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 WeightedVolume2(SoundBuffer src, double dbSPL, double dbSPLBase)
{
double v = 0;
uint sr = src.SampleRate;
// Read buffer into array of complex
Complex[][] data = src.ToComplexArray();
// We only have a single channel
Complex[] cdata = data[0];
// FFT in place
Fourier.FFT(cdata.Length, cdata);
// Calculate magnitude, weighted by 80-phon loudness, for each loudness band.
// These are the ISO measured points:
FilterProfile lfg;
if (dbSPLBase == 0)
{
lfg = SPL(dbSPL);
}
else
{
lfg = DifferentialSPL(dbSPL, dbSPLBase);
}
// lfg.Add(new FreqGain(sr / 2, lfg[lfg.Count - 1].Gain));
// Cover the ISO measured range (only...)
int nStart = (int)(lfg[0].Freq * (long)cdata.Length / sr);
int nEnd = (int)(lfg[lfg.Count - 1].Freq * (long)cdata.Length / sr);
// Just use linear interpolation (on a dB scale; linear freq scale) of gain between each measured point
int nfg = 0;
int startp = nStart;
int endp = (int)(lfg[nfg + 1].Freq * (long)cdata.Length / sr); // endpoint of this band
double dB1 = lfg[nfg].Gain; // SPL of the ISO223 curve at this freq
double dB2 = lfg[nfg+1].Gain; // ...and the next point
double vThisBand = 0;
int nThisBand = 0;
for (int j = nStart; j < nEnd; j++)
{
if (j > endp)
{
if (nThisBand > 0) v += Math.Sqrt(vThisBand / nThisBand); // RMS
while (j >= endp)
{
nfg++;
startp = j;
endp = (int)(lfg[nfg + 1].Freq * (long)cdata.Length / sr);
dB1 = lfg[nfg].Gain;
dB2 = lfg[nfg + 1].Gain;
}
vThisBand = 0;
nThisBand = 0;
}
Complex c = cdata[j];
double dbHere = dB1 + ((dB2 - dB1) * (double)(j - startp) / (double)(endp - startp));
vThisBand += (c.Re * c.Re) / MathUtil.gain(dbHere);
nThisBand++;
}
if(nThisBand>0) v += Math.Sqrt(vThisBand / nThisBand);
return v;
}
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 WeightedVolume2(SoundBuffer src, double dbSPL, double dbSPLBase)
{
double v = 0;
uint sr = src.SampleRate;
// Read buffer into array of complex
Complex[][] data = src.ToComplexArray();
// We only have a single channel
Complex[] cdata = data[0];
// FFT in place
Fourier.FFT(cdata.Length, cdata);
// Calculate magnitude, weighted by 80-phon loudness, for each loudness band.
// These are the ISO measured points:
FilterProfile lfg;
if (dbSPLBase == 0)
{
lfg = SPL(dbSPL);
}
else
{
lfg = DifferentialSPL(dbSPL, dbSPLBase);
}
// lfg.Add(new FreqGain(sr / 2, lfg[lfg.Count - 1].Gain));
// Cover the ISO measured range (only...)
int nStart = (int)(lfg[0].Freq * (long)cdata.Length / sr);
int nEnd = (int)(lfg[lfg.Count - 1].Freq * (long)cdata.Length / sr);
// Just use linear interpolation (on a dB scale; linear freq scale) of gain between each measured point
int nfg = 0;
int startp = nStart;
int endp = (int)(lfg[nfg + 1].Freq * (long)cdata.Length / sr); // endpoint of this band
double dB1 = lfg[nfg].Gain; // SPL of the ISO223 curve at this freq
double dB2 = lfg[nfg + 1].Gain; // ...and the next point
double vThisBand = 0;
int nThisBand = 0;
for (int j = nStart; j < nEnd; j++)
{
if (j > endp)
{
if (nThisBand > 0)
{
v += Math.Sqrt(vThisBand / nThisBand); // RMS
}
while (j >= endp)
{
nfg++;
startp = j;
endp = (int)(lfg[nfg + 1].Freq * (long)cdata.Length / sr);
dB1 = lfg[nfg].Gain;
dB2 = lfg[nfg + 1].Gain;
}
vThisBand = 0;
nThisBand = 0;
}
Complex c = cdata[j];
double dbHere = dB1 + ((dB2 - dB1) * (double)(j - startp) / (double)(endp - startp));
vThisBand += (c.Re * c.Re) / MathUtil.gain(dbHere);
nThisBand++;
}
if (nThisBand > 0)
{
v += Math.Sqrt(vThisBand / nThisBand);
}
return(v);
}
static SoundObj Deconvolve(string infile, out Complex[] impulseFFT, out int peakpos)
{
WaveReader reader = new WaveReader(infile);
ushort nChannels = reader.NumChannels;
uint sampleRate = reader.SampleRate;
CallbackSource cs;
SingleChannel[] channels = new SingleChannel[2];
Complex[][][] data = new Complex[2][][];
double[] stdDev = new double[2];
double[] maxLogs = new double[2];
double[] maxs = new double[2];
double[] avgLog = new double[2];
Complex[] swp_data = null;
Complex[] mea_data = null;
if (_fmax == int.MaxValue)
{
_fmax = (int)sampleRate / 2;
}
bool isEstimatedSweepRange = false;
if (nChannels != 2)
{
throw new Exception("Input must have two channels.");
}
peakpos = 0;
int cSwp = 0;
int cMea = 1;
int L = 0;
int Nh = 0;
double mx;
double max = 0;
double maxLog = 0;
double stdev = 0;
double avg = 0;
for (int iteration = 1; iteration <= 2; iteration++)
{
// Read and FFT all the data
// one channel at a time
for (ushort c = 0; c < 2; c++)
{
SingleChannel channel = reader.Channel(c);
Complex[] cdata;
SoundBuffer buff = new SoundBuffer(channel);
buff.ReadAll();
if (iteration==2 && _split)
{
// Split up the input file
string infile2 = Path.ChangeExtension(Path.GetFileName(infile), ".PCM");
if (c == cSwp)
{
WaveWriter wri = new WaveWriter("refchannel_" + infile2, 1, channel.SampleRate, 32, DitherType.NONE, WaveFormat.IEEE_FLOAT);
wri.Input = buff;
wri.Run(); wri.Close();
}
if (c == cMea)
{
WaveWriter wri = new WaveWriter("sweep_" + infile2, 1, channel.SampleRate, 32, DitherType.NONE, WaveFormat.IEEE_FLOAT);
wri.Input = buff;
wri.Run(); wri.Close();
}
}
// 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
data[c] = buff.ToComplexArray();
// Then we're done with the buffer for this channel
buff = null;
GC.Collect();
cdata = data[c][0];
if (iteration==2 && c==cSwp && _power > 0)
{
// Deconvolve against a power of the sweep,
// for distortion measurement of harmonic _power
Complex p = new Complex((double)_power,0);
for (int j = 0; j < cdata.Length; j++)
{
cdata[j].Pow(p);
}
}
// FFT in place
Fourier.FFT(cdata.Length, cdata);
if (false && iteration==1)
{
// write the fft magnitudes to disk
cs = new CallbackSource(1, sampleRate, delegate(long j)
{
if (j >= cdata.Length)
{
return null;
}
Complex si = cdata[j];
Sample s = new Sample(1);
double f = (double)j * sampleRate / cdata.Length;
s[0] = mag(sampleRate, f, si.Magnitude);
return s;
});
// cs.SampleRate = sampleRate;
// cs.NumChannels = 1;
WaveWriter writer = new WaveWriter("fft_" + c + "_" + infile);
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Input = cs;
writer.Normalization = -3;
writer.Run();
writer.Close();
}
// Take a slice of the FFT, std dev of log(|fft|),
// the lower value should be the sweep
int n3 = cdata.Length / 4;
int n1 = n3 / 2;
int n2 = n1 + n3;
get_stddev(sampleRate, n1, n2, cdata, out max, out maxLog, out stdev, out avg);
maxs[c] = max;
maxLogs[c] = maxLog;
stdDev[c] = stdev;
avgLog[c] = avg;
// Trace.WriteLine("Channel {0} standard deviation {1}", c, stdDev[c]);
}
GC.Collect();
if (iteration == 1)
{
if (stdDev[1] < stdDev[0])
{
if (_refchannel == -1)
{
cSwp = 1;
cMea = 0;
stderr.WriteLine(" Right channel seems to be the sweep");
}
else
{
stderr.WriteLine(" Right channel seems to be the sweep");
stderr.WriteLine(" But you said use refchannel {0}, so using that.", _refchannel);
cSwp = _refchannel;
cMea = (_nInFiles - 1) - _refchannel;
}
}
else
{
if (_refchannel == -1)
{
stderr.WriteLine(" Left channel seems to be the sweep");
}
else
{
stderr.WriteLine(" Left channel seems to be the sweep");
stderr.WriteLine(" But you said use refchannel {0}, so using that.", _refchannel);
cSwp = _refchannel;
cMea = (_nInFiles - 1) - _refchannel;
}
}
}
swp_data = data[cSwp][0];
mea_data = data[cMea][0];
L = swp_data.Length;
Nh = L / 2;
// stderr.WriteLine("avgLog=" + avgLog[cSwp] + " maxLog=" + maxLogs[cSwp]);
double hz1 = L / sampleRate;
if (false && _fmin == 0)
{
isEstimatedSweepRange = true;
// Working back from 100Hz,
// Look for the first range where average of a 1Hz window
// is less than 0.7* average for the sweep itself
int kmin = (int)(hz1 * 100);
_fmin = 100;
while (kmin > 0)
{
get_stddev(sampleRate, kmin, (int)(kmin + hz1), swp_data, out max, out maxLog, out stdev, out avg);
if (avg < avgLog[cSwp] * 0.5)
{
break;
}
kmin -= (int)hz1;
_fmin--;
}
// stderr.WriteLine("avg/2: kmin=" + kmin + ", _fmin=" + _fmin);
}
if (false && _fmax == sampleRate / 2)
{
isEstimatedSweepRange = true;
// Working forward from (say) 15kHz,
// Look for the first range where average of a 100Hz window
// is less than 0.7* average for the sweep itself
int kmax = (int)(hz1 * 10000);
_fmax = 10000;
get_stddev(sampleRate, kmax, (int)(kmax + 100 * hz1), swp_data, out max, out maxLog, out stdev, out avg);
double stdTest = stdev;
while (kmax < L / 2)
{
get_stddev(sampleRate, kmax, (int)(kmax + 100 * hz1), swp_data, out max, out maxLog, out stdev, out avg);
if (avg < avgLog[cSwp] * 0.5)
{
break;
}
kmax += (int)(100 * hz1);
_fmax += 100;
}
// stderr.WriteLine("StdDev Peak: kmax=" + kmax + ", _fmax=" + _fmax + ", sdev=" + stdev + " ref " + stdTest + " (1Hz=" + hz1 + "), avgLog=" + avg);
}
if (!_noSubsonicFilter)
{
// Filter LF from the measurement data
// to avoid spurious stuff below 15Hz
int s1 = (int)(7 * hz1);
int s2 = (int)(15 * hz1);
for (int j = 0; j < s1; j++)
{
mea_data[j].Set(0, 0);
mea_data[swp_data.Length - j - 1].Set(0, 0);
}
for (int j = s1; j < s2; j++)
{
double n = (double)(j - s1) / (s2 - s1);
double m = 0.5 * (1 + Math.Cos(Math.PI * (1 + n)));
mea_data[j].mul(m);
mea_data[swp_data.Length - j - 1].mul(m);
}
}
// Divide in complex domain
for (int j = 0; j < swp_data.Length; j++)
{
swp_data[j].idiv(mea_data[j]);
// Make a copy in mea_data, we'll need it later
mea_data[j] = swp_data[j];
}
// IFFT to get the impulse response
Fourier.IFFT(swp_data.Length, swp_data);
// Scan the imp to find maximum
mx = 0;
int mp = 0;
for (int j = 0; j < sampleRate; j++)
{
Complex si = swp_data[j];
double mg = Math.Abs(si.Magnitude);
if (mg > mx) { mx = mg; mp = j; }
}
// Look one sample backwards from max position
// to find the likely "pulse peak" if within 30% of max
peakpos = mp;
if (mp>0 && swp_data[mp - 1].Magnitude / mx > 0.7)
{
peakpos = mp - 1;
}
}
// stderr.WriteLine(" {0} range {1}Hz to {2}Hz", isEstimatedSweepRange ? "Estimated sweep" : "Sweep", _fmin, _fmax);
if (_fmaxSpecified && _fminSpecified)
{
HackSweep(swp_data, mea_data, peakpos, L, sampleRate);
}
else
{
Fourier.FFT(swp_data.Length, swp_data);
}
// Window the extremities of the whole response, finally?
if (true)
{
// Make a copy in mea_data, we'll need it later
for (int j = 0; j < swp_data.Length; j++)
{
mea_data[j] = swp_data[j];
}
// Return FFT of impulse
impulseFFT = mea_data;
}
// IFFT to get the impulse response
Fourier.IFFT(swp_data.Length, swp_data);
// Scan the imp to find maximum
mx = 0;
for (int j = 0; j < sampleRate; j++)
{
Complex si = swp_data[j];
double mg = Math.Abs(si.Magnitude);
if (mg > mx) mx = mg;
}
if (_noNorm)
{
mx = 1.0;
}
// Yield the first half (normalized) as result
cs = new CallbackSource(1, sampleRate, delegate(long j)
{
if (j > (_returnAll ? L-1 : L / 2))
{
return null;
}
Complex si = swp_data[j];
Sample s = new Sample(si.Re / mx);
return s;
});
cs.SampleRate = sampleRate;
cs.NumChannels = 1;
return cs;
}
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);
}
