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 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();
}
}