/// <summary>
/// Create a new sound by changing the speed/pitch of the old sound. 2.0 = an octave up, 1.0 = the same, 0.5 = down an octave
/// </summary>
public static WaveAudio ScalePitchAndDuration(WaveAudio w, double factor)
{
if (factor < 0) throw new ArgumentException("Factor must >= 0");
WaveAudio res = new WaveAudio(w.getSampleRate(), w.getNumChannels());
// do operation for all channels
for (int i = 0; i < w.getNumChannels(); i++)
res.data[i] = scalePitchAndDurationChannel(w.data[i], factor);
return res;
}
public static WaveAudio hiPassFilter(WaveAudio w, double factor) //0.5
{
WaveAudio ret = new WaveAudio(w.getSampleRate(), w.getNumChannels());
ret.LengthInSamples = w.LengthInSamples;
for (int ch = 0; ch < w.getNumChannels(); ch++)
{
ret.data[ch][0] = w.data[ch][0];
for (int i = 1; i < ret.data[ch].Length; i++)
ret.data[ch][i] = factor * ret.data[ch][i - 1] + (factor) * (w.data[ch][i] - w.data[ch][i-1]);
}
return ret;
}
//could also get continuous with window.
public static WaveAudio lowPassFilter(WaveAudio w, double factor) //0.5
{
//http://en.wikipedia.org/wiki/Low-pass_filter
WaveAudio ret = new WaveAudio(w.getSampleRate(), w.getNumChannels());
ret.LengthInSamples = w.LengthInSamples;
for (int ch = 0; ch < w.getNumChannels(); ch++)
{
ret.data[ch][0] = w.data[ch][0];
for (int i=1; i<ret.data[ch].Length; i++)
ret.data[ch][i] = (1-factor)*ret.data[ch][i-1] + (factor)*w.data[ch][i];
}
return ret;
}
public static WaveAudio hiPassFilter(WaveAudio w, double factor) //0.5
{
WaveAudio ret = new WaveAudio(w.getSampleRate(), w.getNumChannels());
ret.LengthInSamples = w.LengthInSamples;
for (int ch = 0; ch < w.getNumChannels(); ch++)
{
ret.data[ch][0] = w.data[ch][0];
for (int i = 1; i < ret.data[ch].Length; i++)
{
ret.data[ch][i] = factor * ret.data[ch][i - 1] + (factor) * (w.data[ch][i] - w.data[ch][i - 1]);
}
}
return(ret);
}
public static WaveAudio Tremolo(WaveAudio w, double tremfreq, double amp)
{
WaveAudio res = new WaveAudio(w.getSampleRate(), w.getNumChannels());
res.LengthInSamples = w.LengthInSamples;
double tremeloFreqScale = 2.0 * Math.PI * tremfreq / (double)w.getSampleRate();
for (int ch = 0; ch < w.data.Length; ch++)
{
for (int i = 0; i < w.data[ch].Length; i++)
{
double val = w.data[ch][i] * (1 + amp * Math.Sin(tremeloFreqScale * i));
if (val > 1.0)
{
val = 1.0;
}
else if (val < -1.0)
{
val = -1.0;
}
res.data[ch][i] = val;
}
}
return(res);
}
public static WaveAudio Vibrato(WaveAudio wave, double freq, double width)
{
if (width < 0)
{
throw new ArgumentException("Factor must >= 0");
}
WaveAudio newwave = new WaveAudio(wave.getSampleRate(), wave.getNumChannels());
// do operation for all channels
for (int i = 0; i < wave.getNumChannels(); i++)
{
newwave.data[i] = vibratoChannel(wave.data[i], wave.getSampleRate(), width, freq);
}
return(newwave);
}
/// <summary>
/// Create a new sound by changing the speed/pitch of the old sound. 2.0 = an octave up, 1.0 = the same, 0.5 = down an octave
/// </summary>
public static WaveAudio ScalePitchAndDuration(WaveAudio w, double factor)
{
if (factor < 0)
{
throw new ArgumentException("Factor must >= 0");
}
WaveAudio res = new WaveAudio(w.getSampleRate(), w.getNumChannels());
// do operation for all channels
for (int i = 0; i < w.getNumChannels(); i++)
{
res.data[i] = scalePitchAndDurationChannel(w.data[i], factor);
}
return(res);
}
//could also get continuous with window.
public static WaveAudio lowPassFilter(WaveAudio w, double factor) //0.5
{
//http://en.wikipedia.org/wiki/Low-pass_filter
WaveAudio ret = new WaveAudio(w.getSampleRate(), w.getNumChannels());
ret.LengthInSamples = w.LengthInSamples;
for (int ch = 0; ch < w.getNumChannels(); ch++)
{
ret.data[ch][0] = w.data[ch][0];
for (int i = 1; i < ret.data[ch].Length; i++)
{
ret.data[ch][i] = (1 - factor) * ret.data[ch][i - 1] + (factor) * w.data[ch][i];
}
}
return(ret);
}
public static double[][] SpectrumContentOverTime(WaveAudio w, int nBins, int nSize)
{
if (w.getNumChannels() != 1)
{
throw new Exception("Only mono supported.");
}
if (!isPowerOfTwo((uint)nSize))
{
throw new Exception("Size must be power of 2.");
}
int nDatapoints = w.LengthInSamples / nSize - 1;
double[][] res = new double[nDatapoints][];
double[] buffer = new double[nSize];
for (int i = 0; i < nDatapoints; i++)
{
// get samples from this slice of time. Put it into the buffer
Array.Copy(w.data[0], i * nSize, buffer, 0, nSize);
res[i] = getSpectrumContent(buffer, nBins);
}
return(res);
}
// The following create new audio without modifying original
public static WaveAudio Concatenate(WaveAudio w1, WaveAudio w2)
{
// make sure sample rates match we could be nicer and convert automatically
if (w1.m_currentSampleRate != w2.m_currentSampleRate) throw new Exception("Sample rates don't match");
if (w2.getNumChannels() != w2.getNumChannels()) throw new Exception("Number of channels don't match");
WaveAudio newwave = new WaveAudio(w1.getSampleRate(), w1.getNumChannels());
newwave.LengthInSamples = w1.LengthInSamples + w2.LengthInSamples;
for (int ch = 0; ch < w1.getNumChannels(); ch++)
{
// source sIndex, destination, destIndex, length
Array.Copy(w1.data[ch], 0, newwave.data[ch], 0, w1.data[ch].Length);
Array.Copy(w2.data[ch], 0, newwave.data[ch], w1.data[ch].Length, w2.data[0].Length);
}
return newwave;
}
public static WaveAudio GetSliceSample(WaveAudio wthis, int nStart, int nEnd)
{
WaveAudio slice = new WaveAudio(wthis.getSampleRate(), wthis.getNumChannels());
if (nEnd <= nStart || nEnd > wthis.LengthInSamples || nStart < 0) throw new Exception("Invalid slice");
for (int ch = 0; ch < slice.data.Length; ch++)
{
slice.data[ch] = new double[nEnd - nStart];
Array.Copy(wthis.data[ch], nStart, slice.data[ch], 0, nEnd - nStart);
}
return slice;
}
// These work by shifting the signal until it seems to correlate with itself.
// In other words if the signal looks very similar to (signal shifted 200 samples) than the fundamental period is probably 200 samples
// Note that the algorithm only works well when there's only one prominent fundamental.
// This could be optimized by looking at the rate of change to determine a maximum without testing all periods.
private static double[] detectPitchCalculation(WaveAudio w, double minHz, double maxHz, int nCandidates, int nResolution, PitchDetectAlgorithm algorithm)
{
// note that higher frequency means lower period
int nLowPeriodInSamples = hzToPeriodInSamples(maxHz, w.getSampleRate());
int nHiPeriodInSamples = hzToPeriodInSamples(minHz, w.getSampleRate());
if (nHiPeriodInSamples <= nLowPeriodInSamples) throw new Exception("Bad range for pitch detection.");
if (w.getNumChannels() != 1) throw new Exception("Only mono supported.");
double[] samples = w.data[0];
if (samples.Length < nHiPeriodInSamples) throw new Exception("Not enough samples.");
// both algorithms work in a similar way
// they yield an array of data, and then we find the index at which the value is highest.
double[] results = new double[nHiPeriodInSamples - nLowPeriodInSamples];
if (algorithm == PitchDetectAlgorithm.Amdf)
{
for (int period = nLowPeriodInSamples; period < nHiPeriodInSamples; period += nResolution)
{
double sum = 0;
// for each sample, see how close it is to a sample n away. Then sum these.
for (int i = 0; i < samples.Length - period; i++)
sum += Math.Abs(samples[i] - samples[i + period]);
double mean = sum / (double)samples.Length;
mean *= -1; //somewhat of a hack. We are trying to find the minimum value, but our findBestCandidates finds the max. value.
results[period - nLowPeriodInSamples] = mean;
}
}
else if (algorithm == PitchDetectAlgorithm.Autocorrelation)
{
for (int period = nLowPeriodInSamples; period < nHiPeriodInSamples; period += nResolution)
{
double sum = 0;
// for each sample, find correlation. (If they are far apart, small)
for (int i = 0; i < samples.Length - period; i++)
sum += samples[i] * samples[i + period];
double mean = sum / (double)samples.Length;
results[period - nLowPeriodInSamples] = mean;
}
}
// find the best indices
int[] bestIndices = findBestCandidates(nCandidates, ref results); //note findBestCandidates modifies parameter
// convert back to Hz
double[] res = new double[nCandidates];
for (int i=0; i<nCandidates;i++)
res[i] = periodInSamplesToHz(bestIndices[i]+nLowPeriodInSamples, w.getSampleRate());
return res;
}
// The following create new audio without modifying original
public static WaveAudio Concatenate(WaveAudio w1, WaveAudio w2)
{
// make sure sample rates match we could be nicer and convert automatically
if (w1.m_currentSampleRate != w2.m_currentSampleRate)
{
throw new Exception("Sample rates don't match");
}
if (w2.getNumChannels() != w2.getNumChannels())
{
throw new Exception("Number of channels don't match");
}
WaveAudio newwave = new WaveAudio(w1.getSampleRate(), w1.getNumChannels());
newwave.LengthInSamples = w1.LengthInSamples + w2.LengthInSamples;
for (int ch = 0; ch < w1.getNumChannels(); ch++)
{
// source sIndex, destination, destIndex, length
Array.Copy(w1.data[ch], 0, newwave.data[ch], 0, w1.data[ch].Length);
Array.Copy(w2.data[ch], 0, newwave.data[ch], w1.data[ch].Length, w2.data[0].Length);
}
return(newwave);
}
public static WaveAudio GetSliceSample(WaveAudio wthis, int nStart, int nEnd)
{
WaveAudio slice = new WaveAudio(wthis.getSampleRate(), wthis.getNumChannels());
if (nEnd <= nStart || nEnd > wthis.LengthInSamples || nStart < 0)
{
throw new Exception("Invalid slice");
}
for (int ch = 0; ch < slice.data.Length; ch++)
{
slice.data[ch] = new double[nEnd - nStart];
Array.Copy(wthis.data[ch], nStart, slice.data[ch], 0, nEnd - nStart);
}
return(slice);
}
/// <summary>
/// Find spectrum content of signal. Returns array, where each element represents energy at frequencies.
/// For example, SpectrumContent(w, 8) returns 8 numbers. The first in the array is the amount of energy at low frequencies, and the last is the energy at highest frequencies.
/// Note that FFT uses a power of 2 samples, and so all of the signal may not be used.
/// </summary>
/// <param name="w">Sound</param>
/// <param name="nBins">Number of bins to return.</param>
public static double[] SpectrumContent(WaveAudio w, int nBins)
{
if (w.getNumChannels() != 1) throw new Exception("Only mono supported.");
double[] buffer;
// FFT uses a power of 2 samples, so we might have to truncate.
if (!isPowerOfTwo((uint) w.LengthInSamples))
{
int nSize = (int)findNearestPowerOfTwo((uint) w.LengthInSamples);
buffer = new double[nSize];
Array.Copy(w.data[0], buffer, nSize);
}
else
buffer = w.data[0];
return getSpectrumContent(buffer, nBins);
}
public static double[][] SpectrumContentOverTime(WaveAudio w, int nBins, int nSize)
{
if (w.getNumChannels() != 1) throw new Exception("Only mono supported.");
if (!isPowerOfTwo((uint) nSize)) throw new Exception("Size must be power of 2.");
int nDatapoints = w.LengthInSamples / nSize - 1;
double[][] res = new double[nDatapoints][];
double[] buffer = new double[nSize];
for (int i = 0; i < nDatapoints; i++)
{
// get samples from this slice of time. Put it into the buffer
Array.Copy(w.data[0], i * nSize, buffer, 0, nSize);
res[i] = getSpectrumContent(buffer, nBins);
}
return res;
}
public static WaveAudio Phaser(WaveAudio wOriginal, double freq, double fb, int depth, int stages, int drywet)
{
double startphaseleft = 0;
double startphaseright = startphaseleft + Math.PI; //note that left and right channels should start pi out of phase
double freq_scaled = 2.0 * Math.PI * freq / (double)wOriginal.getSampleRate();
WaveAudio w = wOriginal.Clone();
if (w.getNumChannels() == 1)
{
effect_phaseraud_impl(w.data[0], freq_scaled, startphaseleft, fb, depth, stages, drywet);
}
else
{
effect_phaseraud_impl(w.data[0], freq_scaled, startphaseleft, fb, depth, stages, drywet);
effect_phaseraud_impl(w.data[1], freq_scaled, startphaseright, fb, depth, stages, drywet);
}
return(w);
}
public static WaveAudio Wahwah(WaveAudio wOriginal, double freq, double depth, double freqofs, double res)
{
double startphaseleft = 0;
double startphaseright = startphaseleft + Math.PI; //note that left and right channels should start pi out of phase
double freq_scaled = 2.0 * Math.PI * freq / (double)wOriginal.getSampleRate();
WaveAudio w = wOriginal.Clone();
if (w.getNumChannels() == 1)
{
effect_wahwahaud_impl(w.data[0], startphaseleft, freq_scaled, depth, freqofs, res);
}
else
{
effect_wahwahaud_impl(w.data[0], startphaseleft, freq_scaled, depth, freqofs, res);
effect_wahwahaud_impl(w.data[1], startphaseright, freq_scaled, depth, freqofs, res);
}
return(w);
}
public static WaveAudio Derivative(WaveAudio wOriginal)
{
WaveAudio w = wOriginal.Clone();
for (int ch = 0; ch < w.getNumChannels(); ch++)
{
for (int i = 0; i < w.data[ch].Length; i++)
{
if (i + 1 < w.data[ch].Length)
{
w.data[ch][i] = w.data[ch][i] - w.data[ch][i + 1];
}
else
{
w.data[ch][i] = w.data[ch][i] - 0;
}
}
}
return(w);
}
public static void propertytests()
{
// these aren't the best tests.
WaveAudio w1 = new WaveAudio(44100, 2);
asserteq(w1.data.Length, 2, "channels");
asserteq(w1.getNumChannels(), 2, "channels");
assert(w1.data[0] != null && w1.data[1] != null, "channels");
assert(w1.data[0].Length == 1 && w1.data[1].Length == 1, "004");
asserteq(w1.getSampleRate(), 44100, "005");
WaveAudio w1m = new WaveAudio(22050, 1);
asserteq(w1m.data.Length, 1, "channels");
assert(w1m.data[0] != null, "channels");
asserteq(w1m.data[0].Length, 1, "004");
asserteq(w1m.getSampleRate(), 22050, "005");
// now set some properties
w1m.LengthInSamples = 100;
asserteq(w1m.data[0].Length, 100);
asserteqf(w1m.LengthInSeconds, 100 / (double)w1m.getSampleRate(), 0.001);
}
/// <summary>
/// Find spectrum content of signal. Returns array, where each element represents energy at frequencies.
/// For example, SpectrumContent(w, 8) returns 8 numbers. The first in the array is the amount of energy at low frequencies, and the last is the energy at highest frequencies.
/// Note that FFT uses a power of 2 samples, and so all of the signal may not be used.
/// </summary>
/// <param name="w">Sound</param>
/// <param name="nBins">Number of bins to return.</param>
public static double[] SpectrumContent(WaveAudio w, int nBins)
{
if (w.getNumChannels() != 1)
{
throw new Exception("Only mono supported.");
}
double[] buffer;
// FFT uses a power of 2 samples, so we might have to truncate.
if (!isPowerOfTwo((uint)w.LengthInSamples))
{
int nSize = (int)findNearestPowerOfTwo((uint)w.LengthInSamples);
buffer = new double[nSize];
Array.Copy(w.data[0], buffer, nSize);
}
else
{
buffer = w.data[0];
}
return(getSpectrumContent(buffer, nBins));
}
// These work by shifting the signal until it seems to correlate with itself.
// In other words if the signal looks very similar to (signal shifted 200 samples) than the fundamental period is probably 200 samples
// Note that the algorithm only works well when there's only one prominent fundamental.
// This could be optimized by looking at the rate of change to determine a maximum without testing all periods.
private static double[] detectPitchCalculation(WaveAudio w, double minHz, double maxHz, int nCandidates, int nResolution, PitchDetectAlgorithm algorithm)
{
// note that higher frequency means lower period
int nLowPeriodInSamples = hzToPeriodInSamples(maxHz, w.getSampleRate());
int nHiPeriodInSamples = hzToPeriodInSamples(minHz, w.getSampleRate());
if (nHiPeriodInSamples <= nLowPeriodInSamples)
{
throw new Exception("Bad range for pitch detection.");
}
if (w.getNumChannels() != 1)
{
throw new Exception("Only mono supported.");
}
double[] samples = w.data[0];
if (samples.Length < nHiPeriodInSamples)
{
throw new Exception("Not enough samples.");
}
// both algorithms work in a similar way
// they yield an array of data, and then we find the index at which the value is highest.
double[] results = new double[nHiPeriodInSamples - nLowPeriodInSamples];
if (algorithm == PitchDetectAlgorithm.Amdf)
{
for (int period = nLowPeriodInSamples; period < nHiPeriodInSamples; period += nResolution)
{
double sum = 0;
// for each sample, see how close it is to a sample n away. Then sum these.
for (int i = 0; i < samples.Length - period; i++)
{
sum += Math.Abs(samples[i] - samples[i + period]);
}
double mean = sum / (double)samples.Length;
mean *= -1; //somewhat of a hack. We are trying to find the minimum value, but our findBestCandidates finds the max. value.
results[period - nLowPeriodInSamples] = mean;
}
}
else if (algorithm == PitchDetectAlgorithm.Autocorrelation)
{
for (int period = nLowPeriodInSamples; period < nHiPeriodInSamples; period += nResolution)
{
double sum = 0;
// for each sample, find correlation. (If they are far apart, small)
for (int i = 0; i < samples.Length - period; i++)
{
sum += samples[i] * samples[i + period];
}
double mean = sum / (double)samples.Length;
results[period - nLowPeriodInSamples] = mean;
}
}
// find the best indices
int[] bestIndices = findBestCandidates(nCandidates, ref results); //note findBestCandidates modifies parameter
// convert back to Hz
double[] res = new double[nCandidates];
for (int i = 0; i < nCandidates; i++)
{
res[i] = periodInSamplesToHz(bestIndices[i] + nLowPeriodInSamples, w.getSampleRate());
}
return(res);
}
// Helper function. It's long and gross because either sound could be longer.
// I could be more clever and use Math.Max / Min to have WaveAudio longer, WaveAudio shorter
// , but at least now it is readable
/// <summary>
/// Element-wise combination of two audio clips. For example, adding, or modulation.
/// </summary>
internal static WaveAudio elementWiseCombination(WaveAudio w1, WaveAudio w2, ElementWiseCombinationFn fn)
{
if (w1.m_currentSampleRate != w2.m_currentSampleRate)
{
throw new Exception("Sample rates don't match");
}
if (w1.getNumChannels() != w2.getNumChannels())
{
throw new Exception("Number of channels don't match");
}
WaveAudio newwave = new WaveAudio(w1.getSampleRate(), w1.getNumChannels());
newwave.LengthInSamples = Math.Max(w1.LengthInSamples, w2.LengthInSamples);
double val;
for (int ch = 0; ch < w1.getNumChannels(); ch++)
{
if (w1.LengthInSamples > w2.LengthInSamples)
{
for (int i = 0; i < w1.LengthInSamples; i++)
{
if (i >= w2.LengthInSamples)
{
val = fn(w1.data[ch][i], 0);
}
else
{
val = fn(w1.data[ch][i], w2.data[ch][i]);
}
if (val > 1.0)
{
val = 1.0;
}
else if (val < -1.0)
{
val = -1.0;
}
newwave.data[ch][i] = val;
}
}
else
{
for (int i = 0; i < w2.LengthInSamples; i++)
{
if (i >= w1.LengthInSamples)
{
val = fn(0, w2.data[ch][i]);
}
else
{
val = fn(w1.data[ch][i], w2.data[ch][i]);
}
if (val > 1.0)
{
val = 1.0;
}
else if (val < -1.0)
{
val = -1.0;
}
newwave.data[ch][i] = val;
}
}
}
return(newwave);
}
public static WaveAudio Vibrato(WaveAudio wave, double freq, double width)
{
if (width < 0) throw new ArgumentException("Factor must >= 0");
WaveAudio newwave = new WaveAudio(wave.getSampleRate(), wave.getNumChannels());
// do operation for all channels
for (int i = 0; i < wave.getNumChannels(); i++)
newwave.data[i] = vibratoChannel(wave.data[i], wave.getSampleRate(), width, freq);
return newwave;
}
static void iotests_perfile(string strFilename, int nBits, int nChannels, int nRate)
{
WaveAudio w01 = new WaveAudio(strFilename);
asserteq(w01.getNumChannels(), nChannels);
asserteq(w01.getSampleRate(), nRate, "011");
asserteqf(w01.LengthInSamples, 90725 * (nRate / 22050), 1.0, "012"); //note give 1.0 tolerance
asserteqf(w01.LengthInSeconds, 4.1145124, "013");
asserteq(w01.data.Length, nChannels);
asserteq(w01.data[0].Length, w01.LengthInSamples);
for (int i = 0; i < nChannels; i++)
asserteq(w01.data[i].Length, w01.LengthInSamples);
// test converting to other rates / quality
w01.SaveWaveFile("..\\..\\testout\\o_" + nRate + "_" + nBits + "_" + nRate + "_" + nChannels + ".wav", nBits);
nBits = (nBits == 8) ? 16 : 8;
w01.SaveWaveFile("..\\..\\testout\\ot_" + nRate + "_" + nBits + "_" + nRate + "_" + nChannels + ".wav", nBits);
}
public static WaveAudio Tremolo(WaveAudio w, double tremfreq, double amp)
{
WaveAudio res = new WaveAudio(w.getSampleRate(), w.getNumChannels());
res.LengthInSamples = w.LengthInSamples;
double tremeloFreqScale = 2.0 * Math.PI * tremfreq / (double)w.getSampleRate();
for (int ch=0;ch<w.data.Length; ch++)
{
for (int i = 0; i < w.data[ch].Length; i++)
{
double val = w.data[ch][i] * (1 + amp * Math.Sin(tremeloFreqScale * i));
if (val > 1.0) val = 1.0;
else if (val < -1.0) val = -1.0;
res.data[ch][i] = val;
}
}
return res;
}
// Test between 20Hz and 500Hz. This works very well, although it is hard to eliminate octave errors
// This test will play original, then a sine at the frequency it detected. The two should line up.
private static void pitchdetectwav(AudioPlayer pl, string strFilename, PitchDetection.PitchDetectAlgorithm algorithm)
{
string strInstdir = mediadirpitch;
WaveAudio w = new WaveAudio(strInstdir + strFilename);
if (w.getNumChannels() != 1) w.setNumChannels(1, true);
double dfreq = PitchDetection.DetectPitch(w, 50,500,algorithm);
WaveAudio testPitch = new Triangle(dfreq, 0.7).CreateWaveAudio(1.0);
pl.Play(w);
pl.Play(testPitch);
}
// Helper function. It's long and gross because either sound could be longer.
// I could be more clever and use Math.Max / Min to have WaveAudio longer, WaveAudio shorter
// , but at least now it is readable
/// <summary>
/// Element-wise combination of two audio clips. For example, adding, or modulation.
/// </summary>
internal static WaveAudio elementWiseCombination(WaveAudio w1, WaveAudio w2, ElementWiseCombinationFn fn)
{
if (w1.m_currentSampleRate != w2.m_currentSampleRate) throw new Exception("Sample rates don't match");
if (w1.getNumChannels() != w2.getNumChannels()) throw new Exception("Number of channels don't match");
WaveAudio newwave = new WaveAudio(w1.getSampleRate(), w1.getNumChannels());
newwave.LengthInSamples = Math.Max(w1.LengthInSamples, w2.LengthInSamples);
double val;
for (int ch = 0; ch < w1.getNumChannels(); ch++)
{
if (w1.LengthInSamples > w2.LengthInSamples)
{
for (int i = 0; i < w1.LengthInSamples; i++)
{
if (i >= w2.LengthInSamples)
val = fn(w1.data[ch][i], 0);
else
val = fn(w1.data[ch][i], w2.data[ch][i]);
if (val > 1.0) val = 1.0;
else if (val < -1.0) val = -1.0;
newwave.data[ch][i] = val;
}
}
else
{
for (int i = 0; i < w2.LengthInSamples; i++)
{
if (i >= w1.LengthInSamples)
val = fn(0, w2.data[ch][i]);
else
val = fn(w1.data[ch][i], w2.data[ch][i]);
if (val > 1.0) val = 1.0;
else if (val < -1.0) val = -1.0;
newwave.data[ch][i] = val;
}
}
}
return newwave;
}