/// <summary>
/// Noise synthesis mode
/// </summary>
/// <param name="d">Image</param>
/// <param name="Xsize">Specifies the desired width of the spectrogram</param>
/// <param name="bands">Specifies the desired height of the spectrogram (bands)</param>
/// <param name="samplecount">Number of samples</param>
/// <param name="samplerate">Sample rate</param>
/// <param name="basefreq">Base frequency in Hertz</param>
/// <param name="pixpersec">Time resolution in Pixels Per Second</param>
/// <param name="bpo">Frequency resolution in Bands Per Octave</param>
/// <returns></returns>
public static double[] SynthesizeNoise(ref double[][] d, ref int Xsize, ref int bands, ref int samplecount, ref int samplerate, ref double basefreq, ref double pixpersec, ref double bpo)
{
double[] s; // final signal
double coef;
double[] noise; // filtered looped noise
double loop_size_sec = LOOP_SIZE_SEC; // size of the filter bank loop, in seconds. Later to be taken from user input
int loop_size = 0; // size of the filter bank loop, in samples. Deduced from loop_size_sec
int loop_size_min = 0; // minimum required size for the filter bank loop, in samples. Calculated from the longest windowed sinc's length
double[] pink_noise; // original pink noise (in the frequency domain)
double mag; // parameters for the creation of pink_noise's samples
double phase;
double[] envelope; // interpolated envelope
double[] lut; // Blackman Sqaure look-up table
double[] freq; // frequency look-up table
double maxfreq; // central frequency of the last band
int i = 0; // general purpose iterator
int ib = 0; // bands iterator
int il = 0; // loop iterator
int Fa = 0; // Fa is the index of the band's start in the frequency domain
int Fd = 0; // Fd is the index of the band's end in the frequency domain
double La; // La is the log2 of the frequency of Fa
double Ld; // Ld is the log2 of the frequency of Fd
double Li; // Li is the iterative frequency between La and Ld defined logarithmically
freq = GlobalMembersDsp.FrequencyArray(basefreq, bands, bpo);
if (LOGBASE == 1.0)
{
maxfreq = bpo; // in linear mode we use bpo to store the maxfreq since we couldn't deduce maxfreq otherwise
}
else
{
maxfreq = basefreq * Math.Pow(LOGBASE, ((double)(bands - 1) / bpo));
}
clockA = GlobalMembersUtil.GetTime();
samplecount = (int)GlobalMembersUtil.RoundOff(Xsize / pixpersec); // calculation of the length of the final signal
Console.Write("Sound duration : {0:f3} s\n", (double)samplecount / samplerate);
// allocation of the final signal
s = new double[samplecount];
// allocation of the interpolated envelope
envelope = new double[samplecount];
//********Loop size calculation********
loop_size = (int)loop_size_sec * samplerate;
if (LOGBASE == 1.0)
{
loop_size_min = (int)GlobalMembersUtil.RoundOff(4.0 * 5.0 / freq[1] - freq[0]); // linear mode
}
else
{
loop_size_min = (int)GlobalMembersUtil.RoundOff(2.0 * 5.0 / ((freq[0] * Math.Pow(2.0, -1.0 / (bpo))) * (1.0 - Math.Pow(2.0, -1.0 / bpo)))); // this is the estimate of how many samples the longest FIR will take up in the time domain
}
if (loop_size_min > loop_size)
{
loop_size = loop_size_min;
}
loop_size = GlobalMembersUtil.NextPrime(loop_size); // enlarge the loop_size to the next multiple of short primes in order to make IFFTs faster
//--------Loop size calculation--------
//********Pink noise generation********
pink_noise = new double[loop_size];
for (i = 1; i < (loop_size + 1) >> 1; i++)
{
mag = Math.Pow((double)i, 0.5 - 0.5 * LOGBASE); // FIXME something's not necessarily right with that formula
phase = GlobalMembersUtil.DoubleRandom() * PI; // random phase between -pi and +pi
pink_noise[i] = mag * Math.Cos(phase); // real part
pink_noise[loop_size - i] = mag * Math.Sin(phase); // imaginary part
}
//--------Pink noise generation--------
// allocate noise
noise = new double[loop_size];
lut = GlobalMembersDsp.BlackmanSquareLookupTable(ref BMSQ_LUT_SIZE); // Blackman Square look-up table initalisation
for (ib = 0; ib < bands; ib++)
{
Console.Write("{0,4:D}/{1:D}\r", ib + 1, bands);
// reset filtered noise
for (i = 0; i < loop_size; i++)
{
noise[i] = 0; // reset sband
}
//********Filtering********
Fa = (int)GlobalMembersUtil.RoundOff(GlobalMembersDsp.LogPositionToFrequency((double)(ib - 1) / (double)(bands - 1), basefreq, maxfreq) * loop_size);
Fd = (int)GlobalMembersUtil.RoundOff(GlobalMembersDsp.LogPositionToFrequency((double)(ib + 1) / (double)(bands - 1), basefreq, maxfreq) * loop_size);
La = GlobalMembersDsp.FrequencyToLogPosition((double)Fa / (double)loop_size, basefreq, maxfreq);
Ld = GlobalMembersDsp.FrequencyToLogPosition((double)Fd / (double)loop_size, basefreq, maxfreq);
if (Fd > loop_size / 2)
{
Fd = loop_size / 2; // stop reading if reaching the Nyquist frequency
}
if (Fa < 1)
{
Fa = 1;
}
Console.Write("{0,4:D}/{1:D} {2:f2} Hz - {3:f2} Hz\r", ib + 1, bands, (double)Fa * samplerate / loop_size, (double)Fd * samplerate / loop_size);
for (i = Fa; i < Fd; i++)
{
Li = GlobalMembersDsp.FrequencyToLogPosition((double)i / (double)loop_size, basefreq, maxfreq); // calculation of the logarithmic position
Li = (Li - La) / (Ld - La);
coef = 0.5 - 0.5 * Math.Cos(2.0 * PI * Li); // Hann function
noise[i + 1] = pink_noise[i + 1] * coef;
noise[loop_size - 1 - i] = pink_noise[loop_size - 1 - i] * coef;
}
//--------Filtering--------
GlobalMembersDsp.FFT(ref noise, ref noise, loop_size, FFTMethod.IDFT); // IFFT of the filtered noise
// blank the envelope
for (i = 0; i < samplecount; i++)
{
envelope[i] = 0; // reset sband
}
GlobalMembersDsp.BlackmanSquareInterpolation(ref d[bands - ib - 1], ref envelope, ref Xsize, ref samplecount, ref lut, BMSQ_LUT_SIZE); // interpolation of the envelope
il = 0;
for (i = 0; i < samplecount; i++)
{
s[i] += envelope[i] * noise[il]; // modulation
il++; // increment loop iterator
if (il == loop_size) // if the array iterator has reached the end of the array, it's reset
{
il = 0;
}
}
}
Console.Write("\n");
GlobalMembersDsp.Normalize(ref s, ref samplecount, 1.0);
return(s);
}
/// <summary>
/// Sine synthesis mode
/// </summary>
/// <param name="d">Image (d is the original image - spectrogram)</param>
/// <param name="Xsize">Specifies the desired width of the spectrogram</param>
/// <param name="bands">Specifies the desired height of the spectrogram (total count of bands)</param>
/// <param name="samplecount">Number of samples (samplecount is the output sound's length)</param>
/// <param name="samplerate">Sample rate</param>
/// <param name="basefreq">Minimum frequency in Hertz</param>
/// <param name="pixpersec">Time resolution in Pixels Per Second</param>
/// <param name="bpo">Frequency resolution in Bands Per Octave</param>
/// <returns></returns>
public static double[] SynthesizeSine(ref double[][] d, ref int Xsize, ref int bands, ref int samplecount, ref int samplerate, ref double basefreq, ref double pixpersec, ref double bpo)
{
double[] s; // s is the output sound
double[] freq; // freq is the band's central frequency
double[] filter;
double[] sband; // sband is the band's envelope upsampled and shifted up in frequency
int sbsize = 0; // sbsize is the length of sband
double[] sine = new double[4]; // sine is the random sine look-up table
double rphase; // rphase is the band's sine's random phase
int i = 0; // i is a general purpose iterator
int ib = 0; // ib is the band iterator
int Fc = 0; // Fc is the index of the band's centre in the frequency domain on the new signal
int Bc = 0; // Bc is the index of the band's centre in the frequency domain on sband (its imaginary match being sbsize-Bc)
int Mh = 0; // Mh is the length of the real or imaginary part of the envelope's FFT, DC element included and Nyquist element excluded
int Mn = 0; // Mn is the length of the real or imaginary part of the sound's FFT, DC element included and Nyquist element excluded
freq = GlobalMembersDsp.FrequencyArray(basefreq, bands, bpo);
clockA = GlobalMembersUtil.GetTime();
sbsize = GlobalMembersUtil.NextPrime(Xsize * 2); // In Circular mode keep it to sbsize = Xsize * 2;
samplecount = (int)GlobalMembersUtil.RoundOff(Xsize / pixpersec);
Console.Write("Sound duration : {0:f3} s\n", (double)samplecount / samplerate);
samplecount = (int)GlobalMembersUtil.RoundOff(0.5 * sbsize / pixpersec); // Do not change this value as it would stretch envelopes
s = new double[samplecount];
// allocation of the shifted band
sband = new double[sbsize];
// Bc is the index of the band's centre in the frequency domain on sband (its imaginary match being sbsize-Bc)
Bc = (int)GlobalMembersUtil.RoundOff(0.25 * (double)sbsize);
// Mh is the length of the real or imaginary part of the envelope's FFT, DC element included and Nyquist element excluded
Mh = (sbsize + 1) >> 1;
// Mn is the length of the real or imaginary part of the sound's FFT, DC element included and Nyquist element excluded
Mn = (samplecount + 1) >> 1;
filter = GlobalMembersDsp.WindowedSinc_max(Mh, 1.0 / GlobalMembersArss.TRANSITION_BW_SYNT); // generation of the frequency-domain filter
for (ib = 0; ib < bands; ib++)
{
// reset sband
for (i = 0; i < sbsize; i++)
{
sband[i] = 0; // reset sband
}
//********Frequency shifting********
rphase = GlobalMembersUtil.DoubleRandom() * PI; // random phase between -pi and +pi
for (i = 0; i < 4; i++)
{
// generating the random sine LUT
sine[i] = Math.Cos(i * 2.0 * PI * 0.25 + rphase);
}
for (i = 0; i < Xsize; i++) // envelope sampling rate * 2 and frequency shifting by 0.25
{
if ((i & 1) == 0)
{
sband[i << 1] = d[bands - ib - 1][i] * sine[0];
sband[(i << 1) + 1] = d[bands - ib - 1][i] * sine[1];
}
else
{
sband[i << 1] = d[bands - ib - 1][i] * sine[2];
sband[(i << 1) + 1] = d[bands - ib - 1][i] * sine[3];
}
}
//--------Frequency shifting--------
GlobalMembersDsp.FFT(ref sband, ref sband, sbsize, FFTMethod.DFT); // FFT of the envelope
// Fc is the index of the band's centre in the frequency domain on the new signal
Fc = (int)GlobalMembersUtil.RoundOff(freq[ib] * samplecount); // band's centre index (envelope's DC element)
Console.Write("{0,4:D}/{1:D} {2:f2} Hz\r", ib + 1, bands, (double)Fc * samplerate / samplecount);
//********Write FFT********
for (i = 1; i < Mh; i++)
{
if (Fc - Bc + i > 0 && Fc - Bc + i < Mn) // if we're between frequencies 0 and 0.5 of the new signal and that we're not at Fc
{
s[i + Fc - Bc] += sband[i] * filter[i]; // Real part
s[samplecount - (i + Fc - Bc)] += sband[sbsize - i] * filter[i]; // Imaginary part
}
}
//--------Write FFT--------
}
Console.Write("\n");
GlobalMembersDsp.FFT(ref s, ref s, samplecount, FFTMethod.IDFT); // IFFT of the final sound
samplecount = (int)GlobalMembersUtil.RoundOff(Xsize / pixpersec); // chopping tails by ignoring them
GlobalMembersDsp.Normalize(ref s, ref samplecount, 1.0);
return(s);
}