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);
}
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;
}
static void WriteImpulsePCM(ISoundObj stereoImpulse, string fileNameL, string fileNameR)
{
ISoundObj sb;
WaveWriter writer;
ushort bits = 32;
int nTot = 196607;
int nBef = 98300;
// need padding before the sample.
// window the sample first, then pad it.
// Total length 65536, includes nPad and _peakPosL before impulse
int nPad = nBef - _peakPosL;
int usableLength = nTot - nPad;
int windowCenter = usableLength / 2;
int windowSide = _peakPosL * 2 / 3;
int windowFlat = windowCenter - windowSide;
ISoundObj ch = new SingleChannel(stereoImpulse, 0);
ISoundObj wn = new BlackmanHarris(windowCenter, windowSide, windowFlat);
wn.Input = ch;
sb = new SampleBuffer(wn).PaddedSubset(-nPad, nTot);
writer = new WaveWriter(fileNameL);
writer.Input = sb;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = bits;
writer.SampleRate = _sampleRate;
writer.Raw = true;
writer.Normalization = -6.0;
writer.NormalizationType = NormalizationType.PEAK_DBFS;
writer.Run();
writer.Close();
double g = writer.Gain;
writer = null;
nPad = nBef - _peakPosR;
usableLength = nTot - nPad;
windowCenter = usableLength / 2;
windowSide = _peakPosR * 2 / 3;
windowFlat = windowCenter - windowSide;
ch = new SingleChannel(stereoImpulse, 1);
wn = new BlackmanHarris(windowCenter, windowSide, windowFlat);
wn.Input = ch;
sb = new SampleBuffer(wn).PaddedSubset(-nPad, nTot);
writer = new WaveWriter(fileNameR);
writer.Input = sb;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = bits;
writer.SampleRate = _sampleRate;
writer.Raw = true;
writer.Gain = g;
writer.Run();
writer.Close();
writer = null;
}
static ISoundObj DecodeBFormatUHJ(ISoundObj source)
{
ISoundObj input = source;
uint sr = input.SampleRate;
/*
if (_ambiUseShelf)
{
// Shelf-filters
// boost W at high frequencies, and boost X, Y at low frequencies
FilterProfile lfgXY = new FilterProfile();
lfgXY.Add(new FreqGain(_ambiShelfFreq / 2, 0));
lfgXY.Add(new FreqGain(_ambiShelfFreq * 2, -1.25));
FilterImpulse fiXY = new FilterImpulse(0, lfgXY, FilterInterpolation.COSINE, sr);
FilterProfile lfgW = new FilterProfile();
lfgW.Add(new FreqGain(_ambiShelfFreq / 2, 0));
lfgW.Add(new FreqGain(_ambiShelfFreq * 2, 1.76));
FilterImpulse fiW = new FilterImpulse(0, lfgW, FilterInterpolation.COSINE, sr);
}
if (_ambiUseDistance)
{
// Distance compensation filters
// apply phase shift to X, Y at (very) low frequencies
double fc = MathUtil.FcFromMetres(_ambiDistance);
IIR1 discomp = new IIR1LP(sr, fc, 8192); // tbd: chain this
}
*/
// Transformation filters
//
// Primary reference:
// Gerzon 1985 "Ambisonics in Multichannel Broadcasting and Video"
//
// Coefficients from: http://en.wikipedia.org/wiki/Ambisonic_UHJ_format:
// S = 0.9396926*W + 0.1855740*X
// D = j(-0.3420201*W + 0.5098604*X) + 0.6554516*Y
// Left = (S + D)/2.0
// Right = (S - D)/2.0
// which makes
// Left = (0.092787 + 0.2549302j)X + (0.4698463 - 0.17101005j)W + (0.3277258)Y
// Right= (0.092787 - 0.2549302j)X + (0.4698463 + 0.17101005j)W - (0.3277258)Y
//
// Coefficients from: http://www.york.ac.uk/inst/mustech/3d_audio/ambis2.htm
// Left = (0.0928 + 0.255j)X + (0.4699 - 0.171j)W + (0.3277)Y
// Right= (0.0928 - 0.255j)X + (0.4699 + 0.171j)W - (0.3277)Y
// The Mid-Side versions are simpler
// L+R = (0.0928 + 0.255j)X + (0.4699 - 0.171j)W + (0.3277)Y + ((0.0928 - 0.255j)X + (0.4699 + 0.171j)W - (0.3277)Y)
// = (0.1856)X + (0.9398)W
// L-R = (0.0928 + 0.255j)X + (0.4699 - 0.171j)W + (0.3277)Y - ((0.0928 - 0.255j)X + (0.4699 + 0.171j)W - (0.3277)Y)
// = (0.510j)X + (0.342j)W + (0.6554)Y
// but since we're delaying signal via convolution anyway, not *too* much extra processing to do in LR mode...
// Separate the WXY channels
ISoundObj channelW = new SingleChannel(input, 0);
ISoundObj channelX = new SingleChannel(input, 1, true);
ISoundObj channelY = new SingleChannel(input, 2, true);
// Z not used; height is discarded in UHJ conversion.
// Don't assume it's there; horizontal-only .AMB files won't have a fourth channel
// ISoundObj channelZ = new SingleChannel(input, 3);
// Phase shift j is implemented with Hilbert transforms
// so let's load up some filters, multiply by the appropriate coefficients.
int len = 8191;
PhaseMultiplier xl = new PhaseMultiplier(new Complex(0.0927870, 0.25493020), len, sr);
PhaseMultiplier wl = new PhaseMultiplier(new Complex(0.4698463, -0.17101005), len, sr);
PhaseMultiplier yl = new PhaseMultiplier(new Complex(0.3277258, 0.00000000), len, sr);
PhaseMultiplier xr = new PhaseMultiplier(new Complex(0.0927870, -0.25493020), len, sr);
PhaseMultiplier wr = new PhaseMultiplier(new Complex(0.4698463, 0.17101005), len, sr);
PhaseMultiplier yr = new PhaseMultiplier(new Complex(-0.3277258, 0.00000000), len, sr);
// The convolvers to filter
FastConvolver cwl = new FastConvolver(channelW, wl);
FastConvolver cxl = new FastConvolver(channelX, xl);
FastConvolver cyl = new FastConvolver(channelY, yl);
FastConvolver cwr = new FastConvolver(channelW, wr);
FastConvolver cxr = new FastConvolver(channelX, xr);
FastConvolver cyr = new FastConvolver(channelY, yr);
// Sum to get the final output of these things:
Mixer mixerL = new Mixer();
mixerL.Add(cwl, 1.0);
mixerL.Add(cxl, 1.0);
mixerL.Add(cyl, 1.0);
Mixer mixerR = new Mixer();
mixerR.Add(cwr, 1.0);
mixerR.Add(cxr, 1.0);
mixerR.Add(cyr, 1.0);
// output in stereo
ChannelSplicer uhj = new ChannelSplicer();
uhj.Add(mixerL);
uhj.Add(mixerR);
return uhj;
}
static ISoundObj DecodeBFormatCrossed(ISoundObj source)
{
// Stereo decode:
// A shuffle of the X (front-back) and Y (left-right) channels
// with some W mixed, so the effective mics become (hyper)cardioid instead of figure-8.
// If _ambiCardioid=0, this is the same as Blumlein.
// If _ambiCardioid=1, this is cardioid
// Of any value between, for hypercardioid patterns.
// Separate the WXY channels
ISoundObj channelW = new SingleChannel(source, 0);
ISoundObj channelX = new SingleChannel(source, 1, true);
ISoundObj channelY = new SingleChannel(source, 2, true);
// The _ambiMicAngle says angle between the virtual microphones (degrees)
// e.g. if this is 100
// Left and right are each 50 degrees from forward.
// These are implemented by mixing appropriate amounts of X and Y;
// X * cos(angle/2)
// Y * sin(angle/2)
//
// For default mic angle of 90 degrees, mulX=mulY=sqrt(2)/2
//
double mulX = Math.Cos(MathUtil.Radians(_ambiMicAngle / 2));
double mulY = Math.Sin(MathUtil.Radians(_ambiMicAngle / 2));
// Mix back together, adding appropriate amounts of W.
// The W channel gain is conventionally sqrt(2)/2 relative to X and Y,
Mixer mixerL = new Mixer();
mixerL.Add(channelW, _ambiCardioid * MathUtil.INVSQRT2);
mixerL.Add(channelX, mulX);
mixerL.Add(channelY, mulY);
Mixer mixerR = new Mixer();
mixerR.Add(channelW, _ambiCardioid * MathUtil.INVSQRT2);
mixerR.Add(channelX, mulX);
mixerR.Add(channelY, -mulY);
// output in stereo
ChannelSplicer stereo = new ChannelSplicer();
stereo.Add(mixerL);
stereo.Add(mixerR);
return stereo;
}
static ISoundObj RotateBFormat(ISoundObj source)
{
// Rotate a B-Format source
ISoundObj channelW = new SingleChannel(source, 0);
ISoundObj channelX = new SingleChannel(source, 1, true);
ISoundObj channelY = new SingleChannel(source, 2, true);
ISoundObj channelZ = new SingleChannel(source, 3, true);
double rx = MathUtil.Radians(_ambiRotateX);
double ry = MathUtil.Radians(_ambiRotateY);
double rz = MathUtil.Radians(_ambiRotateZ);
// Mixer W = new Mixer();
Mixer X = new Mixer();
Mixer Y = new Mixer();
Mixer Z = new Mixer();
// W is unchanged (omni)
// W.Add(channelW, 1.0);
// http://www.muse.demon.co.uk/fmhrotat.html
// tilt
// x1 = x * 1 + y * 0 + z * 0
// y1 = x * 0 + y * Cos(rx) - z * Sin(rx)
// z1 = x * 0 + y * Sin(rx) + z * Cos(rx)
// tumble
// x2 = x * Cos(ry) + y * 0 - z * Sin(ry)
// y2 = x * 0 + y * 1 + z * 0
// z2 = x * Sin(ry) + y * 0 + z * Cos(ry)
// rotate
// x3 = x * Cos(rz) - y * Sin(rz) + z * 0
// y3 = x * Sin(rz) + y * Cos(rz) + z * 0
// z3 = x * 0 + y * 0 + z * 1
// (read that downwards to get:)
X.Add(channelX, Math.Cos(ry) * Math.Cos(rz));
X.Add(channelY, -Math.Sin(rz));
X.Add(channelZ, -Math.Sin(ry));
Y.Add(channelX, Math.Sin(rz));
Y.Add(channelY, Math.Cos(rx) * Math.Cos(rz));
Y.Add(channelZ, -Math.Sin(rx));
Z.Add(channelX, Math.Sin(ry));
Z.Add(channelY, Math.Sin(rz));
Z.Add(channelZ, Math.Cos(rz) * Math.Cos(ry));
ChannelSplicer ret = new ChannelSplicer();
ret.Add(channelW);
ret.Add(X);
ret.Add(Y);
ret.Add(Z);
return ret;
}
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);
}
