private void Start() { _process = new System.Diagnostics.Process(); System.Diagnostics.ProcessStartInfo processInfo = new System.Diagnostics.ProcessStartInfo(); processInfo.Arguments = _exeArgs; processInfo.FileName = _exeName; processInfo.UseShellExecute = false; processInfo.RedirectStandardInput = true; processInfo.RedirectStandardOutput = true; processInfo.RedirectStandardError = false; _in = _process.StandardInput; _out = _process.StandardOutput; _writer = new WaveWriter(_in.BaseStream); _writer.Input = _input; _writer.BitsPerSample = 16; _process.StartInfo = processInfo; try { _process.Start(); } catch (Exception e) { Trace.WriteLine("That didn't seem to work: {0}", e.Message); _process = null; } }
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 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 FindPeaks(ISoundObj impulse) { // Input: a single-channel impulse // Find the peak positions: // - unfiltered // - filtered with various bandpass filters uint sr = impulse.SampleRate; ushort nc = impulse.NumChannels; double peakM = Math.Round(MathUtil.Metres(_peakPosL, sr), 2); double peakFt = Math.Round(MathUtil.Feet(_peakPosL, sr), 2); stderr.WriteLine(" Impulse peak at sample {0} ({1}m, {2}ft)", _peakPosL, peakM, peakFt); FilterImpulse fi; WaveWriter wri; FastConvolver co; fi = new FilterImpulse(2048, bandpass(400,sr), FilterInterpolation.COSINE, sr); co = new FastConvolver(impulse, fi); wri = new WaveWriter("bp_400.wav", nc, sr, 16, DitherType.NONE, WaveFormat.PCM); wri.Input = co; wri.Normalization = -1; wri.Run(); wri.Close(); fi = new FilterImpulse(2048, bandpass(6000, sr), FilterInterpolation.COSINE, sr); co = new FastConvolver(impulse, fi); wri = new WaveWriter("bp_6000.wav", nc, sr, 16, DitherType.NONE, WaveFormat.PCM); wri.Input = co; wri.Normalization = -1; wri.Run(); wri.Close(); // and fi = new FilterImpulse(2048, bandpass(160, sr), FilterInterpolation.COSINE, sr); co = new FastConvolver(impulse, fi); wri = new WaveWriter("bp_160.wav", nc, sr, 16, DitherType.NONE, WaveFormat.PCM); wri.Input = co; wri.Normalization = -1; wri.Run(); wri.Close(); fi = new FilterImpulse(2048, bandpass(2560, sr), FilterInterpolation.COSINE, sr); co = new FastConvolver(impulse, fi); wri = new WaveWriter("bp_2560.wav", nc, sr, 16, DitherType.NONE, WaveFormat.PCM); wri.Input = co; wri.Normalization = -1; wri.Run(); wri.Close(); fi = new FilterImpulse(2048, bandpass(18000, sr), FilterInterpolation.COSINE, sr); co = new FastConvolver(impulse, fi); wri = new WaveWriter("bp_18k.wav", nc, sr, 16, DitherType.NONE, WaveFormat.PCM); wri.Input = co; wri.Normalization = -1; wri.Run(); wri.Close(); }
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 void WriteImpulsePCM(ISoundObj stereoImpulse, string fileNameL, string fileNameR) { ISoundObj sb; WaveWriter writer; ushort bits = 32; int nTot = 196607; int nBef = 98300; // need padding before the sample. // window the sample first, then pad it. // Total length 65536, includes nPad and _peakPosL before impulse int nPad = nBef - _peakPosL; int usableLength = nTot - nPad; int windowCenter = usableLength / 2; int windowSide = _peakPosL * 2 / 3; int windowFlat = windowCenter - windowSide; ISoundObj ch = new SingleChannel(stereoImpulse, 0); ISoundObj wn = new BlackmanHarris(windowCenter, windowSide, windowFlat); wn.Input = ch; sb = new SampleBuffer(wn).PaddedSubset(-nPad, nTot); writer = new WaveWriter(fileNameL); writer.Input = sb; writer.Format = WaveFormat.IEEE_FLOAT; writer.BitsPerSample = bits; writer.SampleRate = _sampleRate; writer.Raw = true; writer.Normalization = -6.0; writer.NormalizationType = NormalizationType.PEAK_DBFS; writer.Run(); writer.Close(); double g = writer.Gain; writer = null; nPad = nBef - _peakPosR; usableLength = nTot - nPad; windowCenter = usableLength / 2; windowSide = _peakPosR * 2 / 3; windowFlat = windowCenter - windowSide; ch = new SingleChannel(stereoImpulse, 1); wn = new BlackmanHarris(windowCenter, windowSide, windowFlat); wn.Input = ch; sb = new SampleBuffer(wn).PaddedSubset(-nPad, nTot); writer = new WaveWriter(fileNameR); writer.Input = sb; writer.Format = WaveFormat.IEEE_FLOAT; writer.BitsPerSample = bits; writer.SampleRate = _sampleRate; writer.Raw = true; writer.Gain = g; writer.Run(); writer.Close(); writer = 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 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; }
private static ISoundObj GetQuietnessFilter(uint nSampleRate, double nQuietness) { if (nQuietness == 0) { return null; } // Construct an impulse for the quietness setting. // nQuietness is value from 0 to 100 (percentage); // map this to phon values from 20 to 90 (approx) DateTime dtStart = DateTime.Now; FilterProfile spl = Loudness.DifferentialSPL(20, 20 + (nQuietness / 2)); // sound ISoundObj filterImpulse = new FilterImpulse(4096, spl, FilterInterpolation.COSINE, nSampleRate); if (_debug) { TimeSpan ts = DateTime.Now.Subtract(dtStart); Trace.WriteLine("GetQuietnessFilter1 " + ts.TotalMilliseconds); // DEBUG: Write the quietness impulse as wav16 string sPath = Path.Combine(_tempFolder, "QT_" + nQuietness); WaveWriter wri = new WaveWriter(sPath + ".wav"); wri.Input = filterImpulse; wri.Format = WaveFormat.PCM; wri.BitsPerSample = 16; wri.Normalization = -1.0; wri.Dither = DitherType.NONE;//.TRIANGULAR; wri.Run(); wri.Close(); ts = DateTime.Now.Subtract(dtStart); Trace.WriteLine("GetQuietnessFilter " + ts.TotalMilliseconds); } return filterImpulse; }
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; }
private static ISoundObj GenerateFlatnessFilter(string impulsePath, ISoundObj impulseObj, double nFlatness) { // Flatness-filters are single channel // nFlatness is from 0 to 100 // 0: follow the contours of the impulse // 100: completely flat DateTime dtStart = DateTime.Now; ISoundObj filterImpulse = null; uint nSR = impulseObj.SampleRate; uint nSR2 = nSR / 2; string sPath = FlatnessFilterPath(impulsePath, nSR, nFlatness); // Low flatness values (to 0) => very un-smooth // High flatness values (to 100) => very smooth double detail = (nFlatness / 50) + 0.05; // Get profile of the impulse FilterProfile lfg = new FilterProfile(impulseObj, detail); // Scale by the flatness values lfg = lfg * ((100 - nFlatness) / 100); // Invert lfg = lfg.Inverse(20); // Zero at HF lfg.Add(new FreqGain(nSR2 - 100, 0)); // Build the flatness filter filterImpulse = new FilterImpulse(8192, lfg, FilterInterpolation.COSINE, nSR); try { // Write the filter impulse to disk WaveWriter wri = new WaveWriter(sPath); wri.Input = filterImpulse; wri.Format = WaveFormat.IEEE_FLOAT; wri.BitsPerSample = 64; wri.Run(); wri.Close(); if (_debug) { // DEBUG: Write the flatness impulse as wav16 wri = new WaveWriter(sPath + ".wav"); wri.Input = filterImpulse; wri.Format = WaveFormat.PCM; wri.BitsPerSample = 16; wri.Normalization = -1.0; wri.Dither = DitherType.NONE;//.TRIANGULAR; wri.Run(); wri.Close(); } } catch (Exception e) { if (_debug) { Trace.WriteLine("GenerateFlatnessFilter: " + e.Message); } } impulseObj.Reset(); if (_debug) { TimeSpan ts = DateTime.Now.Subtract(dtStart); Trace.WriteLine("GenerateFlatnessFilter " + ts.TotalMilliseconds); } 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(); } }