File content as of revision 1:db0c24aebb8a:

#include "STMstation_synth.h"
#include "mbed.h"
#include "arm_math.h"
#include "FastPWM.h"

const float     freqs[] = {0,46.2493,48.9995,51.913,55,58.2705,61.7354,65.4064,69.2957,73.4162,77.7817,82.4069,87.3071,92.4986,97.9989,103.826,110,116.541,123.471,130.813,138.591,146.832,155.563,164.814,174.614,184.997,195.998,207.652,220,233.082,246.942,261.626,277.183,293.665,311.127,329.628,349.228,369.994,391.995,415.305,440,466.164,493.883,523.251,554.365,587.33,622.254,659.255,698.456,739.989,783.991,830.609,880,932.328,987.767,1046.5,1108.73,1174.66,1244.51,1318.51,1396.91,1479.98,1567.98,1661.22,1760,1864.66,1975.53,2093,2217.46,2349.32,2489.02,2637.02,2793.83,2959.96,3135.96,3322.44,3520,3729.31,3951.07,4186.01};
const uint8_t   freqLength = sizeof(freqs)/sizeof(freqs[0]);
const float     pi = 3.14159265359;

//Sampling rate, uncomment one line only!
//const uint16_t FSAMP = 44100; const float period = 0.000023;
//const uint16_t FSAMP = 22050; const float period = 0.000045;
const uint16_t FSAMP = 11025; const float period = 0.000091;

//Constructor
STMstation_synth::STMstation_synth():tone(AUDIO_PIN,1)
{     
    begin();
}

//Constructor
STMstation_synth::STMstation_synth(PinName audio_pin):tone(audio_pin,1)
{     
    begin();
}

//Set up PWM and timer interrupt
void STMstation_synth::begin(){
    tone.pulsewidth_ticks(128);
    tone.period_ticks(256);
    sample.attach(this,&STMstation_synth::note,period);
}

//Calculate the coefficients
void STMstation_synth::calc_coefs(int i){  
    Master.timbre[i] = Master.notes[i][Master.index[i]]/freqLength;
    
    uint8_t _freq_index = Master.notes[i][Master.index[i]] % freqLength;
    float _freq = freqs[_freq_index];
    uint32_t _duration = floor(0.5f + FSAMP*60.0f/Master.bpm[i]*(Master.durations[i][Master.index[i]]+1.0f)/16.0f);
    uint8_t _attack = Master.AR[i][Master.index[i]] >> 4;
    uint8_t _release = Master.AR[i][Master.index[i]] & 0b00001111;
    
    Master.freqIndex[i] = _freq_index;
    Master.sineCoef[i] = 2*pi/FSAMP*_freq;
    if(_freq == 0){Master.halfPeriod[i] = 0;}
    else{Master.halfPeriod[i] = FSAMP/(2*_freq);}
    Master.triSlope[i] = 2/Master.halfPeriod[i];
    Master.envAtkEnd[i] = _attack/16.0f*_duration;
    Master.envRelStart[i] = (1-_release/16.0f)*_duration;
    if(_attack == 0){Master.atkSlope[i] = 0;}
    else{Master.atkSlope[i] = 16.0f/(_attack*_duration);}
    if(_release == 0){Master.relSlope[i] = 0;}
    else{Master.relSlope[i] = 16.0f/(_release*_duration);}
    Master.relOffset[i] = 16.0f/_release;
    Master.volCoef[i] = Master.vol[i][Master.index[i]]/Master.vSum;
}

//Calculate vSum
void STMstation_synth::calc_vSum(){
    uint8_t active = 0; //Number of active channels
    uint16_t accum = 0; //Sum of volume channel values
    bool over = 0;      //Is any value over 127?
    for(int i=0; i<CHANNELS; i++){
        if(Master.notes[i] != NULL){
            active++;
            accum += Master.vol[i][Master.index[i]];
            if(Master.vol[i][Master.index[i]] > 127){over = 1;}
        }
    }
    if(accum == 0)      {Master.vSum = 1;}
    else if(over == 0)  {Master.vSum = active*127;}
    else                {Master.vSum = accum;}
}

// Calculate the envelope
void STMstation_synth::calc_env(){
    for(int i=0; i<CHANNELS; i++){
        if(Master.notes[i] != NULL){
            if(Master.counter[i] < Master.envAtkEnd[i]){
                if(Master.atkSlope == 0){Master.env[i] = 1;}
                else{Master.env[i] = Master.atkSlope[i]*Master.counter[i];}
            }
            else if(Master.counter[i] > Master.envRelStart[i]){
                if(Master.relSlope == 0){Master.env[i] = 1;}
                else{Master.env[i] = Master.relOffset[i] - Master.relSlope[i]*Master.counter[i];}
            }
            else{
                Master.env[i] = 1;
            }
        }
    }
}

//Square wave switching function
int8_t STMstation_synth::square(float _halfperiod, uint16_t _counter){
    if(_halfperiod == 0){
        return 0;;
    }
    
    uint32_t _div = _counter/_halfperiod;
    uint32_t _div2 = _div % 2;
    if(_div2 == 0){
        return -1;
    }
    else{
        return 1;
    }
}

//Triangle wave function
void STMstation_synth::calc_triangle(uint8_t _channel, float _halfperiod, float _trislope, uint32_t _counter){
    if(_halfperiod == 0){
        Master.triVal[_channel] = 0;
        return;
    }
    
    int8_t _sign = 2*(uint32_t(_counter/_halfperiod) % 2) - 1;
    Master.triVal[_channel] += _sign*_trislope;
    if(_counter == 0){
        Master.triVal[_channel] = 1;
    }
    if(Master.triVal[_channel] < -1){
        Master.triVal[_channel] = -1;
    }
    else if(Master.triVal[_channel] > 1){
        Master.triVal[_channel] = 1;
    }
}

//Calculate noise
void STMstation_synth::calc_noise(uint8_t _channel, uint8_t _freq, uint32_t _counter){
    if(_freq == 0){
        Master.noiseVal[_channel] = 0;
        return;
    }
    
    uint32_t _noisePeriod = (16-_freq);
    if(_counter % _noisePeriod == 0){
        Master.noiseVal[_channel] = 1-(rand() % 257)/128.0f;
    }
}

//Calculate values
void STMstation_synth::calc_val(){
    Master.val = 128;
    for(int i=0; i<CHANNELS; i++){
        if(Master.notes[i] != NULL){
            if(Master.timbre[i] == 0){
                Master.val += 127*arm_sin_f32(Master.sineCoef[i]*Master.counter[i])*Master.env[i]*Master.volCoef[i];
            }
            else if(Master.timbre[i] == 1){
                Master.val += 127*square(Master.halfPeriod[i], Master.counter[i])*Master.env[i]*Master.volCoef[i];
            }
            else if(Master.timbre[i] == 2){
                calc_triangle(i,Master.halfPeriod[i], Master.triSlope[i], Master.counter[i]);
                Master.val += 127*Master.triVal[i]*Master.env[i]*Master.volCoef[i];
            }
            else if(Master.timbre[i] == 3){
                calc_noise(i,Master.freqIndex[i],Master.counter[i]);
                Master.val += 127*Master.noiseVal[i]*Master.env[i]*Master.volCoef[i];
            }
        }
    }
}

//Remove channel
void STMstation_synth::clear_channel(uint8_t _channel){
    Master.notes[_channel] = NULL;
    Master.durations[_channel] = NULL;
    Master.AR[_channel] = NULL;
    Master.vol[_channel] = NULL;
    Master.max[_channel] = NULL;
    Master.repeat[_channel] = NULL;
    Master.counter[_channel] = NULL;
    Master.index[_channel] = NULL;
    *(Master.endptr[_channel]) = 1;
}

//Stop track
void STMstation_synth::stop_track(melody &newMelody){
    for(uint16_t i=0; i<CHANNELS; i++){
        if(newMelody.notes[i] != NULL){
            clear_channel(i);
        }
    }
}

//Check if track is playing, returns 0 if all channels ended, 1 otherwise
bool STMstation_synth::check_track(melody &newMelody){
    for(uint16_t i=0; i<CHANNELS; i++){
        if(newMelody.ended[i] == 0){
            return 1;
        }
    }
    return 0;
}

//Check if notes are finished playing
void STMstation_synth::check_end(){
    uint32_t _duration;
    for(int i=0; i<CHANNELS; i++){
        _duration = floor(0.5f + FSAMP*60.0f/Master.bpm[i]*(Master.durations[i][Master.index[i]]+1.0f)/16.0f);
        if(Master.counter[i] >= _duration){
            if(Master.index[i] < Master.max[i]){
                Master.index[i]++;
            }
            else{
                if(Master.repeat[i] == 1){
                    Master.index[i] = 0;
                    *(Master.endptr[i]) = 1;
                }
                else{
                    clear_channel(i);
                    *(Master.endptr[i]) = 1;
                }
            }
            Master.counter[i] = 0;
        }
        else{
            Master.counter[i]++;
        }
    }
}

//Check if a new note is starting and if we need to recalculate coefficients
void STMstation_synth::check_start(){
    for(int i=0; i<CHANNELS; i++){
        if(Master.counter[i] == 0){
            calc_vSum();
            calc_coefs(i);
        }
    }
}

//Play dat funky music
void STMstation_synth::note(){
    check_start();
    calc_env();
    calc_val();
    tone.pulsewidth_ticks(Master.val);
    check_end();
}

//Start playing a tune
void STMstation_synth::play(melody &newMelody, uint8_t refChannel, uint16_t newIndex){
    uint32_t pos = 0;
    uint32_t rem;
    uint16_t chanIndex;
    
    //pc.printf("Playing track...\n");
    //pc.printf("refChannel = %d, newIndex = %d\n", refChannel, newIndex);
    
    //Calculate how far into the track we're at, in terms of samples
    for(uint16_t i=0; i<newIndex; i++){
        pos += floor(0.5f + FSAMP*60.0f/newMelody.bpm*(newMelody.durations[refChannel][i]+1.0f)/16.0f);
    }
    
    //pc.printf("pos = %d\n",pos);
    
    for(int i=0; i<CHANNELS; i++){
        if(newMelody.notes[i] != NULL){
            chanIndex = 0;
            rem = pos;
            for(uint16_t j=0; j<=newMelody.max[i]; j++){
                if(rem < floor(0.5f + FSAMP*60.0f/newMelody.bpm*(newMelody.durations[i][j]+1.0f)/16.0f)){                
                    *(Master.endptr[i]) = 1;                                            //Important! Overwrite the old .ended back to zero!
                    Master.notes[i] = newMelody.notes[i];
                    Master.durations[i] = newMelody.durations[i];
                    Master.AR[i] = newMelody.AR[i];
                    Master.vol[i] = newMelody.vol[i];
                    Master.max[i] = newMelody.max[i];
                    Master.repeat[i] = newMelody.repeat[i];
                    Master.endptr[i] = &newMelody.ended[i];                             //Now we set the .ended pointer to the new track's
                    *(Master.endptr[i]) = 0;                                            //Now we set the new track's .ended back to zero.
                    Master.counter[i] = rem;
                    Master.index[i] = chanIndex;
                    Master.bpm[i] = newMelody.bpm;
                    //pc.printf("Channel %d: chanIndex = %d, rem = %d\n",i,chanIndex,rem);
                    break;
                }
                else{
                    rem -= floor(0.5f + FSAMP*60.0f/newMelody.bpm*(newMelody.durations[i][j]+1.0f)/16.0f);
                    chanIndex++;
                }
            }            
        }
    }
    calc_vSum();
    for(uint16_t i=0; i<CHANNELS; i++){
        if(Master.notes[i]!=NULL){
            calc_coefs(i);
        }
    }
}