static bool ExecDRC(string drcfile, string target, string outfile, string infileL, string infileR, int peakPosL, int peakPosR, string stereoImpulseFile)
{
GC.Collect();
bool ok = false;
string args;
FastConvolver conv;
WaveWriter wri;
if (!File.Exists(drcfile))
{
stderr.WriteLine();
stderr.WriteLine("{0} not found.", drcfile);
return ok;
}
string tmpL;
string tmpR;
tmpL = Path.GetFileNameWithoutExtension(infileL) + ".tmp";
tmpR = Path.GetFileNameWithoutExtension(infileR) + ".tmp";
_tempFiles.Add(tmpL);
_tempFiles.Add(tmpR);
stderr.WriteLine("Exec DRC for {0}, left channel", drcfile);
stderr.WriteLine();
ok = RunDRC(drcfile, infileL, target, tmpL, peakPosL, out args);
if (ok)
{
stderr.WriteLine();
stderr.WriteLine("Exec DRC for {0}, right channel", drcfile);
stderr.WriteLine();
ok = RunDRC(drcfile, infileR, target, tmpR, peakPosR, out args);
}
if (ok)
{
stderr.WriteLine();
if (_noSkew)
{
stderr.WriteLine("Creating stereo filter {0}", outfile + ".wav" );
}
else
{
stderr.WriteLine("Creating stereo filter {0} (skew {1} samples)", outfile + ".wav", peakPosR - peakPosL);
}
ISoundObj stereoFilter = Splice(tmpL, peakPosL, tmpR, peakPosR, outfile + ".wav");
stderr.WriteLine();
// Convolve noise with the stereo filter
/*
NoiseGenerator noise = new NoiseGenerator(NoiseType.WHITE_FLAT, 2, (int)131072, stereoFilter.SampleRate, 1.0);
conv = new FastConvolver();
conv.impulse = stereoFilter;
conv.Input = noise;
wri = new WaveWriter(drcfile + "_Test.wav");
wri.Input = conv;
wri.Format = WaveFormat.IEEE_FLOAT;
wri.BitsPerSample = 32;
wri.SampleRate = _sampleRate;
wri.Normalization = -1;
wri.Run();
wri.Close();
wri = null;
conv = null;
noise = null;
* */
// Convolve filter with the in-room impulse response
WaveReader rea = new WaveReader(stereoImpulseFile);
conv = new FastConvolver();
conv.impulse = rea;
conv.Input = stereoFilter;
if (_pcm)
{
_impulseFiles.Add("LCorrected_" + outfile + ".pcm: corrected test convolution, raw data (32-bit float), left channel");
_impulseFiles.Add("RCorrected_" + outfile + ".pcm: corrected test convolution, raw data (32-bit float), right channel");
WriteImpulsePCM(conv, "LCorrected_" + outfile + ".pcm", "RCorrected_" + outfile + ".pcm");
}
wri = new WaveWriter(outfile + "_TestConvolution.wav");
wri.Input = conv;
wri.Format = WaveFormat.PCM;
wri.Dither = DitherType.TRIANGULAR;
wri.BitsPerSample = 16;
wri.SampleRate = _sampleRate;
wri.Normalization = -1;
wri.Run();
wri.Close();
rea.Close();
wri = null;
rea = null;
conv = null;
GC.Collect();
}
return ok;
}
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 Prep(string infileL, string infileR, string stereoImpulseFile, string outFile)
{
// Input files are complex
// 0=mag, 1=phase/pi (so it looks OK in a wave editor!)
// FFTs of the room impulse response
// Take two half-FFT-of-impulse WAV files
// Average them, into an array
int n;
SoundBuffer buff;
WaveWriter wri;
// NoiseGenerator noise;
FastConvolver conv;
/*
// Convolve noise with the in-room impulse
noise = new NoiseGenerator(NoiseType.WHITE_FLAT, 2, (int)131072, stereoImpulse.SampleRate, 1.0);
conv = new FastConvolver();
conv.impulse = stereoImpulse;
conv.Input = noise;
wri = new WaveWriter("ImpulseResponse_InRoom.wav");
wri.Input = conv;
wri.Format = WaveFormat.IEEE_FLOAT;
wri.BitsPerSample = 32;
wri.Normalization = 0;
wri.Run();
wri.Close();
wri = null;
conv = null;
*/
WaveReader rdrL = new WaveReader(infileL);
buff = new SoundBuffer(rdrL);
n = (int)buff.ReadAll();
uint sampleRate = buff.SampleRate;
uint nyquist = sampleRate / 2;
double binw = (nyquist / (double)n);
WaveReader rdrR = new WaveReader(infileR);
IEnumerator<ISample> enumL = buff.Samples;
IEnumerator<ISample> enumR = rdrR.Samples;
// For easier processing and visualisation
// read this in to an ERB-scale (not quite log-scale) array
// then we can smooth by convolving with a single half-cosine.
//
int nn = (int)ERB.f2bin(nyquist, sampleRate) + 1;
double[] muff = new double[nn];
int prevbin = 0;
int nbin = 0;
double v = 0;
int j = 0;
while (true)
{
double f = (double)j * binw; // equiv freq, Hz
int bin = (int)ERB.f2bin(f, sampleRate); // the bin we drop this sample in
if (bin > nn)
{
// One of the channels has more, but we're overrun so stop now
break;
}
j++;
bool more = false;
more |= enumL.MoveNext();
more |= enumR.MoveNext();
if (!more)
{
muff[prevbin] = v / nbin;
break;
}
v += enumL.Current[0]; // magnitude
v += enumR.Current[0]; // magnitude
nbin++;
if (bin > prevbin)
{
muff[prevbin] = v / nbin;
v = 0;
nbin = 0;
prevbin = bin;
}
}
double[] smoo = ERB.smooth(muff, 38);
// Pull out the freq response at ERB centers
FilterProfile lfg = ERB.profile(smoo, sampleRate);
// Write this response as a 'target' file
/*
FileStream fs = new FileStream("target_full.txt", FileMode.Create);
StreamWriter sw = new StreamWriter(fs, Encoding.ASCII);
foreach (FreqGain fg in lfg)
{
sw.WriteLine("{0} {1:f4}", Math.Round(fg.Freq), fg.Gain);
}
sw.Close();
*/
/*
fs = new FileStream("target_half.txt", FileMode.Create);
sw = new StreamWriter(fs, Encoding.ASCII);
foreach (FreqGain fg in lfg)
{
sw.WriteLine("{0} {1:f4}", Math.Round(fg.Freq), fg.Gain/2);
}
sw.Close();
*/
// Create a filter to invert this response
FilterProfile ifg = new FilterProfile();
foreach (FreqGain fg in lfg)
{
ifg.Add(new FreqGain(fg.Freq, -fg.Gain));
}
ISoundObj filterImpulse = new FilterImpulse(0, ifg, FilterInterpolation.COSINE, sampleRate);
filterImpulse.SampleRate = sampleRate;
// Write the filter impulse to disk
string sNoCorr = outFile + ".wav";
wri = new WaveWriter(sNoCorr);
wri.Input = filterImpulse; // invertor;
wri.Format = WaveFormat.IEEE_FLOAT;
wri.BitsPerSample = 32;
wri.SampleRate = _sampleRate;
wri.Normalization = -1;
wri.Run();
wri.Close();
_filterFiles.Add(sNoCorr);
/*
// Convolve noise with the NoCorrection filter
noise = new NoiseGenerator(NoiseType.WHITE_FLAT, 2, (int)131072, stereoImpulse.SampleRate, 1.0);
conv = new FastConvolver();
conv.impulse = invertor;
conv.Input = noise;
wri = new WaveWriter("NoCorrection_Test.wav");
wri.Input = conv;
wri.Format = WaveFormat.IEEE_FLOAT;
wri.BitsPerSample = 32;
wri.SampleRate = _sampleRate;
wri.Normalization = 0;
wri.Run();
wri.Close();
wri = null;
conv = null;
*/
// Convolve this with the in-room impulse response
WaveReader rea = new WaveReader(outFile + ".wav");
conv = new FastConvolver();
conv.impulse = rea;
conv.Input = new WaveReader(stereoImpulseFile);
wri = new WaveWriter(outFile + "_TestConvolution.wav");
wri.Input = conv;
wri.Format = WaveFormat.PCM;
wri.Dither = DitherType.TRIANGULAR;
wri.BitsPerSample = 16;
wri.SampleRate = _sampleRate;
wri.Normalization = -1;
wri.Run();
wri.Close();
rea.Close();
wri = null;
conv = null;
}
static ISoundObj Splice(string infileL, int peakPosL, string infileR, int peakPosR, string outfile)
{
//int tmp1;
//bool tmp2;
WaveReader reader1 = new WaveReader(infileL, WaveFormat.IEEE_FLOAT, 32, 1);
// reader1.Skip(peakPosL, out tmp1, out tmp2);
SoundBuffer buff1 = new SoundBuffer(reader1);
buff1.ReadAll();
// Normalize loudness for each channel
double g = Loudness.WeightedVolume(buff1);
buff1.ApplyGain(1/g);
WaveReader reader2 = new WaveReader(infileR, WaveFormat.IEEE_FLOAT, 32, 1);
// reader2.Skip(peakPosR, out tmp1, out tmp2);
SoundBuffer buff2 = new SoundBuffer(reader2);
buff2.ReadAll();
g = Loudness.WeightedVolume(buff2);
buff2.ApplyGain(1/g);
ChannelSplicer splicer = new ChannelSplicer();
splicer.Add(buff1);
splicer.Add(buff2);
// General-purpose:
// Find the extremities of the DRC impulse,
// window asymmetrically.
//
// But, since we specifically used linear-phase filters on target and mic,
// we know that the impulse is centered.
// We want an impulse length (_filterLen)
// so window the 2048 samples at each end (which are pretty low to begin with - less than -100dB)
ISoundObj output;
int nCount = (int)(buff1.Count / 2);
if (nCount > _filterLen/2)
{
BlackmanHarris bhw = new BlackmanHarris(nCount, 2048, (nCount / 2) - 2048);
bhw.Input = splicer;
SampleBuffer sb = new SampleBuffer(bhw);
output = sb.Subset(_filterLen/2, _filterLen);
}
else
{
output = splicer;
}
ISoundObj result = output;
if (!_noSkew)
{
// Apply skew to compensate for time alignment in the impulse responses
Skewer skewer = new Skewer(true);
skewer.Input = output;
skewer.Skew = peakPosL - peakPosR;
result = skewer;
}
WaveWriter writer = new WaveWriter(outfile);
writer.Input = result;
writer.Dither = DitherType.NONE;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.NumChannels = splicer.NumChannels;
if (Double.IsNaN(_gain))
{
writer.Normalization = 0;
}
else
{
writer.Gain = MathUtil.gain(_gain);
}
writer.Run();
writer.Close();
reader1.Close();
reader2.Close();
return result;
}
static ISoundObj GetSignalGenerator(double dBfs, out string desc)
{
double gain = MathUtil.gain(dBfs);
ISoundObj signalGenerator = null;
Sequencer seq;
string description = "Unknown";
switch (_siggen)
{
case "IDENT":
// Left-right identification: embedded resource
Assembly ass = Assembly.GetExecutingAssembly();
foreach (string s in ass.GetManifestResourceNames())
{
if (s.Contains("LeftRight"))
{
Stream res = ass.GetManifestResourceStream(s);
WaveReader rdr = new WaveReader(res);
// The stream is stereo, but we want to alternate
seq = new Sequencer();
for (int j = 0; j < 10; j++)
{
seq.Add(rdr, new List<double>(new double[] { gain, 0 }));
seq.Add(new NoiseGenerator(NoiseType.SILENCE, 2, 1.0, _inputSampleRate, 0.0, false));
seq.Add(rdr, new List<double>(new double[] { 0, gain }));
seq.Add(new NoiseGenerator(NoiseType.SILENCE, 2, 1.0, _inputSampleRate, 0.0, false));
}
signalGenerator = seq;
break;
}
}
/*
// Left-right identification signal: morse code
MorseCode envL = new MorseCode(" " + _sigparamA, 10, true);
ISoundObj sigL = new SweepGenerator(1, envL.LengthSeconds * 5, 220, 7040, _inputSampleRate, 0, false, gain, true);
envL.Input = sigL;
MorseCode envR = new MorseCode(" " + _sigparamB, 10, true);
ISoundObj sigR = new SweepGenerator(1, envR.LengthSeconds * 5, 7040, 220, _inputSampleRate, 0, false, gain, true);
envR.Input = sigR;
signalGenerator = new ChannelSplicer();
(signalGenerator as ChannelSplicer).Add(envL);
(signalGenerator as ChannelSplicer).Add(envR);
*/
description = String.Format("Left/Right channel identification");
break;
case "SWEEP":
seq = new Sequencer();
if (_sigparam1 == 0)
{
_sigparam1 = 45;
}
int lengthSamples = (int)(_sigparam1 * _inputSampleRate);
if (lengthSamples < 8388608)
{
// High-accuracy logarithmic sweep starting at 10Hz
int fade = (int)(_inputSampleRate / 20);
FFTSweepGenerator sg = new FFTSweepGenerator(2, lengthSamples, 10, _inputSampleRate / 2, _inputSampleRate, gain, false);
seq.Add(sg);
description = String.Format("Logarithmic sine sweep 10Hz to {0}Hz in {1} seconds", _inputSampleRate / 2, _sigparam1);
}
else
{
// Simple logarithmic sweep starting at 10Hz, windowed (uses much less memory!)
int fade = (int)(_inputSampleRate / 20);
BlackmanHarris bhwf = new BlackmanHarris(lengthSamples / 2, fade, (int)((lengthSamples / 2) - fade));
SweepGenerator sg = new SweepGenerator(2, lengthSamples, 10, _inputSampleRate / 2, _inputSampleRate, gain, false);
bhwf.Input = sg;
seq.Add(bhwf);
description = String.Format("Log sine sweep 10Hz to {0}Hz in {1} seconds", _inputSampleRate / 2, _sigparam1);
}
// Follow by 3 seconds of silence
seq.Add(new NoiseGenerator(NoiseType.SILENCE, 2, 3.0, _inputSampleRate, 0.0, false));
signalGenerator = seq;
break;
case "SINE":
signalGenerator = new SineGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz sine", _sigparam1);
break;
case "QUAD":
signalGenerator = new SineQuadGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz quadrature", _sigparam1);
break;
case "SQUARE":
signalGenerator = new SquareGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz non-bandlimited square", _sigparam1);
break;
case "BLSQUARE":
signalGenerator = new BandLimitedSquareGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz bandlimited square", _sigparam1);
break;
case "TRIANGLE":
signalGenerator = new TriangleGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz non-bandlimited triangle", _sigparam1);
break;
case "BLTRIANGLE":
signalGenerator = new BandLimitedTriangleGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz bandlimited triangle", _sigparam1);
break;
case "SAWTOOTH":
signalGenerator = new SawtoothGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz non-bandlimited sawtooth", _sigparam1);
break;
case "BLSAWTOOTH":
signalGenerator = new BandLimitedSawtoothGenerator(2, _inputSampleRate, _sigparam1, gain);
description = String.Format("{0}Hz bandlimited sawtooth", _sigparam1);
break;
case "WHITE":
signalGenerator = new NoiseGenerator(NoiseType.WHITE, 2, int.MaxValue, _inputSampleRate, gain, true);
description = String.Format("White noise");
break;
case "PINK":
bool mono = (_sigparam1 != 0 ? true : false);
signalGenerator = new NoiseGenerator(NoiseType.PINK, 2, int.MaxValue, _inputSampleRate, gain, mono);
description = String.Format("Pink noise {0}", mono ? "(mono)" : "(stereo)" );
break;
case "INTERMODULATION":
double n = 1;
description = String.Format("Intermodulation test {0}Hz", _sigparam1);
if (_sigparam2 != 0) { n++; description = description + " + " + _sigparam2 + "Hz"; }
if (_sigparam3 != 0) { n++; description = description + " + " + _sigparam3 + "Hz"; }
signalGenerator = new Mixer();
(signalGenerator as Mixer).Add(new SineGenerator(2, _inputSampleRate, _sigparam1, gain), 1/n);
if (_sigparam2 != 0) (signalGenerator as Mixer).Add(new SineGenerator(2, _inputSampleRate, _sigparam2, gain), 1 / n);
if (_sigparam3 != 0) (signalGenerator as Mixer).Add(new SineGenerator(2, _inputSampleRate, _sigparam3, gain), 1 / n);
break;
case "SHAPEDBURST":
description = String.Format("{0}Hz windowed (Blackman) over {1} cycles", _sigparam1, _sigparam2);
throw new NotImplementedException();
//break;
default:
_siggen = null;
break;
}
if (IsSigGenEQ())
{
if (IsSigGenEQBoth())
{
description = description + ", with EQ processing";
}
else if (IsSigGenEQL())
{
description = description + ", with EQ processing in left channel";
}
else if (IsSigGenEQR())
{
description = description + ", with EQ processing in right channel";
}
else
{
description = description + ", with EQ processing";
}
}
desc = description;
return signalGenerator;
}
private void init()
{
_running = true;
ComputeImpulseFFT();
nChannels = NumChannels;
// Size for work area
N = _impulseLength;
nImpulseChannels = _impulse.NumChannels;
Nh = N << 1;
K = (_partitions <= 0) ? N : (N / _partitions);
L = MathUtil.NextPowerOfTwo(K << 1);
P = (int)Math.Ceiling((double)N / (double)K);
// Trace.WriteLine("{0} partitions of size {1}, chunk size {2}", P, L, K);
// Allocate buffers for the source and output
src = new Complex[nChannels][];
output = new Complex[nChannels][];
for (ushort c = 0; c < nChannels; c++)
{
src[c] = new Complex[Nh];
output[c] = new Complex[Nh];
}
// For partitioned convolution, allocate arrays for the fft accumulators
if (_partitions > 0)
{
accum = new Complex[nChannels][][];
for (ushort c = 0; c < nChannels; c++)
{
accum[c] = new Complex[P][];
for (p = 0; p < P; p++)
{
accum[c][p] = new Complex[L];
}
}
}
if (IsPersistTail)
{
// If there's any *unprocessed* leftover from a previous convolution,
// load it into prevTailSamples before we begin
if (System.IO.File.Exists(_persistFile))
{
WaveReader tailReader = null;
try
{
tailReader = new WaveReader(_persistFile);
prevTailSamples = new List <ISample>(tailReader.Iterations);
// Trace.WriteLine("Read {0} tail samples", tailReader.Iterations);
if (tailReader.NumChannels == NumChannels)
{
foreach (ISample s in tailReader)
{
prevTailSamples.Add(s);
}
}
prevTailEnum = prevTailSamples.GetEnumerator();
}
catch (Exception e)
{
Trace.WriteLine("Could not read tail {0}: {1}", _persistFile.Replace(_persistPath, ""), e.Message);
}
// finally...
// ThreadPool.QueueUserWorkItem(delegate(object o)
// {
try
{
if (tailReader != null)
{
tailReader.Close();
tailReader = null;
}
System.IO.File.Delete(_persistFile);
}
catch (Exception)
{
// ignore, we're done
}
// });
}
}
// Trace.WriteLine("{0} partitions of size {1}, chunk size {2}", P, L, K);
}
private static ISoundObj GetFlatnessFilter(string impulsePath, ISoundObj impulseObj, double nFlatness)
{
ISoundObj filter = null;
if (impulseObj != null && impulsePath != null && nFlatness!=100)
{
uint sr = impulseObj.SampleRate;
if (_debug)
{
Trace.WriteLine(FlatnessFilterName(impulsePath, sr, nFlatness));
}
// Do we already have a cached flatness-filter?
// (Building these things is expensive...)
string sFile = FlatnessFilterPath(impulsePath, sr, nFlatness);
if (File.Exists(sFile))
{
try
{
filter = new WaveReader(sFile);
}
catch (Exception e)
{
if (_debug)
{
Trace.WriteLine("GetFlatnessFilter: " + e.Message);
}
}
}
if (filter == null)
{
// Generate a new flatness filter then
return GenerateFlatnessFilter(impulsePath, impulseObj, nFlatness);
}
}
return filter;
}
static SoundObj GetEQImpulse(ISoundObj mainImpulse, uint sampleRate, out string filterName)
{
DateTime dtStart = DateTime.Now;
SoundObj filterImpulse = null;
// Construct a string describing the filter
string filterDescription = "EQ" + _eqBands + "_" + _inputSampleRate + "_" + _eqLoudness + "_" + FlatnessFilterPath(_impulsePath, sampleRate, _eqFlatness);
bool nothingToDo = (_eqLoudness==0);
nothingToDo &= (_impulsePath == null) || (_eqFlatness == 100);
List<string> fgd = new List<string>();
foreach (FreqGain fg in _eqValues)
{
fgd.Add(fg.Freq + "@" + fg.Gain);
nothingToDo &= (fg.Gain == 0);
}
filterDescription = filterDescription + String.Join("_", fgd.ToArray());
filterDescription = filterDescription + "_IM_" + _impulsePath;
// Cached filters are named by hash of this string
filterName = "EQ" + filterDescription.GetHashCode().ToString("x10").ToUpperInvariant();
if (nothingToDo)
{
Trace.WriteLine("EQ flat");
WriteJSON(_eqValues, null, null);
return null;
}
else
{
Trace.WriteLine(filterName);
}
string filterFile = Path.Combine(_tempFolder, filterName + ".filter");
// Does the cached filter exist?
if (File.Exists(filterFile))
{
try
{
// Just read the cached EQ filter from disk
filterImpulse = new WaveReader(filterFile);
}
catch (Exception e)
{
if (_debug)
{
Trace.WriteLine("GetEQImpulse1: " + e.Message);
}
}
}
if(filterImpulse==null)
{
// Construct a filter impulse from the list of EQ values
SoundObj eqFilter = new FilterImpulse(0, _eqValues, FilterInterpolation.COSINE, _inputSampleRate);
filterImpulse = eqFilter;
ISoundObj qtFilter = GetQuietnessFilter(_inputSampleRate, _eqLoudness);
if (qtFilter != null)
{
// Convolve the two, to create a EQ-and-loudness filter
FastConvolver tmpConvolver = new FastConvolver();
tmpConvolver.partitions = 0;
tmpConvolver.impulse = qtFilter;
tmpConvolver.Input = eqFilter;
filterImpulse = tmpConvolver;
}
ISoundObj ftFilter = GetFlatnessFilter(_impulsePath, mainImpulse, _eqFlatness);
if (ftFilter != null)
{
// Convolve the two, to create a EQ-and-loudness filter
FastConvolver tmpConvolver2 = new FastConvolver();
tmpConvolver2.partitions = 0;
tmpConvolver2.impulse = filterImpulse;
tmpConvolver2.Input = ftFilter;
filterImpulse = tmpConvolver2;
}
// Blackman window to make the filter smaller?
try
{
// Write the filter impulse to disk
WaveWriter wri = new WaveWriter(filterFile);
wri.Input = filterImpulse;
wri.Format = WaveFormat.IEEE_FLOAT;
wri.BitsPerSample = 64;
wri.Run();
wri.Close();
if (_debug)
{
// DEBUG: Write the filter impulse as wav16
wri = new WaveWriter(filterFile + ".wav");
wri.Input = filterImpulse;
wri.Format = WaveFormat.PCM;
wri.BitsPerSample = 16;
wri.Normalization = -1.0;
wri.Dither = DitherType.NONE;//.TRIANGULAR;
wri.Run();
wri.Close();
}
// Write a JSON description of the filter
WriteJSON(_eqValues, filterImpulse, filterName);
}
catch (Exception e)
{
if (_debug)
{
Trace.WriteLine("GetEQImpulse2: " + e.Message);
}
}
}
filterImpulse.Reset();
if (_debug)
{
TimeSpan ts = DateTime.Now.Subtract(dtStart);
Trace.WriteLine("GetEQImpulse " + ts.TotalMilliseconds);
}
// Copy the filter's JSON description (if available) into "current.json"
CopyJSON(filterName);
return filterImpulse;
}
static WaveReader TryGetAppropriateImpulseReader(string filePath, bool allowResample, out string actualPath)
{
WaveReader rdr = null;
bool isExtensibleFormat = false;
WaveFormat format = WaveFormat.ANY;
bool needResample = false;
string resamplePath = GetResampledImpulsePath(filePath);
actualPath = null;
// Check the impulse file.
if (File.Exists(filePath))
{
try
{
rdr = new WaveReader(filePath);
actualPath = filePath;
if (rdr.SampleRate != 0 && rdr.SampleRate != _inputSampleRate)
{
Trace.WriteLine("Can't use {0}: its sample rate is {1} not {2}", CleanPath(_dataFolder, filePath), rdr.SampleRate, _inputSampleRate);
isExtensibleFormat = (rdr.FormatEx != null);
format = rdr.Format;
rdr.Close();
rdr = null;
actualPath = null;
needResample = true;
}
}
catch (Exception e)
{
Trace.WriteLine("Can't use {0}: {1}", CleanPath(_dataFolder, filePath), e.Message);
}
}
else
{
Trace.WriteLine("Can't use {0}: not found", CleanPath(_dataFolder, filePath));
}
if (rdr == null)
{
// No luck there. Check for an already-resampled version.
if (File.Exists(resamplePath))
{
try
{
rdr = new WaveReader(resamplePath);
if (rdr.SampleRate != 0 && rdr.SampleRate != _inputSampleRate)
{
// Oh dear! This shouldn't happen; after all, the resampled file's name is designed to match sample rates
Trace.WriteLine("Can't use {0}: its sample rate is {1} not {2}", CleanPath(_dataFolder, resamplePath), rdr.SampleRate, _inputSampleRate);
rdr.Close();
rdr = null;
}
else
{
actualPath = resamplePath;
Trace.WriteLine("Using resampled version {0} instead", CleanPath(_dataFolder, resamplePath));
needResample = false;
}
}
catch (Exception e)
{
Trace.WriteLine("Can't use {0}: {1}", CleanPath(_dataFolder, resamplePath), e.Message);
}
}
else if(allowResample)
{
Trace.WriteLine("Can't use {0}: not found", CleanPath(_dataFolder, resamplePath));
}
}
if (needResample && allowResample)
{
// If 'sox' is available,
// use it to resample the filter we're trying to use
// sox <input.wav> -r <new_sample_rate> <output.wav> polyphase
//
// The sox application might be either
// - in the same folder as this application (quite likely...), or
// - somewhere else on the path
// and it might be called "sox" or "sox.exe", depending on the operating system (of course).
//
if (true || isExtensibleFormat)
{
// sox can't handle the WaveFormatExtensible file type, so we need to save a clean copy of the file first
// ALSO: sox goes very wrong if the original is near 0dBfs
// so we normalize ALWAYS before resample.
rdr = new WaveReader(filePath);
string newFile = "RE" + filePath.GetHashCode().ToString("x10").ToUpperInvariant() + ".wav";
string newPath = Path.Combine(_tempFolder, newFile);
WaveWriter tempWriter = new WaveWriter(newPath);
tempWriter.Input = rdr;
tempWriter.Format = rdr.Format;
tempWriter.BitsPerSample = rdr.BitsPerSample;
tempWriter.Normalization = -6;
tempWriter.Run();
tempWriter.Close();
rdr.Close();
rdr = null;
filePath = newPath;
}
// _soxExe = "sox";
string exeName = _soxExe;
if(File.Exists(Path.Combine(_pluginFolder, _soxExe + ".exe")))
{
exeName = "\"" + Path.Combine(_pluginFolder, _soxExe + ".exe") + "\"";
}
else if (File.Exists(Path.Combine(_pluginFolder, _soxExe)))
{
exeName = Path.Combine(_pluginFolder, _soxExe);
}
// _soxFmt = "\"{0}\" -r {1} \"{2}\" polyphase";
string soxArgs = String.Format(_soxFmt, filePath, _inputSampleRate, resamplePath);
Trace.WriteLine("Trying {0} {1}", exeName, soxArgs);
System.Diagnostics.Process soxProcess = new System.Diagnostics.Process();
System.Diagnostics.ProcessStartInfo soxInfo = new System.Diagnostics.ProcessStartInfo();
soxInfo.Arguments = soxArgs;
soxInfo.FileName = exeName;
soxInfo.UseShellExecute = false;
soxInfo.RedirectStandardError = true;
soxProcess.StartInfo = soxInfo;
try
{
soxProcess.Start();
// Wait for the sox process to finish!
string err = soxProcess.StandardError.ReadToEnd();
soxProcess.WaitForExit(500);
if (soxProcess.HasExited)
{
int n = soxProcess.ExitCode;
if (n != 0)
{
Trace.WriteLine("No, that didn't seem to work: {0}", err);
}
else
{
Trace.WriteLine("Yes, that seemed to work.");
rdr = TryGetAppropriateImpulseReader(resamplePath, false, out actualPath);
}
}
}
catch (Exception e)
{
Trace.WriteLine("That didn't seem to work: {0}", e.Message);
}
}
if (rdr == null)
{
Trace.WriteLine("No suitable impulse found ({0}).", filePath);
}
return rdr;
}
static void LoadImpulse()
{
DateTime dtStart = DateTime.Now;
string theImpulsePath = null;
string theEQImpulseName = null;
ISoundObj main = GetMainImpulse(out theImpulsePath);
uint sr = (main == null ? _inputSampleRate : main.SampleRate);
if (sr == 0)
{
if (_debug) { Trace.WriteLine("oops: no sample rate!"); }
sr = 44100;
}
ISoundObj eq = GetEQImpulse(main, sr, out theEQImpulseName);
ISoundObj combinedFilter = null;
if (main == null && eq != null)
{
combinedFilter = eq;
}
else if (main != null && eq == null)
{
combinedFilter = main;
}
else if (main != null && eq != null)
{
// Check whether we have (and can load) a cached version of the combined filter
string tempString = theEQImpulseName + "_" + theImpulsePath;
string filterName = "CC" + tempString.GetHashCode().ToString("x10").ToUpperInvariant();
string filterFile = Path.Combine(_tempFolder, filterName + ".filter");
if (_debug)
{
Trace.WriteLine(filterName);
}
if (File.Exists(filterFile))
{
try
{
// Just read the cached EQ filter from disk
combinedFilter = new WaveReader(filterFile);
}
catch (Exception e)
{
if (_debug)
{
Trace.WriteLine("LoadImpulse1: " + e.Message);
}
}
}
if (combinedFilter == null)
{
// Convolve the room-correction impulse with the EQ impulse to make just one.
// (this is quite slow)
FastConvolver temp = new FastConvolver();
temp.partitions = 0;
temp.impulse = eq;
temp.Input = main;
combinedFilter = temp;
try
{
// Write the combined impulse
temp.Reset();
WaveWriter tempWriter = new WaveWriter(filterFile);
tempWriter.Format = WaveFormat.IEEE_FLOAT;
tempWriter.BitsPerSample = 64;
tempWriter.Input = temp;
tempWriter.Run();
tempWriter.Close();
if (_debug)
{
// DEBUG: Write the combined impulse as WAV16
temp.Reset();
tempWriter = new WaveWriter(filterFile + ".wav");
tempWriter.Format = WaveFormat.PCM;
tempWriter.BitsPerSample = 16;
tempWriter.Gain = 0.1;
tempWriter.Dither = DitherType.NONE;//.TRIANGULAR;
tempWriter.Input = temp;
tempWriter.Run();
tempWriter.Close();
}
}
catch (Exception e)
{
if (_debug)
{
Trace.WriteLine("LoadImpulse2: " + e.Message);
}
}
}
}
_MainConvolver.impulse = combinedFilter;
if (combinedFilter != null)
{
// Calculate loudness-adjusted volume of each channel of the impulse
_impulseVolumes.Clear();
for (ushort j = 0; j < combinedFilter.NumChannels; j++)
{
double v = Loudness.WeightedVolume(combinedFilter.Channel(j));
_impulseVolumes.Add(v);
if (_debug)
{
Trace.WriteLine("WV{0}: {1}", j, v);
}
}
}
combinedFilter = null;
if (_debug)
{
TimeSpan ts = DateTime.Now.Subtract(dtStart);
Trace.WriteLine("Loadmpulse " + ts.TotalMilliseconds);
}
}
static void InguzDSP(bool doRun)
{
DateTime dtStartRun = DateTime.Now;
string sigdesc = null;
// We need a few convolvers, of course
if (_slow)
{
_MatrixConvolver = new SlowConvolver();
_MainConvolver = new SlowConvolver();
_EQConvolver = new SlowConvolver();
}
else
{
_MatrixConvolver = new FastConvolver();
_MainConvolver = new FastConvolver();
_EQConvolver = new FastConvolver();
}
// Shuffler
_widthShuffler = new Shuffler();
// Two skewers
_depthSkewer = new Skewer(true);
_skewSkewer = new Skewer(true);
// Writer
if (_outPath == null)
{
_writer = new WaveWriter(); // stdout
}
else
{
_writer = new WaveWriter(_outPath);
}
_writer.NumChannels = _outChannels;
if (_debug)
{
TimeSpan ts = DateTime.Now.Subtract(dtStartRun);
Trace.WriteLine("Setup " + ts.TotalMilliseconds);
}
/*
DateTime exp = DSPUtil.DSPUtil.EXPIRY;
if (exp != null)
{
if (DateTime.Now.CompareTo(exp) >= 0)
{
Trace.WriteLine("**** THIS EVALUATION VERSION EXPIRED {0}", DSPUtil.DSPUtil.EXPIRY);
Trace.WriteLine("**** SEE http://www.inguzaudio.com/DSP/ FOR DETAILS");
_MatrixConvolver.Enabled = false;
_MainConvolver.Enabled = false;
_EQConvolver.Enabled = false;
Show("", "InguzDSP has expired.", 2);
}
else
{
Trace.WriteLine("This evaluation version will expire {0}", DSPUtil.DSPUtil.EXPIRY);
}
}
*/
// Read the configuration file
doRun = LoadConfig2();
// Do any cleanup required before we start
CleanUp();
// The main convolver should persist and re-use its leftovers between invocations
// under this user's (=squeezebox's) ID
_MainConvolver.partitions = _partitions;
if (_tail && !IsSigGenNonEQ())
{
_MainConvolver.PersistPath = _tempFolder;
_MainConvolver.PersistTail = _userID;
}
// Construct a second convolver for the "tone" (EQ) control.
_EQConvolver.partitions = _partitions;
if (_tail && !IsSigGenNonEQ())
{
_EQConvolver.PersistPath = _tempFolder;
_EQConvolver.PersistTail = _userID + ".eq";
}
// Make a reader
// _inPath is the data stream
WaveReader inputReader = null;
bool ok = false;
try
{
if (_inStream != null)
{
inputReader = new WaveReader(_inStream);
}
else if (_isRawIn)
{
inputReader = new WaveReader(_inPath, _rawtype, _rawbits, _rawchan, _startTime);
}
else
{
inputReader = new WaveReader(_inPath, _startTime);
}
inputReader.BigEndian = _bigEndian;
ok = true;
}
catch (Exception e)
{
Trace.WriteLine("Unable to read: " + e.Message);
// Just stop (no need to report the stack)
}
if (ok)
{
if (inputReader.IsSPDIF)
{
// The wave file is just a SPDIF (IEC61937) stream, we shouldn't touch it
_inputSPDIF = true;
_isBypass = true;
}
if (_isBypass)
{
// The settings file says bypass, we shouldn't touch it
_gain = 0;
_dither = DitherType.NONE;
}
uint sr = _inputSampleRate; // Yes, the commandline overrides the source-file...
if (sr == 0)
{
sr = inputReader.SampleRate;
}
if (sr == 0)
{
sr = 44100;
}
_inputSampleRate = sr;
if (WaveFormatEx.AMBISONIC_B_FORMAT_IEEE_FLOAT.Equals(inputReader.FormatEx) ||
WaveFormatEx.AMBISONIC_B_FORMAT_PCM.Equals(inputReader.FormatEx))
{
_isBFormat = true;
}
}
ISoundObj source = inputReader;
if (IsSigGen())
{
// Signal-generator source instead of music.
_isBFormat = false;
source = GetSignalGenerator(-12, out sigdesc);
Show("Test signal", sigdesc, 20);
}
if (IsSigGenNonEQ() || _isBypass)
{
// Signal-generator mode. Overrides everything else!
_writer.Input = source;
}
else
{
if (ok)
{
// Load the room correction impulse to the convolver
// (NB: don't do this until we've created the reader, otherwise we can't be user of the samplerate yet...)
LoadImpulse();
GC.Collect();
}
if (ok && _isBFormat)
{
source = DecodeBFormat(source);
}
if (ok)
{
ISoundObj nextSrc;
// Perform width (matrix) processing on the signal
// - Shuffle the channels
// - Convolve if there's a filter
// - Apply gain to boost or cut the 'side' channel
// - Shuffle back
_widthShuffler.Input = source;
nextSrc = _widthShuffler;
// Use a convolver for the matrix filter
if (_matrixFilter != null)
{
LoadMatrixFilter();
_MatrixConvolver.Input = TwoChannel(nextSrc);
nextSrc = _MatrixConvolver as ISoundObj;
}
// if (_depth != 0)
// {
// // Time-alignment between the LR and MS
// _depthSkewer.Input = nextSrc;
// nextSrc = _depthSkewer;
// }
// Shuffle back again
Shuffler shMSLR = new Shuffler();
shMSLR.Input = nextSrc;
nextSrc = shMSLR;
// Do the room-correction convolution
_MainConvolver.Input = TwoChannel(shMSLR);
nextSrc = _MainConvolver;
if (_skew != 0)
{
// time-alignment between left and right
_skewSkewer.Input = nextSrc;
nextSrc = _skewSkewer;
}
// Splice EQ and non-EQ channels
if (IsSigGenEQ())
{
ChannelSplicer splice = new ChannelSplicer();
if (IsSigGenEQL())
{
splice.Add(new SingleChannel(nextSrc, 0));
}
else
{
splice.Add(new SingleChannel(source, 0));
}
if (IsSigGenEQR())
{
splice.Add(new SingleChannel(nextSrc, 1));
}
else
{
splice.Add(new SingleChannel(source, 1));
}
nextSrc = splice;
}
// Process externally with aften or equivalent?
if (_aftenNeeded && !_isBypass)
{
nextSrc = AftenProcess(nextSrc);
_outFormat = WaveFormat.PCM;
_outBits = 16;
_dither = DitherType.NONE;
}
// Finally pipe this to the writer
_writer.Input = nextSrc;
}
}
if (ok)
{
//dt = System.DateTime.Now; // time to here is approx 300ms
// Dither and output raw-format override anything earlier in the chain
_writer.Dither = _isBypass ? DitherType.NONE : _dither;
_writer.Raw = _isRawOut;
_writer.Format = (_outFormat == WaveFormat.ANY) ? inputReader.Format : _outFormat;
_writer.BitsPerSample = (_outBits == 0 || _isBypass) ? inputReader.BitsPerSample : _outBits;
_writer.SampleRate = (_outRate == 0 || _isBypass) ? _inputSampleRate : _outRate;
SetWriterGain();
if (IsSigGen())
{
Trace.WriteLine("Test signal: {0}, -12dBfs, {1}/{2} {3} {4}", sigdesc, _writer.BitsPerSample, _writer.SampleRate, _writer.Format, _writer.Dither);
}
string amb1 = "";
string amb2 = "";
if(_isBFormat)
{
amb1 = "B-Format ";
amb2 = _ambiType + " ";
}
string big = inputReader.BigEndian ? "(Big-Endian) " : "";
if (_inputSPDIF)
{
Trace.WriteLine("Stream is SPDIF-wrapped; passing through");
}
else if (_isBypass)
{
Trace.WriteLine("Processing is disabled; passing through");
}
Trace.WriteLine("{0}/{1} {2}{3} {4}=> {5}/{6} {7}{8} {9}, gain {10} dB", inputReader.BitsPerSample, _inputSampleRate, amb1, inputReader.Format, big, _writer.BitsPerSample, _writer.SampleRate, amb2, _writer.Format, _writer.Dither, _gain);
TimeSpan elapsedInit = System.DateTime.Now.Subtract(dtStartRun);
int n = _writer.Run();
TimeSpan elapsedTotal = System.DateTime.Now.Subtract(dtStartRun);
double realtime = n / _writer.SampleRate;
double runtime = elapsedTotal.TotalMilliseconds / 1000;
Trace.WriteLine("{0} samples, {1} ms ({2} init), {3} * realtime, peak {4} dBfs", n, elapsedTotal.TotalMilliseconds, elapsedInit.TotalMilliseconds, Math.Round(realtime / runtime, 4), Math.Round(_writer.dbfsPeak, 4));
StopConfigListening();
_writer.Close();
}
}
