commit 3f055b00d2165903e091947e715f94e60321487c
parent d70b427dcb62906898b0f495fdd3b8ef79b9bcad
Author: fundamental <[email protected]>
Date: Thu, 25 Aug 2011 16:35:07 -0400
Style: using pre-increment in loops
- This matches with yoshimi changes
Diffstat:
42 files changed, 487 insertions(+), 487 deletions(-)
diff --git a/ExternalPrograms/Controller/Controller.C b/ExternalPrograms/Controller/Controller.C
@@ -8,7 +8,7 @@ int Pexitprogram;
Controller::Controller() {
//init
- for(int i = 0; i < 6; i++) {
+ for(int i = 0; i < 6; ++i) {
pars[i].mode = 1;
pars[i].val1 = 0;
pars[i].val2 = 127;
diff --git a/src/DSP/AnalogFilter.cpp b/src/DSP/AnalogFilter.cpp
@@ -34,7 +34,7 @@ AnalogFilter::AnalogFilter(unsigned char Ftype,
unsigned char Fstages)
{
stages = Fstages;
- for(int i = 0; i < 3; i++) {
+ for(int i = 0; i < 3; ++i) {
coeff.c[i] = 0.0;
coeff.d[i] = 0.0;
oldCoeff.c[i] = 0.0;
@@ -61,7 +61,7 @@ AnalogFilter::~AnalogFilter()
void AnalogFilter::cleanup()
{
- for(int i = 0; i < MAX_FILTER_STAGES + 1; i++) {
+ for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i) {
history[i].x1 = 0.0;
history[i].x2 = 0.0;
history[i].y1 = 0.0;
@@ -361,7 +361,7 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist,
const Coeff &coeff)
{
if(order == 1) { //First order filter
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float y0 = smp[i]*coeff.c[0] + hist.x1*coeff.c[1]
+ hist.y1*coeff.d[1];
hist.y1 = y0;
@@ -370,7 +370,7 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist,
}
}
if(order == 2) { //Second order filter
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float y0 = smp[i]*coeff.c[0] + hist.x1*coeff.c[1]
+ hist.x2*coeff.c[2] + hist.y1*coeff.d[1]
+ hist.y2*coeff.d[2];
@@ -384,7 +384,7 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist,
}
void AnalogFilter::filterout(float *smp)
{
- for(int i = 0; i < stages + 1; i++)
+ for(int i = 0; i < stages + 1; ++i)
singlefilterout(smp, history[i], coeff);
if(needsinterpolation) {
@@ -392,10 +392,10 @@ void AnalogFilter::filterout(float *smp)
float *ismp = getTmpBuffer();
memcpy(ismp, smp, sizeof(float) * SOUND_BUFFER_SIZE);
- for(int i = 0; i < stages + 1; i++)
+ for(int i = 0; i < stages + 1; ++i)
singlefilterout(ismp, oldHistory[i], oldCoeff);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float x = i / (float) SOUND_BUFFER_SIZE;
smp[i] = ismp[i] * (1.0 - x) + smp[i] * x;
}
@@ -403,7 +403,7 @@ void AnalogFilter::filterout(float *smp)
needsinterpolation = false;
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
smp[i] *= outgain;
}
@@ -411,14 +411,14 @@ float AnalogFilter::H(float freq)
{
float fr = freq / SAMPLE_RATE * PI * 2.0;
float x = coeff.c[0], y = 0.0;
- for(int n = 1; n < 3; n++) {
+ for(int n = 1; n < 3; ++n) {
x += cos(n * fr) * coeff.c[n];
y -= sin(n * fr) * coeff.c[n];
}
float h = x * x + y * y;
x = 1.0;
y = 0.0;
- for(int n = 1; n < 3; n++) {
+ for(int n = 1; n < 3; ++n) {
x -= cos(n * fr) * coeff.d[n];
y += sin(n * fr) * coeff.d[n];
}
diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp
@@ -30,12 +30,12 @@
FormantFilter::FormantFilter(FilterParams *pars)
{
numformants = pars->Pnumformants;
- for(int i = 0; i < numformants; i++)
+ for(int i = 0; i < numformants; ++i)
formant[i] = new AnalogFilter(4 /*BPF*/, 1000.0, 10.0, pars->Pstages);
cleanup();
- for(int j = 0; j < FF_MAX_VOWELS; j++)
- for(int i = 0; i < numformants; i++) {
+ for(int j = 0; j < FF_MAX_VOWELS; ++j)
+ for(int i = 0; i < numformants; ++i) {
formantpar[j][i].freq = pars->getformantfreq(
pars->Pvowels[j].formants[i].freq);
formantpar[j][i].amp = pars->getformantamp(
@@ -44,9 +44,9 @@ FormantFilter::FormantFilter(FilterParams *pars)
pars->Pvowels[j].formants[i].q);
}
- for(int i = 0; i < FF_MAX_FORMANTS; i++)
+ for(int i = 0; i < FF_MAX_FORMANTS; ++i)
oldformantamp[i] = 1.0;
- for(int i = 0; i < numformants; i++) {
+ for(int i = 0; i < numformants; ++i) {
currentformants[i].freq = 1000.0;
currentformants[i].amp = 1.0;
currentformants[i].q = 2.0;
@@ -57,7 +57,7 @@ FormantFilter::FormantFilter(FilterParams *pars)
sequencesize = pars->Psequencesize;
if(sequencesize == 0)
sequencesize = 1;
- for(int k = 0; k < sequencesize; k++)
+ for(int k = 0; k < sequencesize; ++k)
sequence[k].nvowel = pars->Psequence[k].nvowel;
vowelclearness = pow(10.0, (pars->Pvowelclearness - 32.0) / 48.0);
@@ -76,13 +76,13 @@ FormantFilter::FormantFilter(FilterParams *pars)
FormantFilter::~FormantFilter()
{
- for(int i = 0; i < numformants; i++)
+ for(int i = 0; i < numformants; ++i)
delete (formant[i]);
}
void FormantFilter::cleanup()
{
- for(int i = 0; i < numformants; i++)
+ for(int i = 0; i < numformants; ++i)
formant[i]->cleanup();
}
@@ -129,7 +129,7 @@ void FormantFilter::setpos(float input)
p2 = sequence[p2].nvowel;
if(firsttime != 0) {
- for(int i = 0; i < numformants; i++) {
+ for(int i = 0; i < numformants; ++i) {
currentformants[i].freq = formantpar[p1][i].freq
* (1.0
- pos) + formantpar[p2][i].freq * pos;
@@ -145,7 +145,7 @@ void FormantFilter::setpos(float input)
firsttime = 0;
}
else {
- for(int i = 0; i < numformants; i++) {
+ for(int i = 0; i < numformants; ++i) {
currentformants[i].freq = currentformants[i].freq
* (1.0 - formantslowness)
+ (formantpar[p1][i].freq
@@ -183,7 +183,7 @@ void FormantFilter::setfreq(float frequency)
void FormantFilter::setq(float q_)
{
Qfactor = q_;
- for(int i = 0; i < numformants; i++)
+ for(int i = 0; i < numformants; ++i)
formant[i]->setq(Qfactor * currentformants[i].q);
}
@@ -208,21 +208,21 @@ void FormantFilter::filterout(float *smp)
memcpy(inbuffer, smp, sizeof(float) * SOUND_BUFFER_SIZE);
memset(smp, 0, sizeof(float) * SOUND_BUFFER_SIZE);
- for(int j = 0; j < numformants; j++) {
+ for(int j = 0; j < numformants; ++j) {
float *tmpbuf = getTmpBuffer();
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpbuf[i] = inbuffer[i] * outgain;
formant[j]->filterout(tmpbuf);
if(ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j], currentformants[j].amp))
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
smp[i] += tmpbuf[i]
* INTERPOLATE_AMPLITUDE(oldformantamp[j],
currentformants[j].amp,
i,
SOUND_BUFFER_SIZE);
else
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
smp[i] += tmpbuf[i] * currentformants[j].amp;
returnTmpBuffer(tmpbuf);
oldformantamp[j] = currentformants[j].amp;
diff --git a/src/DSP/SVFilter.cpp b/src/DSP/SVFilter.cpp
@@ -53,7 +53,7 @@ SVFilter::~SVFilter()
void SVFilter::cleanup()
{
- for(int i = 0; i < MAX_FILTER_STAGES + 1; i++) {
+ for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i) {
st[i].low = 0.0;
st[i].high = 0.0;
st[i].band = 0.0;
@@ -151,7 +151,7 @@ void SVFilter::singlefilterout(float *smp, fstage &x, parameters &par)
errx(1, "Impossible SVFilter type encountered [%d]", type);
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
x.low = x.low + par.f * x.band;
x.high = par.q_sqrt * smp[i] - x.low - par.q * x.band;
x.band = par.f * x.high + x.band;
@@ -163,17 +163,17 @@ void SVFilter::singlefilterout(float *smp, fstage &x, parameters &par)
void SVFilter::filterout(float *smp)
{
- for(int i = 0; i < stages + 1; i++)
+ for(int i = 0; i < stages + 1; ++i)
singlefilterout(smp, st[i], par);
if(needsinterpolation) {
float *ismp = getTmpBuffer();
memcpy(ismp, smp, sizeof(float) * SOUND_BUFFER_SIZE);
- for(int i = 0; i < stages + 1; i++)
+ for(int i = 0; i < stages + 1; ++i)
singlefilterout(ismp, st[i], ipar);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float x = i / (float) SOUND_BUFFER_SIZE;
smp[i] = ismp[i] * (1.0 - x) + smp[i] * x;
}
@@ -181,7 +181,7 @@ void SVFilter::filterout(float *smp)
needsinterpolation = false;
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
smp[i] *= outgain;
}
diff --git a/src/DSP/Unison.cpp b/src/DSP/Unison.cpp
@@ -83,7 +83,7 @@ void Unison::update_parameters() {
float increments_per_second = SAMPLE_RATE
/ (float) update_period_samples;
// printf("#%g, %g\n",increments_per_second,base_freq);
- for(int i = 0; i < unison_size; i++) {
+ for(int i = 0; i < unison_size; ++i) {
float base = pow(UNISON_FREQ_SPAN, RND * 2.0 - 1.0);
uv[i].relative_amplitude = base;
float period = base / base_freq;
@@ -116,7 +116,7 @@ void Unison::process(int bufsize, float *inbuf, float *outbuf) {
float volume = 1.0 / sqrt(unison_size);
float xpos_step = 1.0 / (float) update_period_samples;
float xpos = (float) update_period_sample_k * xpos_step;
- for(int i = 0; i < bufsize; i++) {
+ for(int i = 0; i < bufsize; ++i) {
if((update_period_sample_k++) >= update_period_samples) {
update_unison_data();
update_period_sample_k = 0;
@@ -126,7 +126,7 @@ void Unison::process(int bufsize, float *inbuf, float *outbuf) {
float in = inbuf[i], out = 0.0;
float sign = 1.0;
- for(int k = 0; k < unison_size; k++) {
+ for(int k = 0; k < unison_size; ++k) {
float vpos = uv[k].realpos1
* (1.0 - xpos) + uv[k].realpos2 * xpos; //optimize
float pos = delay_k + max_delay - vpos - 1.0; //optimize
@@ -154,7 +154,7 @@ void Unison::update_unison_data() {
if(!uv)
return;
- for(int k = 0; k < unison_size; k++) {
+ for(int k = 0; k < unison_size; ++k) {
float pos = uv[k].position;
float step = uv[k].step;
pos += step;
diff --git a/src/Effects/Alienwah.cpp b/src/Effects/Alienwah.cpp
@@ -60,7 +60,7 @@ void Alienwah::out(const Stereo<float *> &smp)
clfol = complex<float>(cos(lfol + phase) * fb, sin(lfol + phase) * fb); //rework
clfor = complex<float>(cos(lfor + phase) * fb, sin(lfor + phase) * fb); //rework
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float x = ((float) i) / SOUND_BUFFER_SIZE;
float x1 = 1.0 - x;
//left
@@ -98,7 +98,7 @@ void Alienwah::out(const Stereo<float *> &smp)
*/
void Alienwah::cleanup()
{
- for(int i = 0; i < Pdelay; i++) {
+ for(int i = 0; i < Pdelay; ++i) {
oldl[i] = complex<float>(0.0, 0.0);
oldr[i] = complex<float>(0.0, 0.0);
}
@@ -175,7 +175,7 @@ void Alienwah::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
if(insertion == 0)
changepar(0, presets[npreset][0] / 2); //lower the volume if this is system effect
diff --git a/src/Effects/Chorus.cpp b/src/Effects/Chorus.cpp
@@ -78,7 +78,7 @@ void Chorus::out(const Stereo<float *> &input)
dl2 = getdelay(lfol);
dr2 = getdelay(lfor);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float inl = input.l[i];
float inr = input.r[i];
@@ -123,12 +123,12 @@ void Chorus::out(const Stereo<float *> &input)
}
if(Poutsub != 0)
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] *= -1.0;
efxoutr[i] *= -1.0;
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] *= pangainL;
efxoutr[i] *= pangainR;
}
@@ -203,7 +203,7 @@ void Chorus::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
Ppreset = npreset;
}
diff --git a/src/Effects/Distorsion.cpp b/src/Effects/Distorsion.cpp
@@ -99,13 +99,13 @@ void Distorsion::out(const Stereo<float *> &smp)
inputvol *= -1.0;
if(Pstereo != 0) { //Stereo
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] = smp.l[i] * inputvol * pangainL;
efxoutr[i] = smp.r[i] * inputvol * pangainR;
}
}
else {
- for(i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
efxoutl[i] = (smp.l[i] * pangainL + smp.r[i] * pangainR) * inputvol;
}
@@ -121,11 +121,11 @@ void Distorsion::out(const Stereo<float *> &smp)
applyfilters(efxoutl, efxoutr);
if(Pstereo == 0)
- for(i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
efxoutr[i] = efxoutl[i];
float level = dB2rap(60.0 * Plevel / 127.0 - 40.0);
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
lout = efxoutl[i];
rout = efxoutr[i];
l = lout * (1.0 - lrcross) + rout * lrcross;
@@ -196,7 +196,7 @@ void Distorsion::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
if(insertion == 0)
changepar(0, (int) (presets[npreset][0] / 1.5)); //lower the volume if this is system effect
diff --git a/src/Effects/DynamicFilter.cpp b/src/Effects/DynamicFilter.cpp
@@ -61,7 +61,7 @@ void DynamicFilter::out(const Stereo<float *> &smp)
const float freq = filterpars->getfreq();
const float q = filterpars->getq();
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] = smp.l[i];
efxoutr[i] = smp.r[i];
@@ -86,7 +86,7 @@ void DynamicFilter::out(const Stereo<float *> &smp)
filterr->filterout(efxoutr);
//panning
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] *= pangainL;
efxoutr[i] *= pangainR;
}
@@ -164,7 +164,7 @@ void DynamicFilter::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
filterpars->defaults();
diff --git a/src/Effects/EQ.cpp b/src/Effects/EQ.cpp
@@ -27,7 +27,7 @@
EQ::EQ(const int &insertion_, float *efxoutl_, float *efxoutr_)
:Effect(insertion_, efxoutl_, efxoutr_, NULL, 0)
{
- for(int i = 0; i < MAX_EQ_BANDS; i++) {
+ for(int i = 0; i < MAX_EQ_BANDS; ++i) {
filter[i].Ptype = 0;
filter[i].Pfreq = 64;
filter[i].Pgain = 64;
@@ -48,7 +48,7 @@ EQ::~EQ()
void EQ::cleanup()
{
- for(int i = 0; i < MAX_EQ_BANDS; i++) {
+ for(int i = 0; i < MAX_EQ_BANDS; ++i) {
filter[i].l->cleanup();
filter[i].r->cleanup();
}
@@ -57,12 +57,12 @@ void EQ::cleanup()
void EQ::out(const Stereo<float *> &smp)
{
int i;
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] = smp.l[i] * volume;
efxoutr[i] = smp.r[i] * volume;
}
- for(i = 0; i < MAX_EQ_BANDS; i++) {
+ for(i = 0; i < MAX_EQ_BANDS; ++i) {
if(filter[i].Ptype == 0)
continue;
filter[i].l->filterout(efxoutl);
@@ -100,7 +100,7 @@ void EQ::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
Ppreset = npreset;
}
@@ -203,7 +203,7 @@ float EQ::getfreqresponse(float freq)
{
float resp = 1.0;
- for(int i = 0; i < MAX_EQ_BANDS; i++) {
+ for(int i = 0; i < MAX_EQ_BANDS; ++i) {
if(filter[i].Ptype == 0)
continue;
resp *= filter[i].l->H(freq);
diff --git a/src/Effects/Echo.cpp b/src/Effects/Echo.cpp
@@ -191,7 +191,7 @@ void Echo::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
if(insertion)
setvolume(presets[npreset][0] / 2); //lower the volume if this is insertion effect
diff --git a/src/Effects/EffectMgr.cpp b/src/Effects/EffectMgr.cpp
@@ -48,7 +48,7 @@ EffectMgr::EffectMgr(int insertion_, pthread_mutex_t *mutex_)
// mutex=mutex_;
// efxoutl=new float[SOUND_BUFFER_SIZE];
// efxoutr=new float[SOUND_BUFFER_SIZE];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] = 0.0;
efxoutr[i] = 0.0;
}
@@ -81,7 +81,7 @@ void EffectMgr::changeeffect(int nefx_)
if(nefx == nefx_)
return;
nefx = nefx_;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] = 0.0;
efxoutr[i] = 0.0;
}
@@ -212,7 +212,7 @@ void EffectMgr::out(float *smpsl, float *smpsr)
int i;
if(efx == NULL) {
if(insertion == 0)
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] = 0.0;
smpsr[i] = 0.0;
efxoutl[i] = 0.0;
@@ -221,7 +221,7 @@ void EffectMgr::out(float *smpsl, float *smpsr)
;
return;
}
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] += denormalkillbuf[i];
smpsr[i] += denormalkillbuf[i];
efxoutl[i] = 0.0;
@@ -233,7 +233,7 @@ void EffectMgr::out(float *smpsl, float *smpsr)
if(nefx == 7) { //this is need only for the EQ effect
/**\todo figure out why*/
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] = efxoutl[i];
smpsr[i] = efxoutr[i];
}
@@ -255,7 +255,7 @@ void EffectMgr::out(float *smpsl, float *smpsr)
v2 *= v2; //for Reverb and Echo, the wet function is not liniar
if(dryonly) { //this is used for instrument effect only
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] *= v1;
smpsr[i] *= v1;
efxoutl[i] *= v2;
@@ -263,14 +263,14 @@ void EffectMgr::out(float *smpsl, float *smpsr)
}
}
else { //normal instrument/insertion effect
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
smpsl[i] = smpsl[i] * v1 + efxoutl[i] * v2;
smpsr[i] = smpsr[i] * v1 + efxoutr[i] * v2;
}
}
}
else { //System effect
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] *= 2.0 * volume;
efxoutr[i] *= 2.0 * volume;
smpsl[i] = efxoutl[i];
@@ -317,7 +317,7 @@ void EffectMgr::add2XML(XMLwrapper *xml)
xml->addpar("preset", efx->Ppreset);
xml->beginbranch("EFFECT_PARAMETERS");
- for(int n = 0; n < 128; n++) {
+ for(int n = 0; n < 128; ++n) {
/**\todo evaluate who should oversee saving
* and loading of parameters*/
int par = geteffectpar(n);
@@ -345,7 +345,7 @@ void EffectMgr::getfromXML(XMLwrapper *xml)
efx->Ppreset = xml->getpar127("preset", efx->Ppreset);
if(xml->enterbranch("EFFECT_PARAMETERS")) {
- for(int n = 0; n < 128; n++) {
+ for(int n = 0; n < 128; ++n) {
seteffectpar_nolock(n, 0); //erase effect parameter
if(xml->enterbranch("par_no", n) == 0)
continue;
diff --git a/src/Effects/Phaser.cpp b/src/Effects/Phaser.cpp
@@ -128,7 +128,7 @@ void Phaser::AnalogPhase(const Stereo<float *> &input)
g = oldgain;
oldgain = mod;
- for (int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for (int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
g.l += diff.l;// Linear interpolation between LFO samples
g.r += diff.r;
@@ -158,7 +158,7 @@ void Phaser::AnalogPhase(const Stereo<float *> &input)
float Phaser::applyPhase(float x, float g, float fb,
float &hpf, float *yn1, float *xn1)
{
- for(int j = 0; j < Pstages; j++) { //Phasing routine
+ for(int j = 0; j < Pstages; ++j) { //Phasing routine
mis = 1.0f + offsetpct*offset[j];
//This is symmetrical.
@@ -197,7 +197,7 @@ void Phaser::normalPhase(const Stereo<float *> &input)
gain.l = limit(gain.l, ZERO_, ONE_);
gain.r = limit(gain.r, ZERO_, ONE_);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float x = (float) i / SOUND_BUFFER_SIZE;
float x1 = 1.0 - x;
//TODO think about making panning an external feature
@@ -229,7 +229,7 @@ void Phaser::normalPhase(const Stereo<float *> &input)
float Phaser::applyPhase(float x, float g, float *old)
{
- for(int j = 0; j < Pstages * 2; j++) { //Phasing routine
+ for(int j = 0; j < Pstages * 2; ++j) { //Phasing routine
float tmp = old[j];
old[j] = g * tmp + x;
x = tmp - g *old[j];
@@ -243,11 +243,11 @@ float Phaser::applyPhase(float x, float g, float *old)
void Phaser::cleanup()
{
fb = oldgain = Stereo<float>(0.0);
- for(int i = 0; i < Pstages * 2; i++) {
+ for(int i = 0; i < Pstages * 2; ++i) {
old.l[i] = 0.0;
old.r[i] = 0.0;
}
- for(int i = 0; i < Pstages; i++) {
+ for(int i = 0; i < Pstages; ++i) {
xn1.l[i] = 0.0;
yn1.l[i] = 0.0;
xn1.r[i] = 0.0;
@@ -355,7 +355,7 @@ void Phaser::setpreset(unsigned char npreset)
};
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
Ppreset = npreset;
}
diff --git a/src/Effects/Reverb.cpp b/src/Effects/Reverb.cpp
@@ -48,7 +48,7 @@ Reverb::Reverb(const int &insertion_, float *efxoutl_, float *efxoutr_)
roomsize = 1.0;
rs = 1.0;
- for(int i = 0; i < REV_COMBS * 2; i++) {
+ for(int i = 0; i < REV_COMBS * 2; ++i) {
comblen[i] = 800 + (int)(RND * 1400);
combk[i] = 0;
lpcomb[i] = 0;
@@ -56,7 +56,7 @@ Reverb::Reverb(const int &insertion_, float *efxoutl_, float *efxoutr_)
comb[i] = NULL;
}
- for(int i = 0; i < REV_APS * 2; i++) {
+ for(int i = 0; i < REV_APS * 2; ++i) {
aplen[i] = 500 + (int)(RND * 500);
apk[i] = 0;
ap[i] = NULL;
@@ -81,9 +81,9 @@ Reverb::~Reverb()
if(lpf != NULL)
delete lpf;
- for(i = 0; i < REV_APS * 2; i++)
+ for(i = 0; i < REV_APS * 2; ++i)
delete [] ap[i];
- for(i = 0; i < REV_COMBS * 2; i++)
+ for(i = 0; i < REV_COMBS * 2; ++i)
delete [] comb[i];
if(bandwidth)
@@ -96,18 +96,18 @@ Reverb::~Reverb()
void Reverb::cleanup()
{
int i, j;
- for(i = 0; i < REV_COMBS * 2; i++) {
+ for(i = 0; i < REV_COMBS * 2; ++i) {
lpcomb[i] = 0.0;
- for(j = 0; j < comblen[i]; j++)
+ for(j = 0; j < comblen[i]; ++j)
comb[i][j] = 0.0;
}
- for(i = 0; i < REV_APS * 2; i++)
- for(j = 0; j < aplen[i]; j++)
+ for(i = 0; i < REV_APS * 2; ++i)
+ for(j = 0; j < aplen[i]; ++j)
ap[i][j] = 0.0;
if(idelay != NULL)
- for(i = 0; i < idelaylen; i++)
+ for(i = 0; i < idelaylen; ++i)
idelay[i] = 0.0;
if(hpf != NULL)
@@ -122,12 +122,12 @@ void Reverb::cleanup()
void Reverb::processmono(int ch, float *output, float *inputbuf)
{
/**\todo: implement the high part from lohidamp*/
- for(int j = REV_COMBS * ch; j < REV_COMBS * (ch + 1); j++) {
+ for(int j = REV_COMBS * ch; j < REV_COMBS * (ch + 1); ++j) {
int &ck = combk[j];
const int comblength = comblen[j];
float &lpcombj = lpcomb[j];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float fbout = comb[j][ck] * combfb[j];
fbout = fbout * (1.0 - lohifb) + lpcombj * lohifb;
lpcombj = fbout;
@@ -140,10 +140,10 @@ void Reverb::processmono(int ch, float *output, float *inputbuf)
}
}
- for(int j = REV_APS * ch; j < REV_APS * (1 + ch); j++) {
+ for(int j = REV_APS * ch; j < REV_APS * (1 + ch); ++j) {
int &ak = apk[j];
const int aplength = aplen[j];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmp = ap[j][ak];
ap[j][ak] = 0.7 * tmp + output[i];
output[i] = tmp - 0.7 * ap[j][ak];
@@ -162,11 +162,11 @@ void Reverb::out(const Stereo<float *> &smp)
return;
float *inputbuf = getTmpBuffer();
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
inputbuf[i] = (smp.l[i] + smp.r[i]) / 2.0;
if(idelay != NULL) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
//Initial delay r
float tmp = inputbuf[i] + idelay[idelayk] * idelayfb;
inputbuf[i] = idelay[idelayk];
@@ -195,7 +195,7 @@ void Reverb::out(const Stereo<float *> &smp)
lvol *= 2;
rvol *= 2;
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
efxoutl[i] *= lvol;
efxoutr[i] *= rvol;
}
@@ -226,7 +226,7 @@ void Reverb::settime(unsigned char Ptime)
this->Ptime = Ptime;
t = pow(60.0, (float)Ptime / 127.0) - 0.97;
- for(i = 0; i < REV_COMBS * 2; i++)
+ for(i = 0; i < REV_COMBS * 2; ++i)
combfb[i] =
-exp((float)comblen[i] / (float)SAMPLE_RATE * log(0.001) / t);
//the feedback is negative because it removes the DC
@@ -266,7 +266,7 @@ void Reverb::setidelay(unsigned char Pidelay)
if(idelaylen > 1) {
idelayk = 0;
idelay = new float[idelaylen];
- for(int i = 0; i < idelaylen; i++)
+ for(int i = 0; i < idelaylen; ++i)
idelay[i] = 0.0;
}
}
@@ -336,7 +336,7 @@ void Reverb::settype(unsigned char Ptype)
this->Ptype = Ptype;
float tmp;
- for(int i = 0; i < REV_COMBS * 2; i++) {
+ for(int i = 0; i < REV_COMBS * 2; ++i) {
if(Ptype == 0)
tmp = 800.0 + (int)(RND * 1400.0);
else
@@ -356,7 +356,7 @@ void Reverb::settype(unsigned char Ptype)
comb[i] = new float[comblen[i]];
}
- for(int i = 0; i < REV_APS * 2; i++) {
+ for(int i = 0; i < REV_APS * 2; ++i) {
if(Ptype == 0)
tmp = 500 + (int)(RND * 500);
else
@@ -441,7 +441,7 @@ void Reverb::setpreset(unsigned char npreset)
if(npreset >= NUM_PRESETS)
npreset = NUM_PRESETS - 1;
- for(int n = 0; n < PRESET_SIZE; n++)
+ for(int n = 0; n < PRESET_SIZE; ++n)
changepar(n, presets[npreset][n]);
if(insertion != 0)
changepar(0, presets[npreset][0] / 2); //lower the volume if reverb is insertion effect
diff --git a/src/Misc/Bank.cpp b/src/Misc/Bank.cpp
@@ -96,7 +96,7 @@ void Bank::setname(unsigned int ninstrument, const string &newname, int newslot)
snprintf(tmpfilename, 100, "%4d-%s", ninstrument + 1, newname.c_str());
//add the zeroes at the start of filename
- for(int i = 0; i < 4; i++)
+ for(int i = 0; i < 4; ++i)
if(tmpfilename[i] == ' ')
tmpfilename[i] = '0';
@@ -154,7 +154,7 @@ void Bank::savetoslot(unsigned int ninstrument, Part *part)
(char *)part->Pname);
//add the zeroes at the start of filename
- for(int i = 0; i < 4; i++)
+ for(int i = 0; i < 4; ++i)
if(tmpfilename[i] == ' ')
tmpfilename[i] = '0';
@@ -206,7 +206,7 @@ int Bank::loadbank(string bankdirname)
int no = 0;
unsigned int startname = 0;
- for(unsigned int i = 0; i < 4; i++) {
+ for(unsigned int i = 0; i < 4; ++i) {
if(strlen(filename) <= i)
break;
@@ -317,7 +317,7 @@ void Bank::rescanforbanks()
//remove old banks
banks.clear();
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++)
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
if(!config.cfg.bankRootDirList[i].empty())
scanrootdir(config.cfg.bankRootDirList[i]);
@@ -326,8 +326,8 @@ void Bank::rescanforbanks()
//remove duplicate bank names
int dupl = 0;
- for(int j = 0; j < (int) banks.size() - 1; j++) {
- for(int i = j + 1; i <(int) banks.size(); i++) {
+ for(int j = 0; j < (int) banks.size() - 1; ++j) {
+ for(int i = j + 1; i <(int) banks.size(); ++i) {
if(banks[i].name == banks[j].name) {
//add a [1] to the first bankname and [n] to others
banks[i].name = banks[i].name + '[' + stringFrom(dupl +2) + ']';
diff --git a/src/Misc/Config.cpp b/src/Misc/Config.cpp
@@ -66,9 +66,9 @@ void Config::init()
winmidimax = 1;
//try to find out how many input midi devices are there
winmididevices = new winmidionedevice[winmidimax];
- for(int i = 0; i < winmidimax; i++) {
+ for(int i = 0; i < winmidimax; ++i) {
winmididevices[i].name = new char[MAX_STRING_SIZE];
- for(int j = 0; j < MAX_STRING_SIZE; j++)
+ for(int j = 0; j < MAX_STRING_SIZE; ++j)
winmididevices[i].name[j] = '\0';
}
@@ -107,7 +107,7 @@ Config::~Config()
delete [] cfg.LinuxOSSWaveOutDev;
delete [] cfg.LinuxOSSSeqInDev;
- for(int i = 0; i < winmidimax; i++)
+ for(int i = 0; i < winmidimax; ++i)
delete [] winmididevices[i].name;
delete [] winmididevices;
}
@@ -122,13 +122,13 @@ void Config::save()
void Config::clearbankrootdirlist()
{
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++)
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
cfg.bankRootDirList[i].clear();
}
void Config::clearpresetsdirlist()
{
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++)
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
cfg.presetsDirList[i].clear();
}
@@ -196,7 +196,7 @@ void Config::readConfig(const char *filename)
10);
//get bankroot dirs
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++) {
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i) {
if(xmlcfg.enterbranch("BANKROOT", i)) {
cfg.bankRootDirList[i] = xmlcfg.getparstr("bank_root", "");
xmlcfg.exitbranch();
@@ -204,7 +204,7 @@ void Config::readConfig(const char *filename)
}
//get preset root dirs
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++) {
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i) {
if(xmlcfg.enterbranch("PRESETSROOT", i)) {
cfg.presetsDirList[i] = xmlcfg.getparstr("presets_root", "");
xmlcfg.exitbranch();
@@ -261,14 +261,14 @@ void Config::saveConfig(const char *filename)
xmlcfg->addpar("virtual_keyboard_layout", cfg.VirKeybLayout);
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++)
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
if(!cfg.bankRootDirList[i].empty()) {
xmlcfg->beginbranch("BANKROOT", i);
xmlcfg->addparstr("bank_root", cfg.bankRootDirList[i]);
xmlcfg->endbranch();
}
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++)
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i)
if(!cfg.presetsDirList[i].empty()) {
xmlcfg->beginbranch("PRESETSROOT", i);
xmlcfg->addparstr("presets_root", cfg.presetsDirList[i]);
diff --git a/src/Misc/Master.cpp b/src/Misc/Master.cpp
@@ -47,20 +47,20 @@ Master::Master()
fft = new FFTwrapper(OSCIL_SIZE);
shutup = 0;
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
vuoutpeakpart[npart] = 1e-9;
fakepeakpart[npart] = 0;
}
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++)
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
part[npart] = new Part(µtonal, fft, &mutex);
//Insertion Effects init
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
insefx[nefx] = new EffectMgr(1, &mutex);
//System Effects init
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx)
sysefx[nefx] = new EffectMgr(0, &mutex);
@@ -73,25 +73,25 @@ void Master::defaults()
setPvolume(80);
setPkeyshift(64);
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
part[npart]->defaults();
part[npart]->Prcvchn = npart % NUM_MIDI_CHANNELS;
}
partonoff(0, 1); //enable the first part
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx) {
insefx[nefx]->defaults();
Pinsparts[nefx] = -1;
}
//System Effects init
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx) {
sysefx[nefx]->defaults();
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++)
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
setPsysefxvol(npart, nefx, 0);
- for(int nefxto = 0; nefxto < NUM_SYS_EFX; nefxto++)
+ for(int nefxto = 0; nefxto < NUM_SYS_EFX; ++nefxto)
setPsysefxsend(nefx, nefxto, 0);
}
@@ -125,7 +125,7 @@ Master &Master::getInstance()
void Master::noteOn(char chan, char note, char velocity)
{
if(velocity) {
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
if(chan == part[npart]->Prcvchn) {
fakepeakpart[npart] = velocity * 2;
if(part[npart]->Penabled)
@@ -143,7 +143,7 @@ void Master::noteOn(char chan, char note, char velocity)
*/
void Master::noteOff(char chan, char note)
{
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++)
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
if((chan == part[npart]->Prcvchn) && part[npart]->Penabled)
part[npart]->NoteOff(note);
}
@@ -175,7 +175,7 @@ void Master::setController(char chan, int type, int par)
;
}
else { //other controllers
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) //Send the controller to all part assigned to the channel
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) //Send the controller to all part assigned to the channel
if((chan == part[npart]->Prcvchn) && (part[npart]->Penabled != 0))
part[npart]->SetController(type, par);
;
@@ -194,7 +194,7 @@ void Master::vuUpdate(const float *outl, const float *outr)
//Peak computation (for vumeters)
vu.outpeakl = 1e-12;
vu.outpeakr = 1e-12;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
if(fabs(outl[i]) > vu.outpeakl)
vu.outpeakl = fabs(outl[i]);
if(fabs(outr[i]) > vu.outpeakr)
@@ -210,7 +210,7 @@ void Master::vuUpdate(const float *outl, const float *outr)
//RMS Peak computation (for vumeters)
vu.rmspeakl = 1e-12;
vu.rmspeakr = 1e-12;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
vu.rmspeakl += outl[i] * outl[i];
vu.rmspeakr += outr[i] * outr[i];
}
@@ -218,12 +218,12 @@ void Master::vuUpdate(const float *outl, const float *outr)
vu.rmspeakr = sqrt(vu.rmspeakr / SOUND_BUFFER_SIZE);
//Part Peak computation (for Part vumeters or fake part vumeters)
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
vuoutpeakpart[npart] = 1.0e-12;
if(part[npart]->Penabled != 0) {
float *outl = part[npart]->partoutl,
*outr = part[npart]->partoutr;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmp = fabs(outl[i] + outr[i]);
if(tmp > vuoutpeakpart[npart])
vuoutpeakpart[npart] = tmp;
@@ -247,7 +247,7 @@ void Master::partonoff(int npart, int what)
fakepeakpart[npart] = 0;
part[npart]->Penabled = 0;
part[npart]->cleanup();
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx) {
if(Pinsparts[nefx] == npart)
insefx[nefx]->cleanup();
;
@@ -273,12 +273,12 @@ void Master::AudioOut(float *outl, float *outr)
memset(outr, 0, sizeof(float) * SOUND_BUFFER_SIZE);
//Compute part samples and store them part[npart]->partoutl,partoutr
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++)
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
if(part[npart]->Penabled != 0)
part[npart]->ComputePartSmps();
//Insertion effects
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx) {
if(Pinsparts[nefx] >= 0) {
int efxpart = Pinsparts[nefx];
if(part[efxpart]->Penabled)
@@ -289,7 +289,7 @@ void Master::AudioOut(float *outl, float *outr)
//Apply the part volumes and pannings (after insertion effects)
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
if(part[npart]->Penabled == 0)
continue;
@@ -306,7 +306,7 @@ void Master::AudioOut(float *outl, float *outr)
//the volume or the panning has changed and needs interpolation
if(ABOVE_AMPLITUDE_THRESHOLD(oldvol.l, newvol.l)
|| ABOVE_AMPLITUDE_THRESHOLD(oldvol.r, newvol.r)) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
Stereo<float> vol(INTERPOLATE_AMPLITUDE(oldvol.l, newvol.l,
i, SOUND_BUFFER_SIZE),
INTERPOLATE_AMPLITUDE(oldvol.r, newvol.r,
@@ -318,7 +318,7 @@ void Master::AudioOut(float *outl, float *outr)
part[npart]->oldvolumer = newvol.r;
}
else {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //the volume did not changed
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //the volume did not changed
part[npart]->partoutl[i] *= newvol.l;
part[npart]->partoutr[i] *= newvol.r;
}
@@ -327,7 +327,7 @@ void Master::AudioOut(float *outl, float *outr)
//System effects
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx) {
if(sysefx[nefx]->geteffect() == 0)
continue; //the effect is disabled
@@ -338,7 +338,7 @@ void Master::AudioOut(float *outl, float *outr)
memset(tmpmixr, 0, sizeof(float) * SOUND_BUFFER_SIZE);
//Mix the channels according to the part settings about System Effect
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
//skip if the part has no output to effect
if(Psysefxvol[nefx][npart] == 0)
continue;
@@ -349,17 +349,17 @@ void Master::AudioOut(float *outl, float *outr)
//the output volume of each part to system effect
const float vol = sysefxvol[nefx][npart];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
tmpmixl[i] += part[npart]->partoutl[i] * vol;
tmpmixr[i] += part[npart]->partoutr[i] * vol;
}
}
// system effect send to next ones
- for(int nefxfrom = 0; nefxfrom < nefx; nefxfrom++) {
+ for(int nefxfrom = 0; nefxfrom < nefx; ++nefxfrom) {
if(Psysefxsend[nefxfrom][nefx] != 0) {
const float vol = sysefxsend[nefxfrom][nefx];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
tmpmixl[i] += sysefx[nefxfrom]->efxoutl[i] * vol;
tmpmixr[i] += sysefx[nefxfrom]->efxoutr[i] * vol;
}
@@ -370,7 +370,7 @@ void Master::AudioOut(float *outl, float *outr)
//Add the System Effect to sound output
const float outvol = sysefx[nefx]->sysefxgetvolume();
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] += tmpmixl[i] * outvol;
outr[i] += tmpmixr[i] * outvol;
}
@@ -380,9 +380,9 @@ void Master::AudioOut(float *outl, float *outr)
}
//Mix all parts
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
if(part[npart]->Penabled) { //only mix active parts
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //the volume did not changed
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //the volume did not changed
outl[i] += part[npart]->partoutl[i];
outr[i] += part[npart]->partoutr[i];
}
@@ -390,13 +390,13 @@ void Master::AudioOut(float *outl, float *outr)
}
//Insertion effects for Master Out
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
if(Pinsparts[nefx] == -2)
insefx[nefx]->out(outl, outr);
//Master Volume
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= volume;
outr[i] *= volume;
}
@@ -408,7 +408,7 @@ void Master::AudioOut(float *outl, float *outr)
//Shutup if it is asked (with fade-out)
if(shutup) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmp =
(SOUND_BUFFER_SIZE - i) / (float) SOUND_BUFFER_SIZE;
outl[i] *= tmp;
@@ -468,11 +468,11 @@ void Master::GetAudioOutSamples(size_t nsamples,
Master::~Master()
{
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++)
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
delete part[npart];
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
delete insefx[nefx];
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx)
delete sysefx[nefx];
delete fft;
@@ -517,13 +517,13 @@ void Master::setPsysefxsend(int Pefxfrom, int Pefxto, char Pvol)
*/
void Master::ShutUp()
{
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
part[npart]->cleanup();
fakepeakpart[npart] = 0;
}
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
insefx[nefx]->cleanup();
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx)
sysefx[nefx]->cleanup();
vuresetpeaks();
shutup = 0;
@@ -555,7 +555,7 @@ vuData Master::getVuData()
void Master::applyparameters(bool lockmutex)
{
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++)
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
part[npart]->applyparameters(lockmutex);
}
@@ -569,26 +569,26 @@ void Master::add2XML(XMLwrapper *xml)
microtonal.add2XML(xml);
xml->endbranch();
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
xml->beginbranch("PART", npart);
part[npart]->add2XML(xml);
xml->endbranch();
}
xml->beginbranch("SYSTEM_EFFECTS");
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx) {
xml->beginbranch("SYSTEM_EFFECT", nefx);
xml->beginbranch("EFFECT");
sysefx[nefx]->add2XML(xml);
xml->endbranch();
- for(int pefx = 0; pefx < NUM_MIDI_PARTS; pefx++) {
+ for(int pefx = 0; pefx < NUM_MIDI_PARTS; ++pefx) {
xml->beginbranch("VOLUME", pefx);
xml->addpar("vol", Psysefxvol[nefx][pefx]);
xml->endbranch();
}
- for(int tonefx = nefx + 1; tonefx < NUM_SYS_EFX; tonefx++) {
+ for(int tonefx = nefx + 1; tonefx < NUM_SYS_EFX; ++tonefx) {
xml->beginbranch("SENDTO", tonefx);
xml->addpar("send_vol", Psysefxsend[nefx][tonefx]);
xml->endbranch();
@@ -600,7 +600,7 @@ void Master::add2XML(XMLwrapper *xml)
xml->endbranch();
xml->beginbranch("INSERTION_EFFECTS");
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx) {
xml->beginbranch("INSERTION_EFFECT", nefx);
xml->addpar("part", Pinsparts[nefx]);
@@ -691,7 +691,7 @@ void Master::getfromXML(XMLwrapper *xml)
part[0]->Penabled = 0;
- for(int npart = 0; npart < NUM_MIDI_PARTS; npart++) {
+ for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart) {
if(xml->enterbranch("PART", npart) == 0)
continue;
part[npart]->getfromXML(xml);
@@ -705,7 +705,7 @@ void Master::getfromXML(XMLwrapper *xml)
sysefx[0]->changeeffect(0);
if(xml->enterbranch("SYSTEM_EFFECTS")) {
- for(int nefx = 0; nefx < NUM_SYS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx) {
if(xml->enterbranch("SYSTEM_EFFECT", nefx) == 0)
continue;
if(xml->enterbranch("EFFECT")) {
@@ -713,7 +713,7 @@ void Master::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
- for(int partefx = 0; partefx < NUM_MIDI_PARTS; partefx++) {
+ for(int partefx = 0; partefx < NUM_MIDI_PARTS; ++partefx) {
if(xml->enterbranch("VOLUME", partefx) == 0)
continue;
setPsysefxvol(partefx, nefx,
@@ -721,7 +721,7 @@ void Master::getfromXML(XMLwrapper *xml)
xml->exitbranch();
}
- for(int tonefx = nefx + 1; tonefx < NUM_SYS_EFX; tonefx++) {
+ for(int tonefx = nefx + 1; tonefx < NUM_SYS_EFX; ++tonefx) {
if(xml->enterbranch("SENDTO", tonefx) == 0)
continue;
setPsysefxsend(nefx, tonefx,
@@ -736,7 +736,7 @@ void Master::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("INSERTION_EFFECTS")) {
- for(int nefx = 0; nefx < NUM_INS_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx) {
if(xml->enterbranch("INSERTION_EFFECT", nefx) == 0)
continue;
Pinsparts[nefx] = xml->getpar("part",
diff --git a/src/Misc/Microtonal.cpp b/src/Misc/Microtonal.cpp
@@ -49,10 +49,10 @@ void Microtonal::defaults()
Pmapsize = 12;
Pmappingenabled = 0;
- for(int i = 0; i < 128; i++)
+ for(int i = 0; i < 128; ++i)
Pmapping[i] = i;
- for(int i = 0; i < MAX_OCTAVE_SIZE; i++) {
+ for(int i = 0; i < MAX_OCTAVE_SIZE; ++i) {
octave[i].tuning = tmpoctave[i].tuning = pow(
2,
(i % octavesize
@@ -64,7 +64,7 @@ void Microtonal::defaults()
octave[11].type = 2;
octave[11].x1 = 2;
octave[11].x2 = 1;
- for(int i = 0; i < MICROTONAL_MAX_NAME_LEN; i++) {
+ for(int i = 0; i < MICROTONAL_MAX_NAME_LEN; ++i) {
Pname[i] = '\0';
Pcomment[i] = '\0';
}
@@ -137,7 +137,7 @@ float Microtonal::getnotefreq(int note, int keyshift) const
minus = 1;
}
int deltanote = 0;
- for(int i = 0; i < tmp; i++)
+ for(int i = 0; i < tmp; ++i)
if(Pmapping[i % Pmapsize] >= 0)
deltanote++;
float rap_anote_middlenote =
@@ -227,10 +227,10 @@ bool Microtonal::operator!=(const Microtonal µ) const
MCREQ(Pmapsize);
MCREQ(Pmappingenabled);
- for(int i = 0; i < 128; i++)
+ for(int i = 0; i < 128; ++i)
MCREQ(Pmapping[i]);
- for(int i = 0; i < octavesize; i++) {
+ for(int i = 0; i < octavesize; ++i) {
FMCREQ(octave[i].tuning);
MCREQ(octave[i].type);
MCREQ(octave[i].x1);
@@ -317,7 +317,7 @@ int Microtonal::texttotunings(const char *text)
char *lin;
lin = new char[MAX_LINE_SIZE + 1];
while(k < strlen(text)) {
- for(i = 0; i < MAX_LINE_SIZE; i++) {
+ for(i = 0; i < MAX_LINE_SIZE; ++i) {
lin[i] = text[k++];
if(lin[i] < 0x20)
break;
@@ -338,7 +338,7 @@ int Microtonal::texttotunings(const char *text)
if(nl == 0)
return -2; //the input is empty
octavesize = nl;
- for(i = 0; i < octavesize; i++) {
+ for(i = 0; i < octavesize; ++i) {
octave[i].tuning = tmpoctave[i].tuning;
octave[i].type = tmpoctave[i].type;
octave[i].x1 = tmpoctave[i].x1;
@@ -355,11 +355,11 @@ void Microtonal::texttomapping(const char *text)
unsigned int i, k = 0;
char *lin;
lin = new char[MAX_LINE_SIZE + 1];
- for(i = 0; i < 128; i++)
+ for(i = 0; i < 128; ++i)
Pmapping[i] = -1;
int tx = 0;
while(k < strlen(text)) {
- for(i = 0; i < MAX_LINE_SIZE; i++) {
+ for(i = 0; i < MAX_LINE_SIZE; ++i) {
lin[i] = text[k++];
if(lin[i] < 0x20)
break;
@@ -420,7 +420,7 @@ int Microtonal::loadscl(const char *filename)
//loads the short description
if(loadline(file, &tmp[0]) != 0)
return 2;
- for(int i = 0; i < 500; i++)
+ for(int i = 0; i < 500; ++i)
if(tmp[i] < 32)
tmp[i] = 0;
snprintf((char *) Pname, MICROTONAL_MAX_NAME_LEN, "%s", tmp);
@@ -433,7 +433,7 @@ int Microtonal::loadscl(const char *filename)
if(nnotes > MAX_OCTAVE_SIZE)
return 2;
//load the tunnings
- for(int nline = 0; nline < nnotes; nline++) {
+ for(int nline = 0; nline < nnotes; ++nline) {
if(loadline(file, &tmp[0]) != 0)
return 2;
linetotunings(nline, &tmp[0]);
@@ -441,7 +441,7 @@ int Microtonal::loadscl(const char *filename)
fclose(file);
octavesize = nnotes;
- for(int i = 0; i < octavesize; i++) {
+ for(int i = 0; i < octavesize; ++i) {
octave[i].tuning = tmpoctave[i].tuning;
octave[i].type = tmpoctave[i].type;
octave[i].x1 = tmpoctave[i].x1;
@@ -525,7 +525,7 @@ int Microtonal::loadkbm(const char *filename)
//load the mappings
if(Pmapsize != 0) {
- for(int nline = 0; nline < Pmapsize; nline++) {
+ for(int nline = 0; nline < Pmapsize; ++nline) {
if(loadline(file, &tmp[0]) != 0)
return 2;
if(sscanf(&tmp[0], "%d", &x) == 0)
@@ -571,7 +571,7 @@ void Microtonal::add2XML(XMLwrapper *xml) const
xml->beginbranch("OCTAVE");
xml->addpar("octave_size", octavesize);
- for(int i = 0; i < octavesize; i++) {
+ for(int i = 0; i < octavesize; ++i) {
xml->beginbranch("DEGREE", i);
if(octave[i].type == 1)
xml->addparreal("cents", octave[i].tuning);
@@ -587,7 +587,7 @@ void Microtonal::add2XML(XMLwrapper *xml) const
xml->beginbranch("KEYBOARD_MAPPING");
xml->addpar("map_size", Pmapsize);
xml->addpar("mapping_enabled", Pmappingenabled);
- for(int i = 0; i < Pmapsize; i++) {
+ for(int i = 0; i < Pmapsize; ++i) {
xml->beginbranch("KEYMAP", i);
xml->addpar("degree", Pmapping[i]);
xml->endbranch();
@@ -620,7 +620,7 @@ void Microtonal::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("OCTAVE")) {
octavesize = xml->getpar127("octave_size", octavesize);
- for(int i = 0; i < octavesize; i++) {
+ for(int i = 0; i < octavesize; ++i) {
if(xml->enterbranch("DEGREE", i) == 0)
continue;
octave[i].x2 = 0;
@@ -641,7 +641,7 @@ void Microtonal::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("KEYBOARD_MAPPING")) {
Pmapsize = xml->getpar127("map_size", Pmapsize);
Pmappingenabled = xml->getpar127("mapping_enabled", Pmappingenabled);
- for(int i = 0; i < Pmapsize; i++) {
+ for(int i = 0; i < Pmapsize; ++i) {
if(xml->enterbranch("KEYMAP", i) == 0)
continue;
Pmapping[i] = xml->getpar127("degree", Pmapping[i]);
diff --git a/src/Misc/Part.cpp b/src/Misc/Part.cpp
@@ -43,7 +43,7 @@ Part::Part(Microtonal *microtonal_, FFTwrapper *fft_, pthread_mutex_t *mutex_)
partoutl = new float [SOUND_BUFFER_SIZE];
partoutr = new float [SOUND_BUFFER_SIZE];
- for(int n = 0; n < NUM_KIT_ITEMS; n++) {
+ for(int n = 0; n < NUM_KIT_ITEMS; ++n) {
kit[n].Pname = new unsigned char [PART_MAX_NAME_LEN];
kit[n].adpars = NULL;
kit[n].subpars = NULL;
@@ -55,12 +55,12 @@ Part::Part(Microtonal *microtonal_, FFTwrapper *fft_, pthread_mutex_t *mutex_)
kit[0].padpars = new PADnoteParameters(fft, mutex);
//Part's Insertion Effects init
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) {
partefx[nefx] = new EffectMgr(1, mutex);
Pefxbypass[nefx] = false;
}
- for(int n = 0; n < NUM_PART_EFX + 1; n++) {
+ for(int n = 0; n < NUM_PART_EFX + 1; ++n) {
partfxinputl[n] = new float [SOUND_BUFFER_SIZE];
partfxinputr[n] = new float [SOUND_BUFFER_SIZE];
}
@@ -68,11 +68,11 @@ Part::Part(Microtonal *microtonal_, FFTwrapper *fft_, pthread_mutex_t *mutex_)
killallnotes = 0;
oldfreq = -1.0;
- for(int i = 0; i < POLIPHONY; i++) {
+ for(int i = 0; i < POLIPHONY; ++i) {
partnote[i].status = KEY_OFF;
partnote[i].note = -1;
partnote[i].itemsplaying = 0;
- for(int j = 0; j < NUM_KIT_ITEMS; j++) {
+ for(int j = 0; j < NUM_KIT_ITEMS; ++j) {
partnote[i].kititem[j].adnote = NULL;
partnote[i].kititem[j].subnote = NULL;
partnote[i].kititem[j].padnote = NULL;
@@ -121,7 +121,7 @@ void Part::defaultsinstrument()
Pkitmode = 0;
Pdrummode = 0;
- for(int n = 0; n < NUM_KIT_ITEMS; n++) {
+ for(int n = 0; n < NUM_KIT_ITEMS; ++n) {
kit[n].Penabled = 0;
kit[n].Pmuted = 0;
kit[n].Pminkey = 0;
@@ -140,7 +140,7 @@ void Part::defaultsinstrument()
kit[0].subpars->defaults();
kit[0].padpars->defaults();
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) {
partefx[nefx]->defaults();
Pefxroute[nefx] = 0; //route to next effect
}
@@ -153,17 +153,17 @@ void Part::defaultsinstrument()
*/
void Part::cleanup(bool final)
{
- for(int k = 0; k < POLIPHONY; k++)
+ for(int k = 0; k < POLIPHONY; ++k)
KillNotePos(k);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partoutl[i] = final ? 0.0 : denormalkillbuf[i];
partoutr[i] = final ? 0.0 : denormalkillbuf[i];
}
ctl.resetall();
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx)
partefx[nefx]->cleanup();
- for(int n = 0; n < NUM_PART_EFX + 1; n++) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int n = 0; n < NUM_PART_EFX + 1; ++n) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[n][i] = final ? 0.0 : denormalkillbuf[i];
partfxinputr[n][i] = final ? 0.0 : denormalkillbuf[i];
}
@@ -173,7 +173,7 @@ void Part::cleanup(bool final)
Part::~Part()
{
cleanup(true);
- for(int n = 0; n < NUM_KIT_ITEMS; n++) {
+ for(int n = 0; n < NUM_KIT_ITEMS; ++n) {
if(kit[n].adpars != NULL)
delete (kit[n].adpars);
if(kit[n].subpars != NULL)
@@ -189,9 +189,9 @@ Part::~Part()
delete [] Pname;
delete [] partoutl;
delete [] partoutr;
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx)
delete (partefx[nefx]);
- for(int n = 0; n < NUM_PART_EFX + 1; n++) {
+ for(int n = 0; n < NUM_PART_EFX + 1; ++n) {
delete [] partfxinputl[n];
delete [] partfxinputr[n];
}
@@ -237,7 +237,7 @@ void Part::NoteOn(unsigned char note,
lastnote = note;
pos = -1;
- for(i = 0; i < POLIPHONY; i++) {
+ for(i = 0; i < POLIPHONY; ++i) {
if(partnote[i].status == KEY_OFF) {
pos = i;
break;
@@ -262,14 +262,14 @@ void Part::NoteOn(unsigned char note,
}
else {
// Legato mode is valid, but this is only a first note.
- for(i = 0; i < POLIPHONY; i++)
+ for(i = 0; i < POLIPHONY; ++i)
if((partnote[i].status == KEY_PLAYING)
|| (partnote[i].status == KEY_RELASED_AND_SUSTAINED))
RelaseNotePos(i);
// Set posb
posb = (pos + 1) % POLIPHONY; //We really want it (if the following fails)
- for(i = 0; i < POLIPHONY; i++)
+ for(i = 0; i < POLIPHONY; ++i)
if((partnote[i].status == KEY_OFF) && (pos != i)) {
posb = i;
break;
@@ -280,7 +280,7 @@ void Part::NoteOn(unsigned char note,
}
else // Legato mode is either off or non-applicable.
if(Ppolymode == 0) { //if the mode is 'mono' turn off all other notes
- for(i = 0; i < POLIPHONY; i++)
+ for(i = 0; i < POLIPHONY; ++i)
if(partnote[i].status == KEY_PLAYING)
RelaseNotePos(i);
RelaseSustainedKeys();
@@ -380,7 +380,7 @@ void Part::NoteOn(unsigned char note,
}
else { // "kit mode" legato note
int ci = 0;
- for(int item = 0; item < NUM_KIT_ITEMS; item++) {
+ for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
if(kit[item].Pmuted != 0)
continue;
if((note < kit[item].Pminkey) || (note > kit[item].Pmaxkey))
@@ -480,7 +480,7 @@ void Part::NoteOn(unsigned char note,
}
}
else { //init the notes for the "kit mode"
- for(int item = 0; item < NUM_KIT_ITEMS; item++) {
+ for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
if(kit[item].Pmuted != 0)
continue;
if((note < kit[item].Pminkey) || (note > kit[item].Pmaxkey))
@@ -631,7 +631,7 @@ void Part::SetController(unsigned int type, int par)
setPvolume(Pvolume); //update the volume
setPpanning(Ppanning); //update the panning
- for(int item = 0; item < NUM_KIT_ITEMS; item++) {
+ for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
if(kit[item].adpars == NULL)
continue;
kit[item].adpars->GlobalPar.Reson->
@@ -647,7 +647,7 @@ void Part::SetController(unsigned int type, int par)
break;
case C_resonance_center:
ctl.setresonancecenter(par);
- for(int item = 0; item < NUM_KIT_ITEMS; item++) {
+ for(int item = 0; item < NUM_KIT_ITEMS; ++item) {
if(kit[item].adpars == NULL)
continue;
kit[item].adpars->GlobalPar.Reson->
@@ -672,7 +672,7 @@ void Part::RelaseSustainedKeys()
if(monomemnotes.back() != lastnote) // Sustain controller manipulation would cause repeated same note respawn without this check.
MonoMemRenote(); // To play most recent still held note.
- for(int i = 0; i < POLIPHONY; i++)
+ for(int i = 0; i < POLIPHONY; ++i)
if(partnote[i].status == KEY_RELASED_AND_SUSTAINED)
RelaseNotePos(i);
}
@@ -683,7 +683,7 @@ void Part::RelaseSustainedKeys()
void Part::RelaseAllKeys()
{
- for(int i = 0; i < POLIPHONY; i++)
+ for(int i = 0; i < POLIPHONY; ++i)
if((partnote[i].status != KEY_RELASED)
&& (partnote[i].status != KEY_OFF)) //thanks to Frank Neumann
RelaseNotePos(i);
@@ -707,7 +707,7 @@ void Part::MonoMemRenote()
*/
void Part::RelaseNotePos(int pos)
{
- for(int j = 0; j < NUM_KIT_ITEMS; j++) {
+ for(int j = 0; j < NUM_KIT_ITEMS; ++j) {
if(partnote[pos].kititem[j].adnote != NULL)
if(partnote[pos].kititem[j].adnote)
partnote[pos].kititem[j].adnote->relasekey();
@@ -734,7 +734,7 @@ void Part::KillNotePos(int pos)
partnote[pos].time = 0;
partnote[pos].itemsplaying = 0;
- for(int j = 0; j < NUM_KIT_ITEMS; j++) {
+ for(int j = 0; j < NUM_KIT_ITEMS; ++j) {
if(partnote[pos].kititem[j].adnote != NULL) {
delete (partnote[pos].kititem[j].adnote);
partnote[pos].kititem[j].adnote = NULL;
@@ -768,14 +768,14 @@ void Part::setkeylimit(unsigned char Pkeylimit)
//release old keys if the number of notes>keylimit
if(Ppolymode != 0) {
int notecount = 0;
- for(int i = 0; i < POLIPHONY; i++)
+ for(int i = 0; i < POLIPHONY; ++i)
if((partnote[i].status == KEY_PLAYING)
|| (partnote[i].status == KEY_RELASED_AND_SUSTAINED))
notecount++;
int oldestnotepos = -1;
if(notecount > keylimit) { //find out the oldest note
- for(int i = 0; i < POLIPHONY; i++) {
+ for(int i = 0; i < POLIPHONY; ++i) {
int maxtime = 0;
if(((partnote[i].status == KEY_PLAYING)
|| (partnote[i].status == KEY_RELASED_AND_SUSTAINED))
@@ -802,7 +802,7 @@ void Part::AllNotesOff()
void Part::RunNote(unsigned int k)
{
unsigned noteplay = 0;
- for(int item = 0; item < partnote[k].itemsplaying; item++) {
+ for(int item = 0; item < partnote[k].itemsplaying; ++item) {
int sendcurrenttofx = partnote[k].kititem[item].sendtoparteffect;
for(unsigned type = 0; type < 3; ++type) {
@@ -828,7 +828,7 @@ void Part::RunNote(unsigned int k)
delete (*note);
(*note) = NULL;
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //add the note to part(mix)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //add the note to part(mix)
partfxinputl[sendcurrenttofx][i] += tmpoutl[i];
partfxinputr[sendcurrenttofx][i] += tmpoutr[i];
}
@@ -847,14 +847,14 @@ void Part::RunNote(unsigned int k)
*/
void Part::ComputePartSmps()
{
- for(unsigned nefx = 0; nefx < NUM_PART_EFX + 1; nefx++) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(unsigned nefx = 0; nefx < NUM_PART_EFX + 1; ++nefx) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[nefx][i] = 0.0;
partfxinputr[nefx][i] = 0.0;
}
}
- for(unsigned k = 0; k < POLIPHONY; k++) {
+ for(unsigned k = 0; k < POLIPHONY; ++k) {
if(partnote[k].status == KEY_OFF)
continue;
partnote[k].time++;
@@ -864,39 +864,39 @@ void Part::ComputePartSmps()
//Apply part's effects and mix them
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) {
if(!Pefxbypass[nefx]) {
partefx[nefx]->out(partfxinputl[nefx], partfxinputr[nefx]);
if(Pefxroute[nefx] == 2) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[nefx + 1][i] += partefx[nefx]->efxoutl[i];
partfxinputr[nefx + 1][i] += partefx[nefx]->efxoutr[i];
}
}
}
int routeto = ((Pefxroute[nefx] == 0) ? nefx + 1 : NUM_PART_EFX);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partfxinputl[routeto][i] += partfxinputl[nefx][i];
partfxinputr[routeto][i] += partfxinputr[nefx][i];
}
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
partoutl[i] = partfxinputl[NUM_PART_EFX][i];
partoutr[i] = partfxinputr[NUM_PART_EFX][i];
}
//Kill All Notes if killallnotes!=0
if(killallnotes != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmp =
(SOUND_BUFFER_SIZE - i) / (float) SOUND_BUFFER_SIZE;
partoutl[i] *= tmp;
partoutr[i] *= tmp;
}
- for(int k = 0; k < POLIPHONY; k++)
+ for(int k = 0; k < POLIPHONY; ++k)
KillNotePos(k);
killallnotes = 0;
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++)
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx)
partefx[nefx]->cleanup();
}
@@ -957,7 +957,7 @@ void Part::setkititemstatus(int kititem, int Penabled_)
}
if(resetallnotes)
- for(int k = 0; k < POLIPHONY; k++)
+ for(int k = 0; k < POLIPHONY; ++k)
KillNotePos(k);
}
@@ -975,7 +975,7 @@ void Part::add2XMLinstrument(XMLwrapper *xml)
xml->addpar("kit_mode", Pkitmode);
xml->addparbool("drum_mode", Pdrummode);
- for(int i = 0; i < NUM_KIT_ITEMS; i++) {
+ for(int i = 0; i < NUM_KIT_ITEMS; ++i) {
xml->beginbranch("INSTRUMENT_KIT_ITEM", i);
xml->addparbool("enabled", kit[i].Penabled);
if(kit[i].Penabled != 0) {
@@ -1013,7 +1013,7 @@ void Part::add2XMLinstrument(XMLwrapper *xml)
xml->endbranch();
xml->beginbranch("INSTRUMENT_EFFECTS");
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) {
xml->beginbranch("INSTRUMENT_EFFECT", nefx);
xml->beginbranch("EFFECT");
partefx[nefx]->add2XML(xml);
@@ -1092,7 +1092,7 @@ int Part::loadXMLinstrument(const char *filename)/*{*/
void Part::applyparameters(bool lockmutex)/*{*/
{
- for(int n = 0; n < NUM_KIT_ITEMS; n++)
+ for(int n = 0; n < NUM_KIT_ITEMS; ++n)
if((kit[n].padpars != NULL) && (kit[n].Ppadenabled != 0))
kit[n].padpars->applyparameters(lockmutex);
}/*}*/
@@ -1113,7 +1113,7 @@ void Part::getfromXMLinstrument(XMLwrapper *xml)
Pdrummode = xml->getparbool("drum_mode", Pdrummode);
setkititemstatus(0, 0);
- for(int i = 0; i < NUM_KIT_ITEMS; i++) {
+ for(int i = 0; i < NUM_KIT_ITEMS; ++i) {
if(xml->enterbranch("INSTRUMENT_KIT_ITEM", i) == 0)
continue;
setkititemstatus(i, xml->getparbool("enabled", kit[i].Penabled));
@@ -1161,7 +1161,7 @@ void Part::getfromXMLinstrument(XMLwrapper *xml)
if(xml->enterbranch("INSTRUMENT_EFFECTS")) {
- for(int nefx = 0; nefx < NUM_PART_EFX; nefx++) {
+ for(int nefx = 0; nefx < NUM_PART_EFX; ++nefx) {
if(xml->enterbranch("INSTRUMENT_EFFECT", nefx) == 0)
continue;
if(xml->enterbranch("EFFECT")) {
diff --git a/src/Misc/Util.cpp b/src/Misc/Util.cpp
@@ -133,7 +133,7 @@ void os_sleep(long length)
std::string legalizeFilename(std::string filename)
{
- for(int i = 0; i < (int) filename.size(); i++) {
+ for(int i = 0; i < (int) filename.size(); ++i) {
char c = filename[i];
if(!(isdigit(c) || isalpha(c) || (c == '-') || (c == ' ')))
filename[i] = '_';
@@ -143,7 +143,7 @@ std::string legalizeFilename(std::string filename)
void invSignal(float *sig, size_t len)
{
- for(size_t i = 0; i < len; i++)
+ for(size_t i = 0; i < len; ++i)
sig[i] *= -1.0f;
}
diff --git a/src/Misc/WaveShapeSmps.cpp b/src/Misc/WaveShapeSmps.cpp
@@ -35,7 +35,7 @@ void waveShapeSmps(int n,
switch(type) {
case 1:
ws = pow(10, ws * ws * 3.0) - 1.0 + 0.001; //Arctangent
- for(i = 0; i < n; i++)
+ for(i = 0; i < n; ++i)
smps[i] = atan(smps[i] * ws) / atan(ws);
break;
case 2:
@@ -44,13 +44,13 @@ void waveShapeSmps(int n,
tmpv = sin(ws) + 0.1;
else
tmpv = 1.1;
- for(i = 0; i < n; i++)
+ for(i = 0; i < n; ++i)
smps[i] = sin(smps[i] * (0.1 + ws - ws * smps[i])) / tmpv;
;
break;
case 3:
ws = ws * ws * ws * 20.0 + 0.0001; //Pow
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
smps[i] *= ws;
if(fabs(smps[i]) < 1.0) {
smps[i] = (smps[i] - pow(smps[i], 3.0)) * 3.0;
@@ -67,12 +67,12 @@ void waveShapeSmps(int n,
tmpv = sin(ws);
else
tmpv = 1.0;
- for(i = 0; i < n; i++)
+ for(i = 0; i < n; ++i)
smps[i] = sin(smps[i] * ws) / tmpv;
break;
case 5:
ws = ws * ws + 0.000001; //Quantisize
- for(i = 0; i < n; i++)
+ for(i = 0; i < n; ++i)
smps[i] = floor(smps[i] / ws + 0.5) * ws;
break;
case 6:
@@ -81,12 +81,12 @@ void waveShapeSmps(int n,
tmpv = sin(ws);
else
tmpv = 1.0;
- for(i = 0; i < n; i++)
+ for(i = 0; i < n; ++i)
smps[i] = asin(sin(smps[i] * ws)) / tmpv;
break;
case 7:
ws = pow(2.0, -ws * ws * 8.0); //Limiter
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i];
if(fabs(tmp) > ws) {
if(tmp >= 0.0)
@@ -100,7 +100,7 @@ void waveShapeSmps(int n,
break;
case 8:
ws = pow(2.0, -ws * ws * 8.0); //Upper Limiter
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i];
if(tmp > ws)
smps[i] = ws;
@@ -109,7 +109,7 @@ void waveShapeSmps(int n,
break;
case 9:
ws = pow(2.0, -ws * ws * 8.0); //Lower Limiter
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i];
if(tmp < -ws)
smps[i] = -ws;
@@ -118,7 +118,7 @@ void waveShapeSmps(int n,
break;
case 10:
ws = (pow(2.0, ws * 6.0) - 1.0) / pow(2.0, 6.0); //Inverse Limiter
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i];
if(fabs(tmp) > ws) {
if(tmp >= 0.0)
@@ -132,7 +132,7 @@ void waveShapeSmps(int n,
break;
case 11:
ws = pow(5, ws * ws * 1.0) - 1.0; //Clip
- for(i = 0; i < n; i++)
+ for(i = 0; i < n; ++i)
smps[i] = smps[i]
* (ws + 0.5) * 0.9999 - floor(
0.5 + smps[i] * (ws + 0.5) * 0.9999);
@@ -143,7 +143,7 @@ void waveShapeSmps(int n,
tmpv = ws;
else
tmpv = 1.0;
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i] * ws;
if((tmp > -2.0) && (tmp < 1.0))
smps[i] = tmp * (1.0 - tmp) * (tmp + 2.0) / tmpv;
@@ -157,7 +157,7 @@ void waveShapeSmps(int n,
tmpv = ws * (1 + ws) / 2.0;
else
tmpv = 1.0;
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i] * ws;
if((tmp > -1.0) && (tmp < 1.618034))
smps[i] = tmp * (1.0 - tmp) / tmpv;
@@ -174,7 +174,7 @@ void waveShapeSmps(int n,
tmpv = 0.5;
else
tmpv = 0.5 - 1.0 / (exp(ws) + 1.0);
- for(i = 0; i < n; i++) {
+ for(i = 0; i < n; ++i) {
float tmp = smps[i] * ws;
if(tmp < -10.0)
tmp = -10.0;
diff --git a/src/Nio/OssEngine.cpp b/src/Nio/OssEngine.cpp
@@ -212,7 +212,7 @@ void *OssEngine::thread()
const Stereo<float *> smps = getNext();
float l, r;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
l = smps.l[i];
r = smps.r[i];
diff --git a/src/Nio/PaEngine.cpp b/src/Nio/PaEngine.cpp
@@ -100,7 +100,7 @@ int PaEngine::process(float *out, unsigned long framesPerBuffer)
// cerr << "Bug: PaEngine::process SOUND_BUFFER_SIZE!=framesPerBuffer"
// << framesPerBuffer << ' ' << smp.l.size() << endl;
- for(int i = 0; i < framesPerBuffer; i++) {
+ for(int i = 0; i < framesPerBuffer; ++i) {
*out++ = smp.l[i];
*out++ = smp.r[i];
}
diff --git a/src/Output/DSSIaudiooutput.cpp b/src/Output/DSSIaudiooutput.cpp
@@ -674,7 +674,7 @@ bool DSSIaudiooutput::mapNextBank()
else
{
bank.loadbank(bank.banks[bankNoToMap].dir);
- for(unsigned long instrument = 0; instrument < BANK_SIZE; instrument++)
+ for(unsigned long instrument = 0; instrument < BANK_SIZE; ++instrument)
{
string insName = bank.getname(instrument);
if(!insName.empty() && insName[0] != '\0' && insName[0] != ' ')
diff --git a/src/Params/ADnoteParameters.cpp b/src/Params/ADnoteParameters.cpp
@@ -43,7 +43,7 @@ ADnoteParameters::ADnoteParameters(FFTwrapper *fft_)
fft = fft_;
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)
EnableVoice(nvoice);
defaults();
@@ -71,7 +71,7 @@ void ADnoteParameters::defaults()
//Default Parameters
GlobalPar.defaults();
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)
defaults(nvoice);
VoicePar[0].Enabled = 1;
@@ -280,7 +280,7 @@ ADnoteGlobalParam::~ADnoteGlobalParam()
ADnoteParameters::~ADnoteParameters()
{
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)
KillVoice(nvoice);
}
@@ -304,7 +304,7 @@ int ADnoteParameters::get_unison_size_index(int nvoice) {
void ADnoteParameters::set_unison_size_index(int nvoice, int index) {
int unison = 1;
- for(int i = 0; i <= index; i++) {
+ for(int i = 0; i <= index; ++i) {
unison = ADnote_unison_sizes[i];
if(unison == 0) {
unison = ADnote_unison_sizes[i - 1];
@@ -325,7 +325,7 @@ void ADnoteParameters::add2XMLsection(XMLwrapper *xml, int n)
int oscilused = 0, fmoscilused = 0; //if the oscil or fmoscil are used by another voice
- for(int i = 0; i < NUM_VOICES; i++) {
+ for(int i = 0; i < NUM_VOICES; ++i) {
if(VoicePar[i].Pextoscil == nvoice)
oscilused = 1;
if(VoicePar[i].PextFMoscil == nvoice)
@@ -544,7 +544,7 @@ void ADnoteGlobalParam::add2XML(XMLwrapper *xml)
void ADnoteParameters::add2XML(XMLwrapper *xml)
{
GlobalPar.add2XML(xml);
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
xml->beginbranch("VOICE", nvoice);
add2XMLsection(xml, nvoice);
xml->endbranch();
@@ -631,7 +631,7 @@ void ADnoteParameters::getfromXML(XMLwrapper *xml)
{
GlobalPar.getfromXML(xml);
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
VoicePar[nvoice].Enabled = 0;
if(xml->enterbranch("VOICE", nvoice) == 0)
continue;
diff --git a/src/Params/EnvelopeParams.cpp b/src/Params/EnvelopeParams.cpp
@@ -39,7 +39,7 @@ EnvelopeParams::EnvelopeParams(unsigned char Penvstretch_,
PS_val = 64;
PR_val = 64;
- for(i = 0; i < MAX_ENVELOPE_POINTS; i++) {
+ for(i = 0; i < MAX_ENVELOPE_POINTS; ++i) {
Penvdt[i] = 32;
Penvval[i] = 64;
}
@@ -223,7 +223,7 @@ void EnvelopeParams::add2XML(XMLwrapper *xml)
xml->addpar("R_val", PR_val);
if((Pfreemode != 0) || (!xml->minimal)) {
- for(int i = 0; i < Penvpoints; i++) {
+ for(int i = 0; i < Penvpoints; ++i) {
xml->beginbranch("POINT", i);
if(i != 0)
xml->addpar("dt", Penvdt[i]);
@@ -252,7 +252,7 @@ void EnvelopeParams::getfromXML(XMLwrapper *xml)
PS_val = xml->getpar127("S_val", PS_val);
PR_val = xml->getpar127("R_val", PR_val);
- for(int i = 0; i < Penvpoints; i++) {
+ for(int i = 0; i < Penvpoints; ++i) {
if(xml->enterbranch("POINT", i) == 0)
continue;
if(i != 0)
diff --git a/src/Params/FilterParams.cpp b/src/Params/FilterParams.cpp
@@ -56,12 +56,12 @@ void FilterParams::defaults()
Pnumformants = 3;
Pformantslowness = 64;
- for(int j = 0; j < FF_MAX_VOWELS; j++)
+ for(int j = 0; j < FF_MAX_VOWELS; ++j)
defaults(j);
;
Psequencesize = 3;
- for(int i = 0; i < FF_MAX_SEQUENCE; i++)
+ for(int i = 0; i < FF_MAX_SEQUENCE; ++i)
Psequence[i].nvowel = i % FF_MAX_VOWELS;
Psequencestretch = 40;
@@ -74,7 +74,7 @@ void FilterParams::defaults()
void FilterParams::defaults(int n)
{
int j = n;
- for(int i = 0; i < FF_MAX_FORMANTS; i++) {
+ for(int i = 0; i < FF_MAX_FORMANTS; ++i) {
Pvowels[j].formants[i].freq = (int)(RND * 127.0); //some random freqs
Pvowels[j].formants[i].q = 64;
Pvowels[j].formants[i].amp = 127;
@@ -104,8 +104,8 @@ void FilterParams::getfromFilterParams(FilterParams *pars)
Pnumformants = pars->Pnumformants;
Pformantslowness = pars->Pformantslowness;
- for(int j = 0; j < FF_MAX_VOWELS; j++) {
- for(int i = 0; i < FF_MAX_FORMANTS; i++) {
+ for(int j = 0; j < FF_MAX_VOWELS; ++j) {
+ for(int i = 0; i < FF_MAX_FORMANTS; ++i) {
Pvowels[j].formants[i].freq = pars->Pvowels[j].formants[i].freq;
Pvowels[j].formants[i].q = pars->Pvowels[j].formants[i].q;
Pvowels[j].formants[i].amp = pars->Pvowels[j].formants[i].amp;
@@ -113,7 +113,7 @@ void FilterParams::getfromFilterParams(FilterParams *pars)
}
Psequencesize = pars->Psequencesize;
- for(int i = 0; i < FF_MAX_SEQUENCE; i++)
+ for(int i = 0; i < FF_MAX_SEQUENCE; ++i)
Psequence[i].nvowel = pars->Psequence[i].nvowel;
Psequencestretch = pars->Psequencestretch;
@@ -191,11 +191,11 @@ void FilterParams::formantfilterH(int nvowel, int nfreqs, float *freqs)
float filter_freq, filter_q, filter_amp;
float omega, sn, cs, alpha;
- for(int i = 0; i < nfreqs; i++)
+ for(int i = 0; i < nfreqs; ++i)
freqs[i] = 0.0;
//for each formant...
- for(int nformant = 0; nformant < Pnumformants; nformant++) {
+ for(int nformant = 0; nformant < Pnumformants; ++nformant) {
//compute formant parameters(frequency,amplitude,etc.)
filter_freq = getformantfreq(Pvowels[nvowel].formants[nformant].freq);
filter_q = getformantq(Pvowels[nvowel].formants[nformant].q) * getq();
@@ -222,23 +222,23 @@ void FilterParams::formantfilterH(int nvowel, int nfreqs, float *freqs)
continue;
- for(int i = 0; i < nfreqs; i++) {
+ for(int i = 0; i < nfreqs; ++i) {
float freq = getfreqx(i / (float) nfreqs);
if(freq > SAMPLE_RATE / 2) {
- for(int tmp = i; tmp < nfreqs; tmp++)
+ for(int tmp = i; tmp < nfreqs; ++tmp)
freqs[tmp] = 0.0;
break;
}
float fr = freq / SAMPLE_RATE * PI * 2.0;
float x = c[0], y = 0.0;
- for(int n = 1; n < 3; n++) {
+ for(int n = 1; n < 3; ++n) {
x += cos(n * fr) * c[n];
y -= sin(n * fr) * c[n];
}
float h = x * x + y * y;
x = 1.0;
y = 0.0;
- for(int n = 1; n < 3; n++) {
+ for(int n = 1; n < 3; ++n) {
x -= cos(n * fr) * d[n];
y += sin(n * fr) * d[n];
}
@@ -247,7 +247,7 @@ void FilterParams::formantfilterH(int nvowel, int nfreqs, float *freqs)
freqs[i] += pow(h, (Pstages + 1.0) / 2.0) * filter_amp;
}
}
- for(int i = 0; i < nfreqs; i++) {
+ for(int i = 0; i < nfreqs; ++i) {
if(freqs[i] > 0.000000001)
freqs[i] = rap2dB(freqs[i]) + getgain();
else
@@ -282,7 +282,7 @@ float FilterParams::getformantq(unsigned char q)
void FilterParams::add2XMLsection(XMLwrapper *xml, int n)
{
int nvowel = n;
- for(int nformant = 0; nformant < FF_MAX_FORMANTS; nformant++) {
+ for(int nformant = 0; nformant < FF_MAX_FORMANTS; ++nformant) {
xml->beginbranch("FORMANT", nformant);
xml->addpar("freq", Pvowels[nvowel].formants[nformant].freq);
xml->addpar("amp", Pvowels[nvowel].formants[nformant].amp);
@@ -310,7 +310,7 @@ void FilterParams::add2XML(XMLwrapper *xml)
xml->addpar("vowel_clearness", Pvowelclearness);
xml->addpar("center_freq", Pcenterfreq);
xml->addpar("octaves_freq", Poctavesfreq);
- for(int nvowel = 0; nvowel < FF_MAX_VOWELS; nvowel++) {
+ for(int nvowel = 0; nvowel < FF_MAX_VOWELS; ++nvowel) {
xml->beginbranch("VOWEL", nvowel);
add2XMLsection(xml, nvowel);
xml->endbranch();
@@ -318,7 +318,7 @@ void FilterParams::add2XML(XMLwrapper *xml)
xml->addpar("sequence_size", Psequencesize);
xml->addpar("sequence_stretch", Psequencestretch);
xml->addparbool("sequence_reversed", Psequencereversed);
- for(int nseq = 0; nseq < FF_MAX_SEQUENCE; nseq++) {
+ for(int nseq = 0; nseq < FF_MAX_SEQUENCE; ++nseq) {
xml->beginbranch("SEQUENCE_POS", nseq);
xml->addpar("vowel_id", Psequence[nseq].nvowel);
xml->endbranch();
@@ -331,7 +331,7 @@ void FilterParams::add2XML(XMLwrapper *xml)
void FilterParams::getfromXMLsection(XMLwrapper *xml, int n)
{
int nvowel = n;
- for(int nformant = 0; nformant < FF_MAX_FORMANTS; nformant++) {
+ for(int nformant = 0; nformant < FF_MAX_FORMANTS; ++nformant) {
if(xml->enterbranch("FORMANT", nformant) == 0)
continue;
Pvowels[nvowel].formants[nformant].freq = xml->getpar127(
@@ -367,7 +367,7 @@ void FilterParams::getfromXML(XMLwrapper *xml)
Pcenterfreq = xml->getpar127("center_freq", Pcenterfreq);
Poctavesfreq = xml->getpar127("octaves_freq", Poctavesfreq);
- for(int nvowel = 0; nvowel < FF_MAX_VOWELS; nvowel++) {
+ for(int nvowel = 0; nvowel < FF_MAX_VOWELS; ++nvowel) {
if(xml->enterbranch("VOWEL", nvowel) == 0)
continue;
getfromXMLsection(xml, nvowel);
@@ -377,7 +377,7 @@ void FilterParams::getfromXML(XMLwrapper *xml)
Psequencestretch = xml->getpar127("sequence_stretch", Psequencestretch);
Psequencereversed = xml->getparbool("sequence_reversed",
Psequencereversed);
- for(int nseq = 0; nseq < FF_MAX_SEQUENCE; nseq++) {
+ for(int nseq = 0; nseq < FF_MAX_SEQUENCE; ++nseq) {
if(xml->enterbranch("SEQUENCE_POS", nseq) == 0)
continue;
Psequence[nseq].nvowel = xml->getpar("vowel_id",
diff --git a/src/Params/PADnoteParameters.cpp b/src/Params/PADnoteParameters.cpp
@@ -48,7 +48,7 @@ PADnoteParameters::PADnoteParameters(FFTwrapper *fft_,
FilterEnvelope->ADSRinit_filter(64, 40, 64, 70, 60, 64);
FilterLfo = new LFOParams(80, 0, 64, 0, 0, 0, 0, 2);
- for(int i = 0; i < PAD_MAX_SAMPLES; i++)
+ for(int i = 0; i < PAD_MAX_SAMPLES; ++i)
sample[i].smp = NULL;
newsample.smp = NULL;
@@ -147,7 +147,7 @@ void PADnoteParameters::deletesample(int n)
void PADnoteParameters::deletesamples()
{
- for(int i = 0; i < PAD_MAX_SAMPLES; i++)
+ for(int i = 0; i < PAD_MAX_SAMPLES; ++i)
deletesample(i);
}
@@ -156,7 +156,7 @@ void PADnoteParameters::deletesamples()
*/
float PADnoteParameters::getprofile(float *smp, int size)
{
- for(int i = 0; i < size; i++)
+ for(int i = 0; i < size; ++i)
smp[i] = 0.0;
const int supersample = 16;
float basepar = pow(2.0, (1.0 - Php.base.par1 / 127.0) * 12.0);
@@ -173,7 +173,7 @@ float PADnoteParameters::getprofile(float *smp, int size)
float amppar2 = (1.0 - Php.amp.par2 / 127.0) * 0.998 + 0.001;
float width = pow(150.0 / (Php.width + 22.0), 2.0);
- for(int i = 0; i < size * supersample; i++) {
+ for(int i = 0; i < size * supersample; ++i) {
bool makezero = false;
float x = i * 1.0 / (size * (float) supersample);
@@ -272,7 +272,7 @@ float PADnoteParameters::getprofile(float *smp, int size)
//normalize the profile (make the max. to be equal to 1.0)
float max = 0.0;
- for(int i = 0; i < size; i++) {
+ for(int i = 0; i < size; ++i) {
if(smp[i] < 0.0)
smp[i] = 0.0;
if(smp[i] > max)
@@ -280,7 +280,7 @@ float PADnoteParameters::getprofile(float *smp, int size)
}
if(max < 0.00001)
max = 1.0;
- for(int i = 0; i < size; i++)
+ for(int i = 0; i < size; ++i)
smp[i] /= max;
if(!Php.autoscale)
@@ -289,7 +289,7 @@ float PADnoteParameters::getprofile(float *smp, int size)
//compute the estimated perceived bandwidth
float sum = 0.0;
int i;
- for(i = 0; i < size / 2 - 2; i++) {
+ for(i = 0; i < size / 2 - 2; ++i) {
sum += smp[i] * smp[i] + smp[size - i - 1] * smp[size - i - 1];
if(sum >= 4.0)
break;
@@ -381,26 +381,26 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
int profilesize,
float bwadjust)
{
- for(int i = 0; i < size; i++)
+ for(int i = 0; i < size; ++i)
spectrum[i] = 0.0;
float harmonics[OSCIL_SIZE / 2];
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
harmonics[i] = 0.0;
//get the harmonic structure from the oscillator (I am using the frequency amplitudes, only)
oscilgen->get(harmonics, basefreq, false);
//normalize
float max = 0.0;
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
if(harmonics[i] > max)
max = harmonics[i];
if(max < 0.000001)
max = 1;
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
harmonics[i] /= max;
- for(int nh = 1; nh < OSCIL_SIZE / 2; nh++) { //for each harmonic
+ for(int nh = 1; nh < OSCIL_SIZE / 2; ++nh) { //for each harmonic
float realfreq = getNhr(nh) * basefreq;
if(realfreq > SAMPLE_RATE * 0.49999)
break;
@@ -451,7 +451,7 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
float rap = sqrt((float)profilesize / (float)ibw);
int cfreq =
(int) (realfreq / (SAMPLE_RATE * 0.5) * size) - ibw / 2;
- for(int i = 0; i < ibw; i++) {
+ for(int i = 0; i < ibw; ++i) {
int src = (int)(i * rap * rap);
int spfreq = i + cfreq;
if(spfreq < 0)
@@ -464,7 +464,7 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
else { //if the bandwidth is smaller than the profilesize
float rap = sqrt((float)ibw / (float)profilesize);
float ibasefreq = realfreq / (SAMPLE_RATE * 0.5) * size;
- for(int i = 0; i < profilesize; i++) {
+ for(int i = 0; i < profilesize; ++i) {
float idfreq = i / (float)profilesize - 0.5;
idfreq *= ibw;
int spfreq = (int) (idfreq + ibasefreq);
@@ -487,26 +487,26 @@ void PADnoteParameters::generatespectrum_otherModes(float *spectrum,
int size,
float basefreq)
{
- for(int i = 0; i < size; i++)
+ for(int i = 0; i < size; ++i)
spectrum[i] = 0.0;
float harmonics[OSCIL_SIZE / 2];
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
harmonics[i] = 0.0;
//get the harmonic structure from the oscillator (I am using the frequency amplitudes, only)
oscilgen->get(harmonics, basefreq, false);
//normalize
float max = 0.0;
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
if(harmonics[i] > max)
max = harmonics[i];
if(max < 0.000001)
max = 1;
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
harmonics[i] /= max;
- for(int nh = 1; nh < OSCIL_SIZE / 2; nh++) { //for each harmonic
+ for(int nh = 1; nh < OSCIL_SIZE / 2; ++nh) { //for each harmonic
float realfreq = getNhr(nh) * basefreq;
///sa fac aici interpolarea si sa am grija daca frecv descresc
@@ -528,13 +528,13 @@ void PADnoteParameters::generatespectrum_otherModes(float *spectrum,
if(Pmode != 1) {
int old = 0;
- for(int k = 1; k < size; k++) {
+ for(int k = 1; k < size; ++k) {
if((spectrum[k] > 1e-10) || (k == (size - 1))) {
int delta = k - old;
float val1 = spectrum[old];
float val2 = spectrum[k];
float idelta = 1.0 / delta;
- for(int i = 0; i < delta; i++) {
+ for(int i = 0; i < delta; ++i) {
float x = idelta * i;
spectrum[old + i] = val1 * (1.0 - x) + val2 * x;
}
@@ -580,9 +580,9 @@ void PADnoteParameters::applyparameters(bool lockmutex)
fft_t *fftfreqs = new fft_t[samplesize / 2];
float adj[samplemax]; //this is used to compute frequency relation to the base frequency
- for(int nsample = 0; nsample < samplemax; nsample++)
+ for(int nsample = 0; nsample < samplemax; ++nsample)
adj[nsample] = (Pquality.oct + 1.0) * (float)nsample / samplemax;
- for(int nsample = 0; nsample < samplemax; nsample++) {
+ for(int nsample = 0; nsample < samplemax; ++nsample) {
float tmp = adj[nsample] - adj[samplemax - 1] * 0.5;
float basefreqadjust = pow(2.0, tmp);
@@ -601,24 +601,24 @@ void PADnoteParameters::applyparameters(bool lockmutex)
newsample.smp = new float[samplesize + extra_samples];
newsample.smp[0] = 0.0;
- for(int i = 1; i < spectrumsize; i++) //randomize the phases
+ for(int i = 1; i < spectrumsize; ++i) //randomize the phases
fftfreqs[i] = std::polar(spectrum[i], (float)RND * 6.29f);
fft->freqs2smps(fftfreqs, newsample.smp); //that's all; here is the only ifft for the whole sample; no windows are used ;-)
//normalize(rms)
float rms = 0.0;
- for(int i = 0; i < samplesize; i++)
+ for(int i = 0; i < samplesize; ++i)
rms += newsample.smp[i] * newsample.smp[i];
rms = sqrt(rms);
if(rms < 0.000001)
rms = 1.0;
rms *= sqrt(262144.0 / samplesize);
- for(int i = 0; i < samplesize; i++)
+ for(int i = 0; i < samplesize; ++i)
newsample.smp[i] *= 1.0 / rms * 50.0;
//prepare extra samples used by the linear or cubic interpolation
- for(int i = 0; i < extra_samples; i++)
+ for(int i = 0; i < extra_samples; ++i)
newsample.smp[i + samplesize] = newsample.smp[i];
//replace the current sample with the new computed sample
@@ -644,12 +644,12 @@ void PADnoteParameters::applyparameters(bool lockmutex)
//delete the additional samples that might exists and are not useful
if(lockmutex) {
pthread_mutex_lock(mutex);
- for(int i = samplemax; i < PAD_MAX_SAMPLES; i++)
+ for(int i = samplemax; i < PAD_MAX_SAMPLES; ++i)
deletesample(i);
pthread_mutex_unlock(mutex);
}
else
- for(int i = samplemax; i < PAD_MAX_SAMPLES; i++)
+ for(int i = samplemax; i < PAD_MAX_SAMPLES; ++i)
deletesample(i);
;
}
@@ -658,7 +658,7 @@ void PADnoteParameters::export2wav(std::string basefilename)
{
applyparameters(true);
basefilename += "_PADsynth_";
- for(int k = 0; k < PAD_MAX_SAMPLES; k++) {
+ for(int k = 0; k < PAD_MAX_SAMPLES; ++k) {
if(sample[k].smp == NULL)
continue;
char tmpstr[20];
@@ -668,7 +668,7 @@ void PADnoteParameters::export2wav(std::string basefilename)
if(wav.good()) {
int nsmps = sample[k].size;
short int *smps = new short int[nsmps];
- for(int i = 0; i < nsmps; i++)
+ for(int i = 0; i < nsmps; ++i)
smps[i] = (short int)(sample[k].smp[i] * 32767.0);
wav.writeMonoSamples(nsmps, smps);
}
diff --git a/src/Params/PresetsStore.cpp b/src/Params/PresetsStore.cpp
@@ -96,7 +96,7 @@ void PresetsStore::rescanforpresets(const string &type)
clearpresets();
string ftype = "." + type + ".xpz";
- for(int i = 0; i < MAX_BANK_ROOT_DIRS; i++) {
+ for(int i = 0; i < MAX_BANK_ROOT_DIRS; ++i) {
if(config.cfg.presetsDirList[i].empty())
continue;
diff --git a/src/Params/SUBnoteParameters.cpp b/src/Params/SUBnoteParameters.cpp
@@ -63,7 +63,7 @@ void SUBnoteParameters::defaults()
PFreqEnvelopeEnabled = 0;
PBandWidthEnvelopeEnabled = 0;
- for(int n = 0; n < MAX_SUB_HARMONICS; n++) {
+ for(int n = 0; n < MAX_SUB_HARMONICS; ++n) {
Phmag[n] = 0;
Phrelbw[n] = 64;
}
@@ -101,7 +101,7 @@ void SUBnoteParameters::add2XML(XMLwrapper *xml)
xml->addpar("start", Pstart);
xml->beginbranch("HARMONICS");
- for(int i = 0; i < MAX_SUB_HARMONICS; i++) {
+ for(int i = 0; i < MAX_SUB_HARMONICS; ++i) {
if((Phmag[i] == 0) && (xml->minimal))
continue;
xml->beginbranch("HARMONIC", i);
@@ -174,7 +174,7 @@ void SUBnoteParameters::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("HARMONICS")) {
Phmag[0] = 0;
- for(int i = 0; i < MAX_SUB_HARMONICS; i++) {
+ for(int i = 0; i < MAX_SUB_HARMONICS; ++i) {
if(xml->enterbranch("HARMONIC", i) == 0)
continue;
Phmag[i] = xml->getpar127("mag", Phmag[i]);
diff --git a/src/Synth/ADnote.cpp b/src/Synth/ADnote.cpp
@@ -92,7 +92,7 @@ ADnote::ADnote(ADnoteParameters *pars,
else
NoteGlobalPar.Punch.Enabled = 0;
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
pars->VoicePar[nvoice].OscilSmp->newrandseed(rand());
NoteVoicePar[nvoice].OscilSmp = NULL;
NoteVoicePar[nvoice].FMSmp = NULL;
@@ -137,13 +137,13 @@ ADnote::ADnote(ADnoteParameters *pars,
default: { //unison for more than 2 subvoices
float unison_values[unison];
float min = -1e-6, max = 1e-6;
- for(int k = 0; k < unison; k++) {
+ for(int k = 0; k < unison; ++k) {
float step = (k / (float) (unison - 1)) * 2.0 - 1.0; //this makes the unison spread more uniform
float val = step + (RND * 2.0 - 1.0) / (unison - 1);
unison_values[k] = limit(val, min, max);
}
float diff = max - min;
- for(int k = 0; k < unison; k++) {
+ for(int k = 0; k < unison; ++k) {
unison_values[k] =
(unison_values[k] - (max + min) * 0.5) / diff; //the lowest value will be -1 and the highest will be 1
unison_base_freq_rap[nvoice][k] =
@@ -154,7 +154,7 @@ ADnote::ADnote(ADnoteParameters *pars,
//unison vibrattos
if(unison > 1) {
- for(int k = 0; k < unison; k++) //reduce the frequency difference for larger vibrattos
+ for(int k = 0; k < unison; ++k) //reduce the frequency difference for larger vibrattos
unison_base_freq_rap[nvoice][k] = 1.0
+ (unison_base_freq_rap[nvoice][k] - 1.0)
* (1.0 - unison_vibratto_a);
@@ -171,7 +171,7 @@ ADnote::ADnote(ADnoteParameters *pars,
(1.0
- pars->VoicePar[nvoice].
Unison_vibratto_speed / 127.0) * 4.0);
- for(int k = 0; k < unison; k++) {
+ for(int k = 0; k < unison; ++k) {
unison_vibratto[nvoice].position[k] = RND * 1.8 - 0.9;
//make period to vary randomly from 50% to 200% vibratto base period
float vibratto_period = vibratto_base_period
@@ -194,13 +194,13 @@ ADnote::ADnote(ADnoteParameters *pars,
if(unison != 1) {
int inv = pars->VoicePar[nvoice].Unison_invert_phase;
switch(inv) {
- case 0: for(int k = 0; k < unison; k++)
+ case 0: for(int k = 0; k < unison; ++k)
unison_invert_phase[nvoice][k] = false;
break;
- case 1: for(int k = 0; k < unison; k++)
+ case 1: for(int k = 0; k < unison; ++k)
unison_invert_phase[nvoice][k] = (RND > 0.5);
break;
- default: for(int k = 0; k < unison; k++)
+ default: for(int k = 0; k < unison; ++k)
unison_invert_phase[nvoice][k] =
(k % inv == 0) ? true : false;
break;
@@ -259,7 +259,7 @@ ADnote::ADnote(ADnoteParameters *pars,
- for(int k = 0; k < unison; k++) {
+ for(int k = 0; k < unison; ++k) {
oscposhi[nvoice][k] = 0;
oscposlo[nvoice][k] = 0.0;
oscposhiFM[nvoice][k] = 0;
@@ -282,7 +282,7 @@ ADnote::ADnote(ADnoteParameters *pars,
pars->VoicePar[nvoice].Presonance);
//I store the first elments to the last position for speedups
- for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; i++)
+ for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE
+ i] =
NoteVoicePar[nvoice].OscilSmp[i];
@@ -292,7 +292,7 @@ ADnote::ADnote(ADnoteParameters *pars,
- 64.0) / 128.0 * OSCIL_SIZE + OSCIL_SIZE * 4);
oscposhi_start %= OSCIL_SIZE;
- for(int k = 0; k < unison; k++) {
+ for(int k = 0; k < unison; ++k) {
oscposhi[nvoice][k] = oscposhi_start;
oscposhi_start = (int)(RND * (OSCIL_SIZE - 1)); //put random starting point for other subvoices
}
@@ -371,7 +371,7 @@ ADnote::ADnote(ADnoteParameters *pars,
partparams->VoicePar[nvoice].PFMVelocityScaleFunction);
FMoldsmp[nvoice] = new float [unison];
- for(int k = 0; k < unison; k++)
+ for(int k = 0; k < unison; ++k)
FMoldsmp[nvoice][k] = 0.0; //this is for FM (integration)
firsttick[nvoice] = 1;
@@ -382,13 +382,13 @@ ADnote::ADnote(ADnoteParameters *pars,
}
max_unison = 1;
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)
if(unison_size[nvoice] > max_unison)
max_unison = unison_size[nvoice];
tmpwave_unison = new float *[max_unison];
- for(int k = 0; k < max_unison; k++) {
+ for(int k = 0; k < max_unison; ++k) {
tmpwave_unison[k] = new float[SOUND_BUFFER_SIZE];
memset(tmpwave_unison[k], 0, SOUND_BUFFER_SIZE * sizeof(float));
}
@@ -433,7 +433,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
* (VelF(velocity, pars->GlobalPar.PFilterVelocityScaleFunction) - 1);
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
if(NoteVoicePar[nvoice].Enabled == OFF)
continue; //(gf) Stay the same as first note in legato.
@@ -485,7 +485,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
pars->VoicePar[nvoice].Presonance);//(gf)Modif of the above line.
//I store the first elments to the last position for speedups
- for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; i++)
+ for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
NoteVoicePar[nvoice].OscilSmp[OSCIL_SIZE
+ i] =
NoteVoicePar[nvoice].OscilSmp[i];
@@ -562,12 +562,12 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
partparams->GlobalPar.GlobalFilter->getfreqtracking(basefreq);
// Forbids the Modulation Voice to be greater or equal than voice
- for(int i = 0; i < NUM_VOICES; i++)
+ for(int i = 0; i < NUM_VOICES; ++i)
if(NoteVoicePar[i].FMVoice >= i)
NoteVoicePar[i].FMVoice = -1;
// Voice Parameter init
- for(unsigned nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
if(NoteVoicePar[nvoice].Enabled == 0)
continue;
@@ -616,7 +616,7 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
if(!partparams->GlobalPar.Hrandgrouping)
partparams->VoicePar[vc].FMSmp->newrandseed(rand());
- for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; i++)
+ for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
NoteVoicePar[nvoice].FMSmp[OSCIL_SIZE + i] =
NoteVoicePar[nvoice].FMSmp[i];
}
@@ -631,10 +631,10 @@ void ADnote::legatonote(float freq, float velocity, int portamento_,
}
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
- for(unsigned i = nvoice + 1; i < NUM_VOICES; i++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
+ for(unsigned i = nvoice + 1; i < NUM_VOICES; ++i)
tmp[i] = 0;
- for(unsigned i = nvoice + 1; i < NUM_VOICES; i++)
+ for(unsigned i = nvoice + 1; i < NUM_VOICES; ++i)
if((NoteVoicePar[i].FMVoice == nvoice) && (tmp[i] == 0))
tmp[i] = 1;
@@ -671,7 +671,7 @@ void ADnote::KillVoice(int nvoice)
*/
void ADnote::KillNote()
{
- for(unsigned nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
if(NoteVoicePar[nvoice].Enabled == ON)
KillVoice(nvoice);
@@ -693,7 +693,7 @@ ADnote::~ADnote()
delete [] tmpwaver;
delete [] bypassl;
delete [] bypassr;
- for(int k = 0; k < max_unison; k++)
+ for(int k = 0; k < max_unison; ++k)
delete[] tmpwave_unison[k];
delete[] tmpwave_unison;
}
@@ -716,12 +716,12 @@ void ADnote::initparameters()
* NoteGlobalPar.AmpLfo->amplfoout();
// Forbids the Modulation Voice to be greater or equal than voice
- for(int i = 0; i < NUM_VOICES; i++)
+ for(int i = 0; i < NUM_VOICES; ++i)
if(NoteVoicePar[i].FMVoice >= i)
NoteVoicePar[i].FMVoice = -1;
// Voice Parameter init
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
Voice &vce = NoteVoicePar[nvoice];
ADnoteVoiceParam ¶m = partparams->VoicePar[nvoice];
@@ -795,17 +795,17 @@ void ADnote::initparameters()
if(!partparams->GlobalPar.Hrandgrouping)
partparams->VoicePar[vc].FMSmp->newrandseed(rand());
- for(int k = 0; k < unison_size[nvoice]; k++)
+ for(int k = 0; k < unison_size[nvoice]; ++k)
oscposhiFM[nvoice][k] = (oscposhi[nvoice][k]
+ partparams->VoicePar[vc].FMSmp->get(vce.FMSmp, tmp))
% OSCIL_SIZE;
- for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; i++)
+ for(int i = 0; i < OSCIL_SMP_EXTRA_SAMPLES; ++i)
vce.FMSmp[OSCIL_SIZE + i] = vce.FMSmp[i];
int oscposhiFM_add =
(int)((param.PFMoscilphase
- 64.0) / 128.0 * OSCIL_SIZE + OSCIL_SIZE * 4);
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
oscposhiFM[nvoice][k] += oscposhiFM_add;
oscposhiFM[nvoice][k] %= OSCIL_SIZE;
}
@@ -822,10 +822,10 @@ void ADnote::initparameters()
}
}
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
- for(int i = nvoice + 1; i < NUM_VOICES; i++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
+ for(int i = nvoice + 1; i < NUM_VOICES; ++i)
tmp[i] = 0;
- for(int i = nvoice + 1; i < NUM_VOICES; i++)
+ for(int i = nvoice + 1; i < NUM_VOICES; ++i)
if((NoteVoicePar[i].FMVoice == nvoice) && (tmp[i] == 0)) {
NoteVoicePar[nvoice].VoiceOut = new float[SOUND_BUFFER_SIZE];
tmp[i] = 1;
@@ -847,7 +847,7 @@ void ADnote::compute_unison_freq_rap(int nvoice) {
return;
}
float relbw = ctl->bandwidth.relbw * bandwidthDetuneMultiplier;
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float pos = unison_vibratto[nvoice].position[k];
float step = unison_vibratto[nvoice].step[k];
pos += step;
@@ -877,7 +877,7 @@ void ADnote::compute_unison_freq_rap(int nvoice) {
*/
void ADnote::setfreq(int nvoice, float in_freq)
{
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float freq = fabs(in_freq) * unison_freq_rap[nvoice][k];
float speed = freq * float(OSCIL_SIZE) / (float) SAMPLE_RATE;
if(speed > OSCIL_SIZE)
@@ -893,7 +893,7 @@ void ADnote::setfreq(int nvoice, float in_freq)
*/
void ADnote::setfreqFM(int nvoice, float in_freq)
{
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float freq = fabs(in_freq) * unison_freq_rap[nvoice][k];
float speed = freq * float(OSCIL_SIZE) / (float) SAMPLE_RATE;
if(speed > OSCIL_SIZE)
@@ -981,7 +981,7 @@ void ADnote::computecurrentparameters()
}
//compute parameters for all voices
- for(nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
if(NoteVoicePar[nvoice].Enabled != ON)
continue;
NoteVoicePar[nvoice].DelayTicks -= 1;
@@ -1071,7 +1071,7 @@ void ADnote::computecurrentparameters()
inline void ADnote::fadein(float *smps) const
{
int zerocrossings = 0;
- for(int i = 1; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 1; i < SOUND_BUFFER_SIZE; ++i)
if((smps[i - 1] < 0.0) && (smps[i] > 0.0))
zerocrossings++; //this is only the possitive crossings
@@ -1083,7 +1083,7 @@ inline void ADnote::fadein(float *smps) const
F2I(tmp, n); //how many samples is the fade-in
if(n > SOUND_BUFFER_SIZE)
n = SOUND_BUFFER_SIZE;
- for(int i = 0; i < n; i++) { //fade-in
+ for(int i = 0; i < n; ++i) { //fade-in
float tmp = 0.5 - cos((float)i / (float) n * PI) * 0.5;
smps[i] *= tmp;
}
@@ -1097,14 +1097,14 @@ inline void ADnote::ComputeVoiceOscillator_LinearInterpolation(int nvoice)
int i, poshi;
float poslo;
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
poshi = oscposhi[nvoice][k];
poslo = oscposlo[nvoice][k];
int freqhi = oscfreqhi[nvoice][k];
float freqlo = oscfreqlo[nvoice][k];
float *smps = NoteVoicePar[nvoice].OscilSmp;
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
tw[i] = smps[poshi] * (1.0 - poslo) + smps[poshi + 1] * poslo;
poslo += freqlo;
if(poslo >= 1.0) {
@@ -1172,9 +1172,9 @@ inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
if(NoteVoicePar[nvoice].FMVoice >= 0) {
//if I use VoiceOut[] as modullator
int FMVoice = NoteVoicePar[nvoice].FMVoice;
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
FMnewamplitude[nvoice],
i,
@@ -1185,14 +1185,14 @@ inline void ADnote::ComputeVoiceOscillatorMorph(int nvoice)
}
}
else {
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
int poshiFM = oscposhiFM[nvoice][k];
float posloFM = oscposloFM[nvoice][k];
int freqhiFM = oscfreqhiFM[nvoice][k];
float freqloFM = oscfreqloFM[nvoice][k];
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
FMnewamplitude[nvoice],
i,
@@ -1228,9 +1228,9 @@ inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
FMoldamplitude[nvoice] = 1.0;
if(NoteVoicePar[nvoice].FMVoice >= 0) {
// if I use VoiceOut[] as modullator
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
FMnewamplitude[nvoice],
i,
@@ -1241,14 +1241,14 @@ inline void ADnote::ComputeVoiceOscillatorRingModulation(int nvoice)
}
}
else {
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
int poshiFM = oscposhiFM[nvoice][k];
float posloFM = oscposloFM[nvoice][k];
int freqhiFM = oscfreqhiFM[nvoice][k];
float freqloFM = oscfreqloFM[nvoice][k];
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
amp = INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
FMnewamplitude[nvoice],
i,
@@ -1285,7 +1285,7 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
if(NoteVoicePar[nvoice].FMVoice >= 0) {
//if I use VoiceOut[] as modulator
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
memcpy(tw, NoteVoicePar[NoteVoicePar[nvoice].FMVoice].VoiceOut,
SOUND_BUFFER_SIZE * sizeof(float));
@@ -1293,14 +1293,14 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
}
else {
//Compute the modulator and store it in tmpwave_unison[][]
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
int poshiFM = oscposhiFM[nvoice][k];
float posloFM = oscposloFM[nvoice][k];
int freqhiFM = oscfreqhiFM[nvoice][k];
float freqloFM = oscfreqloFM[nvoice][k];
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
tw[i] =
(NoteVoicePar[nvoice].FMSmp[poshiFM] * (1.0 - posloFM)
+ NoteVoicePar[nvoice].FMSmp[poshiFM + 1] * posloFM);
@@ -1319,9 +1319,9 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
// Amplitude interpolation
if(ABOVE_AMPLITUDE_THRESHOLD(FMoldamplitude[nvoice],
FMnewamplitude[nvoice])) {
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
tw[i] *= INTERPOLATE_AMPLITUDE(FMoldamplitude[nvoice],
FMnewamplitude[nvoice],
i,
@@ -1330,9 +1330,9 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
}
}
else {
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
tw[i] *= FMnewamplitude[nvoice];
}
}
@@ -1342,10 +1342,10 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
if(FMmode != 0) { //Frequency modulation
float normalize = OSCIL_SIZE / 262144.0 * 44100.0
/ (float)SAMPLE_RATE;
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
float fmold = FMoldsmp[nvoice][k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
fmold = fmod(fmold + tw[i] * normalize, OSCIL_SIZE);
tw[i] = fmold;
}
@@ -1354,22 +1354,22 @@ inline void ADnote::ComputeVoiceOscillatorFrequencyModulation(int nvoice,
}
else { //Phase modulation
float normalize = OSCIL_SIZE / 262144.0;
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i)
tw[i] *= normalize;
}
}
//do the modulation
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
int poshi = oscposhi[nvoice][k];
float poslo = oscposlo[nvoice][k];
int freqhi = oscfreqhi[nvoice][k];
float freqlo = oscfreqlo[nvoice][k];
- for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(i = 0; i < SOUND_BUFFER_SIZE; ++i) {
F2I(tw[i], FMmodfreqhi);
FMmodfreqlo = fmod(tw[i] + 0.0000000001, 1.0);
if(FMmodfreqhi < 0)
@@ -1416,9 +1416,9 @@ inline void ADnote::ComputeVoiceOscillatorPitchModulation(int /*nvoice*/)
*/
inline void ADnote::ComputeVoiceNoise(int nvoice)
{
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tw[i] = RND * 2.0 - 1.0;
}
}
@@ -1441,7 +1441,7 @@ int ADnote::noteout(float *outl, float *outr)
memset(bypassr, 0, SOUND_BUFFER_SIZE * sizeof(float));
computecurrentparameters();
- for(unsigned nvoice = 0; nvoice < NUM_VOICES; nvoice++) {
+ for(unsigned nvoice = 0; nvoice < NUM_VOICES; ++nvoice) {
if((NoteVoicePar[nvoice].Enabled != ON)
|| (NoteVoicePar[nvoice].DelayTicks > 0))
continue;
@@ -1473,7 +1473,7 @@ int ADnote::noteout(float *outl, float *outr)
memset(tmpwavel, 0, SOUND_BUFFER_SIZE * sizeof(float));
if(stereo)
memset(tmpwaver, 0, SOUND_BUFFER_SIZE * sizeof(float));
- for(int k = 0; k < unison_size[nvoice]; k++) {
+ for(int k = 0; k < unison_size[nvoice]; ++k) {
float *tw = tmpwave_unison[k];
if(stereo) {
float stereo_pos = 0;
@@ -1510,13 +1510,13 @@ int ADnote::noteout(float *outl, float *outr)
rvol = -rvol;
}
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwavel[i] += tw[i] * lvol;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwaver[i] += tw[i] * rvol;
}
else
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwavel[i] += tw[i];
}
@@ -1534,14 +1534,14 @@ int ADnote::noteout(float *outl, float *outr)
rest = 10;
if(rest > SOUND_BUFFER_SIZE)
rest = SOUND_BUFFER_SIZE;
- for(int i = 0; i < SOUND_BUFFER_SIZE - rest; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE - rest; ++i)
tmpwavel[i] *= oldam;
if(stereo)
- for(int i = 0; i < SOUND_BUFFER_SIZE - rest; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE - rest; ++i)
tmpwaver[i] *= oldam;
}
// Amplitude interpolation
- for(int i = 0; i < rest; i++) {
+ for(int i = 0; i < rest; ++i) {
float amp = INTERPOLATE_AMPLITUDE(oldam, newam, i, rest);
tmpwavel[i + (SOUND_BUFFER_SIZE - rest)] *= amp;
if(stereo)
@@ -1549,10 +1549,10 @@ int ADnote::noteout(float *outl, float *outr)
}
}
else {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwavel[i] *= newam;
if(stereo)
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwaver[i] *= newam;
}
@@ -1574,11 +1574,11 @@ int ADnote::noteout(float *outl, float *outr)
//check if the amplitude envelope is finished, if yes, the voice will be fadeout
if(NoteVoicePar[nvoice].AmpEnvelope != NULL) {
if(NoteVoicePar[nvoice].AmpEnvelope->finished() != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwavel[i] *= 1.0 - (float)i
/ (float)SOUND_BUFFER_SIZE;
if(stereo)
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmpwaver[i] *= 1.0 - (float)i
/ (float)SOUND_BUFFER_SIZE;
}
@@ -1589,11 +1589,11 @@ int ADnote::noteout(float *outl, float *outr)
// Put the ADnote samples in VoiceOut (without appling Global volume, because I wish to use this voice as a modullator)
if(NoteVoicePar[nvoice].VoiceOut != NULL) {
if(stereo)
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i]
+ tmpwaver[i];
else //mono
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
NoteVoicePar[nvoice].VoiceOut[i] = tmpwavel[i];
}
@@ -1602,7 +1602,7 @@ int ADnote::noteout(float *outl, float *outr)
// Add the voice that do not bypass the filter to out
if(NoteVoicePar[nvoice].filterbypass == 0) { //no bypass
if(stereo) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //stereo
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //stereo
outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume
* NoteVoicePar[nvoice].Panning * 2.0;
outr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume
@@ -1610,13 +1610,13 @@ int ADnote::noteout(float *outl, float *outr)
}
}
else
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)//mono
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)//mono
outl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;
}
else { //bypass the filter
if(stereo) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //stereo
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //stereo
bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume
* NoteVoicePar[nvoice].Panning * 2.0;
bypassr[i] += tmpwaver[i] * NoteVoicePar[nvoice].Volume
@@ -1624,7 +1624,7 @@ int ADnote::noteout(float *outl, float *outr)
}
}
else
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) //mono
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) //mono
bypassl[i] += tmpwavel[i] * NoteVoicePar[nvoice].Volume;
}
@@ -1646,14 +1646,14 @@ int ADnote::noteout(float *outl, float *outr)
else
NoteGlobalPar.GlobalFilterR->filterout(&outr[0]);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] += bypassl[i];
outr[i] += bypassr[i];
}
if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude)) {
// Amplitude Interpolation
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,
globalnewamplitude,
i,
@@ -1663,7 +1663,7 @@ int ADnote::noteout(float *outl, float *outr)
}
}
else {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;
outr[i] *= globalnewamplitude * (1.0 - NoteGlobalPar.Panning);
}
@@ -1671,7 +1671,7 @@ int ADnote::noteout(float *outl, float *outr)
//Apply the punch
if(NoteGlobalPar.Punch.Enabled != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float punchamp = NoteGlobalPar.Punch.initialvalue
* NoteGlobalPar.Punch.t + 1.0;
outl[i] *= punchamp;
@@ -1692,7 +1692,7 @@ int ADnote::noteout(float *outl, float *outr)
// Check if the global amplitude is finished.
// If it does, disable the note
if(NoteGlobalPar.AmpEnvelope->finished()) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //fade-out
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //fade-out
float tmp = 1.0 - (float)i / (float)SOUND_BUFFER_SIZE;
outl[i] *= tmp;
outr[i] *= tmp;
@@ -1708,7 +1708,7 @@ int ADnote::noteout(float *outl, float *outr)
*/
void ADnote::relasekey()
{
- for(int nvoice = 0; nvoice < NUM_VOICES; nvoice++)
+ for(int nvoice = 0; nvoice < NUM_VOICES; ++nvoice)
NoteVoicePar[nvoice].releasekey();
NoteGlobalPar.FreqEnvelope->relasekey();
NoteGlobalPar.FilterEnvelope->relasekey();
diff --git a/src/Synth/Envelope.cpp b/src/Synth/Envelope.cpp
@@ -48,7 +48,7 @@ Envelope::Envelope(EnvelopeParams *envpars, float basefreq)
if((mode == 2) && (linearenvelope != 0))
mode = 1; //change to linear
- for(i = 0; i < MAX_ENVELOPE_POINTS; i++) {
+ for(i = 0; i < MAX_ENVELOPE_POINTS; ++i) {
float tmp = envpars->getdt(i) / 1000.0 * envstretch;
if(tmp > bufferdt)
envdt[i] = bufferdt / tmp;
diff --git a/src/Synth/OscilGen.cpp b/src/Synth/OscilGen.cpp
@@ -93,7 +93,7 @@ void rmsNormalize(fft_t *freqs)
const float gain = 1.0 / sqrt(sum);
- for(int i = 1; i < OSCIL_SIZE / 2; i++)
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i)
freqs[i] *= gain;
}
@@ -145,7 +145,7 @@ void OscilGen::defaults()
oldmodulationpar2 = 0;
oldmodulationpar3 = 0;
- for(int i = 0; i < MAX_AD_HARMONICS; i++) {
+ for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
hmag[i] = 0.0;
hphase[i] = 0.0;
Phmag[i] = 64;
@@ -214,14 +214,14 @@ void OscilGen::convert2sine()
mag[0] = 0;
phase[0] = 0;
- for(int i = 0; i < MAX_AD_HARMONICS; i++) {
+ for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
mag[i] = abs(freqs, i + 1);
phase[i] = arg(freqs, i + 1);
}
defaults();
- for(int i = 0; i < MAX_AD_HARMONICS - 1; i++) {
+ for(int i = 0; i < MAX_AD_HARMONICS - 1; ++i) {
float newmag = mag[i];
float newphase = phase[i];
@@ -279,7 +279,7 @@ void OscilGen::getbasefunction(float *smps)
base_func func = getBaseFunction(Pcurrentbasefunc);
- for(i = 0; i < OSCIL_SIZE; i++) {
+ for(i = 0; i < OSCIL_SIZE; ++i) {
float t = i * 1.0 / OSCIL_SIZE;
switch(Pbasefuncmodulation) {
@@ -322,7 +322,7 @@ void OscilGen::oscilfilter()
const float par2 = Pfilterpar2 / 127.0;
filter_func filter = getFilter(Pfiltertype);
- for(int i = 1; i < OSCIL_SIZE / 2; i++)
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i)
oscilFFTfreqs[i] *= filter(i,par,par2);
normalize(oscilFFTfreqs);
@@ -354,14 +354,14 @@ inline void normalize(float *smps, size_t N)
{
//Find max
float max = 0.0;
- for(size_t i = 0; i < N; i++)
+ for(size_t i = 0; i < N; ++i)
if(max < fabs(smps[i]))
max = fabs(smps[i]);
if(max < 0.00001)
max = 1.0;
//Normalize to +-1
- for(size_t i = 0; i < N; i++)
+ for(size_t i = 0; i < N; ++i)
smps[i] /= max;
}
@@ -377,7 +377,7 @@ void OscilGen::waveshape()
clearDC(oscilFFTfreqs);
//reduce the amplitude of the freqs near the nyquist
- for(int i = 1; i < OSCIL_SIZE / 8; i++) {
+ for(int i = 1; i < OSCIL_SIZE / 8; ++i) {
float gain = i / (OSCIL_SIZE / 8.0);
oscilFFTfreqs[OSCIL_SIZE / 2 - i] *= gain;
}
@@ -431,7 +431,7 @@ void OscilGen::modulation()
clearDC(oscilFFTfreqs); //remove the DC
//reduce the amplitude of the freqs near the nyquist
- for(i = 1; i < OSCIL_SIZE / 8; i++) {
+ for(i = 1; i < OSCIL_SIZE / 8; ++i) {
float tmp = i / (OSCIL_SIZE / 8.0);
oscilFFTfreqs[OSCIL_SIZE / 2 - i] *= tmp;
}
@@ -442,13 +442,13 @@ void OscilGen::modulation()
//Normalize
normalize(tmpsmps, OSCIL_SIZE);
- for(i = 0; i < OSCIL_SIZE; i++)
+ for(i = 0; i < OSCIL_SIZE; ++i)
in[i] = tmpsmps[i];
- for(i = 0; i < extra_points; i++)
+ for(i = 0; i < extra_points; ++i)
in[i + OSCIL_SIZE] = tmpsmps[i];
//Do the modulation
- for(i = 0; i < OSCIL_SIZE; i++) {
+ for(i = 0; i < OSCIL_SIZE; ++i) {
float t = i * 1.0 / OSCIL_SIZE;
switch(Pmodulation) {
@@ -508,7 +508,7 @@ void OscilGen::spectrumadjust()
normalize(oscilFFTfreqs);
- for(int i = 0; i < OSCIL_SIZE / 2; i++) {
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i) {
float mag = abs(oscilFFTfreqs, i);
float phase = arg(oscilFFTfreqs, i);
@@ -549,7 +549,7 @@ void OscilGen::shiftharmonics()
}
}
else {
- for(int i = 0; i < OSCIL_SIZE / 2 - 1; i++) {
+ for(int i = 0; i < OSCIL_SIZE / 2 - 1; ++i) {
int oldh = i + abs(harmonicshift);
if(oldh >= (OSCIL_SIZE / 2 - 1))
h = 0.0;
@@ -576,10 +576,10 @@ void OscilGen::prepare()
|| DIFF(basefuncmodulationpar2) || DIFF(basefuncmodulationpar3))
changebasefunction();
- for(int i = 0; i < MAX_AD_HARMONICS; i++)
+ for(int i = 0; i < MAX_AD_HARMONICS; ++i)
hphase[i] = (Phphase[i] - 64.0) / 64.0 * PI / (i + 1);
- for(int i = 0; i < MAX_AD_HARMONICS; i++) {
+ for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
const float hmagnew = 1.0 - fabs(Phmag[i] / 64.0 - 1.0);
switch(Phmagtype) {
case 1:
@@ -604,23 +604,23 @@ void OscilGen::prepare()
}
//remove the harmonics where Phmag[i]==64
- for(int i = 0; i < MAX_AD_HARMONICS; i++)
+ for(int i = 0; i < MAX_AD_HARMONICS; ++i)
if(Phmag[i] == 64)
hmag[i] = 0.0;
clearAll(oscilFFTfreqs);
if(Pcurrentbasefunc == 0) { //the sine case
- for(int i = 0; i < MAX_AD_HARMONICS; i++) {
+ for(int i = 0; i < MAX_AD_HARMONICS; ++i) {
oscilFFTfreqs[i + 1].real() = -hmag[i] * sin(hphase[i] * (i + 1)) / 2.0;
oscilFFTfreqs[i + 1].imag() = hmag[i] * cos(hphase[i] * (i + 1)) / 2.0;
}
}
else {
- for(int j = 0; j < MAX_AD_HARMONICS; j++) {
+ for(int j = 0; j < MAX_AD_HARMONICS; ++j) {
if(Phmag[j] == 64)
continue;
- for(int i = 1; i < OSCIL_SIZE / 2; i++) {
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i) {
int k = i * (j + 1);
if(k >= OSCIL_SIZE / 2)
break;
@@ -662,7 +662,7 @@ void OscilGen::adaptiveharmonic(fft_t *f, float freq)
freq = 440.0;
fft_t *inf = new fft_t[OSCIL_SIZE / 2];
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
inf[i] = f[i];
clearAll(f);
clearDC(inf);
@@ -681,7 +681,7 @@ void OscilGen::adaptiveharmonic(fft_t *f, float freq)
down = true;
}
- for(int i = 0; i < OSCIL_SIZE / 2 - 2; i++) {
+ for(int i = 0; i < OSCIL_SIZE / 2 - 2; ++i) {
float h = i * rap;
int high = (int)(i * rap);
float low = fmod(h, 1.0);
@@ -727,14 +727,14 @@ void OscilGen::adaptiveharmonicpostprocess(fft_t *f, int size)
float par = Padaptiveharmonicspar * 0.01;
par = 1.0 - pow((1.0 - par), 1.5);
- for(int i = 0; i < size; i++) {
+ for(int i = 0; i < size; ++i) {
inf[i] = f[i] * double(par);
f[i] *= (1.0 - par);
}
if(Padaptiveharmonics == 2) { //2n+1
- for(int i = 0; i < size; i++)
+ for(int i = 0; i < size; ++i)
if((i % 2) == 0)
f[i] += inf[i]; //i=0 pt prima armonica,etc.
}
@@ -742,13 +742,13 @@ void OscilGen::adaptiveharmonicpostprocess(fft_t *f, int size)
int nh = (Padaptiveharmonics - 3) / 2 + 2;
int sub_vs_add = (Padaptiveharmonics - 3) % 2;
if(sub_vs_add == 0) {
- for(int i = 0; i < size; i++) {
+ for(int i = 0; i < size; ++i) {
if(((i + 1) % nh) == 0)
f[i] += inf[i];
}
}
else {
- for(int i = 0; i < size / nh - 1; i++)
+ for(int i = 0; i < size / nh - 1; ++i)
f[(i + 1) * nh - 1] += inf[i];
}
}
@@ -828,7 +828,7 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
if(Padaptiveharmonics != 0)
nyquist = OSCIL_SIZE / 2;
- for(int i = 1; i < nyquist - 1; i++)
+ for(int i = 1; i < nyquist - 1; ++i)
outoscilFFTfreqs[i] = oscilFFTfreqs[i];
adaptiveharmonic(outoscilFFTfreqs, freqHz);
@@ -838,7 +838,7 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
}
if(Padaptiveharmonics) { //do the antialiasing in the case of adaptive harmonics
- for(int i = nyquist; i < OSCIL_SIZE / 2; i++)
+ for(int i = nyquist; i < OSCIL_SIZE / 2; ++i)
outoscilFFTfreqs[i] = fft_t(0.0f, 0.0f);
}
@@ -846,7 +846,7 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
// in ADnote by setting start position according to this setting
if((Prand > 64) && (freqHz >= 0.0) && (!ADvsPAD)) {
const float rnd = PI * pow((Prand - 64.0) / 64.0, 2.0);
- for(int i = 1; i < nyquist - 1; i++) //to Nyquist only for AntiAliasing
+ for(int i = 1; i < nyquist - 1; ++i) //to Nyquist only for AntiAliasing
outoscilFFTfreqs[i] *= std::polar<fftw_real>(1.0f, (float)(rnd * i * RND));
}
@@ -860,14 +860,14 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
case 1:
power = power * 2.0 - 0.5;
power = pow(15.0, power);
- for(int i = 1; i < nyquist - 1; i++)
+ for(int i = 1; i < nyquist - 1; ++i)
outoscilFFTfreqs[i] *= pow(RND, power) * normalize;
break;
case 2:
power = power * 2.0 - 0.5;
power = pow(15.0, power) * 2.0;
float rndfreq = 2 * PI * RND;
- for(int i = 1; i < nyquist - 1; i++)
+ for(int i = 1; i < nyquist - 1; ++i)
outoscilFFTfreqs[i] *= pow(fabs(sin(i * rndfreq)), power)
* normalize;
break;
@@ -881,11 +881,11 @@ short int OscilGen::get(float *smps, float freqHz, int resonance)
rmsNormalize(outoscilFFTfreqs);
if((ADvsPAD) && (freqHz > 0.1)) //in this case the smps will contain the freqs
- for(int i = 1; i < OSCIL_SIZE / 2; i++)
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i)
smps[i - 1] = abs(outoscilFFTfreqs, i);
else {
fft->freqs2smps(outoscilFFTfreqs, smps);
- for(int i = 0; i < OSCIL_SIZE; i++)
+ for(int i = 0; i < OSCIL_SIZE; ++i)
smps[i] *= 0.25; //correct the amplitude
}
@@ -904,7 +904,7 @@ void OscilGen::getspectrum(int n, float *spc, int what)
if(n > OSCIL_SIZE / 2)
n = OSCIL_SIZE / 2;
- for(int i = 1; i < n; i++) {
+ for(int i = 1; i < n; ++i) {
if(what == 0)
spc[i - 1] = abs(oscilFFTfreqs, i);
else {
@@ -916,12 +916,12 @@ void OscilGen::getspectrum(int n, float *spc, int what)
}
if(what == 0) {
- for(int i = 0; i < n; i++)
+ for(int i = 0; i < n; ++i)
outoscilFFTfreqs[i] = fft_t(spc[i], spc[i]);
memset(outoscilFFTfreqs+n, 0, (OSCIL_SIZE / 2 - n) * sizeof(fft_t));
adaptiveharmonic(outoscilFFTfreqs, 0.0);
adaptiveharmonicpostprocess(outoscilFFTfreqs, n - 1);
- for(int i = 0; i < n; i++)
+ for(int i = 0; i < n; ++i)
spc[i] = outoscilFFTfreqs[i].imag();
}
}
@@ -932,7 +932,7 @@ void OscilGen::getspectrum(int n, float *spc, int what)
*/
void OscilGen::useasbase()
{
- for(int i = 0; i < OSCIL_SIZE / 2; i++)
+ for(int i = 0; i < OSCIL_SIZE / 2; ++i)
basefuncFFTfreqs[i] = oscilFFTfreqs[i];
oldbasefunc = Pcurrentbasefunc = 127;
@@ -990,7 +990,7 @@ void OscilGen::add2XML(XMLwrapper *xml)
xml->addpar("adaptive_harmonics_power", Padaptiveharmonicspower);
xml->beginbranch("HARMONICS");
- for(int n = 0; n < MAX_AD_HARMONICS; n++) {
+ for(int n = 0; n < MAX_AD_HARMONICS; ++n) {
if((Phmag[n] == 64) && (Phphase[n] == 64))
continue;
xml->beginbranch("HARMONIC", n + 1);
@@ -1004,7 +1004,7 @@ void OscilGen::add2XML(XMLwrapper *xml)
normalize(basefuncFFTfreqs);
xml->beginbranch("BASE_FUNCTION");
- for(int i = 1; i < OSCIL_SIZE / 2; i++) {
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i) {
float xc = basefuncFFTfreqs[i].real();
float xs = basefuncFFTfreqs[i].imag();
if((fabs(xs) > 0.00001) && (fabs(xs) > 0.00001)) {
@@ -1084,7 +1084,7 @@ void OscilGen::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("HARMONICS")) {
Phmag[0] = 64;
Phphase[0] = 64;
- for(int n = 0; n < MAX_AD_HARMONICS; n++) {
+ for(int n = 0; n < MAX_AD_HARMONICS; ++n) {
if(xml->enterbranch("HARMONIC", n + 1) == 0)
continue;
Phmag[n] = xml->getpar127("mag", 64);
@@ -1099,7 +1099,7 @@ void OscilGen::getfromXML(XMLwrapper *xml)
if(xml->enterbranch("BASE_FUNCTION")) {
- for(int i = 1; i < OSCIL_SIZE / 2; i++) {
+ for(int i = 1; i < OSCIL_SIZE / 2; ++i) {
if(xml->enterbranch("BF_HARMONIC", i)) {
basefuncFFTfreqs[i].real() = xml->getparreal("cos", 0.0);
basefuncFFTfreqs[i].imag() = xml->getparreal("sin", 0.0);
diff --git a/src/Synth/PADnote.cpp b/src/Synth/PADnote.cpp
@@ -75,7 +75,7 @@ void PADnote::setup(float freq, float velocity,int portamento_, int midinote, bo
float logfreq = log(basefreq * pow(2.0, NoteGlobalPar.Detune / 1200.0));
float mindist = fabs(logfreq - log(pars->sample[0].basefreq + 0.0001));
nsample = 0;
- for(int i = 1; i < PAD_MAX_SAMPLES; i++) {
+ for(int i = 1; i < PAD_MAX_SAMPLES; ++i) {
if(pars->sample[i].smp == NULL)
break;
float dist = fabs(logfreq - log(pars->sample[i].basefreq + 0.0001));
@@ -192,7 +192,7 @@ PADnote::~PADnote()
inline void PADnote::fadein(float *smps)
{
int zerocrossings = 0;
- for(int i = 1; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 1; i < SOUND_BUFFER_SIZE; ++i)
if((smps[i - 1] < 0.0) && (smps[i] > 0.0))
zerocrossings++; //this is only the possitive crossings
@@ -204,7 +204,7 @@ inline void PADnote::fadein(float *smps)
F2I(tmp, n); //how many samples is the fade-in
if(n > SOUND_BUFFER_SIZE)
n = SOUND_BUFFER_SIZE;
- for(int i = 0; i < n; i++) { //fade-in
+ for(int i = 0; i < n; ++i) { //fade-in
float tmp = 0.5 - cos((float)i / (float) n * PI) * 0.5;
smps[i] *= tmp;
}
@@ -260,7 +260,7 @@ int PADnote::Compute_Linear(float *outl,
return 1;
}
int size = pars->sample[nsample].size;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
poshi_l += freqhi;
poshi_r += freqhi;
poslo += freqlo;
@@ -291,7 +291,7 @@ int PADnote::Compute_Cubic(float *outl,
}
int size = pars->sample[nsample].size;
float xm1, x0, x1, x2, a, b, c;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
poshi_l += freqhi;
poshi_r += freqhi;
poslo += freqlo;
@@ -334,7 +334,7 @@ int PADnote::noteout(float *outl, float *outr)
computecurrentparameters();
float *smps = pars->sample[nsample].smp;
if(smps == NULL) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] = 0.0;
outr[i] = 0.0;
}
@@ -365,7 +365,7 @@ int PADnote::noteout(float *outl, float *outr)
//Apply the punch
if(NoteGlobalPar.Punch.Enabled != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float punchamp = NoteGlobalPar.Punch.initialvalue
* NoteGlobalPar.Punch.t + 1.0;
outl[i] *= punchamp;
@@ -380,7 +380,7 @@ int PADnote::noteout(float *outl, float *outr)
if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude)) {
// Amplitude Interpolation
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,
globalnewamplitude,
i,
@@ -390,7 +390,7 @@ int PADnote::noteout(float *outl, float *outr)
}
}
else {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;
outr[i] *= globalnewamplitude * (1.0 - NoteGlobalPar.Panning);
}
@@ -403,7 +403,7 @@ int PADnote::noteout(float *outl, float *outr)
// Check if the global amplitude is finished.
// If it does, disable the note
if(NoteGlobalPar.AmpEnvelope->finished() != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //fade-out
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //fade-out
float tmp = 1.0 - (float)i / (float)SOUND_BUFFER_SIZE;
outl[i] *= tmp;
outr[i] *= tmp;
diff --git a/src/Synth/Resonance.cpp b/src/Synth/Resonance.cpp
@@ -42,7 +42,7 @@ void Resonance::defaults()
Pprotectthefundamental = 0;
ctlcenter = 1.0;
ctlbw = 1.0;
- for(int i = 0; i < N_RES_POINTS; i++)
+ for(int i = 0; i < N_RES_POINTS; ++i)
Prespoints[i] = 64;
}
@@ -67,13 +67,13 @@ void Resonance::applyres(int n, fft_t *fftdata, float freq)
l1 = log(getfreqx(0.0) * ctlcenter),
l2 = log(2.0) * getoctavesfreq() * ctlbw;
- for(int i = 0; i < N_RES_POINTS; i++)
+ for(int i = 0; i < N_RES_POINTS; ++i)
if(sum < Prespoints[i])
sum = Prespoints[i];
if(sum < 1.0)
sum = 1.0;
- for(int i = 1; i < n; i++) {
+ for(int i = 1; i < n; ++i) {
float x = (log(freq * i) - l1) / l2; //compute where the n-th hamonics fits to the graph
if(x < 0.0)
x = 0.0;
@@ -109,7 +109,7 @@ float Resonance::getfreqresponse(float freq)
float l1 = log(getfreqx(0.0) * ctlcenter),
l2 = log(2.0) * getoctavesfreq() * ctlbw, sum = 0.0;
- for(int i = 0; i < N_RES_POINTS; i++)
+ for(int i = 0; i < N_RES_POINTS; ++i)
if(sum < Prespoints[i])
sum = Prespoints[i];
if(sum < 1.0)
@@ -141,7 +141,7 @@ float Resonance::getfreqresponse(float freq)
void Resonance::smooth()
{
float old = Prespoints[0];
- for(int i = 0; i < N_RES_POINTS; i++) {
+ for(int i = 0; i < N_RES_POINTS; ++i) {
old = old * 0.4 + Prespoints[i] * 0.6;
Prespoints[i] = (int) old;
}
@@ -160,7 +160,7 @@ void Resonance::smooth()
void Resonance::randomize(int type)
{
int r = (int)(RND * 127.0);
- for(int i = 0; i < N_RES_POINTS; i++) {
+ for(int i = 0; i < N_RES_POINTS; ++i) {
Prespoints[i] = r;
if((RND < 0.1) && (type == 0))
r = (int)(RND * 127.0);
@@ -178,10 +178,10 @@ void Resonance::randomize(int type)
void Resonance::interpolatepeaks(int type)
{
int x1 = 0, y1 = Prespoints[0];
- for(int i = 1; i < N_RES_POINTS; i++) {
+ for(int i = 1; i < N_RES_POINTS; ++i) {
if((Prespoints[i] != 64) || (i + 1 == N_RES_POINTS)) {
int y2 = Prespoints[i];
- for(int k = 0; k < i - x1; k++) {
+ for(int k = 0; k < i - x1; ++k) {
float x = (float) k / (i - x1);
if(type == 0)
x = (1 - cos(x * PI)) * 0.5;
@@ -251,7 +251,7 @@ void Resonance::add2XML(XMLwrapper *xml)
xml->addpar("octaves_freq", Poctavesfreq);
xml->addparbool("protect_fundamental_frequency", Pprotectthefundamental);
xml->addpar("resonance_points", N_RES_POINTS);
- for(int i = 0; i < N_RES_POINTS; i++) {
+ for(int i = 0; i < N_RES_POINTS; ++i) {
xml->beginbranch("RESPOINT", i);
xml->addpar("val", Prespoints[i]);
xml->endbranch();
@@ -268,7 +268,7 @@ void Resonance::getfromXML(XMLwrapper *xml)
Poctavesfreq = xml->getpar127("octaves_freq", Poctavesfreq);
Pprotectthefundamental = xml->getparbool("protect_fundamental_frequency",
Pprotectthefundamental);
- for(int i = 0; i < N_RES_POINTS; i++) {
+ for(int i = 0; i < N_RES_POINTS; ++i) {
if(xml->enterbranch("RESPOINT", i) == 0)
continue;
Prespoints[i] = xml->getpar127("val", Prespoints[i]);
diff --git a/src/Synth/SUBnote.cpp b/src/Synth/SUBnote.cpp
@@ -97,7 +97,7 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
//select only harmonics that desire to compute
int harmonics = 0;
- for(int n = 0; n < MAX_SUB_HARMONICS; n++) {
+ for(int n = 0; n < MAX_SUB_HARMONICS; ++n) {
if(pars->Phmag[n] == 0)
continue;
if(n * basefreq > SAMPLE_RATE / 2.0)
@@ -129,7 +129,7 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
//how much the amplitude is normalised (because the harmonics)
float reduceamp = 0.0;
- for(int n = 0; n < numharmonics; n++) {
+ for(int n = 0; n < numharmonics; ++n) {
float freq = basefreq * (pos[n] + 1);
//the bandwidth is not absolute(Hz); it is relative to frequency
@@ -170,7 +170,7 @@ void SUBnote::setup(float freq, float velocity, int portamento_, int midinote, b
gain *= hgain;
reduceamp += hgain;
- for(int nph = 0; nph < numstages; nph++) {
+ for(int nph = 0; nph < numstages; ++nph) {
float amp = 1.0;
if(nph == 0)
amp = gain;
@@ -400,8 +400,8 @@ void SUBnote::computecurrentparameters()
float tmpgain = 1.0 / sqrt(envbw * envfreq);
- for(int n = 0; n < numharmonics; n++) {
- for(int nph = 0; nph < numstages; nph++) {
+ for(int n = 0; n < numharmonics; ++n) {
+ for(int nph = 0; nph < numstages; ++nph) {
if(nph == 0)
gain = tmpgain;
else
@@ -413,8 +413,8 @@ void SUBnote::computecurrentparameters()
}
}
if(stereo != 0)
- for(int n = 0; n < numharmonics; n++) {
- for(int nph = 0; nph < numstages; nph++) {
+ for(int n = 0; n < numharmonics; ++n) {
+ for(int nph = 0; nph < numstages; ++nph) {
if(nph == 0)
gain = tmpgain;
else
@@ -465,13 +465,13 @@ int SUBnote::noteout(float *outl, float *outr)
float *tmprnd = getTmpBuffer();
float *tmpsmp = getTmpBuffer();
//left channel
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmprnd[i] = RND * 2.0 - 1.0;
- for(int n = 0; n < numharmonics; n++) {
+ for(int n = 0; n < numharmonics; ++n) {
memcpy(tmpsmp, tmprnd, SOUND_BUFFER_SIZE * sizeof(float));
- for(int nph = 0; nph < numstages; nph++)
+ for(int nph = 0; nph < numstages; ++nph)
filter(lfilter[nph + n * numstages], tmpsmp);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
outl[i] += tmpsmp[i];
}
@@ -480,13 +480,13 @@ int SUBnote::noteout(float *outl, float *outr)
//right channel
if(stereo != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
tmprnd[i] = RND * 2.0 - 1.0;
- for(int n = 0; n < numharmonics; n++) {
+ for(int n = 0; n < numharmonics; ++n) {
memcpy(tmpsmp, tmprnd, SOUND_BUFFER_SIZE * sizeof(float));
- for(int nph = 0; nph < numstages; nph++)
+ for(int nph = 0; nph < numstages; ++nph)
filter(rfilter[nph + n * numstages], tmpsmp);
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
outr[i] += tmpsmp[i];
}
if(GlobalFilterR != NULL)
@@ -501,7 +501,7 @@ int SUBnote::noteout(float *outl, float *outr)
int n = 10;
if(n > SOUND_BUFFER_SIZE)
n = SOUND_BUFFER_SIZE;
- for(int i = 0; i < n; i++) {
+ for(int i = 0; i < n; ++i) {
float ampfadein = 0.5 - 0.5 * cos(
(float) i / (float) n * PI);
outl[i] *= ampfadein;
@@ -512,7 +512,7 @@ int SUBnote::noteout(float *outl, float *outr)
if(ABOVE_AMPLITUDE_THRESHOLD(oldamplitude, newamplitude)) {
// Amplitude interpolation
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(oldamplitude,
newamplitude,
i,
@@ -522,7 +522,7 @@ int SUBnote::noteout(float *outl, float *outr)
}
}
else {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
outl[i] *= newamplitude * panning;
outr[i] *= newamplitude * (1.0 - panning);
}
@@ -536,7 +536,7 @@ int SUBnote::noteout(float *outl, float *outr)
// Check if the note needs to be computed more
if(AmpEnvelope->finished() != 0) {
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //fade-out
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) { //fade-out
float tmp = 1.0 - (float)i / (float)SOUND_BUFFER_SIZE;
outl[i] *= tmp;
outr[i] *= tmp;
diff --git a/src/Synth/SynthNote.cpp b/src/Synth/SynthNote.cpp
@@ -64,7 +64,7 @@ void SynthNote::Legato::apply(SynthNote ¬e, float *outl, float *outr)
if(decounter == -10)
decounter = fade.length;
//Yea, could be done without the loop...
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
decounter--;
if(decounter < 1) {
// Catching-up done, we can finally set
@@ -81,7 +81,7 @@ void SynthNote::Legato::apply(SynthNote ¬e, float *outl, float *outr)
if(decounter == -10)
decounter = fade.length;
silent = false;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
decounter--;
if(decounter < 1) {
decounter = -10;
@@ -96,10 +96,10 @@ void SynthNote::Legato::apply(SynthNote ¬e, float *outl, float *outr)
case LM_FadeOut: // Fade-out, then set the catch-up
if(decounter == -10)
decounter = fade.length;
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i) {
decounter--;
if(decounter < 1) {
- for(int j = i; j < SOUND_BUFFER_SIZE; j++) {
+ for(int j = i; j < SOUND_BUFFER_SIZE; ++j) {
outl[j] = 0.0;
outr[j] = 0.0;
}
diff --git a/src/Tests/AdNoteTest.h b/src/Tests/AdNoteTest.h
@@ -38,7 +38,7 @@ class AdNoteTest:public CxxTest::TestSuite
//next the bad global variables that for some reason have not been properly placed in some
//initialization routine, but rather exist as cryptic oneliners in main.cpp:
denormalkillbuf = new float[SOUND_BUFFER_SIZE];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
denormalkillbuf[i] = 0;
//phew, glad to get thouse out of my way. took me a lot of sweat and gdb to get this far...
diff --git a/src/Tests/MicrotonalTest.h b/src/Tests/MicrotonalTest.h
@@ -60,7 +60,7 @@ class MicrotonalTest:public CxxTest::TestSuite
(const char *)testMicro->Pcomment),
"Equal Temperament 12 notes per octave");
- for(int i = 0; i < 128; i++)
+ for(int i = 0; i < 128; ++i)
TS_ASSERT_EQUALS(testMicro->Pmapping[i], i);
TS_ASSERT_DELTA(testMicro->getnotefreq(19, 0), 24.4997, 0.0001);
diff --git a/src/Tests/OscilGenTest.h b/src/Tests/OscilGenTest.h
@@ -25,7 +25,7 @@ class OscilGenTest:public CxxTest::TestSuite
//next the bad global variables that for some reason have not been properly placed in some
//initialization routine, but rather exist as cryptic oneliners in main.cpp:
denormalkillbuf = new float[SOUND_BUFFER_SIZE];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
denormalkillbuf[i] = 0;
//prepare the default settings
diff --git a/src/Tests/SubNoteTest.h b/src/Tests/SubNoteTest.h
@@ -38,7 +38,7 @@ class SubNoteTest:public CxxTest::TestSuite
//next the bad global variables that for some reason have not been properly placed in some
//initialization routine, but rather exist as cryptic oneliners in main.cpp:
denormalkillbuf = new float[SOUND_BUFFER_SIZE];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
denormalkillbuf[i] = 0;
//prepare the default settings
diff --git a/src/main.cpp b/src/main.cpp
@@ -197,7 +197,7 @@ int main(int argc, char *argv[])
srand(time(NULL));
//produce denormal buf
denormalkillbuf = new float [SOUND_BUFFER_SIZE];
- for(int i = 0; i < SOUND_BUFFER_SIZE; i++)
+ for(int i = 0; i < SOUND_BUFFER_SIZE; ++i)
denormalkillbuf[i] = (RND - 0.5) * 1e-16;
/* Parse command-line options */
