synth_render.c

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#include "adsr.h"
#include "ct-head/random.h"
#include "delay.h"
#include "foldback.h"
#include "formant.h"
#include "iir.h"
#include "node_ops.h"
#include "osc.h"
#include "synth.h"
#include "wavfile.h"

#define ITER ((uint32_t)(80.0f * SAMPLE_RATE / AUDIO_BUFFER_SIZE))

static const uint8_t scale[]  = {36, 38, 40, 43, 45, 60, 50, 52};
static const uint8_t scale2[] = {33, 31, 26, 31, 29, 33, 31, 29};

static CT_Smush rnd;

static void init_voice(CTSS_Synth *synth, CTSS_DSPStack *stack, float freq) {
  ctss_init_stack(stack);
  CTSS_DSPNode *env = ctss_adsr("env", synth->lfo[0]);
  ctss_configure_adsr(env, 2.0f, 0.7f, 1.5f, 0.5f, 0.5f, false);
  CTSS_DSPNode *lfo = ctss_osc("lfoPitch", ctss_process_osc_square, 0.0f,
                               HZ_TO_RAD(2.0f), HZ_TO_RAD(1.0f), 0.0f);
  CTSS_DSPNode *lfo2 = ctss_osc("lfoGain", ctss_process_osc_saw, 0.0f,
                                HZ_TO_RAD(6.666666f), 0.4f, 0.6f);
  CTSS_DSPNode *osc1 = ctss_osc("osc1", ctss_process_osc_saw, 0.0f,
                                HZ_TO_RAD(freq * 0.51f), 0.35f, 0.0f);
  ctss_set_osc_lfo(osc1, lfo, 1.0f);
  CTSS_DSPNode *osc2 = ctss_osc("osc2", ctss_process_osc_saw, 0.0f,
                                HZ_TO_RAD(freq), 0.35f, 0.0f);
  ctss_set_osc_lfo(osc2, lfo, 1.0f);
  ctss_set_osc_env(osc2, env, HZ_TO_RAD(10.0f));
  CTSS_DSPNode *sum  = ctss_op4("sum", osc1, env, osc2, env, ctss_process_madd);
  CTSS_DSPNode *gate = ctss_op2("gate", sum, lfo2, ctss_process_mult);
  // CTSS_DSPNode *filter = ctss_filter_iir("filter", IIR_LP, gate, NULL, ct_smush_minmax(&rnd, 300.0f, 1200.0f), 0.85f);
  CTSS_DSPNode *filter =
      ctss_filter_formant("filter", ct_smush(&rnd) % 5, gate);
  CTSS_DSPNode *delay = ctss_delay("delay", filter, SAMPLE_RATE >> 2, 0.4f, 1);
  CTSS_DSPNode *filter2 =
      ctss_filter_iir("filter2", IIR_HP, delay, NULL, 1800.0f, 0.0f);
  CTSS_DSPNode *delay2 =
      ctss_delay("delay2", filter2, (uint32_t)(SAMPLE_RATE * 0.333f), 0.4, 1);
  CTSS_DSPNode *delay3 =
      ctss_delay("delay3", delay2, (uint32_t)(SAMPLE_RATE * 0.4781f), 0.4f, 1);
  CTSS_DSPNode *sum2 = ctss_op4_const("sum2", filter, 0.2f, delay3, 0.8f,
                                      ctss_process_madd_const);
  CTSS_DSPNode *nodes[] = {env,    lfo,   lfo2,    osc1,   osc2,   sum, gate,
                           filter, delay, filter2, delay2, delay3, sum2};
  ctss_build_stack(stack, nodes, 13);
}

int main(int argc, char *argv[]) {
  CTSS_Synth synth;
  int16_t out[ITER * AUDIO_BUFFER_SIZE];
  uint8_t voiceID = 0;

  ct_smush_init(&rnd, 0xdecafbad);
  ctss_init(&synth, 8);

  ctss_add_global_lfo(&synth, ctss_osc("lfo1", ctss_process_osc_sin, 0.0f,
                                       HZ_TO_RAD(0.1f), 0.5f, 1.0f));

  for (uint8_t i = 0; i < synth.numStacks; i++) {
    CTSS_DSPStack *s = &synth.stacks[i];
    init_voice(&synth, s, ctss_notes[scale[0]]);
    // NODE_ID_STATE(CTSS_ADSRState, s, "env")->phase = IDLE;
    s->flags = 0;
  }
  ctss_collect_stacks(&synth);

  for (uint32_t i = 0, j = 0; i < ITER; i++) {
    if (((i % (ITER / 48)) == 0) && j < 32) {
      float freq       = ctss_notes[scale[j % 8]];
      float freq2      = ctss_notes[scale2[(j % 8)] + 12];
      CTSS_DSPStack *s = &synth.stacks[voiceID];
      ctss_reset_adsr(NODE_ID(s, "env"));
      NODE_ID_STATE(CTSS_OscState, s, "osc1")->freq = HZ_TO_RAD(freq2);
      NODE_ID_STATE(CTSS_OscState, s, "osc2")->freq = HZ_TO_RAD(freq);
      // ctss_calculate_iir_coeff(NODE_ID(s, "filter"), ct_smush_minmax(&rnd, 300.0f, 1200.0f), 0.8f);
      ctss_activate_stack(s);
      j++;
      voiceID = (voiceID + 1) % synth.numStacks;
    }
    ctss_update(&synth);
    ctss_mixdown_i16(synth.stackOutputs, &out[i * AUDIO_BUFFER_SIZE], 0,
                     AUDIO_BUFFER_SIZE, synth.numStacks, 1);
  }
  return ctss_wavfile_save("sound.wav", out, SAMPLE_RATE, 16, 1,
                           ITER * AUDIO_BUFFER_SIZE);
}