Sound Sound_resample (Sound me, double samplingFrequency, long precision) {
double upfactor = samplingFrequency * my dx;
if (fabs (upfactor - 2) < 1e-6) return Sound_upsample (me);
if (fabs (upfactor - 1) < 1e-6) return Data_copy (me);
try {
long numberOfSamples = lround ((my xmax - my xmin) * samplingFrequency);
if (numberOfSamples < 1)
Melder_throw (U"The resampled Sound would have no samples.");
autoSound filtered = NULL;
if (upfactor < 1.0) { // need anti-aliasing filter?
long nfft = 1, antiTurnAround = 1000;
while (nfft < my nx + antiTurnAround * 2) nfft *= 2;
autoNUMvector <double> data (1, nfft);
filtered.reset (Sound_create (my ny, my xmin, my xmax, my nx, my dx, my x1));
for (long channel = 1; channel <= my ny; channel ++) {
for (long i = 1; i <= nfft; i ++) {
data [i] = 0;
}
NUMvector_copyElements (my z [channel], & data [antiTurnAround], 1, my nx);
NUMrealft (data.peek(), nfft, 1); // go to the frequency domain
for (long i = (long) floor (upfactor * nfft); i <= nfft; i ++) {
data [i] = 0; // filter away high frequencies
}
data [2] = 0.0;
NUMrealft (data.peek(), nfft, -1); // return to the time domain
double factor = 1.0 / nfft;
double *to = filtered -> z [channel];
for (long i = 1; i <= my nx; i ++) {
to [i] = data [i + antiTurnAround] * factor;
}
}
me = filtered.peek(); // reference copy; remove at end
}
autoSound thee = Sound_create (my ny, my xmin, my xmax, numberOfSamples, 1.0 / samplingFrequency,
0.5 * (my xmin + my xmax - (numberOfSamples - 1) / samplingFrequency));
for (long channel = 1; channel <= my ny; channel ++) {
double *from = my z [channel];
double *to = thy z [channel];
if (precision <= 1) {
for (long i = 1; i <= numberOfSamples; i ++) {
double x = Sampled_indexToX (thee.peek(), i);
double index = Sampled_xToIndex (me, x);
long leftSample = (long) floor (index);
double fraction = index - leftSample;
to [i] = leftSample < 1 || leftSample >= my nx ? 0.0 :
(1 - fraction) * from [leftSample] + fraction * from [leftSample + 1];
}
} else {
for (long i = 1; i <= numberOfSamples; i ++) {
double x = Sampled_indexToX (thee.peek(), i);
double index = Sampled_xToIndex (me, x);
to [i] = NUM_interpolate_sinc (my z [channel], my nx, index, precision);
}
}
}
return thee.transfer();
} catch (MelderError) {
Melder_throw (me, U": not resampled.");
}
}
Sound Sound_resample (Sound me, double samplingFrequency, long precision) {
double *data = NULL;
double upfactor = samplingFrequency * my dx;
long numberOfSamples = floor ((my xmax - my xmin) * samplingFrequency + 0.5), i;
Sound thee = NULL, filtered = NULL;
if (fabs (upfactor - 2) < 1e-6) return Sound_upsample (me);
if (fabs (upfactor - 1) < 1e-6) return (structSound *)Data_copy (me);
if (numberOfSamples < 1)
return (structSound *)Melder_errorp ("Cannot resample to 0 samples.");
thee = Sound_create (my ny, my xmin, my xmax, numberOfSamples, 1.0 / samplingFrequency,
0.5 * (my xmin + my xmax - (numberOfSamples - 1) / samplingFrequency)); cherror
if (upfactor < 1.0) { /* Need anti-aliasing filter? */
long nfft = 1, antiTurnAround = 1000;
while (nfft < my nx + antiTurnAround * 2) nfft *= 2;
data = NUMdvector (1, nfft); cherror
filtered = Sound_create (my ny, my xmin, my xmax, my nx, my dx, my x1); cherror
for (long channel = 1; channel <= my ny; channel ++) {
for (long i = 1; i <= nfft; i ++) {
data [i] = 0;
}
NUMdvector_copyElements (my z [channel], data + antiTurnAround, 1, my nx);
NUMrealft (data, nfft, 1); cherror /* Go to the frequency domain. */
for (long i = floor (upfactor * nfft); i <= nfft; i ++) {
data [i] = 0; /* Filter away high frequencies. */
}
data [2] = 0.0;
NUMrealft (data, nfft, -1); cherror /* Return to the time domain. */
double factor = 1.0 / nfft;
double *to = filtered -> z [channel];
for (long i = 1; i <= my nx; i ++) {
to [i] = data [i + antiTurnAround] * factor;
}
}
me = filtered; /* Reference copy. Remove at end. */
}
autoIntensity Sound_to_Intensity (Sound me, double minimumPitch, double timeStep, int subtractMeanPressure) {
bool veryAccurate = false;
if (veryAccurate) {
autoSound up = Sound_upsample (me); // because squaring doubles the frequency content, i.e. you get super-Nyquist components
return Sound_to_Intensity_ (up.peek(), minimumPitch, timeStep, subtractMeanPressure);
} else {
return Sound_to_Intensity_ (me, minimumPitch, timeStep, subtractMeanPressure);
}
}