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FP: All code should prefer single precision now
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To better match yoshimi single precision usage two large regular expressions
were applied. The performance difference according to tests is fairly minimal,
as hot spots already make heavy use of single precision variables.

Scripted Changes:
perl -pi -e "s/(\d+\.\d+)([^f^\d])/\1f\2/g" `find . |grep -E "(cpp|h)$"`
perl -pi -e "s/(exp|pow|log|sin|cos|tan)\(/\1f\(/g" `find . |grep -E "(cpp|h)$"`

Minor changes were made for a handful of values
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@fundamental
fundamental committed Sep 10, 2011
1 parent e97c0d7 commit d44dc9b
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Showing 66 changed files with 1,499 additions and 1,499 deletions.
210 changes: 105 additions & 105 deletions src/DSP/AnalogFilter.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -35,15 +35,15 @@ AnalogFilter::AnalogFilter(unsigned char Ftype,
{
stages = Fstages;
for(int i = 0; i < 3; ++i) {
coeff.c[i] = 0.0;
coeff.d[i] = 0.0;
oldCoeff.c[i] = 0.0;
oldCoeff.d[i] = 0.0;
coeff.c[i] = 0.0f;
coeff.d[i] = 0.0f;
oldCoeff.c[i] = 0.0f;
oldCoeff.d[i] = 0.0f;
}
type = Ftype;
freq = Ffreq;
q = Fq;
gain = 1.0;
gain = 1.0f;
if(stages >= MAX_FILTER_STAGES)
stages = MAX_FILTER_STAGES;
cleanup();
Expand All @@ -53,7 +53,7 @@ AnalogFilter::AnalogFilter(unsigned char Ftype,
setfreq_and_q(Ffreq, Fq);
firsttime = true;
coeff.d[0] = 0; //this is not used
outgain = 1.0;
outgain = 1.0f;
}

AnalogFilter::~AnalogFilter()
Expand All @@ -62,10 +62,10 @@ AnalogFilter::~AnalogFilter()
void AnalogFilter::cleanup()
{
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;
history[i].y2 = 0.0;
history[i].x1 = 0.0f;
history[i].x2 = 0.0f;
history[i].y1 = 0.0f;
history[i].y2 = 0.0f;
oldHistory[i] = history[i];
}
needsinterpolation = false;
Expand All @@ -78,23 +78,23 @@ void AnalogFilter::computefiltercoefs()

//do not allow frequencies bigger than samplerate/2
float freq = this->freq;
if(freq > (SAMPLE_RATE / 2 - 500.0)) {
freq = SAMPLE_RATE / 2 - 500.0;
if(freq > (SAMPLE_RATE / 2 - 500.0f)) {
freq = SAMPLE_RATE / 2 - 500.0f;
zerocoefs = true;
}
if(freq < 0.1)
freq = 0.1;
if(freq < 0.1f)
freq = 0.1f;
//do not allow bogus Q
if(q < 0.0)
q = 0.0;
if(q < 0.0f)
q = 0.0f;
float tmpq, tmpgain;
if(stages == 0) {
tmpq = q;
tmpgain = gain;
}
else {
tmpq = (q > 1.0 ? pow(q, 1.0 / (stages + 1)) : q);
tmpgain = pow(gain, 1.0 / (stages + 1));
tmpq = (q > 1.0f ? powf(q, 1.0f / (stages + 1)) : q);
tmpgain = powf(gain, 1.0f / (stages + 1));
}

//Alias Terms
Expand All @@ -103,7 +103,7 @@ void AnalogFilter::computefiltercoefs()

//General Constants
const float omega = 2 * PI * freq / SAMPLE_RATE;
const float sn = sin(omega), cs = cos(omega);
const float sn = sinf(omega), cs = cosf(omega);
float alpha, beta;

//most of theese are implementations of
Expand All @@ -113,63 +113,63 @@ void AnalogFilter::computefiltercoefs()
switch(type) {
case 0: //LPF 1 pole
if(!zerocoefs)
tmp = exp(-2.0 * PI * freq / SAMPLE_RATE);
tmp = expf(-2.0f * PI * freq / SAMPLE_RATE);
else
tmp = 0.0;
c[0] = 1.0 - tmp;
c[1] = 0.0;
c[2] = 0.0;
tmp = 0.0f;
c[0] = 1.0f - tmp;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = tmp;
d[2] = 0.0;
d[2] = 0.0f;
order = 1;
break;
case 1: //HPF 1 pole
if(!zerocoefs)
tmp = exp(-2.0 * PI * freq / SAMPLE_RATE);
tmp = expf(-2.0f * PI * freq / SAMPLE_RATE);
else
tmp = 0.0;
c[0] = (1.0 + tmp) / 2.0;
c[1] = -(1.0 + tmp) / 2.0;
c[2] = 0.0;
tmp = 0.0f;
c[0] = (1.0f + tmp) / 2.0f;
c[1] = -(1.0f + tmp) / 2.0f;
c[2] = 0.0f;
d[1] = tmp;
d[2] = 0.0;
d[2] = 0.0f;
order = 1;
break;
case 2: //LPF 2 poles
if(!zerocoefs) {
alpha = sn / (2 * tmpq);
tmp = 1 + alpha;
c[0] = (1.0 - cs) / 2.0 / tmp;
c[1] = (1.0 - cs) / tmp;
c[2] = (1.0 - cs) / 2.0 / tmp;
c[0] = (1.0f - cs) / 2.0f / tmp;
c[1] = (1.0f - cs) / tmp;
c[2] = (1.0f - cs) / 2.0f / tmp;
d[1] = -2 * cs / tmp * (-1);
d[2] = (1 - alpha) / tmp * (-1);
}
else {
c[0] = 1.0;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[0] = 1.0f;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
case 3: //HPF 2 poles
if(!zerocoefs) {
alpha = sn / (2 * tmpq);
tmp = 1 + alpha;
c[0] = (1.0 + cs) / 2.0 / tmp;
c[1] = -(1.0 + cs) / tmp;
c[2] = (1.0 + cs) / 2.0 / tmp;
c[0] = (1.0f + cs) / 2.0f / tmp;
c[1] = -(1.0f + cs) / tmp;
c[2] = (1.0f + cs) / 2.0f / tmp;
d[1] = -2 * cs / tmp * (-1);
d[2] = (1 - alpha) / tmp * (-1);
}
else {
c[0] = 0.0;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[0] = 0.0f;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
Expand All @@ -184,11 +184,11 @@ void AnalogFilter::computefiltercoefs()
d[2] = (1 - alpha) / tmp * (-1);
}
else {
c[0] = 0.0;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[0] = 0.0f;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
Expand All @@ -203,31 +203,31 @@ void AnalogFilter::computefiltercoefs()
d[2] = (1 - alpha) / tmp * (-1);
}
else {
c[0] = 1.0;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[0] = 1.0f;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
case 6: //PEAK (2 poles)
if(!zerocoefs) {
tmpq *= 3.0;
tmpq *= 3.0f;
alpha = sn / (2 * tmpq);
tmp = 1 + alpha / tmpgain;
c[0] = (1.0 + alpha * tmpgain) / tmp;
c[1] = (-2.0 * cs) / tmp;
c[2] = (1.0 - alpha * tmpgain) / tmp;
c[0] = (1.0f + alpha * tmpgain) / tmp;
c[1] = (-2.0f * cs) / tmp;
c[2] = (1.0f - alpha * tmpgain) / tmp;
d[1] = -2 * cs / tmp * (-1);
d[2] = (1 - alpha / tmpgain) / tmp * (-1);
}
else {
c[0] = 1.0;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[0] = 1.0f;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
Expand All @@ -236,27 +236,27 @@ void AnalogFilter::computefiltercoefs()
tmpq = sqrt(tmpq);
alpha = sn / (2 * tmpq);
beta = sqrt(tmpgain) / tmpq;
tmp = (tmpgain + 1.0) + (tmpgain - 1.0) * cs + beta * sn;
tmp = (tmpgain + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn;

c[0] = tmpgain
* ((tmpgain
+ 1.0) - (tmpgain - 1.0) * cs + beta * sn) / tmp;
c[1] = 2.0 * tmpgain
* ((tmpgain - 1.0) - (tmpgain + 1.0) * cs) / tmp;
+ 1.0f) - (tmpgain - 1.0f) * cs + beta * sn) / tmp;
c[1] = 2.0f * tmpgain
* ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp;
c[2] = tmpgain
* ((tmpgain
+ 1.0) - (tmpgain - 1.0) * cs - beta * sn) / tmp;
d[1] = -2.0 * ((tmpgain - 1.0) + (tmpgain + 1.0) * cs) / tmp * (-1);
+ 1.0f) - (tmpgain - 1.0f) * cs - beta * sn) / tmp;
d[1] = -2.0f * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp * (-1);
d[2] =
((tmpgain
+ 1.0) + (tmpgain - 1.0) * cs - beta * sn) / tmp * (-1);
+ 1.0f) + (tmpgain - 1.0f) * cs - beta * sn) / tmp * (-1);
}
else {
c[0] = tmpgain;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
Expand All @@ -265,27 +265,27 @@ void AnalogFilter::computefiltercoefs()
tmpq = sqrt(tmpq);
alpha = sn / (2 * tmpq);
beta = sqrt(tmpgain) / tmpq;
tmp = (tmpgain + 1.0) - (tmpgain - 1.0) * cs + beta * sn;
tmp = (tmpgain + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn;

c[0] = tmpgain
* ((tmpgain
+ 1.0) + (tmpgain - 1.0) * cs + beta * sn) / tmp;
c[1] = -2.0 * tmpgain
* ((tmpgain - 1.0) + (tmpgain + 1.0) * cs) / tmp;
+ 1.0f) + (tmpgain - 1.0f) * cs + beta * sn) / tmp;
c[1] = -2.0f * tmpgain
* ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp;
c[2] = tmpgain
* ((tmpgain
+ 1.0) + (tmpgain - 1.0) * cs - beta * sn) / tmp;
d[1] = 2.0 * ((tmpgain - 1.0) - (tmpgain + 1.0) * cs) / tmp * (-1);
+ 1.0f) + (tmpgain - 1.0f) * cs - beta * sn) / tmp;
d[1] = 2.0f * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp * (-1);
d[2] =
((tmpgain
+ 1.0) - (tmpgain - 1.0) * cs - beta * sn) / tmp * (-1);
+ 1.0f) - (tmpgain - 1.0f) * cs - beta * sn) / tmp * (-1);
}
else {
c[0] = 1.0;
c[1] = 0.0;
c[2] = 0.0;
d[1] = 0.0;
d[2] = 0.0;
c[0] = 1.0f;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = 0.0f;
d[2] = 0.0f;
}
order = 2;
break;
Expand All @@ -299,20 +299,20 @@ void AnalogFilter::computefiltercoefs()

void AnalogFilter::setfreq(float frequency)
{
if(frequency < 0.1)
frequency = 0.1;
if(frequency < 0.1f)
frequency = 0.1f;
float rap = freq / frequency;
if(rap < 1.0)
rap = 1.0 / rap;
if(rap < 1.0f)
rap = 1.0f / rap;

oldabovenq = abovenq;
abovenq = frequency > (SAMPLE_RATE / 2 - 500.0);
abovenq = frequency > (SAMPLE_RATE / 2 - 500.0f);

bool nyquistthresh = (abovenq ^ oldabovenq);


//if the frequency is changed fast, it needs interpolation
if((rap > 3.0) || nyquistthresh) { //(now, filter and coeficients backup)
if((rap > 3.0f) || nyquistthresh) { //(now, filter and coeficients backup)
oldCoeff = coeff;
for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i)
oldHistory[i] = history[i];
Expand Down Expand Up @@ -397,7 +397,7 @@ void AnalogFilter::filterout(float *smp)

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;
smp[i] = ismp[i] * (1.0f - x) + smp[i] * x;
}
returnTmpBuffer(ismp);
needsinterpolation = false;
Expand All @@ -409,20 +409,20 @@ void AnalogFilter::filterout(float *smp)

float AnalogFilter::H(float freq)
{
float fr = freq / SAMPLE_RATE * PI * 2.0;
float x = coeff.c[0], y = 0.0;
float fr = freq / SAMPLE_RATE * PI * 2.0f;
float x = coeff.c[0], y = 0.0f;
for(int n = 1; n < 3; ++n) {
x += cos(n * fr) * coeff.c[n];
y -= sin(n * fr) * coeff.c[n];
x += cosf(n * fr) * coeff.c[n];
y -= sinf(n * fr) * coeff.c[n];
}
float h = x * x + y * y;
x = 1.0;
y = 0.0;
x = 1.0f;
y = 0.0f;
for(int n = 1; n < 3; ++n) {
x -= cos(n * fr) * coeff.d[n];
y += sin(n * fr) * coeff.d[n];
x -= cosf(n * fr) * coeff.d[n];
y += sinf(n * fr) * coeff.d[n];
}
h = h / (x * x + y * y);
return pow(h, (stages + 1.0) / 2.0);
return powf(h, (stages + 1.0f) / 2.0f);
}

4 changes: 2 additions & 2 deletions src/DSP/FFTwrapper.cpp
View file
Original file line number Diff line number Diff line change
Expand Up @@ -68,8 +68,8 @@ void FFTwrapper::freqs2smps(const fft_t *freqs, float *smps)
memcpy( (void*)fft, (const void*)freqs, fftsize*sizeof(double));

//clear unused freq channel
fft[fftsize/2][0] = 0.0;
fft[fftsize/2][1] = 0.0;
fft[fftsize/2][0] = 0.0f;
fft[fftsize/2][1] = 0.0f;

//IDFT
fftw_execute(planfftw_inv);
Expand Down
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