void FM4Op_keyOff(FM4OP *p)
{
ADSR_keyOff(&p->adsr[0]);
ADSR_keyOff(&p->adsr[1]);
ADSR_keyOff(&p->adsr[2]);
ADSR_keyOff(&p->adsr[3]);
}
void sequencer_newStep_action(void) // User callback function called by sequencer_process()
{
if ((noteG.automaticON || noteG.chRequested))
{
seq_sequence_new();
noteG.chRequested = false;
AdditiveGen_newWaveform();
}
if ( (noteG.someNotesMuted) && (rintf(frand_a_b(0.4f , 1)) == 0) )
ADSR_keyOff(&adsr);
else
ADSR_keyOn(&adsr);
if (autoFilterON)
SVF_directSetFilterValue(&SVFilter, Ts * 600.f * powf(5000.f / 600.f, frand_a_b(0 , 1)));
if (noteG.transpose != 0)
{
noteG.rootNote += noteG.transpose ;
seq_transpose();
}
if (autoSound == 1)
{
switch(rand() % 4) // 4 random timbers
{
case 0 : sound = CHORD15; break;
case 1 : AdditiveGen_newWaveform(); sound = ADDITIVE; break;
case 2 : sound = CHORD13min5; break;
case 3 : sound = VOICES3; break;
}
}
if (autoSound == 2)
{
sound = rand() % LAST_SOUND;
if ((sound == CHORD13min5) || (sound == CHORD135))
sound = VOICES3;
if ( sound == ADDITIVE)
AdditiveGen_newWaveform();
}
f0 = notesFreq[seq.track1.note[seq.step_idx]]; // Main "melody" frequency
vol = frand_a_b(0.4f , .8f); // slightly random volume for each note
}
void make_sound(uint16_t *buf , uint16_t length) // To be used with the Sequencer
{
uint16_t pos;
uint16_t *outp;
float y = 0;
float yL, yR ;
float f1;
uint16_t valueL, valueR;
outp = buf;
for (pos = 0; pos < length; pos++)
{
/*--- Sequencer actions and update ---*/
sequencer_process(); //computes f0 and calls sequencer_newStep_action() and sequencer_newSequence_action()
/*--- compute vibrato modulation ---*/
f1 = f0 * (1 + Osc_WT_SINE_SampleCompute(&vibr_lfo));
/*--- Generate waveform ---*/
y = waveCompute(sound, f1);
/*--- Apply envelop and tremolo ---*/
env = ADSR_computeSample(&adsr) * (1 + Osc_WT_SINE_SampleCompute(&_lfo));
y *= vol * env; // apply volume and envelop
if (adsr.cnt_ >= seq.gateTime) ADSR_keyOff(&adsr);
/*--- Apply filter effect ---*/
/* Update the filters cutoff frequencies */
if ((! autoFilterON)&&(filt_lfo.amp != 0))
SVF_directSetFilterValue(&SVFilter, filterFreq * (1 + OpSampleCompute7bis(&filt_lfo)));
if (filt2_lfo.amp != 0)
SVF_directSetFilterValue(&SVFilter2, filterFreq2 * (1 + OpSampleCompute7bis(&filt2_lfo)));
y = 0.5f * (SVF_calcSample(&SVFilter, y) + SVF_calcSample(&SVFilter2, y)); // Two filters in parallel
/*--- Apply delay effect ----*/
if (delayON) y = Delay_compute(y);
/*--- Apply phaser effect ----*/
if (phaserON) y = Phaser_compute(y);
/*--- Apply chorus/flanger effect ---*/
if (chorusON) stereoChorus_compute (&yL, &yR, y) ;
else yL = yR = y;
/*--- clipping ---*/
yL = (yL > 1.0f) ? 1.0f : yL; //clip too loud left samples
yL = (yL < -1.0f) ? -1.0f : yL;
yR = (yR > 1.0f) ? 1.0f : yR; //clip too loud right samples
yR = (yR < -1.0f) ? -1.0f : yR;
/****** let's hear the new sample *******/
valueL = (uint16_t)((int16_t)((32767.0f) * yL)); // conversion float -> int
valueR = (uint16_t)((int16_t)((32767.0f) * yR));
*outp++ = valueL; // left channel sample
*outp++ = valueR; // right channel sample
}
}
int brass(CSOUND *csound, BRASS *p)
{
MYFLT *ar = p->ar;
uint32_t offset = p->h.insdshead->ksmps_offset;
uint32_t early = p->h.insdshead->ksmps_no_end;
uint32_t n, nsmps = CS_KSMPS;
MYFLT amp = (*p->amp)*AMP_RSCALE; /* Normalise */
MYFLT maxPressure = p->maxPressure = amp;
int v_len = (int)p->vibr->flen;
MYFLT *v_data = p->vibr->ftable;
MYFLT vibGain = *p->vibAmt;
MYFLT vTime = p->v_time;
p->v_rate = *p->vibFreq * v_len * csound->onedsr;
/* vibr->setFreq(6.137); */
/* vibrGain = 0.05; */ /* breath periodic vibrato component */
if (p->kloop>0 && p->h.insdshead->relesing) p->kloop=1;
if ((--p->kloop) == 0) {
ADSR_setReleaseRate(csound, &p->adsr, amp * FL(0.005));
ADSR_keyOff(&p->adsr);
}
if (p->frq != *p->frequency) { /* Set frequency if changed */
p->frq = *p->frequency;
if (p->limit > p->frq) {
p->frq =p->limit;
csound->Warning(csound, Str("frequency too low, set to minimum"));
}
p->slideTarget = (CS_ESR / p->frq * FL(2.0)) + FL(3.0);
/* fudge correction for filter delays */
/* we'll play a harmonic */
if (DLineA_setDelay(csound, &p->delayLine, p->slideTarget)) return OK;
p->lipTarget = p->frq;
p->lipT = FL(0.0); /* So other part is set */
} /* End of set frequency */
if (*p->liptension != p->lipT) {
p->lipT = *p->liptension;
LipFilt_setFreq(csound, &p->lipFilter,
p->lipTarget * (MYFLT)pow(4.0,(2.0* p->lipT) -1.0));
}
if (UNLIKELY(offset)) memset(ar, '\0', offset*sizeof(MYFLT));
if (UNLIKELY(early)) {
nsmps -= early;
memset(&ar[nsmps], '\0', early*sizeof(MYFLT));
}
for (n=offset;n<nsmps;n++) {
MYFLT breathPressure;
MYFLT lastOutput;
int temp;
MYFLT temp_time, alpha;
MYFLT v_lastOutput;
MYFLT ans;
breathPressure = maxPressure * ADSR_tick(&p->adsr);
/* Tick on vibrato table */
vTime += p->v_rate; /* Update current time */
while (vTime >= v_len) /* Check for end of sound */
vTime -= v_len; /* loop back to beginning */
while (vTime < FL(0.0)) /* Check for end of sound */
vTime += v_len; /* loop back to beginning */
temp_time = vTime;
#ifdef phase_offset
if (p->v_phaseOffset != FL(0.0)) {
temp_time += p->v_phaseOffset; /* Add phase offset */
while (temp_time >= v_len) /* Check for end of sound */
temp_time -= v_len; /* loop back to beginning */
while (temp_time < FL(0.0)) /* Check for end of sound */
temp_time += v_len; /* loop back to beginning */
}
#endif
temp = (int) temp_time; /* Integer part of time address */
/* fractional part of time address */
alpha = temp_time - (MYFLT)temp;
v_lastOutput = v_data[temp]; /* Do linear interpolation, same as */
v_lastOutput += /*alpha*data[temp+1]+(1-alpha)data[temp] */
(alpha * (v_data[temp+1] - v_lastOutput));
/* End of vibrato tick */
breathPressure += vibGain * v_lastOutput;
lastOutput =
DLineA_tick(&p->delayLine, /* bore delay */
DCBlock_tick(&p->dcBlock, /* block DC */
LipFilt_tick(&p->lipFilter,
FL(0.3) * breathPressure, /* mouth input */
/* and bore reflection */
FL(0.85) * p->delayLine.lastOutput)));
ans = lastOutput*AMP_SCALE*FL(3.5);
ar[n] = ans;
}
p->v_time = vTime;
return OK;
}
