/// <summary>
/// Calculate the weighted volume of a *single channel* 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 WeightedVolume1(ISoundObj src, double dbSPL, double dbSPLBase)
{
if (src.NumChannels != 1)
{
throw new ArgumentException("Requires single-channel");
}
// Read channel into a buffer
SoundBuffer buff = new SoundBuffer(src);
buff.ReadAll();
// And then double in length to prevent wraparound
buff.PadTo(buff.Count * 2);
// Pad to next higher power of two
buff.PadToPowerOfTwo();
int n = buff.Count;
double wvImpulse = WeightedVolume2(buff, dbSPL, dbSPLBase);
// compare with a Dirac pulse the same length
CallbackSource dirac = new CallbackSource(1, src.SampleRate, delegate(long j)
{
if (j >= n)
{
return(null);
}
double v = 0;
if (j == n / 2)
{
v = 1;
}
return(new Sample(v));
});
buff = new SoundBuffer(dirac);
buff.ReadAll();
double wvDirac = WeightedVolume2(buff, dbSPL, dbSPLBase);
buff = null;
GC.Collect();
return(wvImpulse / wvDirac);
}
/// <summary>
/// Calculate the weighted volume of a *single channel* 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 WeightedVolume1(ISoundObj src, double dbSPL, double dbSPLBase)
{
if (src.NumChannels != 1)
{
throw new ArgumentException("Requires single-channel");
}
// Read channel into a buffer
SoundBuffer buff = new SoundBuffer(src);
buff.ReadAll();
// And then double in length to prevent wraparound
buff.PadTo(buff.Count * 2);
// Pad to next higher power of two
buff.PadToPowerOfTwo();
int n = buff.Count;
double wvImpulse = WeightedVolume2(buff, dbSPL, dbSPLBase);
// compare with a Dirac pulse the same length
CallbackSource dirac = new CallbackSource(1, src.SampleRate, delegate(long j)
{
if (j >= n)
{
return null;
}
double v = 0;
if (j == n / 2)
{
v = 1;
}
return new Sample(v);
});
buff = new SoundBuffer(dirac);
buff.ReadAll();
double wvDirac = WeightedVolume2(buff, dbSPL, dbSPLBase);
buff = null;
GC.Collect();
return wvImpulse / wvDirac;
}
static ISoundObj SlidingLowPass(ISoundObj impulse, int peakPos)
{
// Filter the impulse response with a sliding lowpass filter
// - Take the first (peakpos) samples unchanged
// - Take the next 2.5ms (~110 samples @ 44100) unchanged
// - Fade to a lowpass-filtered version
uint sampleRate = impulse.SampleRate;
int startpos1 = (int)(sampleRate * 0.0025); // 2.5mS, 110 samples
int startpos2 = (int)(sampleRate * 0.0050);
int startpos3 = (int)(sampleRate * 0.0100);
int startpos4 = (int)(sampleRate * 0.0200);
int startpos5 = (int)(sampleRate * 0.0400);
List<IEnumerator<ISample>> filters = null;
int nFilter = 0;
int nFilters = 0;
int x = 0;
int f0len = 0;
ISoundObj filteredImpulse = new CallbackSource(1, sampleRate, delegate(long j)
{
if(j==0)
{
// initialize the list of filters
impulse.Reset();
filters = new List<IEnumerator<ISample>>(6);
ISoundObj f0 = LowPassFiltered(impulse, 20, 0);
f0len = f0.Iterations;
filters.Add(f0.Samples); // unfiltered
filters.Add(LowPassFiltered(impulse, 640, -10).Samples); // LPF from 640Hz, kick in at +110
filters.Add(LowPassFiltered(impulse, 320, -20).Samples); // LPF from 320Hz, kick in at +220 (etc)
filters.Add(LowPassFiltered(impulse, 160, -30).Samples); // +440
filters.Add(LowPassFiltered(impulse, 80, -40).Samples); // +880
filters.Add(LowPassFiltered(impulse, 40, -50).Samples); // +1760 = 40ms
nFilter = 0;
nFilters = filters.Count;
}
// Move the filters along a notch.
// (or, right at the beginning, move all the filters along by a lot, compensating for their center)
// For perf, we don't need to keep enumerating the filters we've finished using.
bool moreData = true;
int nSkip = 1;
if (j == 0)
{
nSkip = (f0len/2)+1;
}
for (int skip = 0; skip < nSkip; skip++)
{
int nStart = (nFilter == 0) ? 0 : nFilter - 1; // Keep moving even the old filters along so mixing works later
for (int f = nStart; f < nFilters; f++)
{
moreData &= filters[f].MoveNext();
}
}
if (!moreData) return null;
// Decide which filter we'll use
x++;
if (j == peakPos + startpos1) { nFilter = 1; x = 0; }
if (j == peakPos + startpos2) { nFilter = 2; x = 0; }
if (j == peakPos + startpos3) { nFilter = 3; x = 0; }
if (j == peakPos + startpos4) { nFilter = 4; x = 0; }
if (j == peakPos + startpos5) { nFilter = 5; x = 0; }
// Pick the sample from the current filter
ISample s = filters[nFilter].Current;
// For a short region after switch-over, mix slowly with the sample from the previous filter
int w = 20;
if (x < w && nFilter > 0)
{
ISample sPrev = filters[nFilter-1].Current;
for (int c = 0; c < s.NumChannels; c++)
{
s[c] = ((double)x/w)*s[c] + ((double)(w-x)/w)*sPrev[c];
}
}
return s;
});
return filteredImpulse;
}
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 ISoundObj DecodeBFormatBinaural(ISoundObj source)
{
throw new NotImplementedException();
ISoundObj input = source;
uint sr = input.SampleRate;
// Convolve the BFormat data with the matrix filter
if (!String.IsNullOrEmpty(_bformatFilter))
{
string ignore;
WaveReader rdr = GetAppropriateImpulseReader(_bformatFilter, out ignore);
FastConvolver ambiConvolver = new FastConvolver(source, rdr);
input = ambiConvolver;
}
// Cardioid directed at four (or six) virtual loudspeakers
IEnumerator<ISample> src = input.Samples;
CallbackSource bin = new CallbackSource(2, sr, delegate(long j)
{
if (src.MoveNext())
{
ISample s = src.Current;
double w = s[0];
double x = s[1];
double y = s[2];
double z = s[3];
double wFactor = -0.5;
double left = x + y + z + (wFactor * w);
double right = x - y + z + (wFactor * w);
ISample sample = new Sample2(left, right);
return sample;
}
return null;
});
return bin;
}