public static FilterProfile Profile(ISoundObj impulse, SmoothingType type, double resolution)
{
uint nSR = impulse.SampleRate;
uint nSR2 = nSR / 2;
ushort nChannels = impulse.NumChannels;
for (ushort c = 0; c < nChannels; c++)
{
// Read channel into a buffer
SingleChannel channel = impulse.Channel(c);
SoundBuffer buff = new SoundBuffer(channel);
buff.ReadAll();
// And then double in length to prevent wraparound
buff.PadTo(buff.Count * 2);
// Pad to next higher power of two
buff.PadToPowerOfTwo();
// Read out into array of complex
Complex[][] data = buff.ToComplexArray();
Complex[] cdata = data[0];
// Then we're done with the buffer for this channel
buff = null;
GC.Collect();
// FFT in place
Fourier.FFT(cdata.Length, cdata);
int n = cdata.Length / 2;
// Now we have an array of complex, from 0Hz to Nyquist and back again.
// We really only care about the first half of the cdata buffer, but
// treat it as circular anyway (i.e. wrap around for negative values).
//
// We're only working with magnitudes from here on,
// so we can save some space by computing mags right away and storing them in the
// real part of the complex array; then we can use the imaginary portion for the
// smoothed data.
for (int j = 0; j < cdata.Length; j++)
{
cdata[j].Re = cdata[j].Magnitude;
cdata[j].Im = 0;
}
// Take a rectangular window of width (resolution)*(octave or ERB band)
// Add up all magnitudes falling within this window
//
// Move the window forward by one thingummajig
//double wMid = 0; // center of the window
//double wLen = 0;
}
return(new FilterProfile()); // temp
}
/// <summary>
/// Calculate the weighted volume of a sound source.
/// NB: this consumes lots of memory for long sources.
/// </summary>
/// <param name="src"></param>
/// <param name="dbSPL"></param>
/// <returns>Volume (units, not dB)</returns>
public static double WeightedVolume(ISoundObj src, double dbSPL, double dbSPLBase)
{
double wv = 0;
for (ushort c = 0; c < src.NumChannels; c++)
{
SingleChannel channel = src.Channel(c);
wv += Loudness.WeightedVolume1(channel, dbSPL, dbSPLBase);
}
src.Reset();
wv = wv / src.NumChannels;
return(wv);
}
/// <summary>
/// Calculate the weighted volume of a sound source.
/// NB: this consumes lots of memory for long sources.
/// </summary>
/// <param name="src"></param>
/// <param name="dbSPL"></param>
/// <returns>Volume (units, not dB)</returns>
public static double WeightedVolume(ISoundObj src, double dbSPL, double dbSPLBase)
{
double wv = 0;
for (ushort c = 0; c < src.NumChannels; c++)
{
SingleChannel channel = src.Channel(c);
wv += Loudness.WeightedVolume1(channel, dbSPL, dbSPLBase);
}
src.Reset();
wv = wv / src.NumChannels;
return wv;
}
public static FilterProfile Profile(ISoundObj impulse, SmoothingType type, double resolution)
{
uint nSR = impulse.SampleRate;
uint nSR2 = nSR / 2;
ushort nChannels = impulse.NumChannels;
for (ushort c = 0; c < nChannels; c++)
{
// Read channel into a buffer
SingleChannel channel = impulse.Channel(c);
SoundBuffer buff = new SoundBuffer(channel);
buff.ReadAll();
// And then double in length to prevent wraparound
buff.PadTo(buff.Count * 2);
// Pad to next higher power of two
buff.PadToPowerOfTwo();
// Read out into array of complex
Complex[][] data = buff.ToComplexArray();
Complex[] cdata = data[0];
// Then we're done with the buffer for this channel
buff = null;
GC.Collect();
// FFT in place
Fourier.FFT(cdata.Length, cdata);
int n = cdata.Length / 2;
// Now we have an array of complex, from 0Hz to Nyquist and back again.
// We really only care about the first half of the cdata buffer, but
// treat it as circular anyway (i.e. wrap around for negative values).
//
// We're only working with magnitudes from here on,
// so we can save some space by computing mags right away and storing them in the
// real part of the complex array; then we can use the imaginary portion for the
// smoothed data.
for (int j = 0; j < cdata.Length; j++)
{
cdata[j].Re = cdata[j].Magnitude;
cdata[j].Im = 0;
}
// Take a rectangular window of width (resolution)*(octave or ERB band)
// Add up all magnitudes falling within this window
//
// Move the window forward by one thingummajig
//double wMid = 0; // center of the window
//double wLen = 0;
}
return new FilterProfile(); // temp
}
private static double[] magbands(ISoundObj impulse, double bins)
{
uint nSR = impulse.SampleRate;
uint nSR2 = nSR / 2;
int nn = (int)bins + 1;
double[] muff = new double[nn];
ushort nChannels = impulse.NumChannels;
for (ushort c = 0; c < nChannels; c++)
{
// Read channel into a buffer
SingleChannel channel = impulse.Channel(c);
SoundBuffer buff = new SoundBuffer(channel);
buff.ReadAll();
// And then double in length to prevent wraparound
buff.PadTo(buff.Count * 2);
// Pad to next higher power of two
buff.PadToPowerOfTwo();
// Read out into array of complex
Complex[][] data = buff.ToComplexArray();
Complex[] cdata = data[0];
// Then we're done with the buffer for this channel
buff = null;
GC.Collect();
// FFT in place
Fourier.FFT(cdata.Length, cdata);
int n = cdata.Length / 2;
// Drop the FFT magnitudes into the 'muff' array
// according to an ERB-based scale (near-logarithmic).
// Then smoothing is easy.
double binw = (nSR2 / (double)n);
int prevbin = 0;
int nbin = 0;
double v = 0;
for (int j = 0; j < n; j++)
{
double f = (double)j * binw; // equiv freq, Hz
int bin = (int)ERB.f2bin(f, nSR, bins); // the bin we drop this sample in
v += cdata[j].Magnitude;
nbin++;
if ((bin > prevbin) || (j == n - 1))
{
muff[prevbin] += (v / nbin);
v = 0;
nbin = 0;
prevbin = bin;
}
}
}
// Now muff is sum(all channels) of average-magnitude-per-bin.
// Divide it all by the number of channels, so our gains are averaged...
for (int j = 0; j < muff.Length; j++)
{
muff[j] = muff[j] / nChannels;
}
return(muff);
}