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 void Main(string[] args)
{
// Find where this executable is launched from
string[] cargs = Environment.GetCommandLineArgs();
_thisFolder = Path.GetDirectoryName(cargs[0]);
if (String.IsNullOrEmpty(_thisFolder))
{
_thisFolder = Environment.CurrentDirectory;
}
string appData = Environment.GetFolderPath(Environment.SpecialFolder.CommonApplicationData);
_impulsesFolder = Path.GetFullPath(Path.Combine(appData, "InguzEQ" + slash + "Impulses" + slash));
string[] inFiles = new string[4];
string inL = "";
string inR = "";
if (!DisplayInfo())
{
return;
}
bool ok = (args.Length > 0);
bool longUsage = false;
for (int j = 0; ok && j < args.Length; j++)
{
string arg = args[j];
switch (args[j].ToUpperInvariant())
{
case "/?":
case "-?":
case "/H":
case "/HELP":
ok = false;
longUsage = true;
break;
case "/L":
case "/0":
inFiles[0] = args[++j];
_nInFiles = Math.Max(_nInFiles, 1);
break;
case "/R":
case "/1":
inFiles[1] = args[++j];
_nInFiles = Math.Max(_nInFiles, 2);
break;
case "/2":
inFiles[2] = args[++j];
_nInFiles = Math.Max(_nInFiles, 3);
break;
case "/3":
inFiles[3] = args[++j];
_nInFiles = Math.Max(_nInFiles, 4);
break;
case "/LENGTH":
_filterLen = int.Parse(args[++j], CultureInfo.InvariantCulture);
if (_filterLen < 16)
{
throw new Exception("Length is too small.");
}
break;
case "/DBL":
_dbl = true;
break;
case "/PCM":
_pcm = true;
break;
case "/NODRC":
_noDRC = true;
break;
case "/NOSKEW":
_noSkew = true;
break;
case "/NONORM":
// No normalization of the impulse response (undocumented)
_noNorm = true;
break;
case "/SPLIT":
_split = true;
break;
case "/COPY":
_copy = true;
break;
case "/GAIN":
_gain = double.Parse(args[++j], CultureInfo.InvariantCulture);
break;
case "/ALL":
// Returns negative-time components as part of the impulse response
// (experimental, to be used for THD measurement)
_returnAll = true;
break;
case "/POWER":
// Raises sweep to power n
// (experimental, to be used for THD measurement)
_power = int.Parse(args[++j], CultureInfo.InvariantCulture);
break;
case "/FMIN":
// (experimental, i.e. broken)
_fmin = int.Parse(args[++j], CultureInfo.InvariantCulture);
_fminSpecified = true;
break;
case "/FMAX":
// (experimental, i.e. broken)
_fmax = int.Parse(args[++j], CultureInfo.InvariantCulture);
_fmaxSpecified = true;
break;
case "/DIRECT":
// Create filtered (direct-sound) filters
_doDirectFilters = true;
break;
case "/NOSUB":
// Don't apply subsonic filter to the impulse response
_noSubsonicFilter = true;
break;
case "/NOOVER":
// Don't override DRC's settings for filter type and length
_noOverrideDRC = true;
break;
case "/KEEPTEMP":
// Undocumented
_keepTempFiles = true;
break;
case "/REFCH":
// Override the reference-channel detection
_refchannel = int.Parse(args[++j], CultureInfo.InvariantCulture);
if (_refchannel<0 || _refchannel > _nInFiles - 1)
{
throw new Exception(String.Format("RefCh can only be from 0 to {0}.", _nInFiles-1));
}
break;
case "/ENV":
// Undocumented. Save the Hilbert envelope
_env = true;
break;
case "-":
// ignore
break;
default:
ok = false;
break;
}
}
if (!ok)
{
DisplayUsage(longUsage);
}
else
{
try
{
if (!_noDRC)
{
if (!File.Exists(GetDRCExe()))
{
stderr.WriteLine("Denis Sbragion's DRC (http://drc-fir.sourceforge.net/) was not found.");
stderr.WriteLine("Only the impulse response will be calculated, not correction filters.");
stderr.WriteLine("");
_noDRC = true;
}
}
if (!_noDRC)
{
FileInfo[] drcfiles = new DirectoryInfo(_thisFolder).GetFiles("*.drc");
if (drcfiles.Length == 0)
{
stderr.WriteLine("No .drc files were found in the current folder.");
stderr.WriteLine("Only the impulse response will be calculated, not correction filters.");
stderr.WriteLine("");
_noDRC = true;
}
}
for(int i=0; i<_nInFiles; i++)
{
string inFile = inFiles[i];
if (String.IsNullOrEmpty(inFile))
{
stderr.WriteLine("Error: The {0} input file was not specified.", FileDescription(i));
return;
}
if (!File.Exists(inFile))
{
stderr.WriteLine("Error: The {0} input file {1} was not found.", FileDescription(i), inFile);
return;
}
for (int j = 0; j < i; j++)
{
if (inFile.Equals(inFiles[j]))
{
stderr.WriteLine("Warning: The same input file ({0}) was specified for both {1} and {2}!", inFile, FileDescription(j), FileDescription(i));
//stderr.WriteLine();
}
}
}
// Temporary
if (_nInFiles != 2)
{
stderr.WriteLine("Error: Two input files must be specified.");
return;
}
inL = inFiles[0];
inR = inFiles[1];
// end temporary
uint sampleRate;
List<SoundObj> impulses;
List<ISoundObj> filteredImpulses;
List<string> impDirects;
List<Complex[]> impulseFFTs;
List<double> maxs;
SoundObj impulseL;
SoundObj impulseR;
ISoundObj filteredImpulseL = null;
ISoundObj filteredImpulseR = null;
string impDirectL = null;
string impDirectR = null;
Complex[] impulseLFFT;
Complex[] impulseRFFT;
WaveWriter writer;
ISoundObj buff;
double g;
if (!_keepTempFiles)
{
_tempFiles.Add("rps.pcm");
_tempFiles.Add("rtc.pcm");
}
// Find the left impulse
stderr.WriteLine("Processing left measurement ({0})...", inL);
impulseL = Deconvolve(inL, out impulseLFFT, out _peakPosL);
sampleRate = impulseL.SampleRate;
_sampleRate = sampleRate;
double peakM = Math.Round(MathUtil.Metres(_peakPosL, sampleRate), 2);
double peakFt = Math.Round(MathUtil.Feet(_peakPosL, sampleRate), 2);
stderr.WriteLine(" Impulse peak at sample {0} ({1}m, {2}ft)", _peakPosL, peakM, peakFt);
// Write to PCM
string impFileL = Path.GetFileNameWithoutExtension(inL) + "_imp" + ".pcm";
if (!_keepTempFiles)
{
_tempFiles.Add(impFileL);
}
writer = new WaveWriter(impFileL);
writer.Input = impulseL;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Raw = true;
writer.Run();
writer.Close();
// Write the impulseFFT to disk
int L = impulseLFFT.Length;
string impTempL = Path.GetFileNameWithoutExtension(inL) + "_imp" + ".dat";
_tempFiles.Add(impTempL);
writer = new WaveWriter(impTempL);
writer.Input = new CallbackSource(2, sampleRate, delegate(long j)
{
if (j >= L / 2)
{
return null;
}
Complex si = impulseLFFT[j]; // +impulseLFFT[L - j - 1];
ISample s = new Sample2();
s[0] = si.Magnitude;
s[1] = si.Phase / Math.PI;
return s;
});
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
impulseLFFT = null;
GC.Collect();
if (_doDirectFilters)
{
// Sliding low-pass filter over the impulse
stderr.WriteLine(" Filtering...");
filteredImpulseL = SlidingLowPass(impulseL, _peakPosL);
// Write PCM for the filtered impulse
impDirectL = Path.GetFileNameWithoutExtension(inL) + "_impfilt" + ".pcm";
if (!_keepTempFiles)
{
_tempFiles.Add(impDirectL);
}
writer = new WaveWriter(impDirectL);
writer.Input = filteredImpulseL;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.SampleRate = _sampleRate;
writer.BitsPerSample = 32;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
filteredImpulseL.Reset();
}
GC.Collect();
stderr.WriteLine(" Deconvolution: left impulse done.");
stderr.WriteLine();
// Find the right impulse
stderr.WriteLine("Processing right measurement ({0})...", inR);
impulseR = Deconvolve(inR, out impulseRFFT, out _peakPosR);
peakM = Math.Round(MathUtil.Metres(_peakPosR, sampleRate), 2);
peakFt = Math.Round(MathUtil.Feet(_peakPosR, sampleRate), 2);
stderr.WriteLine(" Impulse peak at sample {0} ({1}m, {2}ft)", _peakPosR, peakM, peakFt);
// Write to PCM
string impFileR = Path.GetFileNameWithoutExtension(inR) + "_imp" + ".pcm";
if (!_keepTempFiles)
{
_tempFiles.Add(impFileR);
}
writer = new WaveWriter(impFileR);
writer.Input = impulseR;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Raw = true;
writer.Run();
writer.Close();
// Write the impulseFFT magnitude to disk
L = impulseRFFT.Length;
string impTempR = Path.GetFileNameWithoutExtension(inR) + "_imp" + ".dat";
_tempFiles.Add(impTempR);
writer = new WaveWriter(impTempR);
writer.Input = new CallbackSource(2, impulseR.SampleRate, delegate(long j)
{
if (j >= L / 2)
{
return null;
}
Complex si = impulseRFFT[j]; // +impulseRFFT[L - j - 1];
ISample s = new Sample2();
s[0] = si.Magnitude;
s[1] = si.Phase / Math.PI;
return s;
});
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
impulseRFFT = null;
GC.Collect();
if (_doDirectFilters)
{
// Sliding low-pass filter over the impulse
stderr.WriteLine(" Filtering...");
filteredImpulseR = SlidingLowPass(impulseR, _peakPosR);
// Write PCM for the filtered impulse
impDirectR = Path.GetFileNameWithoutExtension(inR) + "_impfilt" + ".pcm";
if (!_keepTempFiles)
{
_tempFiles.Add(impDirectR);
}
writer = new WaveWriter(impDirectR);
writer.Input = filteredImpulseR;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
filteredImpulseR.Reset();
}
GC.Collect();
stderr.WriteLine(" Deconvolution: right impulse done.");
stderr.WriteLine();
// Join the left and right impulse files (truncated at 65536) into a WAV
// and normalize loudness for each channel
stderr.WriteLine("Splicing and normalizing (1)");
ChannelSplicer longstereoImpulse = new ChannelSplicer();
// (Don't normalize each channel's volume separately if _returnAll, it's just too expensive)
if (_returnAll)
{
buff = impulseL;
}
else
{
buff = new SoundBuffer(new SampleBuffer(impulseL).Subset(0, 131071));
g = Loudness.WeightedVolume(buff);
(buff as SoundBuffer).ApplyGain(1 / g);
}
longstereoImpulse.Add(buff);
if (_returnAll)
{
buff = impulseR;
}
else
{
buff = new SoundBuffer(new SampleBuffer(impulseR).Subset(0, 131071));
g = Loudness.WeightedVolume(buff);
(buff as SoundBuffer).ApplyGain(1 / g);
}
longstereoImpulse.Add(buff);
ISoundObj stereoImpulse = longstereoImpulse;
_impulseFiles.Add("Impulse_Response_Measured.wav: stereo impulse response from measurements");
writer = new WaveWriter("Impulse_Response_Measured.wav");
writer.Input = longstereoImpulse;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Normalization = -1;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
if (_env)
{
// Also save the Hilbert envelope
HilbertEnvelope env = new HilbertEnvelope(8191);
env.Input = longstereoImpulse;
_impulseFiles.Add("Impulse_Response_Envelope.wav: Hilbert envelope of the impulse response");
writer = new WaveWriter("Impulse_Response_Envelope.wav");
writer.Input = env;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Normalization = -1;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
}
if (_dbl)
{
// Create DBL files for Acourate
_impulseFiles.Add("PulseL.dbl: impulse response, raw data (64-bit float), left channel ");
_impulseFiles.Add("PulseR.dbl: impulse response, raw data (64-bit float), right channel");
_impulseFiles.Add(" (use skew=" + (_peakPosL - _peakPosR) + " for time alignment)");
WriteImpulseDBL(stereoImpulse, "PulseL.dbl", "PulseR.dbl");
}
if (_pcm)
{
// Create PCM files for Octave (etc)
_impulseFiles.Add("LUncorrected.pcm: impulse response, raw data (32-bit float), left channel");
_impulseFiles.Add("RUncorrected.pcm: impulse response, raw data (32-bit float), right channel");
WriteImpulsePCM(stereoImpulse, "LUncorrected.pcm", "RUncorrected.pcm");
}
stereoImpulse = null;
longstereoImpulse = null;
buff = null;
GC.Collect();
if (_doDirectFilters)
{
// Same for the filtered impulse response
stderr.WriteLine("Splicing and normalizing (2)");
ChannelSplicer longstereoImpulseF = new ChannelSplicer();
buff = new SoundBuffer(new SampleBuffer(filteredImpulseL).Subset(0, 131071));
double gL = Loudness.WeightedVolume(buff);
(buff as SoundBuffer).ApplyGain(1 / gL);
longstereoImpulseF.Add(buff);
FilterProfile lfgDirectL = new FilterProfile(buff, 0.5);
buff = new SoundBuffer(new SampleBuffer(filteredImpulseR).Subset(0, 131071));
double gR = Loudness.WeightedVolume(buff);
(buff as SoundBuffer).ApplyGain(1 / gR);
longstereoImpulseF.Add(buff);
FilterProfile lfgDirectR = new FilterProfile(buff, 0.5);
_impulseFiles.Add("Impulse_Response_Filtered.wav: approximation to direct-sound impulse response");
writer = new WaveWriter("Impulse_Response_Filtered.wav");
writer.Input = longstereoImpulseF;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Normalization = -1;
writer.Raw = false;
writer.Run();
writer.Close();
double gg = writer.Gain;
writer = null;
longstereoImpulseF = null;
ChannelSplicer longstereoImpulseD = new ChannelSplicer();
Mixer diffuse = new Mixer();
diffuse.Add(impulseL, 1.0);
diffuse.Add(filteredImpulseL, -1.0);
buff = new SoundBuffer(new SampleBuffer(diffuse).Subset(0, 131071));
(buff as SoundBuffer).ApplyGain(1 / gL);
longstereoImpulseD.Add(buff);
FilterProfile lfgDiffuseL = new FilterProfile(buff, 0.5);
diffuse = new Mixer();
diffuse.Add(impulseR, 1.0);
diffuse.Add(filteredImpulseR, -1.0);
buff = new SoundBuffer(new SampleBuffer(diffuse).Subset(0, 131071));
(buff as SoundBuffer).ApplyGain(1 / gR);
longstereoImpulseD.Add(buff);
FilterProfile lfgDiffuseR = new FilterProfile(buff, 0.5);
_impulseFiles.Add("Impulse_Response_Diffuse.wav: approximation to diffuse-field remnant");
writer = new WaveWriter("Impulse_Response_Diffuse.wav");
writer.Input = longstereoImpulseD;
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Gain = gg;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
// Filter the diffuse-field curve against double the diffuse-field curve
FilterImpulse fiDiffuse = new FilterImpulse(8192, HRTF.diffuseDiff0() * 2, FilterInterpolation.COSINE, sampleRate);
FastConvolver co = new FastConvolver(longstereoImpulseD, fiDiffuse);
SoundBuffer buffd = new SoundBuffer(co);
_impulseFiles.Add("Impulse_Response_Diffuse_Comp.wav: filtered diffuse-field remnant");
writer = new WaveWriter("Impulse_Response_Diffuse_Comp.wav");
writer.Input = buffd.Subset(4096);
writer.Format = WaveFormat.IEEE_FLOAT;
writer.BitsPerSample = 32;
writer.SampleRate = _sampleRate;
writer.Gain = gg;
writer.Raw = false;
writer.Run();
writer.Close();
writer = null;
longstereoImpulseD = null;
bool any = false;
string jsonFile = "Diff.json";
FileStream fs = new FileStream(jsonFile, FileMode.Create);
StreamWriter sw = new StreamWriter(fs);
sw.WriteLine("{");
FilterProfile lfgDiffL = lfgDirectL - lfgDiffuseL;
if (lfgDiffL != null)
{
if (any) sw.WriteLine(",");
any = true;
sw.Write(lfgDiffL.ToJSONString("DiffL", "Diffuse field relative to direct, left channel"));
}
FilterProfile lfgDiffR = lfgDirectR - lfgDiffuseR;
if (lfgDiffR != null)
{
if (any) sw.WriteLine(",");
any = true;
sw.Write(lfgDiffR.ToJSONString("DiffR", "Diffuse field relative to direct, right channel"));
}
sw.WriteLine("}");
sw.Close();
fs.Close();
}
buff = null;
GC.Collect();
System.Console.Error.WriteLine();
if (!_noDRC)
{
// Analyze the freq response
// and create targets
// target_full.txt and target_half.txt
stderr.WriteLine("Analyzing response curves.");
Prep(impTempL, impTempR, "Impulse_Response_Measured.wav", "NoCorrection");
// Call DRC to create the filters
// then splice the DRC left & right output files together
stderr.WriteLine("Preparing for DRC.");
if (DoDRC(impFileL, impFileR, impDirectL, impDirectR, _peakPosL, _peakPosR, "Impulse_Response_Measured.wav", "Impulse_Response_Filtered.wav"))
{
stderr.WriteLine("Success!");
}
}
// Report names of the impulse files created
if (_impulseFiles.Count == 0)
{
System.Console.Error.WriteLine("No impulse response files were created.");
}
if (_impulseFiles.Count > 0)
{
System.Console.Error.WriteLine("Impulse response files were created:");
foreach (string f in _impulseFiles)
{
string s = " " + f;
System.Console.Error.WriteLine(s);
}
}
// Report names of the filter files created
if (_filterFiles.Count == 0 && !_noDRC)
{
System.Console.Error.WriteLine("No correction filter files were created.");
}
if (_filterFiles.Count > 0)
{
System.Console.Error.WriteLine("Correction filter files were created:");
foreach (string f in _filterFiles)
{
string s = " " + f;
if (_copy)
{
try
{
File.Copy(f, Path.Combine(_impulsesFolder, f), true);
s += " (copied)";
}
catch (Exception e)
{
s += " (not copied: " + e.Message + ")";
}
}
System.Console.Error.WriteLine(s);
}
}
if (_peakPosL == _peakPosR)
{
System.Console.Error.WriteLine();
System.Console.Error.WriteLine("Zero time difference between channels. Are you sure the recordings are correct?");
}
}
catch (Exception e)
{
stderr.WriteLine();
stderr.WriteLine(e.Message);
stderr.WriteLine(e.StackTrace);
}
finally
{
foreach (string tempFile in _tempFiles)
{
try
{
File.Delete(tempFile);
}
catch (Exception) { /* ignore */ }
}
}
}
stderr.Flush();
}
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 TwoChannel(ISoundObj src)
{
if (src.NumChannels == 2)
{
return src;
}
else if (src.NumChannels == 1)
{
ChannelSplicer splicer = new ChannelSplicer();
splicer.Add(src);
splicer.Add(src);
return splicer;
}
else
{
ChannelSplicer splicer = new ChannelSplicer();
splicer.Add(new SingleChannel(src, 0));
splicer.Add(new SingleChannel(src, 1));
return splicer;
}
}
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;
}
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();
}
}