode_36db_hp = 5,
           mode_6db_bp  = 6, mode_12db_bp = 7, mode_18db_bp = 8,
           mode_6db_br  = 9, mode_12db_br = 10, mode_18db_br = 11,
           mode_count
    };
    
public:
    biquad_filter_module() : order(0) {}
    
    void calculate_filter(float freq, float q, int mode, float gain = 1.0)
    {
        if (mode <= mode_36db_lp) {
            order = mode + 1;
            left[0].set_lp_rbj(freq, pow(q, 1.0 / order), srate, gain);
        } else if ( mode_12db_hp <= mode && mode <= mode_36db_hp ) {
            order = mode - mode_12db_hp + 1;
            left[0].set_hp_rbj(freq, pow(q, 1.0 / order), srate, gain);
        } else if ( mode_6db_bp <= mode && mode <= mode_18db_bp ) {
            order = mode - mode_6db_bp + 1;
            left[0].set_bp_rbj(freq, pow(q, 1.0 / order), srate, gain);
        } else { // mode_6db_br <= mode <= mode_18db_br
            order = mode - mode_6db_br + 1;
            left[0].set_br_rbj(freq, order * 0.1 * q, srate, gain);
        }
        
        right[0].copy_coeffs(left[0]);
        for (int i = 1; i < order; i++) {
            left[i].copy_coeffs(left[0]);
            right[i].copy_coeffs(left[0]);
        }
    }
    
    void filter_activate()
    {
        for (int i=0; i < order; i++) {
            left[i].reset();
            right[i].reset();
        }
    }
    
    void  sanitize()
    {
        for (int i=0; i < order; i++) {
            left[i].sanitize();
            right[i].sanitize();
        }
    }
    
    inline int process_channel(uint16_t channel_no, float *in, float *out, uint32_t numsamples, int inmask) {
        dsp::biquad_d1<float> *filter;
        switch (channel_no) {
        case 0:
            filter = left;
            break;
            
        case 1:
            filter = right;
            break;
        
        default:
            assert(false);
            break;
        }
        
        if (inmask) {
            switch(order) {
                case 1:
                    for (uint32_t i = 0; i < numsamples; i++)
                        out[i] = filter[0].process(in[i]);
                    break;
                case 2:
                    for (uint32_t i = 0; i < numsamples; i++)
                        out[i] = filter[1].process(filter[0].process(in[i]));
                    break;
                case 3:
                    for (uint32_t i = 0; i < numsamples; i++)
                        out[i] = filter[2].process(filter[1].process(filter[0].process(in[i])));
                    break;
            }
        } else {
            if (filter[order - 1].empty())
                return 0;
            switch(order) {
                case 1:
                    for (uint32_t i = 0; i < numsamples; i++)
                        out[i] = filter[0].process_zeroin();
                    break;
                case 2:
                    if (filter[0].empty())
                        for (uint32_t i = 0; i < numsamples; i++)
                            out[i] = filter[1].process_zeroin();
                    else
                        for (uint32_t i = 0; i < numsamples; i++)
                            out[i] = filter[1].process(filter[0].process_zeroin());
                    break;
                case 3:
                    if (filter[1].empty())
                        for (uint32_t i = 0; i < numsamples; i++)
                            out[i] = filter[2].process_zeroin();
                    else
                        for (uint32_t i = 0; i < numsamples; i++)
                            out[i] = filter[2].process(filter[1].process(filter[0].process_zeroin()));
                    break;
            }
        }
        for (int i = 0; i < order; i++)
            filter[i].sanitize();
        return filter[order - 1].empty() ? 0 : inmask;
    }
    
    float freq_gain(int subindex, float freq, float srate)
    {
        float level = 1.0;
        for (int j = 0; j < order; j++)
            level *= left[j].freq_gain(freq, srate);
        return level;
    }
};

#if 0
{ to keep editor happy
#endif
}

#endif