void EffectNoiseRemoval::GetProfile(sampleCount len,
float *buffer)
{
float *in = new float[len];
float *out = new float[len];
int i;
for(i=0; i<len; i++)
in[i] = buffer[i];
// Apply window and FFT
/* WindowFunc(3, len, in); // Hanning window */
PowerSpectrum(len, in, out);
for(i=0; i<=len/2; i++) {
float value = log(out[i]);
if (finite(value)) {
sum[i] += value;
sumsq[i] += value*value;
profileCount[i]++;
}
}
delete[] in;
delete[] out;
}
void EffectNoiseRemoval::RemoveNoise(sampleCount len, float *buffer)
{
float *inr = new float[len];
float *ini = new float[len];
float *outr = new float[len];
float *outi = new float[len];
float *power = new float[len];
float *plog = new float[len];
int i;
for(i=0; i<len; i++)
inr[i] = buffer[i];
// Apply window and FFT
/* WindowFunc(3, len, inr); // Hanning window */
FFT(len, false, inr, NULL, outr, outi);
for(i=0; i<len; i++)
inr[i] = buffer[i];
WindowFunc(3, len, inr); // Hanning window
PowerSpectrum(len, inr, power);
for(i=0; i<=len/2; i++)
plog[i] = log(power[i]);
int half = len/2;
for(i=0; i<=half; i++) {
float smooth;
//Factor of 4 here replaces previous 2, giving a
//finer gradation of noise removal choice.
if (plog[i] < mNoiseGate[i] + (mLevel / 4.0))
smooth = float(0.0);
else
smooth = float(1.0);
smoothing[i] = smooth * 0.5 + smoothing[i] * 0.5;
}
/* try to eliminate tinkle bells */
for(i=2; i<=half-2; i++) {
if (smoothing[i]>=0.5 &&
smoothing[i]<=0.55 &&
smoothing[i-1]<0.1 &&
smoothing[i-2]<0.1 &&
smoothing[i+1]<0.1 &&
smoothing[i+2]<0.1)
smoothing[i] = float(0.0);
}
outr[0] *= smoothing[0];
outi[0] *= smoothing[0];
outr[half] *= smoothing[half];
outi[half] *= smoothing[half];
for(i=1; i<half; i++) {
int j = len - i;
float smooth = smoothing[i];
outr[i] *= smooth;
outi[i] *= smooth;
outr[j] *= smooth;
outi[j] *= smooth;
}
// Inverse FFT and normalization
FFT(len, true, outr, outi, inr, ini);
WindowFunc(3, len, inr); // Hanning window */
for(i=0; i<len; i++)
buffer[i] = inr[i];
delete[] inr;
delete[] ini;
delete[] outr;
delete[] outi;
delete[] power;
delete[] plog;
}
void FreqWindow::Recalc()
{
wxLogMessage(wxT("Starting FreqWindow::Recalc()"));
if (mProcessed)
delete[] mProcessed;
mProcessed = NULL;
if (!mData) {
mFreqPlot->Refresh(true);
return;
}
int alg = mAlgChoice->GetSelection();
int windowFunc = mFuncChoice->GetSelection();
long windowSize = 0;
(mSizeChoice->GetStringSelection()).ToLong(&windowSize);
int f = NumWindowFuncs();
if (!(windowSize >= 32 && windowSize <= 65536 &&
alg >= 0 && alg <= 4 && windowFunc >= 0 && windowFunc < f)) {
mFreqPlot->Refresh(true);
return;
}
mWindowSize = windowSize;
if (mDataLen < mWindowSize) {
mFreqPlot->Refresh(true);
return;
}
mProcessed = new float[mWindowSize];
int i;
for (i = 0; i < mWindowSize; i++)
mProcessed[i] = float(0.0);
int half = mWindowSize / 2;
float *in = new float[mWindowSize];
float *in2 = new float[mWindowSize];
float *out = new float[mWindowSize];
float *out2 = new float[mWindowSize];
float *win = new float[mWindowSize];
// initialize the window
for(int i=0; i<mWindowSize; i++)
win[i] = 1.0;
WindowFunc(windowFunc, mWindowSize, win);
// Scale window such that an amplitude of 1.0 in the time domain
// shows an amplitude of 0dB in the frequency domain
double wss = 0;
for(int i=0; i<mWindowSize; i++)
wss += win[i];
if(wss > 0)
wss = 4.0 / (wss*wss);
else
wss = 1.0;
//Progress dialog over FFT operation
wxLogMessage(wxT("Starting progress dialogue in FreqWindow::Recalc()"));
ProgressDialog *mProgress = new ProgressDialog(_("FreqWindow"),_("Drawing Spectrum"));
int start = 0;
int windows = 0;
while (start + mWindowSize <= mDataLen) {
for (i = 0; i < mWindowSize; i++)
in[i] = win[i] * mData[start + i];
switch (alg) {
case 0: // Spectrum
PowerSpectrum(mWindowSize, in, out);
for (i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
case 1:
case 2:
case 3: // Autocorrelation, Cuberoot AC or Enhanced AC
// Take FFT
#ifdef EXPERIMENTAL_USE_REALFFTF
RealFFT(mWindowSize, in, out, out2);
#else
FFT(mWindowSize, false, in, NULL, out, out2);
#endif
// Compute power
for (i = 0; i < mWindowSize; i++)
in[i] = (out[i] * out[i]) + (out2[i] * out2[i]);
if (alg == 1) {
for (i = 0; i < mWindowSize; i++)
in[i] = sqrt(in[i]);
}
if (alg == 2 || alg == 3) {
// Tolonen and Karjalainen recommend taking the cube root
// of the power, instead of the square root
for (i = 0; i < mWindowSize; i++)
in[i] = pow(in[i], 1.0f / 3.0f);
}
// Take FFT
#ifdef EXPERIMENTAL_USE_REALFFTF
RealFFT(mWindowSize, in, out, out2);
#else
FFT(mWindowSize, false, in, NULL, out, out2);
#endif
// Take real part of result
for (i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
case 4: // Cepstrum
#ifdef EXPERIMENTAL_USE_REALFFTF
RealFFT(mWindowSize, in, out, out2);
#else
FFT(mWindowSize, false, in, NULL, out, out2);
#endif
// Compute log power
// Set a sane lower limit assuming maximum time amplitude of 1.0
float power;
float minpower = 1e-20*mWindowSize*mWindowSize;
for (i = 0; i < mWindowSize; i++)
{
power = (out[i] * out[i]) + (out2[i] * out2[i]);
if(power < minpower)
in[i] = log(minpower);
else
in[i] = log(power);
}
// Take IFFT
#ifdef EXPERIMENTAL_USE_REALFFTF
InverseRealFFT(mWindowSize, in, NULL, out);
#else
FFT(mWindowSize, true, in, NULL, out, out2);
#endif
// Take real part of result
for (i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
} //switch
start += half;
windows++;
// only update the progress dialogue infrequently to reduce it's overhead
// If we do it every time, it spends as much time updating X11 as doing
// the calculations. 10 seems a reasonable compromise on Linux that
// doesn't make it unresponsive, but avoids the slowdown.
if ((windows % 10) == 0)
mProgress->Update(1 - static_cast<float>(mDataLen - start) / mDataLen);
}
wxLogMessage(wxT("Finished updating progress dialogue in FreqWindow::Recalc()"));
switch (alg) {
double scale;
case 0: // Spectrum
// Convert to decibels
mYMin = 1000000.;
mYMax = -1000000.;
scale = wss / (double)windows;
for (i = 0; i < half; i++)
{
mProcessed[i] = 10 * log10(mProcessed[i] * scale);
if(mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if(mProcessed[i] < mYMin)
mYMin = mProcessed[i];
}
if(mYMin < -dBRange)
mYMin = -dBRange;
if(mYMax <= -dBRange)
mYMax = -dBRange + 10.; // it's all out of range, but show a scale.
else
mYMax += .5;
mProcessedSize = half;
mYStep = 10;
break;
case 1: // Standard Autocorrelation
case 2: // Cuberoot Autocorrelation
for (i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Find min/max
mYMin = mProcessed[0];
mYMax = mProcessed[0];
for (i = 1; i < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
mYStep = 1;
mProcessedSize = half;
break;
case 3: // Enhanced Autocorrelation
for (i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Peak Pruning as described by Tolonen and Karjalainen, 2000
// Clip at zero, copy to temp array
for (i = 0; i < half; i++) {
if (mProcessed[i] < 0.0)
mProcessed[i] = float(0.0);
out[i] = mProcessed[i];
}
// Subtract a time-doubled signal (linearly interp.) from the original
// (clipped) signal
for (i = 0; i < half; i++)
if ((i % 2) == 0)
mProcessed[i] -= out[i / 2];
else
mProcessed[i] -= ((out[i / 2] + out[i / 2 + 1]) / 2);
// Clip at zero again
for (i = 0; i < half; i++)
if (mProcessed[i] < 0.0)
mProcessed[i] = float(0.0);
// Find new min/max
mYMin = mProcessed[0];
mYMax = mProcessed[0];
for (i = 1; i < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
mYStep = 1;
mProcessedSize = half;
break;
case 4: // Cepstrum
for (i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Find min/max, ignoring first and last few values
int ignore = 4;
mYMin = mProcessed[ignore];
mYMax = mProcessed[ignore];
for (i = ignore + 1; i < half - ignore; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
mYStep = 1;
mProcessedSize = half;
break;
}
delete[]in;
delete[]in2;
delete[]out;
delete[]out2;
delete[]win;
wxLogMessage(wxT("About to draw plot in FreqWindow::Recalc()"));
DrawPlot();
mFreqPlot->Refresh(true);
delete mProgress;
}
/**
* Execute smoothing of a single spectrum.
* @param inputWS :: A workspace to pick a spectrum from.
* @param wsIndex :: An index of a spectrum to smooth.
* @return :: A single-spectrum workspace with the smoothed data.
*/
API::MatrixWorkspace_sptr
WienerSmooth::smoothSingleSpectrum(API::MatrixWorkspace_sptr inputWS,
size_t wsIndex) {
size_t dataSize = inputWS->blocksize();
// it won't work for very small workspaces
if (dataSize < 4) {
g_log.debug() << "No smoothing, spectrum copied." << std::endl;
return copyInput(inputWS, wsIndex);
}
// Due to the way RealFFT works the input should be even-sized
const bool isOddSize = dataSize % 2 != 0;
if (isOddSize) {
// add a fake value to the end to make size even
inputWS = copyInput(inputWS, wsIndex);
wsIndex = 0;
auto &X = inputWS->dataX(wsIndex);
auto &Y = inputWS->dataY(wsIndex);
auto &E = inputWS->dataE(wsIndex);
double dx = X[dataSize - 1] - X[dataSize - 2];
X.push_back(X.back() + dx);
Y.push_back(Y.back());
E.push_back(E.back());
}
// the input vectors
auto &X = inputWS->readX(wsIndex);
auto &Y = inputWS->readY(wsIndex);
auto &E = inputWS->readE(wsIndex);
// Digital fourier transform works best for data oscillating around 0.
// Fit a spline with a small number of break points to the data.
// Make sure that the spline passes through the first and the last points
// of the data.
// The fitted spline will be subtracted from the data and the difference
// will be smoothed with the Wiener filter. After that the spline will be
// added to the smoothed data to produce the output.
// number of spline break points, must be smaller than the data size but
// between 2 and 10
size_t nbreak = 10;
if (nbreak * 3 > dataSize)
nbreak = dataSize / 3;
// NB. The spline mustn't fit too well to the data. If it does smoothing
// doesn't happen.
// TODO: it's possible that the spline is unnecessary and a simple linear
// function will
// do a better job.
g_log.debug() << "Spline break points " << nbreak << std::endl;
// define the spline
API::IFunction_sptr spline =
API::FunctionFactory::Instance().createFunction("BSpline");
auto xInterval = getStartEnd(X, inputWS->isHistogramData());
spline->setAttributeValue("StartX", xInterval.first);
spline->setAttributeValue("EndX", xInterval.second);
spline->setAttributeValue("NBreak", static_cast<int>(nbreak));
// fix the first and last parameters to the first and last data values
spline->setParameter(0, Y.front());
spline->fix(0);
size_t lastParamIndex = spline->nParams() - 1;
spline->setParameter(lastParamIndex, Y.back());
spline->fix(lastParamIndex);
// fit the spline to the data
auto fit = createChildAlgorithm("Fit");
fit->initialize();
fit->setProperty("Function", spline);
fit->setProperty("InputWorkspace", inputWS);
fit->setProperty("WorkspaceIndex", static_cast<int>(wsIndex));
fit->setProperty("CreateOutput", true);
fit->execute();
// get the fit output workspace; spectrum 2 contains the difference that is to
// be smoothed
API::MatrixWorkspace_sptr fitOut = fit->getProperty("OutputWorkspace");
// Fourier transform the difference spectrum
auto fourier = createChildAlgorithm("RealFFT");
fourier->initialize();
fourier->setProperty("InputWorkspace", fitOut);
fourier->setProperty("WorkspaceIndex", 2);
// we don't require bin linearity as we don't need the exact transform
fourier->setProperty("IgnoreXBins", true);
fourier->execute();
API::MatrixWorkspace_sptr fourierOut =
fourier->getProperty("OutputWorkspace");
// spectrum 2 of the transformed workspace has the transform modulus which is
// a square
// root of the power spectrum
auto &powerSpec = fourierOut->dataY(2);
// convert the modulus to power spectrum wich is the base of the Wiener filter
std::transform(powerSpec.begin(), powerSpec.end(), powerSpec.begin(),
PowerSpectrum());
// estimate power spectrum's noise as the average of its high frequency half
size_t n2 = powerSpec.size();
double noise =
std::accumulate(powerSpec.begin() + n2 / 2, powerSpec.end(), 0.0);
noise /= static_cast<double>(n2);
// index of the maximum element in powerSpec
const size_t imax = static_cast<size_t>(std::distance(
powerSpec.begin(), std::max_element(powerSpec.begin(), powerSpec.end())));
if (noise == 0.0) {
noise = powerSpec[imax] / guessSignalToNoiseRatio;
}
g_log.debug() << "Maximum signal " << powerSpec[imax] << std::endl;
g_log.debug() << "Noise " << noise << std::endl;
// storage for the Wiener filter, initialized with 0.0's
std::vector<double> wf(n2);
// The filter consists of two parts:
// 1) low frequency region, from 0 until the power spectrum falls to the
// noise level, filter is calculated
// from the power spectrum
// 2) high frequency noisy region, filter is a smooth function of frequency
// decreasing to 0
// the following code is an adaptation of a fortran routine
// noise starting index
size_t i0 = 0;
// intermediate variables
double xx = 0.0;
double xy = 0.0;
double ym = 0.0;
// low frequency filter values: the higher the power spectrum the closer the
// filter to 1.0
for (size_t i = 0; i < n2; ++i) {
double cd1 = powerSpec[i] / noise;
if (cd1 < 1.0 && i > imax) {
i0 = i;
break;
}
double cd2 = log(cd1);
wf[i] = cd1 / (1.0 + cd1);
double j = static_cast<double>(i + 1);
xx += j * j;
xy += j * cd2;
ym += cd2;
}
// i0 should always be > 0 but in case something goes wrong make a check
if (i0 > 0) {
g_log.debug() << "Noise start index " << i0 << std::endl;
// high frequency filter values: smooth decreasing function
double ri0f = static_cast<double>(i0 + 1);
double xm = (1.0 + ri0f) / 2;
ym /= ri0f;
double a1 = (xy - ri0f * xm * ym) / (xx - ri0f * xm * xm);
double b1 = ym - a1 * xm;
g_log.debug() << "(a1,b1) = (" << a1 << ',' << b1 << ')' << std::endl;
const double dblev = -20.0;
// cut-off index
double ri1 = floor((dblev / 4 - b1) / a1);
if (ri1 < static_cast<double>(i0)) {
g_log.warning() << "Failed to build Wiener filter: no smoothing."
<< std::endl;
ri1 = static_cast<double>(i0);
}
size_t i1 = static_cast<size_t>(ri1);
if (i1 > n2)
i1 = n2;
for (size_t i = i0; i < i1; ++i) {
double s = exp(a1 * static_cast<double>(i + 1) + b1);
wf[i] = s / (1.0 + s);
}
// wf[i] for i1 <= i < n2 are 0.0
g_log.debug() << "Cut-off index " << i1 << std::endl;
} else {
g_log.warning() << "Power spectrum has an unexpected shape: no smoothing"
<< std::endl;
return copyInput(inputWS, wsIndex);
}
// multiply the fourier transform by the filter
auto &re = fourierOut->dataY(0);
auto &im = fourierOut->dataY(1);
std::transform(re.begin(), re.end(), wf.begin(), re.begin(),
std::multiplies<double>());
std::transform(im.begin(), im.end(), wf.begin(), im.begin(),
std::multiplies<double>());
// inverse fourier transform
fourier = createChildAlgorithm("RealFFT");
fourier->initialize();
fourier->setProperty("InputWorkspace", fourierOut);
fourier->setProperty("IgnoreXBins", true);
fourier->setPropertyValue("Transform", "Backward");
fourier->execute();
API::MatrixWorkspace_sptr out = fourier->getProperty("OutputWorkspace");
auto &background = fitOut->readY(1);
auto &y = out->dataY(0);
if (y.size() != background.size()) {
throw std::logic_error("Logic error: inconsistent arrays");
}
// add the spline "background" to the smoothed data
std::transform(y.begin(), y.end(), background.begin(), y.begin(),
std::plus<double>());
// copy the x-values and errors from the original spectrum
// remove the last values for odd-sized inputs
if (isOddSize) {
out->dataX(0).assign(X.begin(), X.end() - 1);
out->dataE(0).assign(E.begin(), E.end() - 1);
out->dataY(0).resize(Y.size() - 1);
} else {
out->setX(0, X);
out->dataE(0).assign(E.begin(), E.end());
}
return out;
}
bool SpectrumAnalyst::Calculate(Algorithm alg, int windowFunc,
size_t windowSize, double rate,
const float *data, size_t dataLen,
float *pYMin, float *pYMax,
FreqGauge *progress)
{
// Wipe old data
mProcessed.resize(0);
mRate = 0.0;
mWindowSize = 0;
// Validate inputs
int f = NumWindowFuncs();
if (!(windowSize >= 32 && windowSize <= 65536 &&
alg >= SpectrumAnalyst::Spectrum &&
alg < SpectrumAnalyst::NumAlgorithms &&
windowFunc >= 0 && windowFunc < f)) {
return false;
}
if (dataLen < windowSize) {
return false;
}
// Now repopulate
mRate = rate;
mWindowSize = windowSize;
mAlg = alg;
auto half = mWindowSize / 2;
mProcessed.resize(mWindowSize);
Floats in{ mWindowSize };
Floats out{ mWindowSize };
Floats out2{ mWindowSize };
Floats win{ mWindowSize };
for (size_t i = 0; i < mWindowSize; i++) {
mProcessed[i] = 0.0f;
win[i] = 1.0f;
}
WindowFunc(windowFunc, mWindowSize, win.get());
// Scale window such that an amplitude of 1.0 in the time domain
// shows an amplitude of 0dB in the frequency domain
double wss = 0;
for (size_t i = 0; i<mWindowSize; i++)
wss += win[i];
if(wss > 0)
wss = 4.0 / (wss*wss);
else
wss = 1.0;
if (progress) {
progress->SetRange(dataLen);
}
size_t start = 0;
int windows = 0;
while (start + mWindowSize <= dataLen) {
for (size_t i = 0; i < mWindowSize; i++)
in[i] = win[i] * data[start + i];
switch (alg) {
case Spectrum:
PowerSpectrum(mWindowSize, in.get(), out.get());
for (size_t i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
case Autocorrelation:
case CubeRootAutocorrelation:
case EnhancedAutocorrelation:
// Take FFT
RealFFT(mWindowSize, in.get(), out.get(), out2.get());
// Compute power
for (size_t i = 0; i < mWindowSize; i++)
in[i] = (out[i] * out[i]) + (out2[i] * out2[i]);
if (alg == Autocorrelation) {
for (size_t i = 0; i < mWindowSize; i++)
in[i] = sqrt(in[i]);
}
if (alg == CubeRootAutocorrelation ||
alg == EnhancedAutocorrelation) {
// Tolonen and Karjalainen recommend taking the cube root
// of the power, instead of the square root
for (size_t i = 0; i < mWindowSize; i++)
in[i] = pow(in[i], 1.0f / 3.0f);
}
// Take FFT
RealFFT(mWindowSize, in.get(), out.get(), out2.get());
// Take real part of result
for (size_t i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
case Cepstrum:
RealFFT(mWindowSize, in.get(), out.get(), out2.get());
// Compute log power
// Set a sane lower limit assuming maximum time amplitude of 1.0
{
float power;
float minpower = 1e-20*mWindowSize*mWindowSize;
for (size_t i = 0; i < mWindowSize; i++)
{
power = (out[i] * out[i]) + (out2[i] * out2[i]);
if(power < minpower)
in[i] = log(minpower);
else
in[i] = log(power);
}
// Take IFFT
InverseRealFFT(mWindowSize, in.get(), NULL, out.get());
// Take real part of result
for (size_t i = 0; i < half; i++)
mProcessed[i] += out[i];
}
break;
default:
wxASSERT(false);
break;
} //switch
// Update the progress bar
if (progress) {
progress->SetValue(start);
}
start += half;
windows++;
}
if (progress) {
// Reset for next time
progress->Reset();
}
float mYMin = 1000000, mYMax = -1000000;
double scale;
switch (alg) {
case Spectrum:
// Convert to decibels
mYMin = 1000000.;
mYMax = -1000000.;
scale = wss / (double)windows;
for (size_t i = 0; i < half; i++)
{
mProcessed[i] = 10 * log10(mProcessed[i] * scale);
if(mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if(mProcessed[i] < mYMin)
mYMin = mProcessed[i];
}
break;
case Autocorrelation:
case CubeRootAutocorrelation:
for (size_t i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Find min/max
mYMin = mProcessed[0];
mYMax = mProcessed[0];
for (size_t i = 1; i < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
break;
case EnhancedAutocorrelation:
for (size_t i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Peak Pruning as described by Tolonen and Karjalainen, 2000
// Clip at zero, copy to temp array
for (size_t i = 0; i < half; i++) {
if (mProcessed[i] < 0.0)
mProcessed[i] = float(0.0);
out[i] = mProcessed[i];
}
// Subtract a time-doubled signal (linearly interp.) from the original
// (clipped) signal
for (size_t i = 0; i < half; i++)
if ((i % 2) == 0)
mProcessed[i] -= out[i / 2];
else
mProcessed[i] -= ((out[i / 2] + out[i / 2 + 1]) / 2);
// Clip at zero again
for (size_t i = 0; i < half; i++)
if (mProcessed[i] < 0.0)
mProcessed[i] = float(0.0);
// Find NEW min/max
mYMin = mProcessed[0];
mYMax = mProcessed[0];
for (size_t i = 1; i < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
break;
case Cepstrum:
for (size_t i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Find min/max, ignoring first and last few values
{
size_t ignore = 4;
mYMin = mProcessed[ignore];
mYMax = mProcessed[ignore];
for (size_t i = ignore + 1; i + ignore < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
}
break;
default:
wxASSERT(false);
break;
}
if (pYMin)
*pYMin = mYMin;
if (pYMax)
*pYMax = mYMax;
return true;
}
void EffectNoiseRemoval::RemoveNoise(sampleCount len,
sampleType *buffer)
{
float *inr = new float[len];
float *ini = new float[len];
float *outr = new float[len];
float *outi = new float[len];
float *power = new float[len];
float *plog = new float[len];
int i;
for(i=0; i<len; i++)
inr[i] = buffer[i]/32767.;
// Apply window and FFT
WindowFunc(3, len, inr); // Hanning window
FFT(len, false, inr, NULL, outr, outi);
for(i=0; i<len; i++)
inr[i] = buffer[i]/32767.;
WindowFunc(3, len, inr); // Hanning window
PowerSpectrum(len, inr, power);
for(i=0; i<=len/2; i++)
plog[i] = log(power[i]);
int half = len/2;
for(i=0; i<=half; i++) {
float smooth;
if (plog[i] < noiseGate[i] + (level/2.0))
smooth = 0.0;
else
smooth = 1.0;
smoothing[i] = smooth * 0.5 + smoothing[i] * 0.5;
}
/* try to eliminate tinkle bells */
for(i=2; i<=half-2; i++) {
if (smoothing[i]>=0.5 &&
smoothing[i]<=0.55 &&
smoothing[i-1]<0.1 &&
smoothing[i-2]<0.1 &&
smoothing[i+1]<0.1 &&
smoothing[i+2]<0.1)
smoothing[i] = 0.0;
}
outr[0] *= smoothing[0];
outi[0] *= smoothing[0];
outr[half] *= smoothing[half];
outi[half] *= smoothing[half];
for(i=1; i<half; i++) {
int j = len - i;
float smooth = smoothing[i];
outr[i] *= smooth;
outi[i] *= smooth;
outr[j] *= smooth;
outi[j] *= smooth;
}
// Inverse FFT and normalization
FFT(len, true, outr, outi, inr, ini);
for(i=0; i<len; i++)
buffer[i] = sampleType(inr[i]*32767);
delete[] inr;
delete[] ini;
delete[] outr;
delete[] outi;
delete[] power;
delete[] plog;
}
bool ComputeSpectrum(float * data, int width, int height,
int maxFreq, int windowSize,
double rate, float *grayscaleOut,
bool autocorrelation)
{
int windowFunc = 3;
if (width < windowSize)
return false;
if (!data || !grayscaleOut)
return true;
float *processed = new float[windowSize];
int i;
for (i = 0; i < windowSize; i++)
processed[i] = float(0.0);
int half = windowSize / 2;
float *in = new float[windowSize];
float *out = new float[windowSize];
float *out2 = new float[windowSize];
int start = 0;
int windows = 0;
while (start + windowSize <= width) {
for (i = 0; i < windowSize; i++)
in[i] = data[start + i];
WindowFunc(windowFunc, windowSize, in);
if (autocorrelation) {
// Take FFT
FFT(windowSize, false, in, NULL, out, out2);
// Compute power
for (i = 0; i < windowSize; i++)
in[i] = (out[i] * out[i]) + (out2[i] * out2[i]);
// Tolonen and Karjalainen recommend taking the cube root
// of the power, instead of the square root
for (i = 0; i < windowSize; i++)
in[i] = pow(in[i], 1.0f / 3.0f);
// Take FFT
FFT(windowSize, false, in, NULL, out, out2);
// Take real part of result
for (i = 0; i < half; i++)
processed[i] += out[i];
} else {
PowerSpectrum(windowSize, in, out);
for (i = 0; i < half; i++)
processed[i] += out[i];
}
start += half;
windows++;
}
int maxSamples = int (maxFreq * windowSize / rate + 0.5);
if (maxSamples > half)
maxSamples = half;
if (autocorrelation) {
maxSamples = half;
// Peak Pruning as described by Tolonen and Karjalainen, 2000
// Clip at zero, copy to temp array
for (i = 0; i < maxSamples; i++) {
if (processed[i] < 0.0)
processed[i] = float(0.0);
out[i] = processed[i];
}
// Subtract a time-doubled signal (linearly interp.) from the original
// (clipped) signal
for (i = 0; i < maxSamples; i++)
if ((i % 2) == 0)
processed[i] -= out[i / 2];
else
processed[i] -= ((out[i / 2] + out[i / 2 + 1]) / 2);
// Clip at zero again
for (i = 0; i < maxSamples; i++)
if (processed[i] < 0.0)
processed[i] = float(0.0);
// Find new max
float max = 0;
for (i = 1; i < maxSamples; i++)
if (processed[i] > max)
max = processed[i];
// Reverse and scale
for (i = 0; i < maxSamples; i++)
in[i] = processed[i] / (windowSize / 4);
for (i = 0; i < maxSamples; i++)
processed[maxSamples - 1 - i] = in[i];
} else {
// Convert to decibels
for (i = 0; i < maxSamples; i++)
processed[i] = 10 * log10(processed[i] / windowSize / windows);
}
// Finally, put it into bins in grayscaleOut[], normalized to a 0.0-1.0 scale
for (i = 0; i < height; i++) {
float bin0 = float (i) * maxSamples / height;
float bin1 = float (i + 1) * maxSamples / height;
float binwidth = bin1 - bin0;
float value = float(0.0);
if (int (bin1) == int (bin0))
value = processed[int (bin0)];
else {
value += processed[int (bin0)] * (int (bin0) + 1 - bin0);
bin0 = 1 + int (bin0);
while (bin0 < int (bin1)) {
value += processed[int (bin0)];
bin0 += 1.0;
}
value += processed[int (bin1)] * (bin1 - int (bin1));
value /= binwidth;
}
if (!autocorrelation) {
// Last step converts dB to a 0.0-1.0 range
value = (value + 80.0) / 80.0;
}
if (value > 1.0)
value = float(1.0);
if (value < 0.0)
value = float(0.0);
grayscaleOut[i] = value;
}
delete[]in;
delete[]out;
delete[]out2;
delete[]processed;
return true;
}
void FreqWindow::Recalc()
{
if (mProcessed)
delete mProcessed;
mProcessed = NULL;
if (!mData) {
mFreqPlot->Refresh(false);
return;
}
int alg = mAlgChoice->GetSelection();
int windowFunc = mFuncChoice->GetSelection();
long windowSize = 0;
(mSizeChoice->GetStringSelection()).ToLong(&windowSize);
if (!(windowSize >= 32 && windowSize <= 65536 &&
alg >= 0 && alg <= 3 && windowFunc >= 0 && windowFunc <= 3)) {
mFreqPlot->Refresh(false);
return;
}
mWindowSize = windowSize;
if (mDataLen < mWindowSize) {
mFreqPlot->Refresh(false);
return;
}
mProcessed = new float[mWindowSize];
int i;
for (i = 0; i < mWindowSize; i++)
mProcessed[i] = 0.0;
int half = mWindowSize / 2;
float *in = new float[mWindowSize];
float *in2 = new float[mWindowSize];
float *out = new float[mWindowSize];
float *out2 = new float[mWindowSize];
int start = 0;
int windows = 0;
while (start + mWindowSize <= mDataLen) {
for (i = 0; i < mWindowSize; i++)
in[i] = mData[start + i];
WindowFunc(windowFunc, mWindowSize, in);
switch (alg) {
case 0: // Spectrum
PowerSpectrum(mWindowSize, in, out);
for (i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
case 1:
case 2:
case 3: // Autocorrelation, Cuberoot AC or Enhanced AC
// Take FFT
FFT(mWindowSize, false, in, NULL, out, out2);
// Compute power
for (i = 0; i < mWindowSize; i++)
in[i] = (out[i] * out[i]) + (out2[i] * out2[i]);
if (alg == 1) {
for (i = 0; i < mWindowSize; i++)
in[i] = sqrt(in[i]);
}
if (alg == 2 || alg == 3) {
// Tolonen and Karjalainen recommend taking the cube root
// of the power, instead of the square root
for (i = 0; i < mWindowSize; i++)
in[i] = pow(in[i], 1.0 / 3.0);
}
// Take FFT
FFT(mWindowSize, false, in, NULL, out, out2);
// Take real part of result
for (i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
case 4: // Cepstrum
FFT(mWindowSize, false, in, NULL, out, out2);
// Compute log power
for (i = 0; i < mWindowSize; i++)
in[i] = log((out[i] * out[i]) + (out2[i] * out2[i]));
// Take IFFT
FFT(mWindowSize, true, in, NULL, out, out2);
// Take real part of result
for (i = 0; i < half; i++)
mProcessed[i] += out[i];
break;
} //switch
start += half;
windows++;
}
switch (alg) {
case 0: // Spectrum
// Convert to decibels
for (i = 0; i < half; i++)
mProcessed[i] = 10 * log10(mProcessed[i] / mWindowSize / windows);
mProcessedSize = half;
mYMin = -90;
mYMax = 10;
mYStep = 10;
break;
case 1: // Standard Autocorrelation
case 2: // Cuberoot Autocorrelation
for (i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Find min/max
mYMin = mProcessed[0];
mYMax = mProcessed[0];
for (i = 1; i < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
mYStep = 1;
mProcessedSize = half;
break;
case 3: // Enhanced Autocorrelation
for (i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Peak Pruning as described by Tolonen and Karjalainen, 2000
// Clip at zero, copy to temp array
for (i = 0; i < half; i++) {
if (mProcessed[i] < 0.0)
mProcessed[i] = 0.0;
out[i] = mProcessed[i];
}
// Subtract a time-doubled signal (linearly interp.) from the original
// (clipped) signal
for (i = 0; i < half; i++)
if ((i % 2) == 0)
mProcessed[i] -= out[i / 2];
else
mProcessed[i] -= ((out[i / 2] + out[i / 2 + 1]) / 2);
// Clip at zero again
for (i = 0; i < half; i++)
if (mProcessed[i] < 0.0)
mProcessed[i] = 0.0;
// Find new min/max
mYMin = mProcessed[0];
mYMax = mProcessed[0];
for (i = 1; i < half; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
mYStep = 1;
mProcessedSize = half;
break;
case 4: // Cepstrum
for (i = 0; i < half; i++)
mProcessed[i] = mProcessed[i] / windows;
// Find min/max, ignoring first and last few values
int ignore = 4;
mYMin = mProcessed[ignore];
mYMax = mProcessed[ignore];
for (i = ignore + 1; i < half - ignore; i++)
if (mProcessed[i] > mYMax)
mYMax = mProcessed[i];
else if (mProcessed[i] < mYMin)
mYMin = mProcessed[i];
mYStep = 1;
mProcessedSize = half;
break;
}
delete[]in;
delete[]in2;
delete[]out;
delete[]out2;
mFreqPlot->Refresh(false);
}
/* Mangle a single window. Each output sample (except the first and last
* half-window) is the result of two distinct calls to this function,
* due to overlapping windows. */
static void reduce_noise(chandata_t* chan, float* window, float level)
{
float *inr = (float*)calloc(WINDOWSIZE, sizeof(float));
float *ini = (float*)calloc(WINDOWSIZE, sizeof(float));
float *outr = (float*)calloc(WINDOWSIZE, sizeof(float));
float *outi = (float*)calloc(WINDOWSIZE, sizeof(float));
float *power = (float*)calloc(WINDOWSIZE, sizeof(float));
float *smoothing = chan->smoothing;
static int callnum = 0;
int i;
callnum ++;
for (i = 0; i < FREQCOUNT; i ++) {
assert(smoothing[i] >= 0 && smoothing[i] <= 1);
}
memcpy(inr, window, WINDOWSIZE*sizeof(float));
FFT(WINDOWSIZE, 0, inr, NULL, outr, outi);
memcpy(inr, window, WINDOWSIZE*sizeof(float));
WindowFunc(HANNING, WINDOWSIZE, inr);
PowerSpectrum(WINDOWSIZE, inr, power);
for(i = 0; i < FREQCOUNT; i ++) {
float smooth;
float plog;
plog = log(power[i]);
if (power[i] != 0 && plog < chan->noisegate[i] + level*8.0)
smooth = 0.0;
else
smooth = 1.0;
smoothing[i] = smooth * 0.5 + smoothing[i] * 0.5;
}
/* Audacity says this code will eliminate tinkle bells.
* I have no idea what that means. */
for (i = 2; i < FREQCOUNT - 2; i ++) {
if (smoothing[i]>=0.5 &&
smoothing[i]<=0.55 &&
smoothing[i-1]<0.1 &&
smoothing[i-2]<0.1 &&
smoothing[i+1]<0.1 &&
smoothing[i+2]<0.1)
smoothing[i] = 0.0;
}
outr[0] *= smoothing[0];
outi[0] *= smoothing[0];
outr[FREQCOUNT-1] *= smoothing[FREQCOUNT-1];
outi[FREQCOUNT-1] *= smoothing[FREQCOUNT-1];
for (i = 1; i < FREQCOUNT-1; i ++) {
int j = WINDOWSIZE - i;
float smooth = smoothing[i];
outr[i] *= smooth;
outi[i] *= smooth;
outr[j] *= smooth;
outi[j] *= smooth;
}
FFT(WINDOWSIZE, 1, outr, outi, inr, ini);
WindowFunc(HANNING, WINDOWSIZE, inr);
memcpy(window, inr, WINDOWSIZE*sizeof(float));
free(inr);
free(ini);
free(outr);
free(outi);
free(power);
for (i = 0; i < FREQCOUNT; i ++) {
assert(smoothing[i] >= 0 && smoothing[i] <= 1);
}
}
bool ComputeSpectrum(const float * data, int width,
int windowSize,
double WXUNUSED(rate), float *output,
bool autocorrelation, int windowFunc)
{
if (width < windowSize)
return false;
if (!data || !output)
return true;
float *processed = new float[windowSize];
int i;
for (i = 0; i < windowSize; i++)
processed[i] = float(0.0);
int half = windowSize / 2;
float *in = new float[windowSize];
float *out = new float[windowSize];
float *out2 = new float[windowSize];
int start = 0;
int windows = 0;
while (start + windowSize <= width) {
for (i = 0; i < windowSize; i++)
in[i] = data[start + i];
WindowFunc(windowFunc, windowSize, in);
if (autocorrelation) {
// Take FFT
RealFFT(windowSize, in, out, out2);
// Compute power
for (i = 0; i < windowSize; i++)
in[i] = (out[i] * out[i]) + (out2[i] * out2[i]);
// Tolonen and Karjalainen recommend taking the cube root
// of the power, instead of the square root
for (i = 0; i < windowSize; i++)
in[i] = powf(in[i], 1.0f / 3.0f);
// Take FFT
RealFFT(windowSize, in, out, out2);
}
else
PowerSpectrum(windowSize, in, out);
// Take real part of result
for (i = 0; i < half; i++)
processed[i] += out[i];
start += half;
windows++;
}
if (autocorrelation) {
// Peak Pruning as described by Tolonen and Karjalainen, 2000
/*
Combine most of the calculations in a single for loop.
It should be safe, as indexes refer only to current and previous elements,
that have already been clipped, etc...
*/
for (i = 0; i < half; i++) {
// Clip at zero, copy to temp array
if (processed[i] < 0.0)
processed[i] = float(0.0);
out[i] = processed[i];
// Subtract a time-doubled signal (linearly interp.) from the original
// (clipped) signal
if ((i % 2) == 0)
processed[i] -= out[i / 2];
else
processed[i] -= ((out[i / 2] + out[i / 2 + 1]) / 2);
// Clip at zero again
if (processed[i] < 0.0)
processed[i] = float(0.0);
}
// Reverse and scale
for (i = 0; i < half; i++)
in[i] = processed[i] / (windowSize / 4);
for (i = 0; i < half; i++)
processed[half - 1 - i] = in[i];
} else {
// Convert to decibels
// But do it safely; -Inf is nobody's friend
for (i = 0; i < half; i++){
float temp=(processed[i] / windowSize / windows);
if (temp > 0.0)
processed[i] = 10*log10(temp);
else
processed[i] = 0;
}
}
for(i=0;i<half;i++)
output[i] = processed[i];
delete[]in;
delete[]out;
delete[]out2;
delete[]processed;
return true;
}
